diff options
| author | Matthias P. Braendli <matthias.braendli@mpb.li> | 2020-03-31 10:03:58 +0200 |
|---|---|---|
| committer | Matthias P. Braendli <matthias.braendli@mpb.li> | 2020-03-31 10:03:58 +0200 |
| commit | a1eb6cf861d3c1cbd4e6c016be3cbd2a1e3d797d (patch) | |
| tree | 2b4790eec8f47fb086e645717f07c53b30ace919 /fdk-aac/libAACenc | |
| parent | 2f84a54ec1d10b10293c7b1f4ab9fee31f3c6327 (diff) | |
| parent | c6a73c219dbfdfe639372d9922f4eb512f06fa2f (diff) | |
| download | ODR-AudioEnc-a1eb6cf861d3c1cbd4e6c016be3cbd2a1e3d797d.tar.gz ODR-AudioEnc-a1eb6cf861d3c1cbd4e6c016be3cbd2a1e3d797d.tar.bz2 ODR-AudioEnc-a1eb6cf861d3c1cbd4e6c016be3cbd2a1e3d797d.zip | |
Merge GStreamer into next
Diffstat (limited to 'fdk-aac/libAACenc')
73 files changed, 36386 insertions, 0 deletions
diff --git a/fdk-aac/libAACenc/include/aacenc_lib.h b/fdk-aac/libAACenc/include/aacenc_lib.h new file mode 100644 index 0000000..231bbb4 --- /dev/null +++ b/fdk-aac/libAACenc/include/aacenc_lib.h @@ -0,0 +1,1733 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M. Lohwasser + + Description: + +*******************************************************************************/ + +/** + * \file aacenc_lib.h + * \brief FDK AAC Encoder library interface header file. + * +\mainpage Introduction + +\section Scope + +This document describes the high-level interface and usage of the ISO/MPEG-2/4 +AAC Encoder library developed by the Fraunhofer Institute for Integrated +Circuits (IIS). + +The library implements encoding on the basis of the MPEG-2 and MPEG-4 AAC +Low-Complexity standard, and depending on the library's configuration, MPEG-4 +High-Efficiency AAC v2 and/or AAC-ELD standard. + +All references to SBR (Spectral Band Replication) are only applicable to HE-AAC +or AAC-ELD versions of the library. All references to PS (Parametric Stereo) are +only applicable to HE-AAC v2 versions of the library. + +\section encBasics Encoder Basics + +This document can only give a rough overview about the ISO/MPEG-2 and ISO/MPEG-4 +AAC audio coding standard. To understand all the terms in this document, you are +encouraged to read the following documents. + +- ISO/IEC 13818-7 (MPEG-2 AAC), which defines the syntax of MPEG-2 AAC audio +bitstreams. +- ISO/IEC 14496-3 (MPEG-4 AAC, subparts 1 and 4), which defines the syntax of +MPEG-4 AAC audio bitstreams. +- Lutzky, Schuller, Gayer, Krämer, Wabnik, "A guideline to audio codec +delay", 116th AES Convention, May 8, 2004 + +MPEG Advanced Audio Coding is based on a time-to-frequency mapping of the +signal. The signal is partitioned into overlapping portions and transformed into +frequency domain. The spectral components are then quantized and coded. \n An +MPEG-2 or MPEG-4 AAC audio bitstream is composed of frames. Contrary to MPEG-1/2 +Layer-3 (mp3), the length of individual frames is not restricted to a fixed +number of bytes, but can take on any length between 1 and 768 bytes. + + +\page LIBUSE Library Usage + +\section InterfaceDescription API Files + +All API header files are located in the folder /include of the release package. +All header files are provided for usage in C/C++ programs. The AAC encoder +library API functions are located in aacenc_lib.h. + +In binary releases the encoder core resides in statically linkable libraries +called for example libAACenc.a/libFDK.a (LINUX) or FDK_fastaaclib.lib (MS Visual +C++) for the plain AAC-LC core encoder and libSBRenc.a (LINUX) or +FDK_sbrEncLib.lib (MS Visual C++) for the SBR (Spectral Band Replication) and PS +(Parametric Stereo) modules. + +\section CallingSequence Calling Sequence + +For encoding of ISO/MPEG-2/4 AAC bitstreams the following sequence is mandatory. +Input read and output write functions as well as the corresponding open and +close functions are left out, since they may be implemented differently +according to the user's specific requirements. The example implementation uses +file-based input/output. + +-# Call aacEncOpen() to allocate encoder instance with required \ref encOpen +"configuration". \code HANDLE_AACENCODER hAacEncoder = NULL; if ( (ErrorStatus = +aacEncOpen(&hAacEncoder,0,0)) != AACENC_OK ) { \endcode +-# Call aacEncoder_SetParam() for each parameter to be set. AOT, samplingrate, +channelMode, bitrate and transport type are \ref encParams "mandatory". \code +ErrorStatus = aacEncoder_SetParam(hAacEncoder, parameter, value); +\endcode +-# Call aacEncEncode() with NULL parameters to \ref encReconf "initialize" +encoder instance with present parameter set. \code ErrorStatus = +aacEncEncode(hAacEncoder, NULL, NULL, NULL, NULL); \endcode +-# Call aacEncInfo() to retrieve a configuration data block to be transmitted +out of band. This is required when using RFC3640 or RFC3016 like transport. +\code +AACENC_InfoStruct encInfo; +aacEncInfo(hAacEncoder, &encInfo); +\endcode +-# Encode input audio data in loop. +\code +do +{ +\endcode +Feed \ref feedInBuf "input buffer" with new audio data and provide input/output +\ref bufDes "arguments" to aacEncEncode(). \code ErrorStatus = +aacEncEncode(hAacEncoder, &inBufDesc, &outBufDesc, &inargs, &outargs); \endcode +Write \ref writeOutData "output data" to file or audio device. +\code +} while (ErrorStatus==AACENC_OK); +\endcode +-# Call aacEncClose() and destroy encoder instance. +\code +aacEncClose(&hAacEncoder); +\endcode + + +\section encOpen Encoder Instance Allocation + +The assignment of the aacEncOpen() function is very flexible and can be used in +the following way. +- If the amount of memory consumption is not an issue, the encoder instance can +be allocated for the maximum number of possible audio channels (for example 6 or +8) with the full functional range supported by the library. This is the default +open procedure for the AAC encoder if memory consumption does not need to be +minimized. \code aacEncOpen(&hAacEncoder,0,0) \endcode +- If the required MPEG-4 AOTs do not call for the full functional range of the +library, encoder modules can be allocated selectively. \verbatim +------------------------------------------------------ + AAC | SBR | PS | MD | FLAGS | value +-----+-----+-----+----+-----------------------+------- + X | - | - | - | (0x01) | 0x01 + X | X | - | - | (0x01|0x02) | 0x03 + X | X | X | - | (0x01|0x02|0x04) | 0x07 + X | - | - | X | (0x01 |0x10) | 0x11 + X | X | - | X | (0x01|0x02 |0x10) | 0x13 + X | X | X | X | (0x01|0x02|0x04|0x10) | 0x17 +------------------------------------------------------ + - AAC: Allocate AAC Core Encoder module. + - SBR: Allocate Spectral Band Replication module. + - PS: Allocate Parametric Stereo module. + - MD: Allocate Meta Data module within AAC encoder. +\endverbatim +\code aacEncOpen(&hAacEncoder,value,0) \endcode +- Specifying the maximum number of channels to be supported in the encoder +instance can be done as follows. + - For example allocate an encoder instance which supports 2 channels for all +supported AOTs. The library itself may be capable of encoding up to 6 or 8 +channels but in this example only 2 channel encoding is required and thus only +buffers for 2 channels are allocated to save data memory. \code +aacEncOpen(&hAacEncoder,0,2) \endcode + - Additionally the maximum number of supported channels in the SBR module can +be denoted separately.\n In this example the encoder instance provides a maximum +of 6 channels out of which up to 2 channels support SBR. This encoder instance +can produce for example 5.1 channel AAC-LC streams or stereo HE-AAC (v2) +streams. HE-AAC 5.1 multi channel is not possible since only 2 out of 6 channels +support SBR, which saves data memory. \code aacEncOpen(&hAacEncoder,0,6|(2<<8)) +\endcode \n + +\section bufDes Input/Output Arguments + +\subsection allocIOBufs Provide Buffer Descriptors +In the present encoder API, the input and output buffers are described with \ref +AACENC_BufDesc "buffer descriptors". This mechanism allows a flexible handling +of input and output buffers without impact to the actual encoding call. Optional +buffers are necessary e.g. for ancillary data, meta data input or additional +output buffers describing superframing data in DAB+ or DRM+.\n At least one +input buffer for audio input data and one output buffer for bitstream data must +be allocated. The input buffer size can be a user defined multiple of the number +of input channels. PCM input data will be copied from the user defined PCM +buffer to an internal input buffer and so input data can be less than one AAC +audio frame. The output buffer size should be 6144 bits per channel excluding +the LFE channel. If the output data does not fit into the provided buffer, an +AACENC_ERROR will be returned by aacEncEncode(). \code static INT_PCM +inputBuffer[8*2048]; static UCHAR ancillaryBuffer[50]; static +AACENC_MetaData metaDataSetup; static UCHAR outputBuffer[8192]; +\endcode + +All input and output buffer must be clustered in input and output buffer arrays. +\code +static void* inBuffer[] = { inputBuffer, ancillaryBuffer, &metaDataSetup +}; static INT inBufferIds[] = { IN_AUDIO_DATA, IN_ANCILLRY_DATA, +IN_METADATA_SETUP }; static INT inBufferSize[] = { sizeof(inputBuffer), +sizeof(ancillaryBuffer), sizeof(metaDataSetup) }; static INT inBufferElSize[] += { sizeof(INT_PCM), sizeof(UCHAR), sizeof(AACENC_MetaData) }; + +static void* outBuffer[] = { outputBuffer }; +static INT outBufferIds[] = { OUT_BITSTREAM_DATA }; +static INT outBufferSize[] = { sizeof(outputBuffer) }; +static INT outBufferElSize[] = { sizeof(UCHAR) }; +\endcode + +Allocate buffer descriptors +\code +AACENC_BufDesc inBufDesc; +AACENC_BufDesc outBufDesc; +\endcode + +Initialize input buffer descriptor +\code +inBufDesc.numBufs = sizeof(inBuffer)/sizeof(void*); +inBufDesc.bufs = (void**)&inBuffer; +inBufDesc.bufferIdentifiers = inBufferIds; +inBufDesc.bufSizes = inBufferSize; +inBufDesc.bufElSizes = inBufferElSize; +\endcode + +Initialize output buffer descriptor +\code +outBufDesc.numBufs = sizeof(outBuffer)/sizeof(void*); +outBufDesc.bufs = (void**)&outBuffer; +outBufDesc.bufferIdentifiers = outBufferIds; +outBufDesc.bufSizes = outBufferSize; +outBufDesc.bufElSizes = outBufferElSize; +\endcode + +\subsection argLists Provide Input/Output Argument Lists +The input and output arguments of an aacEncEncode() call are described in +argument structures. \code AACENC_InArgs inargs; AACENC_OutArgs outargs; +\endcode + +\section feedInBuf Feed Input Buffer +The input buffer should be handled as a modulo buffer. New audio data in the +form of pulse-code- modulated samples (PCM) must be read from external and be +fed to the input buffer depending on its fill level. The required sample bitrate +(represented by the data type INT_PCM which is 16, 24 or 32 bits wide) is fixed +and depends on library configuration (usually 16 bit). \code inargs.numInSamples ++= WAV_InputRead ( wavIn, &inputBuffer[inargs.numInSamples], + FDKmin(encInfo.inputChannels*encInfo.frameLength, + sizeof(inputBuffer) / + sizeof(INT_PCM)-inargs.numInSamples), + SAMPLE_BITS + ); +\endcode + +After the encoder's internal buffer is fed with incoming audio samples, and +aacEncEncode() processed the new input data, update/move remaining samples in +input buffer, simulating a modulo buffer: \code if (outargs.numInSamples>0) { + FDKmemmove( inputBuffer, + &inputBuffer[outargs.numInSamples], + sizeof(INT_PCM)*(inargs.numInSamples-outargs.numInSamples) ); + inargs.numInSamples -= outargs.numInSamples; +} +\endcode + +\section writeOutData Output Bitstream Data +If any AAC bitstream data is available, write it to output file or device. This +can be done once the following condition is true: \code if +(outargs.numOutBytes>0) { + +} +\endcode + +If you use file I/O then for example call mpegFileWrite_Write() from the library +libMpegFileWrite \code mpegFileWrite_Write(hMpegFile, outputBuffer, +outargs.numOutBytes, aacEncoder_GetParam(hAacEncoder, AACENC_GRANULE_LENGTH)); +\endcode + +\section cfgMetaData Meta Data Configuration + +If the present library is configured with Metadata support, it is possible to +insert meta data side info into the generated audio bitstream while encoding. + +To work with meta data the encoder instance has to be \ref encOpen "allocated" +with meta data support. The meta data mode must be be configured with the +::AACENC_METADATA_MODE parameter and aacEncoder_SetParam() function. \code +aacEncoder_SetParam(hAacEncoder, AACENC_METADATA_MODE, 0-3); \endcode + +This configuration indicates how to embed meta data into bitstrem. Either no +insertion, MPEG or ETSI style. The meta data itself must be specified within the +meta data setup structure AACENC_MetaData. + +Changing one of the AACENC_MetaData setup parameters can be achieved from +outside the library within ::IN_METADATA_SETUP input buffer. There is no need to +supply meta data setup structure every frame. If there is no new meta setup data +available, the encoder uses the previous setup or the default configuration in +initial state. + +In general the audio compressor and limiter within the encoder library can be +configured with the ::AACENC_METADATA_DRC_PROFILE parameter +AACENC_MetaData::drc_profile and and AACENC_MetaData::comp_profile. +\n + +\section encReconf Encoder Reconfiguration + +The encoder library allows reconfiguration of the encoder instance with new +settings continuously between encoding frames. Each parameter to be changed must +be set with a single aacEncoder_SetParam() call. The internal status of each +parameter can be retrieved with an aacEncoder_GetParam() call.\n There is no +stand-alone reconfiguration function available. When parameters were modified +from outside the library, an internal control mechanism triggers the necessary +reconfiguration process which will be applied at the beginning of the following +aacEncEncode() call. This state can be observed from external via the +AACENC_INIT_STATUS and aacEncoder_GetParam() function. The reconfiguration +process can also be applied immediately when all parameters of an aacEncEncode() +call are NULL with a valid encoder handle.\n\n The internal reconfiguration +process can be controlled from extern with the following access. \code +aacEncoder_SetParam(hAacEncoder, AACENC_CONTROL_STATE, AACENC_CTRLFLAGS); +\endcode + + +\section encParams Encoder Parametrization + +All parameteres listed in ::AACENC_PARAM can be modified within an encoder +instance. + +\subsection encMandatory Mandatory Encoder Parameters +The following parameters must be specified when the encoder instance is +initialized. \code aacEncoder_SetParam(hAacEncoder, AACENC_AOT, value); +aacEncoder_SetParam(hAacEncoder, AACENC_BITRATE, value); +aacEncoder_SetParam(hAacEncoder, AACENC_SAMPLERATE, value); +aacEncoder_SetParam(hAacEncoder, AACENC_CHANNELMODE, value); +\endcode +Beyond that is an internal auto mode which preinitizializes the ::AACENC_BITRATE +parameter if the parameter was not set from extern. The bitrate depends on the +number of effective channels and sampling rate and is determined as follows. +\code +AAC-LC (AOT_AAC_LC): 1.5 bits per sample +HE-AAC (AOT_SBR): 0.625 bits per sample (dualrate sbr) +HE-AAC (AOT_SBR): 1.125 bits per sample (downsampled sbr) +HE-AAC v2 (AOT_PS): 0.5 bits per sample +\endcode + +\subsection channelMode Channel Mode Configuration +The input audio data is described with the ::AACENC_CHANNELMODE parameter in the +aacEncoder_SetParam() call. It is not possible to use the encoder instance with +a 'number of input channels' argument. Instead, the channelMode must be set as +follows. \code aacEncoder_SetParam(hAacEncoder, AACENC_CHANNELMODE, value); +\endcode The parameter is specified in ::CHANNEL_MODE and can be mapped from the +number of input channels in the following way. \code CHANNEL_MODE chMode = +MODE_INVALID; + +switch (nChannels) { + case 1: chMode = MODE_1; break; + case 2: chMode = MODE_2; break; + case 3: chMode = MODE_1_2; break; + case 4: chMode = MODE_1_2_1; break; + case 5: chMode = MODE_1_2_2; break; + case 6: chMode = MODE_1_2_2_1; break; + case 7: chMode = MODE_6_1; break; + case 8: chMode = MODE_7_1_BACK; break; + default: + chMode = MODE_INVALID; +} +return chMode; +\endcode + +\subsection bitreservoir Bitreservoir Configuration +In AAC, the default bitreservoir configuration depends on the chosen bitrate per +frame and the number of effective channels. The size can be determined as below. +\f[ +bitreservoir = nEffChannels*6144 - (bitrate*framelength/samplerate) +\f] +Due to audio quality concerns it is not recommended to change the bitreservoir +size to a lower value than the default setting! However, for minimizing the +delay for streaming applications or for achieving a constant size of the +bitstream packages in each frame, it may be necessaray to change the +bitreservoir size. This can be done with the ::AACENC_PEAK_BITRATE parameter. +\code +aacEncoder_SetParam(hAacEncoder, AACENC_PEAK_BITRATE, value); +\endcode +By setting ::AACENC_BITRATEMODE to fixed framing, the bitreservoir is disabled. +A disabled bitreservoir results in a constant size for each bitstream package. +Please note that especially at lower bitrates a disabled bitreservoir can +downgrade the audio quality considerably! The default bitreservoir configuration +can be achieved as follows. \code aacEncoder_SetParam(hAacEncoder, +AACENC_BITRESERVOIR, -1); \endcode + +To achieve acceptable audio quality with a reduced bitreservoir size setting at +least 1000 bits per audio channel is recommended. For a multichannel audio file +with 5.1 channels the bitreservoir reduced to 5000 bits results in acceptable +audio quality. + + +\subsection vbrmode Variable Bitrate Mode +The encoder provides various Variable Bitrate Modes that differ in audio quality +and average overall bitrate. The given values are averages over time, different +encoder settings and strongly depend on the type of audio signal. The VBR +configurations can be adjusted via ::AACENC_BITRATEMODE encoder parameter. +\verbatim +-------------------------------------------- + VBR_MODE | Approx. Bitrate in kbps/channel + | AAC-LC | AAC-LD/AC_ELD +----------+---------------+----------------- + VBR_1 | 32 - 48 | 32 - 56 + VBR_2 | 40 - 56 | 40 - 64 + VBR_3 | 48 - 64 | 48 - 72 + VBR_4 | 64 - 80 | 64 - 88 + VBR_5 | 96 - 120 | 112 - 144 +-------------------------------------------- +\endverbatim +The bitrate ranges apply for individual audio channels. In case of multichannel +configurations the average bitrate might be estimated by multiplying with the +number of effective channels. This corresponds to all audio input channels +exclusively the low frequency channel. At configurations which are making use of +downmix modules the AAC core channels respectively downmix channels shall be +considered. For ::AACENC_AOT which are using SBR, the average bitrate can be +estimated by using the ratio of 0.5 for dualrate SBR and 0.75 for downsampled +SBR configurations. + + +\subsection encQual Audio Quality Considerations +The default encoder configuration is suggested to be used. Encoder tools such as +TNS and PNS are activated by default and are internally controlled (see \ref +BEHAVIOUR_TOOLS). + +There is an additional quality parameter called ::AACENC_AFTERBURNER. In the +default configuration this quality switch is deactivated because it would cause +a workload increase which might be significant. If workload is not an issue in +the application we recommended to activate this feature. \code +aacEncoder_SetParam(hAacEncoder, AACENC_AFTERBURNER, 0/1); \endcode + +\subsection encELD ELD Auto Configuration Mode +For ELD configuration a so called auto configurator is available which +configures SBR and the SBR ratio by itself. The configurator is used when the +encoder parameter ::AACENC_SBR_MODE and ::AACENC_SBR_RATIO are not set +explicitly. + +Based on sampling rate and chosen bitrate a reasonable SBR configuration will be +used. \verbatim +------------------------------------------------------------------ + Sampling Rate | Total Bitrate | No. of | SBR | SBR Ratio + [kHz] | [bit/s] | Chan | | + | | | | +---------------+-----------------+--------+-----+----------------- + ]min, 16[ | min - max | 1 | off | --- +---------------+-----------------+--------------+----------------- + [16] | min - 27999 | 1 | on | downsampled SBR + | 28000 - max | 1 | off | --- +---------------+-----------------+--------------+----------------- + ]16 - 24] | min - 39999 | 1 | on | downsampled SBR + | 40000 - max | 1 | off | --- +---------------+-----------------+--------------+----------------- + ]24 - 32] | min - 27999 | 1 | on | dualrate SBR + | 28000 - 55999 | 1 | on | downsampled SBR + | 56000 - max | 1 | off | --- +---------------+-----------------+--------------+----------------- + ]32 - 44.1] | min - 63999 | 1 | on | dualrate SBR + | 64000 - max | 1 | off | --- +---------------+-----------------+--------------+----------------- + ]44.1 - 48] | min - 63999 | 1 | on | dualrate SBR + | 64000 - max | 1 | off | --- + | | | | +---------------+-----------------+--------+-----+----------------- + ]min, 16[ | min - max | 2 | off | --- +---------------+-----------------+--------------+----------------- + [16] | min - 31999 | 2 | on | downsampled SBR + | 32000 - 63999 | 2 | on | downsampled SBR + | 64000 - max | 2 | off | --- +---------------+-----------------+--------------+----------------- + ]16 - 24] | min - 47999 | 2 | on | downsampled SBR + | 48000 - 79999 | 2 | on | downsampled SBR + | 80000 - max | 2 | off | --- +---------------+-----------------+--------------+----------------- + ]24 - 32] | min - 31999 | 2 | on | dualrate SBR + | 32000 - 67999 | 2 | on | dualrate SBR + | 68000 - 95999 | 2 | on | downsampled SBR + | 96000 - max | 2 | off | --- +---------------+-----------------+--------------+----------------- + ]32 - 44.1] | min - 43999 | 2 | on | dualrate SBR + | 44000 - 127999 | 2 | on | dualrate SBR + | 128000 - max | 2 | off | --- +---------------+-----------------+--------------+----------------- + ]44.1 - 48] | min - 43999 | 2 | on | dualrate SBR + | 44000 - 127999 | 2 | on | dualrate SBR + | 128000 - max | 2 | off | --- + | | | +------------------------------------------------------------------ +\endverbatim + +\subsection encDsELD Reduced Delay (Downscaled) Mode +The downscaled mode of AAC-ELD reduces the algorithmic delay of AAC-ELD by +virtually increasing the sampling rate. When using the downscaled mode, the +bitrate should be increased for keeping the same audio quality level. For common +signals, the bitrate should be increased by 25% for a downscale factor of 2. + +Currently, downscaling factors 2 and 4 are supported. +To enable the downscaled mode in the encoder, the framelength parameter +AACENC_GRANULE_LENGTH must be set accordingly to 256 or 240 for a downscale +factor of 2 or 128 or 120 for a downscale factor of 4. The default values of 512 +or 480 mean that no downscaling is applied. \code +aacEncoder_SetParam(hAacEncoder, AACENC_GRANULE_LENGTH, 256); +aacEncoder_SetParam(hAacEncoder, AACENC_GRANULE_LENGTH, 128); +\endcode + +Downscaled bitstreams are fully backwards compatible. However, the legacy +decoder needs to support high sample rate, e.g. 96kHz. The signaled sampling +rate is multiplied by the downscale factor. Although not required, downscaling +should be applied when decoding downscaled bitstreams. It reduces CPU workload +and the output will have the same sampling rate as the input. In an ideal +configuration both encoder and decoder should run with the same downscale +factor. + +The following table shows approximate filter bank delays in ms for common +sampling rates(sr) at framesize(fs), and downscale factor(dsf), based on this +formula: \f[ 1000 * fs / (dsf * sr) \f] + +\verbatim +-------------------------------------- + | 512/2 | 512/4 | 480/2 | 480/4 +------+-------+-------+-------+------- +22050 | 17.41 | 8.71 | 16.33 | 8.16 +32000 | 12.00 | 6.00 | 11.25 | 5.62 +44100 | 8.71 | 4.35 | 8.16 | 4.08 +48000 | 8.00 | 4.00 | 7.50 | 3.75 +-------------------------------------- +\endverbatim + +\section audiochCfg Audio Channel Configuration +The MPEG standard refers often to the so-called Channel Configuration. This +Channel Configuration is used for a fixed Channel Mapping. The configurations +1-7 and 11,12,14 are predefined in MPEG standard and used for implicit +signalling within the encoded bitstream. For user defined Configurations the +Channel Configuration is set to 0 and the Channel Mapping must be explecitly +described with an appropriate Program Config Element. The present Encoder +implementation does not allow the user to configure this Channel Configuration +from extern. The Encoder implementation supports fixed Channel Modes which are +mapped to Channel Configuration as follow. \verbatim +---------------------------------------------------------------------------------------- + ChannelMode | ChCfg | Height | front_El | side_El | back_El | +lfe_El +-----------------------+-------+--------+---------------+----------+----------+--------- +MODE_1 | 1 | NORM | SCE | | | +MODE_2 | 2 | NORM | CPE | | | +MODE_1_2 | 3 | NORM | SCE, CPE | | | +MODE_1_2_1 | 4 | NORM | SCE, CPE | | SCE | +MODE_1_2_2 | 5 | NORM | SCE, CPE | | CPE | +MODE_1_2_2_1 | 6 | NORM | SCE, CPE | | CPE | +LFE MODE_1_2_2_2_1 | 7 | NORM | SCE, CPE, CPE | | CPE +| LFE MODE_6_1 | 11 | NORM | SCE, CPE | | CPE, +SCE | LFE MODE_7_1_BACK | 12 | NORM | SCE, CPE | | +CPE, CPE | LFE +-----------------------+-------+--------+---------------+----------+----------+--------- +MODE_7_1_TOP_FRONT | 14 | NORM | SCE, CPE | | CPE | +LFE | | TOP | CPE | | | +-----------------------+-------+--------+---------------+----------+----------+--------- +MODE_7_1_REAR_SURROUND | 0 | NORM | SCE, CPE | | CPE, CPE | +LFE MODE_7_1_FRONT_CENTER | 0 | NORM | SCE, CPE, CPE | | CPE +| LFE +---------------------------------------------------------------------------------------- +- NORM: Normal Height Layer. - TOP: Top Height Layer. - BTM: Bottom Height +Layer. +- SCE: Single Channel Element. - CPE: Channel Pair. - LFE: Low Frequency +Element. \endverbatim + +The Table describes all fixed Channel Elements for each Channel Mode which are +assigned to a speaker arrangement. The arrangement includes front, side, back +and lfe Audio Channel Elements in the normal height layer, possibly followed by +front, side, and back elements in the top and bottom layer (Channel +Configuration 14). \n This mapping of Audio Channel Elements is defined in MPEG +standard for Channel Config 1-7 and 11,12,14.\n In case of Channel Config 0 or +writing matrix mixdown coefficients, the encoder enables the writing of Program +Config Element itself as described in \ref encPCE. The configuration used in +Program Config Element refers to the denoted Table.\n Beside the Channel Element +assignment the Channel Modes are resposible for audio input data channel +mapping. The Channel Mapping of the audio data depends on the selected +::AACENC_CHANNELORDER which can be MPEG or WAV like order.\n Following table +describes the complete channel mapping for both Channel Order configurations. +\verbatim +--------------------------------------------------------------------------------------- +ChannelMode | MPEG-Channelorder | WAV-Channelorder +-----------------------+---+---+---+---+---+---+---+---+---+---+---+---+---+---+---+--- +MODE_1 | 0 | | | | | | | | 0 | | | | | | +| MODE_2 | 0 | 1 | | | | | | | 0 | 1 | | | | +| | MODE_1_2 | 0 | 1 | 2 | | | | | | 2 | 0 | 1 | | +| | | MODE_1_2_1 | 0 | 1 | 2 | 3 | | | | | 2 | 0 | 1 | 3 +| | | | MODE_1_2_2 | 0 | 1 | 2 | 3 | 4 | | | | 2 | 0 | 1 +| 3 | 4 | | | MODE_1_2_2_1 | 0 | 1 | 2 | 3 | 4 | 5 | | | 2 | 0 +| 1 | 4 | 5 | 3 | | MODE_1_2_2_2_1 | 0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 | 2 +| 6 | 7 | 0 | 1 | 4 | 5 | 3 MODE_6_1 | 0 | 1 | 2 | 3 | 4 | 5 | 6 | +| 2 | 0 | 1 | 4 | 5 | 6 | 3 | MODE_7_1_BACK | 0 | 1 | 2 | 3 | 4 | 5 | 6 +| 7 | 2 | 0 | 1 | 6 | 7 | 4 | 5 | 3 MODE_7_1_TOP_FRONT | 0 | 1 | 2 | 3 | 4 | +5 | 6 | 7 | 2 | 0 | 1 | 4 | 5 | 3 | 6 | 7 +-----------------------+---+---+---+---+---+---+---+---+---+---+---+---+---+---+---+--- +MODE_7_1_REAR_SURROUND | 0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 | 2 | 0 | 1 | 6 | 7 | 4 | +5 | 3 MODE_7_1_FRONT_CENTER | 0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 | 2 | 6 | 7 | 0 | 1 +| 4 | 5 | 3 +--------------------------------------------------------------------------------------- +\endverbatim + +The denoted mapping is important for correct audio channel assignment when using +MPEG or WAV ordering. The incoming audio channels are distributed MPEG like +starting at the front channels and ending at the back channels. The distribution +is used as described in Table concering Channel Config and fix channel elements. +Please see the following example for clarification. + +\verbatim +Example: MODE_1_2_2_1 - WAV-Channelorder 5.1 +------------------------------------------ + Input Channel | Coder Channel +--------------------+--------------------- + 2 (front center) | 0 (SCE channel) + 0 (left center) | 1 (1st of 1st CPE) + 1 (right center) | 2 (2nd of 1st CPE) + 4 (left surround) | 3 (1st of 2nd CPE) + 5 (right surround) | 4 (2nd of 2nd CPE) + 3 (LFE) | 5 (LFE) +------------------------------------------ +\endverbatim + + +\section suppBitrates Supported Bitrates + +The FDK AAC Encoder provides a wide range of supported bitrates. +The minimum and maximum allowed bitrate depends on the Audio Object Type. For +AAC-LC the minimum bitrate is the bitrate that is required to write the most +basic and minimal valid bitstream. It consists of the bitstream format header +information and other static/mandatory information within the AAC payload. The +maximum AAC framesize allowed by the MPEG-4 standard determines the maximum +allowed bitrate for AAC-LC. For HE-AAC and HE-AAC v2 a library internal look-up +table is used. + +A good working point in terms of audio quality, sampling rate and bitrate, is at +1 to 1.5 bits/audio sample for AAC-LC, 0.625 bits/audio sample for dualrate +HE-AAC, 1.125 bits/audio sample for downsampled HE-AAC and 0.5 bits/audio sample +for HE-AAC v2. For example for one channel with a sampling frequency of 48 kHz, +the range from 48 kbit/s to 72 kbit/s achieves reasonable audio quality for +AAC-LC. + +For HE-AAC and HE-AAC v2 the lowest possible audio input sampling frequency is +16 kHz because then the AAC-LC core encoder operates in dual rate mode at its +lowest possible sampling frequency, which is 8 kHz. HE-AAC v2 requires stereo +input audio data. + +Please note that in HE-AAC or HE-AAC v2 mode the encoder supports much higher +bitrates than are appropriate for HE-AAC or HE-AAC v2. For example, at a bitrate +of more than 64 kbit/s for a stereo audio signal at 44.1 kHz it usually makes +sense to use AAC-LC, which will produce better audio quality at that bitrate +than HE-AAC or HE-AAC v2. + +\section reommendedConfig Recommended Sampling Rate and Bitrate Combinations + +The following table provides an overview of recommended encoder configuration +parameters which we determined by virtue of numerous listening tests. + +\subsection reommendedConfigLC AAC-LC, HE-AAC, HE-AACv2 in Dualrate SBR mode. +\verbatim +----------------------------------------------------------------------------------- +Audio Object Type | Bit Rate Range | Supported | Preferred | No. +of | [bit/s] | Sampling Rates | Sampl. | Chan. | +| [kHz] | Rate | | | +| [kHz] | +-------------------+------------------+-----------------------+------------+------- +AAC LC + SBR + PS | 8000 - 11999 | 22.05, 24.00 | 24.00 | 2 +AAC LC + SBR + PS | 12000 - 17999 | 32.00 | 32.00 | 2 +AAC LC + SBR + PS | 18000 - 39999 | 32.00, 44.10, 48.00 | 44.10 | 2 +AAC LC + SBR + PS | 40000 - 64000 | 32.00, 44.10, 48.00 | 48.00 | 2 +-------------------+------------------+-----------------------+------------+------- +AAC LC + SBR | 8000 - 11999 | 22.05, 24.00 | 24.00 | 1 +AAC LC + SBR | 12000 - 17999 | 32.00 | 32.00 | 1 +AAC LC + SBR | 18000 - 39999 | 32.00, 44.10, 48.00 | 44.10 | 1 +AAC LC + SBR | 40000 - 64000 | 32.00, 44.10, 48.00 | 48.00 | 1 +-------------------+------------------+-----------------------+------------+------- +AAC LC + SBR | 16000 - 27999 | 32.00, 44.10, 48.00 | 32.00 | 2 +AAC LC + SBR | 28000 - 63999 | 32.00, 44.10, 48.00 | 44.10 | 2 +AAC LC + SBR | 64000 - 128000 | 32.00, 44.10, 48.00 | 48.00 | 2 +-------------------+------------------+-----------------------+------------+------- +AAC LC + SBR | 64000 - 69999 | 32.00, 44.10, 48.00 | 32.00 | +5, 5.1 AAC LC + SBR | 70000 - 239999 | 32.00, 44.10, 48.00 | 44.10 +| 5, 5.1 AAC LC + SBR | 240000 - 319999 | 32.00, 44.10, 48.00 | +48.00 | 5, 5.1 +-------------------+------------------+-----------------------+------------+------- +AAC LC | 8000 - 15999 | 11.025, 12.00, 16.00 | 12.00 | 1 +AAC LC | 16000 - 23999 | 16.00 | 16.00 | 1 +AAC LC | 24000 - 31999 | 16.00, 22.05, 24.00 | 24.00 | 1 +AAC LC | 32000 - 55999 | 32.00 | 32.00 | 1 +AAC LC | 56000 - 160000 | 32.00, 44.10, 48.00 | 44.10 | 1 +AAC LC | 160001 - 288000 | 48.00 | 48.00 | 1 +-------------------+------------------+-----------------------+------------+------- +AAC LC | 16000 - 23999 | 11.025, 12.00, 16.00 | 12.00 | 2 +AAC LC | 24000 - 31999 | 16.00 | 16.00 | 2 +AAC LC | 32000 - 39999 | 16.00, 22.05, 24.00 | 22.05 | 2 +AAC LC | 40000 - 95999 | 32.00 | 32.00 | 2 +AAC LC | 96000 - 111999 | 32.00, 44.10, 48.00 | 32.00 | 2 +AAC LC | 112000 - 320001 | 32.00, 44.10, 48.00 | 44.10 | 2 +AAC LC | 320002 - 576000 | 48.00 | 48.00 | 2 +-------------------+------------------+-----------------------+------------+------- +AAC LC | 160000 - 239999 | 32.00 | 32.00 | +5, 5.1 AAC LC | 240000 - 279999 | 32.00, 44.10, 48.00 | 32.00 +| 5, 5.1 AAC LC | 280000 - 800000 | 32.00, 44.10, 48.00 | +44.10 | 5, 5.1 +----------------------------------------------------------------------------------- +\endverbatim \n + +\subsection reommendedConfigLD AAC-LD, AAC-ELD, AAC-ELD with SBR in Dualrate SBR +mode. Unlike to HE-AAC configuration the SBR is not covered by ELD audio object +type and needs to be enabled explicitly. Use ::AACENC_SBR_MODE to configure SBR +and its samplingrate ratio with ::AACENC_SBR_RATIO parameter. \verbatim +----------------------------------------------------------------------------------- +Audio Object Type | Bit Rate Range | Supported | Preferred | No. +of | [bit/s] | Sampling Rates | Sampl. | Chan. | +| [kHz] | Rate | | | +| [kHz] | +-------------------+------------------+-----------------------+------------+------- +ELD + SBR | 18000 - 24999 | 32.00 - 44.10 | 32.00 | 1 +ELD + SBR | 25000 - 31999 | 32.00 - 48.00 | 32.00 | 1 +ELD + SBR | 32000 - 64000 | 32.00 - 48.00 | 48.00 | 1 +-------------------+------------------+-----------------------+------------+------- +ELD + SBR | 32000 - 51999 | 32.00 - 48.00 | 44.10 | 2 +ELD + SBR | 52000 - 128000 | 32.00 - 48.00 | 48.00 | 2 +-------------------+------------------+-----------------------+------------+------- +ELD + SBR | 78000 - 160000 | 32.00 - 48.00 | 48.00 | 3 +-------------------+------------------+-----------------------+------------+------- +ELD + SBR | 104000 - 212000 | 32.00 - 48.00 | 48.00 | 4 +-------------------+------------------+-----------------------+------------+------- +ELD + SBR | 130000 - 246000 | 32.00 - 48.00 | 48.00 | +5, 5.1 +-------------------+------------------+-----------------------+------------+------- +LD, ELD | 16000 - 19999 | 16.00 - 24.00 | 16.00 | 1 +LD, ELD | 20000 - 39999 | 16.00 - 32.00 | 24.00 | 1 +LD, ELD | 40000 - 49999 | 22.05 - 32.00 | 32.00 | 1 +LD, ELD | 50000 - 61999 | 24.00 - 44.10 | 32.00 | 1 +LD, ELD | 62000 - 84999 | 32.00 - 48.00 | 44.10 | 1 +LD, ELD | 85000 - 192000 | 44.10 - 48.00 | 48.00 | 1 +-------------------+------------------+-----------------------+------------+------- +LD, ELD | 64000 - 75999 | 24.00 - 32.00 | 32.00 | 2 +LD, ELD | 76000 - 97999 | 24.00 - 44.10 | 32.00 | 2 +LD, ELD | 98000 - 135999 | 32.00 - 48.00 | 44.10 | 2 +LD, ELD | 136000 - 384000 | 44.10 - 48.00 | 48.00 | 2 +-------------------+------------------+-----------------------+------------+------- +LD, ELD | 96000 - 113999 | 24.00 - 32.00 | 32.00 | 3 +LD, ELD | 114000 - 146999 | 24.00 - 44.10 | 32.00 | 3 +LD, ELD | 147000 - 203999 | 32.00 - 48.00 | 44.10 | 3 +LD, ELD | 204000 - 576000 | 44.10 - 48.00 | 48.00 | 3 +-------------------+------------------+-----------------------+------------+------- +LD, ELD | 128000 - 151999 | 24.00 - 32.00 | 32.00 | 4 +LD, ELD | 152000 - 195999 | 24.00 - 44.10 | 32.00 | 4 +LD, ELD | 196000 - 271999 | 32.00 - 48.00 | 44.10 | 4 +LD, ELD | 272000 - 768000 | 44.10 - 48.00 | 48.00 | 4 +-------------------+------------------+-----------------------+------------+------- +LD, ELD | 160000 - 189999 | 24.00 - 32.00 | 32.00 | +5, 5.1 LD, ELD | 190000 - 244999 | 24.00 - 44.10 | 32.00 +| 5, 5.1 LD, ELD | 245000 - 339999 | 32.00 - 48.00 | +44.10 | 5, 5.1 LD, ELD | 340000 - 960000 | 44.10 - 48.00 | +48.00 | 5, 5.1 +----------------------------------------------------------------------------------- +\endverbatim \n + +\subsection reommendedConfigELD AAC-ELD with SBR in Downsampled SBR mode. +\verbatim +----------------------------------------------------------------------------------- +Audio Object Type | Bit Rate Range | Supported | Preferred | No. +of | [bit/s] | Sampling Rates | Sampl. | Chan. | +| [kHz] | Rate | | | +| [kHz] | +-------------------+------------------+-----------------------+------------+------- +ELD + SBR | 18000 - 24999 | 16.00 - 22.05 | 22.05 | 1 +(downsampled SBR) | 25000 - 31999 | 16.00 - 24.00 | 24.00 | 1 + | 32000 - 47999 | 22.05 - 32.00 | 32.00 | 1 + | 48000 - 64000 | 22.05 - 48.00 | 32.00 | 1 +-------------------+------------------+-----------------------+------------+------- +ELD + SBR | 32000 - 51999 | 16.00 - 24.00 | 24.00 | 2 +(downsampled SBR) | 52000 - 59999 | 22.05 - 24.00 | 24.00 | 2 + | 60000 - 95999 | 22.05 - 32.00 | 32.00 | 2 + | 96000 - 128000 | 22.05 - 48.00 | 32.00 | 2 +-------------------+------------------+-----------------------+------------+------- +ELD + SBR | 78000 - 99999 | 22.05 - 24.00 | 24.00 | 3 +(downsampled SBR) | 100000 - 143999 | 22.05 - 32.00 | 32.00 | 3 + | 144000 - 159999 | 22.05 - 48.00 | 32.00 | 3 + | 160000 - 192000 | 32.00 - 48.00 | 32.00 | 3 +-------------------+------------------+-----------------------+------------+------- +ELD + SBR | 104000 - 149999 | 22.05 - 24.00 | 24.00 | 4 +(downsampled SBR) | 150000 - 191999 | 22.05 - 32.00 | 32.00 | 4 + | 192000 - 211999 | 22.05 - 48.00 | 32.00 | 4 + | 212000 - 256000 | 32.00 - 48.00 | 32.00 | 4 +-------------------+------------------+-----------------------+------------+------- +ELD + SBR | 130000 - 171999 | 22.05 - 24.00 | 24.00 | +5, 5.1 (downsampled SBR) | 172000 - 239999 | 22.05 - 32.00 | 32.00 +| 5, 5.1 | 240000 - 320000 | 32.00 - 48.00 | 32.00 | 5, 5.1 +----------------------------------------------------------------------------------- +\endverbatim \n + +\subsection reommendedConfigELDv2 AAC-ELD v2, AAC-ELD v2 with SBR. +The ELD v2 212 configuration must be configured explicitly with +::AACENC_CHANNELMODE parameter according MODE_212 value. SBR can be configured +separately through ::AACENC_SBR_MODE and ::AACENC_SBR_RATIO parameter. Following +configurations shall apply to both framelengths 480 and 512. For ELD v2 +configuration without SBR and framelength 480 the supported sampling rate is +restricted to the range from 16 kHz up to 24 kHz. \verbatim +----------------------------------------------------------------------------------- +Audio Object Type | Bit Rate Range | Supported | Preferred | No. +of | [bit/s] | Sampling Rates | Sampl. | Chan. | +| [kHz] | Rate | | | +| [kHz] | +-------------------+------------------+-----------------------+------------+------- +ELD-212 | 16000 - 19999 | 16.00 - 24.00 | 16.00 | 2 +(without SBR) | 20000 - 39999 | 16.00 - 32.00 | 24.00 | 2 + | 40000 - 49999 | 22.05 - 32.00 | 32.00 | 2 + | 50000 - 61999 | 24.00 - 44.10 | 32.00 | 2 + | 62000 - 84999 | 32.00 - 48.00 | 44.10 | 2 + | 85000 - 192000 | 44.10 - 48.00 | 48.00 | 2 +-------------------+------------------+-----------------------+------------+------- +ELD-212 + SBR | 18000 - 20999 | 32.00 | 32.00 | 2 +(dualrate SBR) | 21000 - 25999 | 32.00 - 44.10 | 32.00 | 2 + | 26000 - 31999 | 32.00 - 48.00 | 44.10 | 2 + | 32000 - 64000 | 32.00 - 48.00 | 48.00 | 2 +-------------------+------------------+-----------------------+------------+------- +ELD-212 + SBR | 18000 - 19999 | 16.00 - 22.05 | 22.05 | 2 +(downsampled SBR) | 20000 - 24999 | 16.00 - 24.00 | 22.05 | 2 + | 25000 - 31999 | 16.00 - 24.00 | 24.00 | 2 + | 32000 - 64000 | 24.00 - 24.00 | 24.00 | 2 +-------------------+------------------+-----------------------+------------+------- +\endverbatim \n + +\page ENCODERBEHAVIOUR Encoder Behaviour + +\section BEHAVIOUR_BANDWIDTH Bandwidth + +The FDK AAC encoder usually does not use the full frequency range of the input +signal, but restricts the bandwidth according to certain library-internal +settings. They can be changed in the table "bandWidthTable" in the file +bandwidth.cpp (if available). + +The encoder API provides the ::AACENC_BANDWIDTH parameter to adjust the +bandwidth explicitly. \code aacEncoder_SetParam(hAacEncoder, AACENC_BANDWIDTH, +value); \endcode + +However it is not recommended to change these settings, because they are based +on numerous listening tests and careful tweaks to ensure the best overall +encoding quality. Also, the maximum bandwidth that can be set manually by the +user is 20kHz or fs/2, whichever value is smaller. + +Theoretically a signal of for example 48 kHz can contain frequencies up to 24 +kHz, but to use this full range in an audio encoder usually does not make sense. +Usually the encoder has a very limited amount of bits to spend (typically 128 +kbit/s for stereo 48 kHz content) and to allow full range bandwidth would waste +a lot of these bits for frequencies the human ear is hardly able to perceive +anyway, if at all. Hence it is wise to use the available bits for the really +important frequency range and just skip the rest. At lower bitrates (e. g. <= 80 +kbit/s for stereo 48 kHz content) the encoder will choose an even smaller +bandwidth, because an encoded signal with smaller bandwidth and hence less +artifacts sounds better than a signal with higher bandwidth but then more coding +artefacts across all frequencies. These artefacts would occur if small bitrates +and high bandwidths are chosen because the available bits are just not enough to +encode all frequencies well. + +Unfortunately some people evaluate encoding quality based on possible bandwidth +as well, but it is a double-edged sword considering the trade-off described +above. + +Another aspect is workload consumption. The higher the allowed bandwidth, the +more frequency lines have to be processed, which in turn increases the workload. + +\section FRAMESIZES_AND_BIT_RESERVOIR Frame Sizes & Bit Reservoir + +For AAC there is a difference between constant bit rate and constant frame +length due to the so-called bit reservoir technique, which allows the encoder to +use less bits in an AAC frame for those audio signal sections which are easy to +encode, and then spend them at a later point in time for more complex audio +sections. The extent to which this "bit exchange" is done is limited to allow +for reliable and relatively low delay real time streaming. Therefore, for +AAC-ELD, the bitreservoir is limited. It varies between 500 and 4000 bits/frame, +depending on the bitrate/channel. +- For a bitrate of 12kbps/channel and below, the AAC-ELD bitreservoir is 500 +bits/frame. +- For a bitrate of 70kbps/channel and above, the AAC-ELD bitreservoir is 4000 +bits/frame. +- Between 12kbps/channel and 70kbps/channel, the AAC-ELD bitrervoir is increased +linearly. +- For AAC-LC, the bitrate is only limited by the maximum AAC frame length. It +is, regardless of the available bit reservoir, defined as 6144 bits per channel. + +Over a longer period in time the bitrate will be constant in the AAC constant +bitrate mode, e.g. for ISDN transmission. This means that in AAC each bitstream +frame will in general have a different length in bytes but over time it +will reach the target bitrate. + + +One could also make an MPEG compliant +AAC encoder which always produces constant length packages for each AAC frame, +but the audio quality would be considerably worse since the bit reservoir +technique would have to be switched off completely. A higher bit rate would have +to be used to get the same audio quality as with an enabled bit reservoir. + +For mp3 by the way, the same bit reservoir technique exists, but there each bit +stream frame has a constant length for a given bit rate (ignoring the +padding byte). In mp3 there is a so-called "back pointer" which tells +the decoder which bits belong to the current mp3 frame - and in general some or +many bits have been transmitted in an earlier mp3 frame. Basically this leads to +the same "bit exchange between mp3 frames" as in AAC but with virtually constant +length frames. + +This variable frame length at "constant bit rate" is not something special +in this Fraunhofer IIS AAC encoder. AAC has been designed in that way. + +\subsection BEHAVIOUR_ESTIM_AVG_FRAMESIZES Estimating Average Frame Sizes + +A HE-AAC v1 or v2 audio frame contains 2048 PCM samples per channel (there is +also one mode with 1920 samples per channel but this is only for special +purposes such as DAB+ digital radio). + +The number of HE-AAC frames \f$N\_FRAMES\f$ per second at 44.1 kHz is: + +\f[ +N\_FRAMES = 44100 / 2048 = 21.5332 +\f] + +At a bit rate of 8 kbps the average number of bits per frame +\f$N\_BITS\_PER\_FRAME\f$ is: + +\f[ +N\_BITS\_PER\_FRAME = 8000 / 21.5332 = 371.52 +\f] + +which is about 46.44 bytes per encoded frame. + +At a bit rate of 32 kbps, which is quite high for single channel HE-AAC v1, it +is: + +\f[ +N\_BITS\_PER\_FRAME = 32000 / 21.5332 = 1486 +\f] + +which is about 185.76 bytes per encoded frame. + +These bits/frame figures are average figures where each AAC frame generally has +a different size in bytes. To calculate the same for AAC-LC just use 1024 +instead of 2048 PCM samples per frame and channel. For AAC-LD/ELD it is either +480 or 512 PCM samples per frame and channel. + + +\section BEHAVIOUR_TOOLS Encoder Tools + +The AAC encoder supports TNS, PNS, MS, Intensity and activates these tools +depending on the audio signal and the encoder configuration (i.e. bitrate or +AOT). It is not required to configure these tools manually. + +PNS improves encoding quality only for certain bitrates. Therefore it makes +sense to activate PNS only for these bitrates and save the processing power +required for PNS (about 10 % of the encoder) when using other bitrates. This is +done automatically inside the encoder library. PNS is disabled inside the +encoder library if an MPEG-2 AOT is choosen since PNS is an MPEG-4 AAC feature. + +If SBR is activated, the encoder automatically deactivates PNS internally. If +TNS is disabled but PNS is allowed, the encoder deactivates PNS calculation +internally. + +*/ + +#ifndef AACENC_LIB_H +#define AACENC_LIB_H + +#include "machine_type.h" +#include "FDK_audio.h" + +#define AACENCODER_LIB_VL0 4 +#define AACENCODER_LIB_VL1 0 +#define AACENCODER_LIB_VL2 0 + +/** + * AAC encoder error codes. + */ +typedef enum { + AACENC_OK = 0x0000, /*!< No error happened. All fine. */ + + AACENC_INVALID_HANDLE = + 0x0020, /*!< Handle passed to function call was invalid. */ + AACENC_MEMORY_ERROR = 0x0021, /*!< Memory allocation failed. */ + AACENC_UNSUPPORTED_PARAMETER = 0x0022, /*!< Parameter not available. */ + AACENC_INVALID_CONFIG = 0x0023, /*!< Configuration not provided. */ + + AACENC_INIT_ERROR = 0x0040, /*!< General initialization error. */ + AACENC_INIT_AAC_ERROR = 0x0041, /*!< AAC library initialization error. */ + AACENC_INIT_SBR_ERROR = 0x0042, /*!< SBR library initialization error. */ + AACENC_INIT_TP_ERROR = 0x0043, /*!< Transport library initialization error. */ + AACENC_INIT_META_ERROR = + 0x0044, /*!< Meta data library initialization error. */ + AACENC_INIT_MPS_ERROR = 0x0045, /*!< MPS library initialization error. */ + + AACENC_ENCODE_ERROR = 0x0060, /*!< The encoding process was interrupted by an + unexpected error. */ + + AACENC_ENCODE_EOF = 0x0080 /*!< End of file reached. */ + +} AACENC_ERROR; + +/** + * AAC encoder buffer descriptors identifier. + * This identifier are used within buffer descriptors + * AACENC_BufDesc::bufferIdentifiers. + */ +typedef enum { + /* Input buffer identifier. */ + IN_AUDIO_DATA = 0, /*!< Audio input buffer, interleaved INT_PCM samples. */ + IN_ANCILLRY_DATA = 1, /*!< Ancillary data to be embedded into bitstream. */ + IN_METADATA_SETUP = 2, /*!< Setup structure for embedding meta data. */ + + /* Output buffer identifier. */ + OUT_BITSTREAM_DATA = 3, /*!< Buffer holds bitstream output data. */ + OUT_AU_SIZES = + 4 /*!< Buffer contains sizes of each access unit. This information + is necessary for superframing. */ + +} AACENC_BufferIdentifier; + +/** + * AAC encoder handle. + */ +typedef struct AACENCODER *HANDLE_AACENCODER; + +/** + * Provides some info about the encoder configuration. + */ +typedef struct { + UINT maxOutBufBytes; /*!< Maximum number of encoder bitstream bytes within one + frame. Size depends on maximum number of supported + channels in encoder instance. For superframing (as + used for example in DAB+), size has to be a multiple + accordingly. */ + + UINT maxAncBytes; /*!< Maximum number of ancillary data bytes which can be + inserted into bitstream within one frame. */ + + UINT inBufFillLevel; /*!< Internal input buffer fill level in samples per + channel. This parameter will automatically be cleared + if samplingrate or channel(Mode/Order) changes. */ + + UINT inputChannels; /*!< Number of input channels expected in encoding + process. */ + + UINT frameLength; /*!< Amount of input audio samples consumed each frame per + channel, depending on audio object type configuration. */ + + UINT nDelay; /*!< Codec delay in PCM samples/channel. Depends on framelength + and AOT. Does not include framing delay for filling up encoder + PCM input buffer. */ + + UINT nDelayCore; /*!< Codec delay in PCM samples/channel, w/o delay caused by + the decoder SBR module. This delay is needed to correctly + write edit lists for gapless playback. The decoder may not + know how much delay is introdcued by SBR, since it may not + know if SBR is active at all (implicit signaling), + therefore the deocder must take into account any delay + caused by the SBR module. */ + + UCHAR confBuf[64]; /*!< Configuration buffer in binary format as an + AudioSpecificConfig or StreamMuxConfig according to the + selected transport type. */ + + UINT confSize; /*!< Number of valid bytes in confBuf. */ + +} AACENC_InfoStruct; + +/** + * Describes the input and output buffers for an aacEncEncode() call. + */ +typedef struct { + INT numBufs; /*!< Number of buffers. */ + void **bufs; /*!< Pointer to vector containing buffer addresses. */ + INT *bufferIdentifiers; /*!< Identifier of each buffer element. See + ::AACENC_BufferIdentifier. */ + INT *bufSizes; /*!< Size of each buffer in 8-bit bytes. */ + INT *bufElSizes; /*!< Size of each buffer element in bytes. */ + +} AACENC_BufDesc; + +/** + * Defines the input arguments for an aacEncEncode() call. + */ +typedef struct { + INT numInSamples; /*!< Number of valid input audio samples (multiple of input + channels). */ + INT numAncBytes; /*!< Number of ancillary data bytes to be encoded. */ + +} AACENC_InArgs; + +/** + * Defines the output arguments for an aacEncEncode() call. + */ +typedef struct { + INT numOutBytes; /*!< Number of valid bitstream bytes generated during + aacEncEncode(). */ + INT numInSamples; /*!< Number of input audio samples consumed by the encoder. + */ + INT numAncBytes; /*!< Number of ancillary data bytes consumed by the encoder. + */ + INT bitResState; /*!< State of the bit reservoir in bits. */ + +} AACENC_OutArgs; + +/** + * Meta Data Compression Profiles. + */ +typedef enum { + AACENC_METADATA_DRC_NONE = 0, /*!< None. */ + AACENC_METADATA_DRC_FILMSTANDARD = 1, /*!< Film standard. */ + AACENC_METADATA_DRC_FILMLIGHT = 2, /*!< Film light. */ + AACENC_METADATA_DRC_MUSICSTANDARD = 3, /*!< Music standard. */ + AACENC_METADATA_DRC_MUSICLIGHT = 4, /*!< Music light. */ + AACENC_METADATA_DRC_SPEECH = 5, /*!< Speech. */ + AACENC_METADATA_DRC_NOT_PRESENT = + 256 /*!< Disable writing gain factor (used for comp_profile only). */ + +} AACENC_METADATA_DRC_PROFILE; + +/** + * Meta Data setup structure. + */ +typedef struct { + AACENC_METADATA_DRC_PROFILE + drc_profile; /*!< MPEG DRC compression profile. See + ::AACENC_METADATA_DRC_PROFILE. */ + AACENC_METADATA_DRC_PROFILE + comp_profile; /*!< ETSI heavy compression profile. See + ::AACENC_METADATA_DRC_PROFILE. */ + + INT drc_TargetRefLevel; /*!< Used to define expected level to: + Scaled with 16 bit. x*2^16. */ + INT comp_TargetRefLevel; /*!< Adjust limiter to avoid overload. + Scaled with 16 bit. x*2^16. */ + + INT prog_ref_level_present; /*!< Flag, if prog_ref_level is present */ + INT prog_ref_level; /*!< Programme Reference Level = Dialogue Level: + -31.75dB .. 0 dB ; stepsize: 0.25dB + Scaled with 16 bit. x*2^16.*/ + + UCHAR PCE_mixdown_idx_present; /*!< Flag, if dmx-idx should be written in + programme config element */ + UCHAR ETSI_DmxLvl_present; /*!< Flag, if dmx-lvl should be written in + ETSI-ancData */ + + SCHAR centerMixLevel; /*!< Center downmix level (0...7, according to table) */ + SCHAR surroundMixLevel; /*!< Surround downmix level (0...7, according to + table) */ + + UCHAR + dolbySurroundMode; /*!< Indication for Dolby Surround Encoding Mode. + - 0: Dolby Surround mode not indicated + - 1: 2-ch audio part is not Dolby surround encoded + - 2: 2-ch audio part is Dolby surround encoded */ + + UCHAR drcPresentationMode; /*!< Indicatin for DRC Presentation Mode. + - 0: Presentation mode not inticated + - 1: Presentation mode 1 + - 2: Presentation mode 2 */ + + struct { + /* extended ancillary data */ + UCHAR extAncDataEnable; /*< Indicates if MPEG4_ext_ancillary_data() exists. + - 0: No MPEG4_ext_ancillary_data(). + - 1: Insert MPEG4_ext_ancillary_data(). */ + + UCHAR + extDownmixLevelEnable; /*< Indicates if ext_downmixing_levels() exists. + - 0: No ext_downmixing_levels(). + - 1: Insert ext_downmixing_levels(). */ + UCHAR extDownmixLevel_A; /*< Downmix level index A (0...7, according to + table) */ + UCHAR extDownmixLevel_B; /*< Downmix level index B (0...7, according to + table) */ + + UCHAR dmxGainEnable; /*< Indicates if ext_downmixing_global_gains() exists. + - 0: No ext_downmixing_global_gains(). + - 1: Insert ext_downmixing_global_gains(). */ + INT dmxGain5; /*< Gain factor for downmix to 5 channels. + -15.75dB .. -15.75dB; stepsize: 0.25dB + Scaled with 16 bit. x*2^16.*/ + INT dmxGain2; /*< Gain factor for downmix to 2 channels. + -15.75dB .. -15.75dB; stepsize: 0.25dB + Scaled with 16 bit. x*2^16.*/ + + UCHAR lfeDmxEnable; /*< Indicates if ext_downmixing_lfe_level() exists. + - 0: No ext_downmixing_lfe_level(). + - 1: Insert ext_downmixing_lfe_level(). */ + UCHAR lfeDmxLevel; /*< Downmix level index for LFE (0..15, according to + table) */ + + } ExtMetaData; + +} AACENC_MetaData; + +/** + * AAC encoder control flags. + * + * In interaction with the ::AACENC_CONTROL_STATE parameter it is possible to + * get information about the internal initialization process. It is also + * possible to overwrite the internal state from extern when necessary. + */ +typedef enum { + AACENC_INIT_NONE = 0x0000, /*!< Do not trigger initialization. */ + AACENC_INIT_CONFIG = + 0x0001, /*!< Initialize all encoder modules configuration. */ + AACENC_INIT_STATES = 0x0002, /*!< Reset all encoder modules history buffer. */ + AACENC_INIT_TRANSPORT = + 0x1000, /*!< Initialize transport lib with new parameters. */ + AACENC_RESET_INBUFFER = + 0x2000, /*!< Reset fill level of internal input buffer. */ + AACENC_INIT_ALL = 0xFFFF /*!< Initialize all. */ +} AACENC_CTRLFLAGS; + +/** + * \brief AAC encoder setting parameters. + * + * Use aacEncoder_SetParam() function to configure, or use aacEncoder_GetParam() + * function to read the internal status of the following parameters. + */ +typedef enum { + AACENC_AOT = + 0x0100, /*!< Audio object type. See ::AUDIO_OBJECT_TYPE in FDK_audio.h. + - 2: MPEG-4 AAC Low Complexity. + - 5: MPEG-4 AAC Low Complexity with Spectral Band Replication + (HE-AAC). + - 29: MPEG-4 AAC Low Complexity with Spectral Band + Replication and Parametric Stereo (HE-AAC v2). This + configuration can be used only with stereo input audio data. + - 23: MPEG-4 AAC Low-Delay. + - 39: MPEG-4 AAC Enhanced Low-Delay. Since there is no + ::AUDIO_OBJECT_TYPE for ELD in combination with SBR defined, + enable SBR explicitely by ::AACENC_SBR_MODE parameter. The ELD + v2 212 configuration can be configured by ::AACENC_CHANNELMODE + parameter. + - 129: MPEG-2 AAC Low Complexity. + - 132: MPEG-2 AAC Low Complexity with Spectral Band + Replication (HE-AAC). + + Please note that the virtual MPEG-2 AOT's basically disables + non-existing Perceptual Noise Substitution tool in AAC encoder + and controls the MPEG_ID flag in adts header. The virtual + MPEG-2 AOT doesn't prohibit specific transport formats. */ + + AACENC_BITRATE = 0x0101, /*!< Total encoder bitrate. This parameter is + mandatory and interacts with ::AACENC_BITRATEMODE. + - CBR: Bitrate in bits/second. + - VBR: Variable bitrate. Bitrate argument will + be ignored. See \ref suppBitrates for details. */ + + AACENC_BITRATEMODE = 0x0102, /*!< Bitrate mode. Configuration can be different + kind of bitrate configurations: + - 0: Constant bitrate, use bitrate according + to ::AACENC_BITRATE. (default) Within none + LD/ELD ::AUDIO_OBJECT_TYPE, the CBR mode makes + use of full allowed bitreservoir. In contrast, + at Low-Delay ::AUDIO_OBJECT_TYPE the + bitreservoir is kept very small. + - 1: Variable bitrate mode, \ref vbrmode + "very low bitrate". + - 2: Variable bitrate mode, \ref vbrmode + "low bitrate". + - 3: Variable bitrate mode, \ref vbrmode + "medium bitrate". + - 4: Variable bitrate mode, \ref vbrmode + "high bitrate". + - 5: Variable bitrate mode, \ref vbrmode + "very high bitrate". */ + + AACENC_SAMPLERATE = 0x0103, /*!< Audio input data sampling rate. Encoder + supports following sampling rates: 8000, 11025, + 12000, 16000, 22050, 24000, 32000, 44100, + 48000, 64000, 88200, 96000 */ + + AACENC_SBR_MODE = 0x0104, /*!< Configure SBR independently of the chosen Audio + Object Type ::AUDIO_OBJECT_TYPE. This parameter + is for ELD audio object type only. + - -1: Use ELD SBR auto configurator (default). + - 0: Disable Spectral Band Replication. + - 1: Enable Spectral Band Replication. */ + + AACENC_GRANULE_LENGTH = + 0x0105, /*!< Core encoder (AAC) audio frame length in samples: + - 1024: Default configuration. + - 960: DRM/DAB+. + - 512: Default length in LD/ELD configuration. + - 480: Length in LD/ELD configuration. + - 256: Length for ELD reduced delay mode (x2). + - 240: Length for ELD reduced delay mode (x2). + - 128: Length for ELD reduced delay mode (x4). + - 120: Length for ELD reduced delay mode (x4). */ + + AACENC_CHANNELMODE = 0x0106, /*!< Set explicit channel mode. Channel mode must + match with number of input channels. + - 1-7, 11,12,14 and 33,34: MPEG channel + modes supported, see ::CHANNEL_MODE in + FDK_audio.h. */ + + AACENC_CHANNELORDER = + 0x0107, /*!< Input audio data channel ordering scheme: + - 0: MPEG channel ordering (e. g. 5.1: C, L, R, SL, SR, LFE). + (default) + - 1: WAVE file format channel ordering (e. g. 5.1: L, R, C, + LFE, SL, SR). */ + + AACENC_SBR_RATIO = + 0x0108, /*!< Controls activation of downsampled SBR. With downsampled + SBR, the delay will be shorter. On the other hand, for + achieving the same quality level, downsampled SBR needs more + bits than dual-rate SBR. With downsampled SBR, the AAC encoder + will work at the same sampling rate as the SBR encoder (single + rate). Downsampled SBR is supported for AAC-ELD and HE-AACv1. + - 1: Downsampled SBR (default for ELD). + - 2: Dual-rate SBR (default for HE-AAC). */ + + AACENC_AFTERBURNER = + 0x0200, /*!< This parameter controls the use of the afterburner feature. + The afterburner is a type of analysis by synthesis algorithm + which increases the audio quality but also the required + processing power. It is recommended to always activate this if + additional memory consumption and processing power consumption + is not a problem. If increased MHz and memory consumption are + an issue then the MHz and memory cost of this optional module + need to be evaluated against the improvement in audio quality + on a case by case basis. + - 0: Disable afterburner (default). + - 1: Enable afterburner. */ + + AACENC_BANDWIDTH = 0x0203, /*!< Core encoder audio bandwidth: + - 0: Determine audio bandwidth internally + (default, see chapter \ref BEHAVIOUR_BANDWIDTH). + - 1 to fs/2: Audio bandwidth in Hertz. Limited + to 20kHz max. Not usable if SBR is active. This + setting is for experts only, better do not touch + this value to avoid degraded audio quality. */ + + AACENC_PEAK_BITRATE = + 0x0207, /*!< Peak bitrate configuration parameter to adjust maximum bits + per audio frame. Bitrate is in bits/second. The peak bitrate + will internally be limited to the chosen bitrate + ::AACENC_BITRATE as lower limit and the + number_of_effective_channels*6144 bit as upper limit. + + Setting the peak bitrate equal to ::AACENC_BITRATE does not + necessarily mean that the audio frames will be of constant + size. Since the peak bitate is in bits/second, the frame sizes + can vary by one byte in one or the other direction over various + frames. However, it is not recommended to reduce the peak + pitrate to ::AACENC_BITRATE - it would disable the + bitreservoir, which would affect the audio quality by a large + amount. */ + + AACENC_TRANSMUX = 0x0300, /*!< Transport type to be used. See ::TRANSPORT_TYPE + in FDK_audio.h. Following types can be configured + in encoder library: + - 0: raw access units + - 1: ADIF bitstream format + - 2: ADTS bitstream format + - 6: Audio Mux Elements (LATM) with + muxConfigPresent = 1 + - 7: Audio Mux Elements (LATM) with + muxConfigPresent = 0, out of band StreamMuxConfig + - 10: Audio Sync Stream (LOAS) */ + + AACENC_HEADER_PERIOD = + 0x0301, /*!< Frame count period for sending in-band configuration buffers + within LATM/LOAS transport layer. Additionally this parameter + configures the PCE repetition period in raw_data_block(). See + \ref encPCE. + - 0xFF: auto-mode default 10 for TT_MP4_ADTS, TT_MP4_LOAS and + TT_MP4_LATM_MCP1, otherwise 0. + - n: Frame count period. */ + + AACENC_SIGNALING_MODE = + 0x0302, /*!< Signaling mode of the extension AOT: + - 0: Implicit backward compatible signaling (default for + non-MPEG-4 based AOT's and for the transport formats ADIF and + ADTS) + - A stream that uses implicit signaling can be decoded + by every AAC decoder, even AAC-LC-only decoders + - An AAC-LC-only decoder will only decode the + low-frequency part of the stream, resulting in a band-limited + output + - This method works with all transport formats + - This method does not work with downsampled SBR + - 1: Explicit backward compatible signaling + - A stream that uses explicit backward compatible + signaling can be decoded by every AAC decoder, even AAC-LC-only + decoders + - An AAC-LC-only decoder will only decode the + low-frequency part of the stream, resulting in a band-limited + output + - A decoder not capable of decoding PS will only decode + the AAC-LC+SBR part. If the stream contained PS, the result + will be a a decoded mono downmix + - This method does not work with ADIF or ADTS. For + LOAS/LATM, it only works with AudioMuxVersion==1 + - This method does work with downsampled SBR + - 2: Explicit hierarchical signaling (default for MPEG-4 + based AOT's and for all transport formats excluding ADIF and + ADTS) + - A stream that uses explicit hierarchical signaling can + be decoded only by HE-AAC decoders + - An AAC-LC-only decoder will not decode a stream that + uses explicit hierarchical signaling + - A decoder not capable of decoding PS will not decode + the stream at all if it contained PS + - This method does not work with ADIF or ADTS. It works + with LOAS/LATM and the MPEG-4 File format + - This method does work with downsampled SBR + + For making sure that the listener always experiences the + best audio quality, explicit hierarchical signaling should be + used. This makes sure that only a full HE-AAC-capable decoder + will decode those streams. The audio is played at full + bandwidth. For best backwards compatibility, it is recommended + to encode with implicit SBR signaling. A decoder capable of + AAC-LC only will then only decode the AAC part, which means the + decoded audio will sound band-limited. + + For MPEG-2 transport types (ADTS,ADIF), only implicit + signaling is possible. + + For LOAS and LATM, explicit backwards compatible signaling + only works together with AudioMuxVersion==1. The reason is + that, for explicit backwards compatible signaling, additional + information will be appended to the ASC. A decoder that is only + capable of decoding AAC-LC will skip this part. Nevertheless, + for jumping to the end of the ASC, it needs to know the ASC + length. Transmitting the length of the ASC is a feature of + AudioMuxVersion==1, it is not possible to transmit the length + of the ASC with AudioMuxVersion==0, therefore an AAC-LC-only + decoder will not be able to parse a LOAS/LATM stream that was + being encoded with AudioMuxVersion==0. + + For downsampled SBR, explicit signaling is mandatory. The + reason for this is that the extension sampling frequency (which + is in case of SBR the sampling frequqncy of the SBR part) can + only be signaled in explicit mode. + + For AAC-ELD, the SBR information is transmitted in the + ELDSpecific Config, which is part of the AudioSpecificConfig. + Therefore, the settings here will have no effect on AAC-ELD.*/ + + AACENC_TPSUBFRAMES = + 0x0303, /*!< Number of sub frames in a transport frame for LOAS/LATM or + ADTS (default 1). + - ADTS: Maximum number of sub frames restricted to 4. + - DAB+: Maximum number of sub frames restricted to 6. + - LOAS/LATM: Maximum number of sub frames restricted to 2.*/ + + AACENC_AUDIOMUXVER = + 0x0304, /*!< AudioMuxVersion to be used for LATM. (AudioMuxVersionA, + currently not implemented): + - 0: Default, no transmission of tara Buffer fullness, no ASC + length and including actual latm Buffer fullnes. + - 1: Transmission of tara Buffer fullness, ASC length and + actual latm Buffer fullness. + - 2: Transmission of tara Buffer fullness, ASC length and + maximum level of latm Buffer fullness. */ + + AACENC_PROTECTION = 0x0306, /*!< Configure protection in transport layer: + - 0: No protection. (default) + - 1: CRC active for ADTS transport format. */ + + AACENC_ANCILLARY_BITRATE = + 0x0500, /*!< Constant ancillary data bitrate in bits/second. + - 0: Either no ancillary data or insert exact number of + bytes, denoted via input parameter, numAncBytes in + AACENC_InArgs. + - else: Insert ancillary data with specified bitrate. */ + + AACENC_METADATA_MODE = 0x0600, /*!< Configure Meta Data. See ::AACENC_MetaData + for further details: + - 0: Do not embed any metadata. + - 1: Embed dynamic_range_info metadata. + - 2: Embed dynamic_range_info and + ancillary_data metadata. + - 3: Embed ancillary_data metadata. */ + + AACENC_CONTROL_STATE = + 0xFF00, /*!< There is an automatic process which internally reconfigures + the encoder instance when a configuration parameter changed or + an error occured. This paramerter allows overwriting or getting + the control status of this process. See ::AACENC_CTRLFLAGS. */ + + AACENC_NONE = 0xFFFF /*!< ------ */ + +} AACENC_PARAM; + +#ifdef __cplusplus +extern "C" { +#endif + +/** + * \brief Open an instance of the encoder. + * + * Allocate memory for an encoder instance with a functional range denoted by + * the function parameters. Preinitialize encoder instance with default + * configuration. + * + * \param phAacEncoder A pointer to an encoder handle. Initialized on return. + * \param encModules Specify encoder modules to be supported in this encoder + * instance: + * - 0x0: Allocate memory for all available encoder + * modules. + * - else: Select memory allocation regarding encoder + * modules. Following flags are possible and can be combined. + * - 0x01: AAC module. + * - 0x02: SBR module. + * - 0x04: PS module. + * - 0x08: MPS module. + * - 0x10: Metadata module. + * - example: (0x01|0x02|0x04|0x08|0x10) allocates + * all modules and is equivalent to default configuration denotet by 0x0. + * \param maxChannels Number of channels to be allocated. This parameter can + * be used in different ways: + * - 0: Allocate maximum number of AAC and SBR channels as + * supported by the library. + * - nChannels: Use same maximum number of channels for + * allocating memory in AAC and SBR module. + * - nChannels | (nSbrCh<<8): Number of SBR channels can be + * different to AAC channels to save data memory. + * + * \return + * - AACENC_OK, on succes. + * - AACENC_INVALID_HANDLE, AACENC_MEMORY_ERROR, AACENC_INVALID_CONFIG, + * on failure. + */ +AACENC_ERROR aacEncOpen(HANDLE_AACENCODER *phAacEncoder, const UINT encModules, + const UINT maxChannels); + +/** + * \brief Close the encoder instance. + * + * Deallocate encoder instance and free whole memory. + * + * \param phAacEncoder Pointer to the encoder handle to be deallocated. + * + * \return + * - AACENC_OK, on success. + * - AACENC_INVALID_HANDLE, on failure. + */ +AACENC_ERROR aacEncClose(HANDLE_AACENCODER *phAacEncoder); + +/** + * \brief Encode audio data. + * + * This function is mainly for encoding audio data. In addition the function can + * be used for an encoder (re)configuration process. + * - PCM input data will be retrieved from external input buffer until the fill + * level allows encoding a single frame. This functionality allows an external + * buffer with reduced size in comparison to the AAC or HE-AAC audio frame + * length. + * - If the value of the input samples argument is zero, just internal + * reinitialization will be applied if it is requested. + * - At the end of a file the flushing process can be triggerd via setting the + * value of the input samples argument to -1. The encoder delay lines are fully + * flushed when the encoder returns no valid bitstream data + * AACENC_OutArgs::numOutBytes. Furthermore the end of file is signaled by the + * return value AACENC_ENCODE_EOF. + * - If an error occured in the previous frame or any of the encoder parameters + * changed, an internal reinitialization process will be applied before encoding + * the incoming audio samples. + * - The function can also be used for an independent reconfiguration process + * without encoding. The first parameter has to be a valid encoder handle and + * all other parameters can be set to NULL. + * - If the size of the external bitbuffer in outBufDesc is not sufficient for + * writing the whole bitstream, an internal error will be the return value and a + * reconfiguration will be triggered. + * + * \param hAacEncoder A valid AAC encoder handle. + * \param inBufDesc Input buffer descriptor, see AACENC_BufDesc: + * - At least one input buffer with audio data is + * expected. + * - Optionally a second input buffer with + * ancillary data can be fed. + * \param outBufDesc Output buffer descriptor, see AACENC_BufDesc: + * - Provide one output buffer for the encoded + * bitstream. + * \param inargs Input arguments, see AACENC_InArgs. + * \param outargs Output arguments, AACENC_OutArgs. + * + * \return + * - AACENC_OK, on success. + * - AACENC_INVALID_HANDLE, AACENC_ENCODE_ERROR, on failure in encoding + * process. + * - AACENC_INVALID_CONFIG, AACENC_INIT_ERROR, AACENC_INIT_AAC_ERROR, + * AACENC_INIT_SBR_ERROR, AACENC_INIT_TP_ERROR, AACENC_INIT_META_ERROR, + * AACENC_INIT_MPS_ERROR, on failure in encoder initialization. + * - AACENC_UNSUPPORTED_PARAMETER, on incorrect input or output buffer + * descriptor initialization. + * - AACENC_ENCODE_EOF, when flushing fully concluded. + */ +AACENC_ERROR aacEncEncode(const HANDLE_AACENCODER hAacEncoder, + const AACENC_BufDesc *inBufDesc, + const AACENC_BufDesc *outBufDesc, + const AACENC_InArgs *inargs, AACENC_OutArgs *outargs); + +/** + * \brief Acquire info about present encoder instance. + * + * This function retrieves information of the encoder configuration. In addition + * to informative internal states, a configuration data block of the current + * encoder settings will be returned. The format is either Audio Specific Config + * in case of Raw Packets transport format or StreamMuxConfig in case of + * LOAS/LATM transport format. The configuration data block is binary coded as + * specified in ISO/IEC 14496-3 (MPEG-4 audio), to be used directly for MPEG-4 + * File Format or RFC3016 or RFC3640 applications. + * + * \param hAacEncoder A valid AAC encoder handle. + * \param pInfo Pointer to AACENC_InfoStruct. Filled on return. + * + * \return + * - AACENC_OK, on succes. + * - AACENC_INIT_ERROR, on failure. + */ +AACENC_ERROR aacEncInfo(const HANDLE_AACENCODER hAacEncoder, + AACENC_InfoStruct *pInfo); + +/** + * \brief Set one single AAC encoder parameter. + * + * This function allows configuration of all encoder parameters specified in + * ::AACENC_PARAM. Each parameter must be set with a separate function call. An + * internal validation of the configuration value range will be done and an + * internal reconfiguration will be signaled. The actual configuration adoption + * is part of the subsequent aacEncEncode() call. + * + * \param hAacEncoder A valid AAC encoder handle. + * \param param Parameter to be set. See ::AACENC_PARAM. + * \param value Parameter value. See parameter description in + * ::AACENC_PARAM. + * + * \return + * - AACENC_OK, on success. + * - AACENC_INVALID_HANDLE, AACENC_UNSUPPORTED_PARAMETER, + * AACENC_INVALID_CONFIG, on failure. + */ +AACENC_ERROR aacEncoder_SetParam(const HANDLE_AACENCODER hAacEncoder, + const AACENC_PARAM param, const UINT value); + +/** + * \brief Get one single AAC encoder parameter. + * + * This function is the complement to aacEncoder_SetParam(). After encoder + * reinitialization with user defined settings, the internal status can be + * obtained of each parameter, specified with ::AACENC_PARAM. + * + * \param hAacEncoder A valid AAC encoder handle. + * \param param Parameter to be returned. See ::AACENC_PARAM. + * + * \return Internal configuration value of specifed parameter ::AACENC_PARAM. + */ +UINT aacEncoder_GetParam(const HANDLE_AACENCODER hAacEncoder, + const AACENC_PARAM param); + +/** + * \brief Get information about encoder library build. + * + * Fill a given LIB_INFO structure with library version information. + * + * \param info Pointer to an allocated LIB_INFO struct. + * + * \return + * - AACENC_OK, on success. + * - AACENC_INVALID_HANDLE, AACENC_INIT_ERROR, on failure. + */ +AACENC_ERROR aacEncGetLibInfo(LIB_INFO *info); + +#ifdef __cplusplus +} +#endif + +#endif /* AACENC_LIB_H */ diff --git a/fdk-aac/libAACenc/src/aacEnc_ram.cpp b/fdk-aac/libAACenc/src/aacEnc_ram.cpp new file mode 100644 index 0000000..77b1131 --- /dev/null +++ b/fdk-aac/libAACenc/src/aacEnc_ram.cpp @@ -0,0 +1,208 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M. Lohwasser, M. Gayer + + Description: + +*******************************************************************************/ + +/*! + \file + \brief Memory layout + \author Markus Lohwasser +*/ + +#include "aacEnc_ram.h" + +C_AALLOC_MEM(AACdynamic_RAM, FIXP_DBL, AAC_ENC_DYN_RAM_SIZE / sizeof(FIXP_DBL)) + +/* + Static memory areas, must not be overwritten in other sections of the decoder + ! +*/ + +/* + The structure AacEncoder contains all Encoder structures. +*/ + +C_ALLOC_MEM(Ram_aacEnc_AacEncoder, struct AAC_ENC, 1) + +/* + The structure PSY_INTERNAl contains all psych configuration and data pointer. + * PsyStatic holds last and current Psych data. + * PsyInputBuffer contains time input. Signal is needed at the beginning of + Psych. Memory can be reused after signal is in time domain. + * PsyData contains spectral, nrg and threshold information. Necessary data + are copied into PsyOut, so memory is available after leaving psych. + * TnsData, ChaosMeasure, PnsData are temporarily necessary, e.g. use memory + from PsyInputBuffer. +*/ + +C_ALLOC_MEM2(Ram_aacEnc_PsyElement, PSY_ELEMENT, 1, ((8))) + +C_ALLOC_MEM(Ram_aacEnc_PsyInternal, PSY_INTERNAL, 1) +C_ALLOC_MEM2(Ram_aacEnc_PsyStatic, PSY_STATIC, 1, (8)) + +C_ALLOC_MEM2(Ram_aacEnc_PsyInputBuffer, INT_PCM, MAX_INPUT_BUFFER_SIZE, (8)) + +PSY_DYNAMIC *GetRam_aacEnc_PsyDynamic(int n, UCHAR *dynamic_RAM) { + FDK_ASSERT(dynamic_RAM != 0); + /* The reinterpret_cast is used to suppress a compiler warning. We know that + * (dynamic_RAM + P_BUF_1 + n*sizeof(PSY_DYNAMIC)) is sufficiently aligned, so + * the cast is safe */ + return reinterpret_cast<PSY_DYNAMIC *>(reinterpret_cast<void *>( + dynamic_RAM + P_BUF_1 + n * sizeof(PSY_DYNAMIC))); +} + +/* + The structure PSY_OUT holds all psychoaccoustic data needed + in quantization module +*/ +C_ALLOC_MEM2(Ram_aacEnc_PsyOut, PSY_OUT, 1, (1)) + +C_ALLOC_MEM2(Ram_aacEnc_PsyOutElements, PSY_OUT_ELEMENT, 1, (1) * ((8))) +C_ALLOC_MEM2(Ram_aacEnc_PsyOutChannel, PSY_OUT_CHANNEL, 1, (1) * (8)) + +/* + The structure QC_STATE contains preinitialized settings and quantizer + structures. + * AdjustThreshold structure contains element-wise settings. + * ElementBits contains elemnt-wise bit consumption settings. + * When CRC is active, lookup table is necessary for fast crc calculation. + * Bitcounter contains buffer to find optimal codebooks and minimal bit + consumption. Values are temporarily, so dynamic memory can be used. +*/ + +C_ALLOC_MEM(Ram_aacEnc_QCstate, QC_STATE, 1) +C_ALLOC_MEM(Ram_aacEnc_AdjustThreshold, ADJ_THR_STATE, 1) + +C_ALLOC_MEM2(Ram_aacEnc_AdjThrStateElement, ATS_ELEMENT, 1, ((8))) +C_ALLOC_MEM2(Ram_aacEnc_ElementBits, ELEMENT_BITS, 1, ((8))) +C_ALLOC_MEM(Ram_aacEnc_BitCntrState, struct BITCNTR_STATE, 1) + +INT *GetRam_aacEnc_BitLookUp(int n, UCHAR *dynamic_RAM) { + FDK_ASSERT(dynamic_RAM != 0); + /* The reinterpret_cast is used to suppress a compiler warning. We know that + * (dynamic_RAM + P_BUF_1) is sufficiently aligned, so the cast is safe */ + return reinterpret_cast<INT *>( + reinterpret_cast<void *>(dynamic_RAM + P_BUF_1)); +} +INT *GetRam_aacEnc_MergeGainLookUp(int n, UCHAR *dynamic_RAM) { + FDK_ASSERT(dynamic_RAM != 0); + /* The reinterpret_cast is used to suppress a compiler warning. We know that + * (dynamic_RAM + P_BUF_1 + sizeof(INT)*(MAX_SFB_LONG*(CODE_BOOK_ESC_NDX+1))) + * is sufficiently aligned, so the cast is safe */ + return reinterpret_cast<INT *>(reinterpret_cast<void *>( + dynamic_RAM + P_BUF_1 + + sizeof(INT) * (MAX_SFB_LONG * (CODE_BOOK_ESC_NDX + 1)))); +} + +/* + The structure QC_OUT contains settings and structures holding all necessary + information needed in bitstreamwriter. +*/ + +C_ALLOC_MEM2(Ram_aacEnc_QCout, QC_OUT, 1, (1)) +C_ALLOC_MEM2(Ram_aacEnc_QCelement, QC_OUT_ELEMENT, 1, (1) * ((8))) +QC_OUT_CHANNEL *GetRam_aacEnc_QCchannel(int n, UCHAR *dynamic_RAM) { + FDK_ASSERT(dynamic_RAM != 0); + /* The reinterpret_cast is used to suppress a compiler warning. We know that + * (dynamic_RAM + P_BUF_0 + n*sizeof(QC_OUT_CHANNEL)) is sufficiently aligned, + * so the cast is safe */ + return reinterpret_cast<QC_OUT_CHANNEL *>(reinterpret_cast<void *>( + dynamic_RAM + P_BUF_0 + n * sizeof(QC_OUT_CHANNEL))); +} diff --git a/fdk-aac/libAACenc/src/aacEnc_ram.h b/fdk-aac/libAACenc/src/aacEnc_ram.h new file mode 100644 index 0000000..0775aae --- /dev/null +++ b/fdk-aac/libAACenc/src/aacEnc_ram.h @@ -0,0 +1,249 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M. Lohwasser, M. Gayer + + Description: + +*******************************************************************************/ + +/*! + \file + \brief Memory layout + \author Markus Lohwasser +*/ + +#ifndef AACENC_RAM_H +#define AACENC_RAM_H + +#include "common_fix.h" + +#include "aacenc.h" +#include "psy_data.h" +#include "interface.h" +#include "psy_main.h" +#include "bitenc.h" +#include "bit_cnt.h" +#include "psy_const.h" + +#define OUTPUTBUFFER_SIZE \ + (8192) /*!< Output buffer size has to be at least 6144 bits per channel \ + (768 bytes). FDK bitbuffer implementation expects buffer of \ + size 2^n. */ + +/* + Moved AAC_ENC struct definition from aac_enc.cpp into aacEnc_ram.h to get size + and respective static memory in aacEnc_ram.cpp. aac_enc.h is the outward + visible header file and putting the struct into would cause necessity of + additional visible header files outside library. +*/ + +/* define hBitstream size: max AAC framelength is 6144 bits/channel */ +/*#define BUFFER_BITSTR_SIZE ((6400*(8)/bbWordSize) +((bbWordSize - 1) / + * bbWordSize))*/ + +struct AAC_ENC { + AACENC_CONFIG *config; + + INT ancillaryBitsPerFrame; /* ancillary bits per frame calculated from + ancillary rate */ + + CHANNEL_MAPPING channelMapping; + + QC_STATE *qcKernel; + QC_OUT *qcOut[(1)]; + + PSY_OUT *psyOut[(1)]; + PSY_INTERNAL *psyKernel; + + /* lifetime vars */ + + CHANNEL_MODE encoderMode; + INT bandwidth90dB; + AACENC_BITRATE_MODE bitrateMode; + + INT dontWriteAdif; /* use: write ADIF header only before 1st frame */ + + FIXP_DBL *dynamic_RAM; + + INT maxChannels; /* used while allocation */ + INT maxElements; + INT maxFrames; + + AUDIO_OBJECT_TYPE aot; /* AOT to be used while encoding. */ +}; + +#define maxSize(a, b) (((a) > (b)) ? (a) : (b)) + +#define BIT_LOOK_UP_SIZE \ + (sizeof(INT) * (MAX_SFB_LONG * (CODE_BOOK_ESC_NDX + 1))) +#define MERGE_GAIN_LOOK_UP_SIZE (sizeof(INT) * MAX_SFB_LONG) + +/* Size of AhFlag buffer in function FDKaacEnc_adaptThresholdsToPe() */ +#define ADJ_THR_AH_FLAG_SIZE (sizeof(UCHAR) * ((8)) * (2) * MAX_GROUPED_SFB) +/* Size of ThrExp buffer in function FDKaacEnc_adaptThresholdsToPe() */ +#define ADJ_THR_THR_EXP_SIZE (sizeof(FIXP_DBL) * ((8)) * (2) * MAX_GROUPED_SFB) +/* Size of sfbNActiveLinesLdData buffer in function FDKaacEnc_correctThresh() */ +#define ADJ_THR_ACT_LIN_LD_DATA_SIZE \ + (sizeof(FIXP_DBL) * ((8)) * (2) * MAX_GROUPED_SFB) +/* Total amount of dynamic buffer needed in adjust thresholds functionality */ +#define ADJ_THR_SIZE \ + (ADJ_THR_AH_FLAG_SIZE + ADJ_THR_THR_EXP_SIZE + ADJ_THR_ACT_LIN_LD_DATA_SIZE) + +/* Dynamic RAM - Allocation */ +/* + +++++++++++++++++++++++++++++++++++++++++++++++++++++ + | P_BUF_0 | P_BUF_1 | + +++++++++++++++++++++++++++++++++++++++++++++++++++++ + | QC_OUT_CH | PSY_DYN | + +++++++++++++++++++++++++++++++++++++++++++++++++++++ + | | BitLookUp+MergeGainLookUp | + +++++++++++++++++++++++++++++++++++++++++++++++++++++ + | | AH_FLAG | THR_EXP | ACT_LIN_LD_DATA | + +++++++++++++++++++++++++++++++++++++++++++++++++++++ + | | Bitstream output buffer | + +++++++++++++++++++++++++++++++++++++++++++++++++++++ +*/ + +#define BUF_SIZE_0 (ALIGN_SIZE(sizeof(QC_OUT_CHANNEL) * (8))) +#define BUF_SIZE_1 \ + (ALIGN_SIZE(maxSize(maxSize(sizeof(PSY_DYNAMIC), \ + (BIT_LOOK_UP_SIZE + MERGE_GAIN_LOOK_UP_SIZE)), \ + ADJ_THR_SIZE))) + +#define P_BUF_0 (0) +#define P_BUF_1 (P_BUF_0 + BUF_SIZE_0) + +#define AAC_ENC_DYN_RAM_SIZE (BUF_SIZE_0 + BUF_SIZE_1) + +H_ALLOC_MEM(AACdynamic_RAM, FIXP_DBL) +/* + ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ +END - Dynamic RAM - Allocation */ + +/* + See further Memory Allocation details in aacEnc_ram.cpp +*/ +H_ALLOC_MEM(Ram_aacEnc_AacEncoder, AAC_ENC) + +H_ALLOC_MEM(Ram_aacEnc_PsyElement, PSY_ELEMENT) + +H_ALLOC_MEM(Ram_aacEnc_PsyInternal, PSY_INTERNAL) +H_ALLOC_MEM(Ram_aacEnc_PsyStatic, PSY_STATIC) +H_ALLOC_MEM(Ram_aacEnc_PsyInputBuffer, INT_PCM) + +PSY_DYNAMIC *GetRam_aacEnc_PsyDynamic(int n, UCHAR *dynamic_RAM); + +H_ALLOC_MEM(Ram_aacEnc_PsyOutChannel, PSY_OUT_CHANNEL) + +H_ALLOC_MEM(Ram_aacEnc_PsyOut, PSY_OUT) +H_ALLOC_MEM(Ram_aacEnc_PsyOutElements, PSY_OUT_ELEMENT) + +H_ALLOC_MEM(Ram_aacEnc_QCstate, QC_STATE) +H_ALLOC_MEM(Ram_aacEnc_AdjustThreshold, ADJ_THR_STATE) + +H_ALLOC_MEM(Ram_aacEnc_AdjThrStateElement, ATS_ELEMENT) +H_ALLOC_MEM(Ram_aacEnc_ElementBits, ELEMENT_BITS) +H_ALLOC_MEM(Ram_aacEnc_BitCntrState, BITCNTR_STATE) + +INT *GetRam_aacEnc_BitLookUp(int n, UCHAR *dynamic_RAM); +INT *GetRam_aacEnc_MergeGainLookUp(int n, UCHAR *dynamic_RAM); +QC_OUT_CHANNEL *GetRam_aacEnc_QCchannel(int n, UCHAR *dynamic_RAM); + +H_ALLOC_MEM(Ram_aacEnc_QCout, QC_OUT) +H_ALLOC_MEM(Ram_aacEnc_QCelement, QC_OUT_ELEMENT) + +#endif /* #ifndef AACENC_RAM_H */ diff --git a/fdk-aac/libAACenc/src/aacEnc_rom.cpp b/fdk-aac/libAACenc/src/aacEnc_rom.cpp new file mode 100644 index 0000000..ac0fa9d --- /dev/null +++ b/fdk-aac/libAACenc/src/aacEnc_rom.cpp @@ -0,0 +1,2486 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M. Lohwasser, M. Gayer + + Description: + +*******************************************************************************/ + +#include "aacEnc_rom.h" + +/* + Huffman Tables +*/ +const ULONG FDKaacEnc_huff_ltab1_2[3][3][3][3] = { + {{{0x000b0009, 0x00090007, 0x000b0009}, + {0x000a0008, 0x00070006, 0x000a0008}, + {0x000b0009, 0x00090008, 0x000b0009}}, + {{0x000a0008, 0x00070006, 0x000a0007}, + {0x00070006, 0x00050005, 0x00070006}, + {0x00090007, 0x00070006, 0x000a0008}}, + {{0x000b0009, 0x00090007, 0x000b0008}, + {0x00090008, 0x00070006, 0x00090008}, + {0x000b0009, 0x00090007, 0x000b0009}}}, + {{{0x00090008, 0x00070006, 0x00090007}, + {0x00070006, 0x00050005, 0x00070006}, + {0x00090007, 0x00070006, 0x00090008}}, + {{0x00070006, 0x00050005, 0x00070006}, + {0x00050005, 0x00010003, 0x00050005}, + {0x00070006, 0x00050005, 0x00070006}}, + {{0x00090008, 0x00070006, 0x00090007}, + {0x00070006, 0x00050005, 0x00070006}, + {0x00090008, 0x00070006, 0x00090008}}}, + {{{0x000b0009, 0x00090007, 0x000b0009}, + {0x00090008, 0x00070006, 0x00090008}, + {0x000b0008, 0x00090007, 0x000b0009}}, + {{0x000a0008, 0x00070006, 0x00090007}, + {0x00070006, 0x00050004, 0x00070006}, + {0x00090008, 0x00070006, 0x000a0007}}, + {{0x000b0009, 0x00090007, 0x000b0009}, + {0x000a0007, 0x00070006, 0x00090008}, + {0x000b0009, 0x00090007, 0x000b0009}}}}; + +const ULONG FDKaacEnc_huff_ltab3_4[3][3][3][3] = { + {{{0x00010004, 0x00040005, 0x00080008}, + {0x00040005, 0x00050004, 0x00080008}, + {0x00090009, 0x00090008, 0x000a000b}}, + {{0x00040005, 0x00060005, 0x00090008}, + {0x00060005, 0x00060004, 0x00090008}, + {0x00090008, 0x00090007, 0x000a000a}}, + {{0x00090009, 0x000a0008, 0x000d000b}, + {0x00090008, 0x00090008, 0x000b000a}, + {0x000b000b, 0x000a000a, 0x000c000b}}}, + {{{0x00040004, 0x00060005, 0x000a0008}, + {0x00060004, 0x00070004, 0x000a0008}, + {0x000a0008, 0x000a0008, 0x000c000a}}, + {{0x00050004, 0x00070004, 0x000b0008}, + {0x00060004, 0x00070004, 0x000a0007}, + {0x00090008, 0x00090007, 0x000b0009}}, + {{0x00090008, 0x000a0008, 0x000d000a}, + {0x00080007, 0x00090007, 0x000c0009}, + {0x000a000a, 0x000b0009, 0x000c000a}}}, + {{{0x00080008, 0x000a0008, 0x000f000b}, + {0x00090008, 0x000b0007, 0x000f000a}, + {0x000d000b, 0x000e000a, 0x0010000c}}, + {{0x00080008, 0x000a0007, 0x000e000a}, + {0x00090007, 0x000a0007, 0x000e0009}, + {0x000c000a, 0x000c0009, 0x000f000b}}, + {{0x000b000b, 0x000c000a, 0x0010000c}, + {0x000a000a, 0x000b0009, 0x000f000b}, + {0x000c000b, 0x000c000a, 0x000f000b}}}}; + +const ULONG FDKaacEnc_huff_ltab5_6[9][9] = { + {0x000d000b, 0x000c000a, 0x000b0009, 0x000b0009, 0x000a0009, 0x000b0009, + 0x000b0009, 0x000c000a, 0x000d000b}, + {0x000c000a, 0x000b0009, 0x000a0008, 0x00090007, 0x00080007, 0x00090007, + 0x000a0008, 0x000b0009, 0x000c000a}, + {0x000c0009, 0x000a0008, 0x00090006, 0x00080006, 0x00070006, 0x00080006, + 0x00090006, 0x000a0008, 0x000b0009}, + {0x000b0009, 0x00090007, 0x00080006, 0x00050004, 0x00040004, 0x00050004, + 0x00080006, 0x00090007, 0x000b0009}, + {0x000a0009, 0x00080007, 0x00070006, 0x00040004, 0x00010004, 0x00040004, + 0x00070006, 0x00080007, 0x000b0009}, + {0x000b0009, 0x00090007, 0x00080006, 0x00050004, 0x00040004, 0x00050004, + 0x00080006, 0x00090007, 0x000b0009}, + {0x000b0009, 0x000a0008, 0x00090006, 0x00080006, 0x00070006, 0x00080006, + 0x00090006, 0x000a0008, 0x000b0009}, + {0x000c000a, 0x000b0009, 0x000a0008, 0x00090007, 0x00080007, 0x00090007, + 0x000a0007, 0x000b0008, 0x000c000a}, + {0x000d000b, 0x000c000a, 0x000c0009, 0x000b0009, 0x000a0009, 0x000a0009, + 0x000b0009, 0x000c000a, 0x000d000b}}; + +const ULONG FDKaacEnc_huff_ltab7_8[8][8] = { + {0x00010005, 0x00030004, 0x00060005, 0x00070006, 0x00080007, 0x00090008, + 0x000a0009, 0x000b000a}, + {0x00030004, 0x00040003, 0x00060004, 0x00070005, 0x00080006, 0x00080007, + 0x00090007, 0x00090008}, + {0x00060005, 0x00060004, 0x00070004, 0x00080005, 0x00080006, 0x00090007, + 0x00090007, 0x000a0008}, + {0x00070006, 0x00070005, 0x00080005, 0x00080006, 0x00090006, 0x00090007, + 0x000a0008, 0x000a0008}, + {0x00080007, 0x00080006, 0x00090006, 0x00090006, 0x000a0007, 0x000a0007, + 0x000a0008, 0x000b0009}, + {0x00090008, 0x00080007, 0x00090006, 0x00090007, 0x000a0007, 0x000a0008, + 0x000b0008, 0x000b000a}, + {0x000a0009, 0x00090007, 0x00090007, 0x000a0008, 0x000a0008, 0x000b0008, + 0x000c0009, 0x000c0009}, + {0x000b000a, 0x000a0008, 0x000a0008, 0x000a0008, 0x000b0009, 0x000b0009, + 0x000c0009, 0x000c000a}}; + +const ULONG FDKaacEnc_huff_ltab9_10[13][13] = { + {0x00010006, 0x00030005, 0x00060006, 0x00080006, 0x00090007, 0x000a0008, + 0x000a0009, 0x000b000a, 0x000b000a, 0x000c000a, 0x000c000b, 0x000d000b, + 0x000d000c}, + {0x00030005, 0x00040004, 0x00060004, 0x00070005, 0x00080006, 0x00080007, + 0x00090007, 0x000a0008, 0x000a0008, 0x000a0009, 0x000b000a, 0x000c000a, + 0x000c000b}, + {0x00060006, 0x00060004, 0x00070005, 0x00080005, 0x00080006, 0x00090006, + 0x000a0007, 0x000a0008, 0x000a0008, 0x000b0009, 0x000c0009, 0x000c000a, + 0x000c000a}, + {0x00080006, 0x00070005, 0x00080005, 0x00090005, 0x00090006, 0x000a0007, + 0x000a0007, 0x000b0008, 0x000b0008, 0x000b0009, 0x000c0009, 0x000c000a, + 0x000d000a}, + {0x00090007, 0x00080006, 0x00090006, 0x00090006, 0x000a0006, 0x000a0007, + 0x000b0007, 0x000b0008, 0x000b0008, 0x000c0009, 0x000c0009, 0x000c000a, + 0x000d000a}, + {0x000a0008, 0x00090007, 0x00090006, 0x000a0007, 0x000b0007, 0x000b0007, + 0x000b0008, 0x000c0008, 0x000b0008, 0x000c0009, 0x000c000a, 0x000d000a, + 0x000d000b}, + {0x000b0009, 0x00090007, 0x000a0007, 0x000b0007, 0x000b0007, 0x000b0008, + 0x000c0008, 0x000c0009, 0x000c0009, 0x000c0009, 0x000d000a, 0x000d000a, + 0x000d000b}, + {0x000b0009, 0x000a0008, 0x000a0008, 0x000b0008, 0x000b0008, 0x000c0008, + 0x000c0009, 0x000d0009, 0x000d0009, 0x000d000a, 0x000d000a, 0x000d000b, + 0x000d000b}, + {0x000b0009, 0x000a0008, 0x000a0008, 0x000b0008, 0x000b0008, 0x000b0008, + 0x000c0009, 0x000c0009, 0x000d000a, 0x000d000a, 0x000e000a, 0x000d000b, + 0x000e000b}, + {0x000b000a, 0x000a0009, 0x000b0009, 0x000b0009, 0x000c0009, 0x000c0009, + 0x000c0009, 0x000c000a, 0x000d000a, 0x000d000a, 0x000e000b, 0x000e000b, + 0x000e000c}, + {0x000c000a, 0x000b0009, 0x000b0009, 0x000c0009, 0x000c0009, 0x000c000a, + 0x000d000a, 0x000d000a, 0x000d000a, 0x000e000b, 0x000e000b, 0x000e000b, + 0x000f000c}, + {0x000c000b, 0x000b000a, 0x000c0009, 0x000c000a, 0x000c000a, 0x000d000a, + 0x000d000a, 0x000d000a, 0x000d000b, 0x000e000b, 0x000e000b, 0x000f000b, + 0x000f000c}, + {0x000d000b, 0x000c000a, 0x000c000a, 0x000c000a, 0x000d000a, 0x000d000a, + 0x000d000a, 0x000d000b, 0x000e000b, 0x000e000c, 0x000e000c, 0x000e000c, + 0x000f000c}}; + +const UCHAR FDKaacEnc_huff_ltab11[17][17] = { + {0x04, 0x05, 0x06, 0x07, 0x08, 0x08, 0x09, 0x0a, 0x0a, 0x0a, 0x0b, 0x0b, + 0x0c, 0x0b, 0x0c, 0x0c, 0x0a}, + {0x05, 0x04, 0x05, 0x06, 0x07, 0x07, 0x08, 0x08, 0x09, 0x09, 0x09, 0x0a, + 0x0a, 0x0a, 0x0a, 0x0b, 0x08}, + {0x06, 0x05, 0x05, 0x06, 0x07, 0x07, 0x08, 0x08, 0x08, 0x09, 0x09, 0x09, + 0x0a, 0x0a, 0x0a, 0x0a, 0x08}, + {0x07, 0x06, 0x06, 0x06, 0x07, 0x07, 0x08, 0x08, 0x08, 0x09, 0x09, 0x09, + 0x0a, 0x0a, 0x0a, 0x0a, 0x08}, + {0x08, 0x07, 0x07, 0x07, 0x07, 0x08, 0x08, 0x08, 0x08, 0x09, 0x09, 0x09, + 0x0a, 0x0a, 0x0a, 0x0a, 0x08}, + {0x08, 0x07, 0x07, 0x07, 0x07, 0x08, 0x08, 0x08, 0x09, 0x09, 0x09, 0x09, + 0x0a, 0x0a, 0x0a, 0x0a, 0x08}, + {0x09, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x09, 0x09, 0x09, 0x0a, + 0x0a, 0x0a, 0x0a, 0x0a, 0x08}, + {0x09, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x09, 0x09, 0x09, 0x0a, 0x0a, + 0x0a, 0x0a, 0x0a, 0x0a, 0x08}, + {0x0a, 0x09, 0x08, 0x08, 0x09, 0x09, 0x09, 0x09, 0x09, 0x0a, 0x0a, 0x0a, + 0x0a, 0x0a, 0x0a, 0x0b, 0x08}, + {0x0a, 0x09, 0x09, 0x09, 0x09, 0x09, 0x09, 0x09, 0x0a, 0x0a, 0x0a, 0x0a, + 0x0a, 0x0a, 0x0b, 0x0b, 0x08}, + {0x0b, 0x09, 0x09, 0x09, 0x09, 0x09, 0x09, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, + 0x0b, 0x0a, 0x0b, 0x0b, 0x08}, + {0x0b, 0x0a, 0x09, 0x09, 0x0a, 0x09, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0b, + 0x0b, 0x0b, 0x0b, 0x0b, 0x08}, + {0x0b, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0b, + 0x0b, 0x0b, 0x0b, 0x0b, 0x09}, + {0x0b, 0x0a, 0x09, 0x09, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0b, 0x0b, + 0x0b, 0x0b, 0x0b, 0x0b, 0x09}, + {0x0b, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0b, 0x0b, + 0x0b, 0x0b, 0x0b, 0x0b, 0x09}, + {0x0c, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0a, 0x0b, 0x0b, 0x0b, 0x0b, + 0x0b, 0x0b, 0x0c, 0x0c, 0x09}, + {0x09, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, 0x08, + 0x08, 0x08, 0x08, 0x09, 0x05}}; + +const UCHAR FDKaacEnc_huff_ltabscf[121] = { + 0x12, 0x12, 0x12, 0x12, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, + 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x12, 0x13, 0x12, + 0x11, 0x11, 0x10, 0x11, 0x10, 0x10, 0x10, 0x10, 0x0f, 0x0f, 0x0e, + 0x0e, 0x0e, 0x0e, 0x0e, 0x0e, 0x0d, 0x0d, 0x0c, 0x0c, 0x0c, 0x0b, + 0x0c, 0x0b, 0x0a, 0x0a, 0x0a, 0x09, 0x09, 0x08, 0x08, 0x08, 0x07, + 0x06, 0x06, 0x05, 0x04, 0x03, 0x01, 0x04, 0x04, 0x05, 0x06, 0x06, + 0x07, 0x07, 0x08, 0x08, 0x09, 0x09, 0x0a, 0x0a, 0x0a, 0x0b, 0x0b, + 0x0b, 0x0b, 0x0c, 0x0c, 0x0d, 0x0d, 0x0d, 0x0e, 0x0e, 0x10, 0x0f, + 0x10, 0x0f, 0x12, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, + 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, + 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13, 0x13}; + +const USHORT FDKaacEnc_huff_ctab1[3][3][3][3] = {{{{0x07f8, 0x01f1, 0x07fd}, + {0x03f5, 0x0068, 0x03f0}, + {0x07f7, 0x01ec, 0x07f5}}, + {{0x03f1, 0x0072, 0x03f4}, + {0x0074, 0x0011, 0x0076}, + {0x01eb, 0x006c, 0x03f6}}, + {{0x07fc, 0x01e1, 0x07f1}, + {0x01f0, 0x0061, 0x01f6}, + {0x07f2, 0x01ea, 0x07fb}}}, + {{{0x01f2, 0x0069, 0x01ed}, + {0x0077, 0x0017, 0x006f}, + {0x01e6, 0x0064, 0x01e5}}, + {{0x0067, 0x0015, 0x0062}, + {0x0012, 0x0000, 0x0014}, + {0x0065, 0x0016, 0x006d}}, + {{0x01e9, 0x0063, 0x01e4}, + {0x006b, 0x0013, 0x0071}, + {0x01e3, 0x0070, 0x01f3}}}, + {{{0x07fe, 0x01e7, 0x07f3}, + {0x01ef, 0x0060, 0x01ee}, + {0x07f0, 0x01e2, 0x07fa}}, + {{0x03f3, 0x006a, 0x01e8}, + {0x0075, 0x0010, 0x0073}, + {0x01f4, 0x006e, 0x03f7}}, + {{0x07f6, 0x01e0, 0x07f9}, + {0x03f2, 0x0066, 0x01f5}, + {0x07ff, 0x01f7, 0x07f4}}}}; + +const USHORT FDKaacEnc_huff_ctab2[3][3][3][3] = {{{{0x01f3, 0x006f, 0x01fd}, + {0x00eb, 0x0023, 0x00ea}, + {0x01f7, 0x00e8, 0x01fa}}, + {{0x00f2, 0x002d, 0x0070}, + {0x0020, 0x0006, 0x002b}, + {0x006e, 0x0028, 0x00e9}}, + {{0x01f9, 0x0066, 0x00f8}, + {0x00e7, 0x001b, 0x00f1}, + {0x01f4, 0x006b, 0x01f5}}}, + {{{0x00ec, 0x002a, 0x006c}, + {0x002c, 0x000a, 0x0027}, + {0x0067, 0x001a, 0x00f5}}, + {{0x0024, 0x0008, 0x001f}, + {0x0009, 0x0000, 0x0007}, + {0x001d, 0x000b, 0x0030}}, + {{0x00ef, 0x001c, 0x0064}, + {0x001e, 0x000c, 0x0029}, + {0x00f3, 0x002f, 0x00f0}}}, + {{{0x01fc, 0x0071, 0x01f2}, + {0x00f4, 0x0021, 0x00e6}, + {0x00f7, 0x0068, 0x01f8}}, + {{0x00ee, 0x0022, 0x0065}, + {0x0031, 0x0002, 0x0026}, + {0x00ed, 0x0025, 0x006a}}, + {{0x01fb, 0x0072, 0x01fe}, + {0x0069, 0x002e, 0x00f6}, + {0x01ff, 0x006d, 0x01f6}}}}; + +const USHORT FDKaacEnc_huff_ctab3[3][3][3][3] = {{{{0x0000, 0x0009, 0x00ef}, + {0x000b, 0x0019, 0x00f0}, + {0x01eb, 0x01e6, 0x03f2}}, + {{0x000a, 0x0035, 0x01ef}, + {0x0034, 0x0037, 0x01e9}, + {0x01ed, 0x01e7, 0x03f3}}, + {{0x01ee, 0x03ed, 0x1ffa}, + {0x01ec, 0x01f2, 0x07f9}, + {0x07f8, 0x03f8, 0x0ff8}}}, + {{{0x0008, 0x0038, 0x03f6}, + {0x0036, 0x0075, 0x03f1}, + {0x03eb, 0x03ec, 0x0ff4}}, + {{0x0018, 0x0076, 0x07f4}, + {0x0039, 0x0074, 0x03ef}, + {0x01f3, 0x01f4, 0x07f6}}, + {{0x01e8, 0x03ea, 0x1ffc}, + {0x00f2, 0x01f1, 0x0ffb}, + {0x03f5, 0x07f3, 0x0ffc}}}, + {{{0x00ee, 0x03f7, 0x7ffe}, + {0x01f0, 0x07f5, 0x7ffd}, + {0x1ffb, 0x3ffa, 0xffff}}, + {{0x00f1, 0x03f0, 0x3ffc}, + {0x01ea, 0x03ee, 0x3ffb}, + {0x0ff6, 0x0ffa, 0x7ffc}}, + {{0x07f2, 0x0ff5, 0xfffe}, + {0x03f4, 0x07f7, 0x7ffb}, + {0x0ff7, 0x0ff9, 0x7ffa}}}}; + +const USHORT FDKaacEnc_huff_ctab4[3][3][3][3] = {{{{0x0007, 0x0016, 0x00f6}, + {0x0018, 0x0008, 0x00ef}, + {0x01ef, 0x00f3, 0x07f8}}, + {{0x0019, 0x0017, 0x00ed}, + {0x0015, 0x0001, 0x00e2}, + {0x00f0, 0x0070, 0x03f0}}, + {{0x01ee, 0x00f1, 0x07fa}, + {0x00ee, 0x00e4, 0x03f2}, + {0x07f6, 0x03ef, 0x07fd}}}, + {{{0x0005, 0x0014, 0x00f2}, + {0x0009, 0x0004, 0x00e5}, + {0x00f4, 0x00e8, 0x03f4}}, + {{0x0006, 0x0002, 0x00e7}, + {0x0003, 0x0000, 0x006b}, + {0x00e3, 0x0069, 0x01f3}}, + {{0x00eb, 0x00e6, 0x03f6}, + {0x006e, 0x006a, 0x01f4}, + {0x03ec, 0x01f0, 0x03f9}}}, + {{{0x00f5, 0x00ec, 0x07fb}, + {0x00ea, 0x006f, 0x03f7}, + {0x07f9, 0x03f3, 0x0fff}}, + {{0x00e9, 0x006d, 0x03f8}, + {0x006c, 0x0068, 0x01f5}, + {0x03ee, 0x01f2, 0x07f4}}, + {{0x07f7, 0x03f1, 0x0ffe}, + {0x03ed, 0x01f1, 0x07f5}, + {0x07fe, 0x03f5, 0x07fc}}}}; + +const USHORT FDKaacEnc_huff_ctab5[9][9] = { + {0x1fff, 0x0ff7, 0x07f4, 0x07e8, 0x03f1, 0x07ee, 0x07f9, 0x0ff8, 0x1ffd}, + {0x0ffd, 0x07f1, 0x03e8, 0x01e8, 0x00f0, 0x01ec, 0x03ee, 0x07f2, 0x0ffa}, + {0x0ff4, 0x03ef, 0x01f2, 0x00e8, 0x0070, 0x00ec, 0x01f0, 0x03ea, 0x07f3}, + {0x07eb, 0x01eb, 0x00ea, 0x001a, 0x0008, 0x0019, 0x00ee, 0x01ef, 0x07ed}, + {0x03f0, 0x00f2, 0x0073, 0x000b, 0x0000, 0x000a, 0x0071, 0x00f3, 0x07e9}, + {0x07ef, 0x01ee, 0x00ef, 0x0018, 0x0009, 0x001b, 0x00eb, 0x01e9, 0x07ec}, + {0x07f6, 0x03eb, 0x01f3, 0x00ed, 0x0072, 0x00e9, 0x01f1, 0x03ed, 0x07f7}, + {0x0ff6, 0x07f0, 0x03e9, 0x01ed, 0x00f1, 0x01ea, 0x03ec, 0x07f8, 0x0ff9}, + {0x1ffc, 0x0ffc, 0x0ff5, 0x07ea, 0x03f3, 0x03f2, 0x07f5, 0x0ffb, 0x1ffe}}; + +const USHORT FDKaacEnc_huff_ctab6[9][9] = { + {0x07fe, 0x03fd, 0x01f1, 0x01eb, 0x01f4, 0x01ea, 0x01f0, 0x03fc, 0x07fd}, + {0x03f6, 0x01e5, 0x00ea, 0x006c, 0x0071, 0x0068, 0x00f0, 0x01e6, 0x03f7}, + {0x01f3, 0x00ef, 0x0032, 0x0027, 0x0028, 0x0026, 0x0031, 0x00eb, 0x01f7}, + {0x01e8, 0x006f, 0x002e, 0x0008, 0x0004, 0x0006, 0x0029, 0x006b, 0x01ee}, + {0x01ef, 0x0072, 0x002d, 0x0002, 0x0000, 0x0003, 0x002f, 0x0073, 0x01fa}, + {0x01e7, 0x006e, 0x002b, 0x0007, 0x0001, 0x0005, 0x002c, 0x006d, 0x01ec}, + {0x01f9, 0x00ee, 0x0030, 0x0024, 0x002a, 0x0025, 0x0033, 0x00ec, 0x01f2}, + {0x03f8, 0x01e4, 0x00ed, 0x006a, 0x0070, 0x0069, 0x0074, 0x00f1, 0x03fa}, + {0x07ff, 0x03f9, 0x01f6, 0x01ed, 0x01f8, 0x01e9, 0x01f5, 0x03fb, 0x07fc}}; + +const USHORT FDKaacEnc_huff_ctab7[8][8] = { + {0x0000, 0x0005, 0x0037, 0x0074, 0x00f2, 0x01eb, 0x03ed, 0x07f7}, + {0x0004, 0x000c, 0x0035, 0x0071, 0x00ec, 0x00ee, 0x01ee, 0x01f5}, + {0x0036, 0x0034, 0x0072, 0x00ea, 0x00f1, 0x01e9, 0x01f3, 0x03f5}, + {0x0073, 0x0070, 0x00eb, 0x00f0, 0x01f1, 0x01f0, 0x03ec, 0x03fa}, + {0x00f3, 0x00ed, 0x01e8, 0x01ef, 0x03ef, 0x03f1, 0x03f9, 0x07fb}, + {0x01ed, 0x00ef, 0x01ea, 0x01f2, 0x03f3, 0x03f8, 0x07f9, 0x07fc}, + {0x03ee, 0x01ec, 0x01f4, 0x03f4, 0x03f7, 0x07f8, 0x0ffd, 0x0ffe}, + {0x07f6, 0x03f0, 0x03f2, 0x03f6, 0x07fa, 0x07fd, 0x0ffc, 0x0fff}}; + +const USHORT FDKaacEnc_huff_ctab8[8][8] = { + {0x000e, 0x0005, 0x0010, 0x0030, 0x006f, 0x00f1, 0x01fa, 0x03fe}, + {0x0003, 0x0000, 0x0004, 0x0012, 0x002c, 0x006a, 0x0075, 0x00f8}, + {0x000f, 0x0002, 0x0006, 0x0014, 0x002e, 0x0069, 0x0072, 0x00f5}, + {0x002f, 0x0011, 0x0013, 0x002a, 0x0032, 0x006c, 0x00ec, 0x00fa}, + {0x0071, 0x002b, 0x002d, 0x0031, 0x006d, 0x0070, 0x00f2, 0x01f9}, + {0x00ef, 0x0068, 0x0033, 0x006b, 0x006e, 0x00ee, 0x00f9, 0x03fc}, + {0x01f8, 0x0074, 0x0073, 0x00ed, 0x00f0, 0x00f6, 0x01f6, 0x01fd}, + {0x03fd, 0x00f3, 0x00f4, 0x00f7, 0x01f7, 0x01fb, 0x01fc, 0x03ff}}; + +const USHORT FDKaacEnc_huff_ctab9[13][13] = { + {0x0000, 0x0005, 0x0037, 0x00e7, 0x01de, 0x03ce, 0x03d9, 0x07c8, 0x07cd, + 0x0fc8, 0x0fdd, 0x1fe4, 0x1fec}, + {0x0004, 0x000c, 0x0035, 0x0072, 0x00ea, 0x00ed, 0x01e2, 0x03d1, 0x03d3, + 0x03e0, 0x07d8, 0x0fcf, 0x0fd5}, + {0x0036, 0x0034, 0x0071, 0x00e8, 0x00ec, 0x01e1, 0x03cf, 0x03dd, 0x03db, + 0x07d0, 0x0fc7, 0x0fd4, 0x0fe4}, + {0x00e6, 0x0070, 0x00e9, 0x01dd, 0x01e3, 0x03d2, 0x03dc, 0x07cc, 0x07ca, + 0x07de, 0x0fd8, 0x0fea, 0x1fdb}, + {0x01df, 0x00eb, 0x01dc, 0x01e6, 0x03d5, 0x03de, 0x07cb, 0x07dd, 0x07dc, + 0x0fcd, 0x0fe2, 0x0fe7, 0x1fe1}, + {0x03d0, 0x01e0, 0x01e4, 0x03d6, 0x07c5, 0x07d1, 0x07db, 0x0fd2, 0x07e0, + 0x0fd9, 0x0feb, 0x1fe3, 0x1fe9}, + {0x07c4, 0x01e5, 0x03d7, 0x07c6, 0x07cf, 0x07da, 0x0fcb, 0x0fda, 0x0fe3, + 0x0fe9, 0x1fe6, 0x1ff3, 0x1ff7}, + {0x07d3, 0x03d8, 0x03e1, 0x07d4, 0x07d9, 0x0fd3, 0x0fde, 0x1fdd, 0x1fd9, + 0x1fe2, 0x1fea, 0x1ff1, 0x1ff6}, + {0x07d2, 0x03d4, 0x03da, 0x07c7, 0x07d7, 0x07e2, 0x0fce, 0x0fdb, 0x1fd8, + 0x1fee, 0x3ff0, 0x1ff4, 0x3ff2}, + {0x07e1, 0x03df, 0x07c9, 0x07d6, 0x0fca, 0x0fd0, 0x0fe5, 0x0fe6, 0x1feb, + 0x1fef, 0x3ff3, 0x3ff4, 0x3ff5}, + {0x0fe0, 0x07ce, 0x07d5, 0x0fc6, 0x0fd1, 0x0fe1, 0x1fe0, 0x1fe8, 0x1ff0, + 0x3ff1, 0x3ff8, 0x3ff6, 0x7ffc}, + {0x0fe8, 0x07df, 0x0fc9, 0x0fd7, 0x0fdc, 0x1fdc, 0x1fdf, 0x1fed, 0x1ff5, + 0x3ff9, 0x3ffb, 0x7ffd, 0x7ffe}, + {0x1fe7, 0x0fcc, 0x0fd6, 0x0fdf, 0x1fde, 0x1fda, 0x1fe5, 0x1ff2, 0x3ffa, + 0x3ff7, 0x3ffc, 0x3ffd, 0x7fff}}; + +const USHORT FDKaacEnc_huff_ctab10[13][13] = { + {0x0022, 0x0008, 0x001d, 0x0026, 0x005f, 0x00d3, 0x01cf, 0x03d0, 0x03d7, + 0x03ed, 0x07f0, 0x07f6, 0x0ffd}, + {0x0007, 0x0000, 0x0001, 0x0009, 0x0020, 0x0054, 0x0060, 0x00d5, 0x00dc, + 0x01d4, 0x03cd, 0x03de, 0x07e7}, + {0x001c, 0x0002, 0x0006, 0x000c, 0x001e, 0x0028, 0x005b, 0x00cd, 0x00d9, + 0x01ce, 0x01dc, 0x03d9, 0x03f1}, + {0x0025, 0x000b, 0x000a, 0x000d, 0x0024, 0x0057, 0x0061, 0x00cc, 0x00dd, + 0x01cc, 0x01de, 0x03d3, 0x03e7}, + {0x005d, 0x0021, 0x001f, 0x0023, 0x0027, 0x0059, 0x0064, 0x00d8, 0x00df, + 0x01d2, 0x01e2, 0x03dd, 0x03ee}, + {0x00d1, 0x0055, 0x0029, 0x0056, 0x0058, 0x0062, 0x00ce, 0x00e0, 0x00e2, + 0x01da, 0x03d4, 0x03e3, 0x07eb}, + {0x01c9, 0x005e, 0x005a, 0x005c, 0x0063, 0x00ca, 0x00da, 0x01c7, 0x01ca, + 0x01e0, 0x03db, 0x03e8, 0x07ec}, + {0x01e3, 0x00d2, 0x00cb, 0x00d0, 0x00d7, 0x00db, 0x01c6, 0x01d5, 0x01d8, + 0x03ca, 0x03da, 0x07ea, 0x07f1}, + {0x01e1, 0x00d4, 0x00cf, 0x00d6, 0x00de, 0x00e1, 0x01d0, 0x01d6, 0x03d1, + 0x03d5, 0x03f2, 0x07ee, 0x07fb}, + {0x03e9, 0x01cd, 0x01c8, 0x01cb, 0x01d1, 0x01d7, 0x01df, 0x03cf, 0x03e0, + 0x03ef, 0x07e6, 0x07f8, 0x0ffa}, + {0x03eb, 0x01dd, 0x01d3, 0x01d9, 0x01db, 0x03d2, 0x03cc, 0x03dc, 0x03ea, + 0x07ed, 0x07f3, 0x07f9, 0x0ff9}, + {0x07f2, 0x03ce, 0x01e4, 0x03cb, 0x03d8, 0x03d6, 0x03e2, 0x03e5, 0x07e8, + 0x07f4, 0x07f5, 0x07f7, 0x0ffb}, + {0x07fa, 0x03ec, 0x03df, 0x03e1, 0x03e4, 0x03e6, 0x03f0, 0x07e9, 0x07ef, + 0x0ff8, 0x0ffe, 0x0ffc, 0x0fff}}; + +const USHORT FDKaacEnc_huff_ctab11[21][17] = { + {0x0000, 0x0006, 0x0019, 0x003d, 0x009c, 0x00c6, 0x01a7, 0x0390, 0x03c2, + 0x03df, 0x07e6, 0x07f3, 0x0ffb, 0x07ec, 0x0ffa, 0x0ffe, 0x038e}, + {0x0005, 0x0001, 0x0008, 0x0014, 0x0037, 0x0042, 0x0092, 0x00af, 0x0191, + 0x01a5, 0x01b5, 0x039e, 0x03c0, 0x03a2, 0x03cd, 0x07d6, 0x00ae}, + {0x0017, 0x0007, 0x0009, 0x0018, 0x0039, 0x0040, 0x008e, 0x00a3, 0x00b8, + 0x0199, 0x01ac, 0x01c1, 0x03b1, 0x0396, 0x03be, 0x03ca, 0x009d}, + {0x003c, 0x0015, 0x0016, 0x001a, 0x003b, 0x0044, 0x0091, 0x00a5, 0x00be, + 0x0196, 0x01ae, 0x01b9, 0x03a1, 0x0391, 0x03a5, 0x03d5, 0x0094}, + {0x009a, 0x0036, 0x0038, 0x003a, 0x0041, 0x008c, 0x009b, 0x00b0, 0x00c3, + 0x019e, 0x01ab, 0x01bc, 0x039f, 0x038f, 0x03a9, 0x03cf, 0x0093}, + {0x00bf, 0x003e, 0x003f, 0x0043, 0x0045, 0x009e, 0x00a7, 0x00b9, 0x0194, + 0x01a2, 0x01ba, 0x01c3, 0x03a6, 0x03a7, 0x03bb, 0x03d4, 0x009f}, + {0x01a0, 0x008f, 0x008d, 0x0090, 0x0098, 0x00a6, 0x00b6, 0x00c4, 0x019f, + 0x01af, 0x01bf, 0x0399, 0x03bf, 0x03b4, 0x03c9, 0x03e7, 0x00a8}, + {0x01b6, 0x00ab, 0x00a4, 0x00aa, 0x00b2, 0x00c2, 0x00c5, 0x0198, 0x01a4, + 0x01b8, 0x038c, 0x03a4, 0x03c4, 0x03c6, 0x03dd, 0x03e8, 0x00ad}, + {0x03af, 0x0192, 0x00bd, 0x00bc, 0x018e, 0x0197, 0x019a, 0x01a3, 0x01b1, + 0x038d, 0x0398, 0x03b7, 0x03d3, 0x03d1, 0x03db, 0x07dd, 0x00b4}, + {0x03de, 0x01a9, 0x019b, 0x019c, 0x01a1, 0x01aa, 0x01ad, 0x01b3, 0x038b, + 0x03b2, 0x03b8, 0x03ce, 0x03e1, 0x03e0, 0x07d2, 0x07e5, 0x00b7}, + {0x07e3, 0x01bb, 0x01a8, 0x01a6, 0x01b0, 0x01b2, 0x01b7, 0x039b, 0x039a, + 0x03ba, 0x03b5, 0x03d6, 0x07d7, 0x03e4, 0x07d8, 0x07ea, 0x00ba}, + {0x07e8, 0x03a0, 0x01bd, 0x01b4, 0x038a, 0x01c4, 0x0392, 0x03aa, 0x03b0, + 0x03bc, 0x03d7, 0x07d4, 0x07dc, 0x07db, 0x07d5, 0x07f0, 0x00c1}, + {0x07fb, 0x03c8, 0x03a3, 0x0395, 0x039d, 0x03ac, 0x03ae, 0x03c5, 0x03d8, + 0x03e2, 0x03e6, 0x07e4, 0x07e7, 0x07e0, 0x07e9, 0x07f7, 0x0190}, + {0x07f2, 0x0393, 0x01be, 0x01c0, 0x0394, 0x0397, 0x03ad, 0x03c3, 0x03c1, + 0x03d2, 0x07da, 0x07d9, 0x07df, 0x07eb, 0x07f4, 0x07fa, 0x0195}, + {0x07f8, 0x03bd, 0x039c, 0x03ab, 0x03a8, 0x03b3, 0x03b9, 0x03d0, 0x03e3, + 0x03e5, 0x07e2, 0x07de, 0x07ed, 0x07f1, 0x07f9, 0x07fc, 0x0193}, + {0x0ffd, 0x03dc, 0x03b6, 0x03c7, 0x03cc, 0x03cb, 0x03d9, 0x03da, 0x07d3, + 0x07e1, 0x07ee, 0x07ef, 0x07f5, 0x07f6, 0x0ffc, 0x0fff, 0x019d}, + {0x01c2, 0x00b5, 0x00a1, 0x0096, 0x0097, 0x0095, 0x0099, 0x00a0, 0x00a2, + 0x00ac, 0x00a9, 0x00b1, 0x00b3, 0x00bb, 0x00c0, 0x018f, 0x0004}, + {0x0018, 0x002e, 0x0000, 0x005a, 0x00a5, 0x00f8, 0x00b7, 0x0094, 0x00f9, + 0x004d, 0x0021, 0x002b, 0x004f, 0x007b, 0x00bc, 0x0046, 0x0015}, + {0x0042, 0x0037, 0x0078, 0x000d, 0x0068, 0x005f, 0x000d, 0x005e, 0x005a, + 0x00be, 0x0063, 0x007e, 0x001f, 0x0092, 0x001a, 0x00ab, 0x0032}, + {0x00e6, 0x0037, 0x0000, 0x0058, 0x000b, 0x005a, 0x00e1, 0x005d, 0x0029, + 0x0017, 0x007e, 0x0069, 0x00aa, 0x0054, 0x0029, 0x0032, 0x0041}, + {0x0046, 0x00ea, 0x0034, 0x00ea, 0x0011, 0x001b, 0x00a9, 0x0094, 0x00e2, + 0x0031, 0x00d0, 0x00e5, 0x0007, 0x0070, 0x0069, 0x003e, 0x0021}}; + +const ULONG FDKaacEnc_huff_ctabscf[121] = { + 0x0003ffe8, 0x0003ffe6, 0x0003ffe7, 0x0003ffe5, 0x0007fff5, 0x0007fff1, + 0x0007ffed, 0x0007fff6, 0x0007ffee, 0x0007ffef, 0x0007fff0, 0x0007fffc, + 0x0007fffd, 0x0007ffff, 0x0007fffe, 0x0007fff7, 0x0007fff8, 0x0007fffb, + 0x0007fff9, 0x0003ffe4, 0x0007fffa, 0x0003ffe3, 0x0001ffef, 0x0001fff0, + 0x0000fff5, 0x0001ffee, 0x0000fff2, 0x0000fff3, 0x0000fff4, 0x0000fff1, + 0x00007ff6, 0x00007ff7, 0x00003ff9, 0x00003ff5, 0x00003ff7, 0x00003ff3, + 0x00003ff6, 0x00003ff2, 0x00001ff7, 0x00001ff5, 0x00000ff9, 0x00000ff7, + 0x00000ff6, 0x000007f9, 0x00000ff4, 0x000007f8, 0x000003f9, 0x000003f7, + 0x000003f5, 0x000001f8, 0x000001f7, 0x000000fa, 0x000000f8, 0x000000f6, + 0x00000079, 0x0000003a, 0x00000038, 0x0000001a, 0x0000000b, 0x00000004, + 0x00000000, 0x0000000a, 0x0000000c, 0x0000001b, 0x00000039, 0x0000003b, + 0x00000078, 0x0000007a, 0x000000f7, 0x000000f9, 0x000001f6, 0x000001f9, + 0x000003f4, 0x000003f6, 0x000003f8, 0x000007f5, 0x000007f4, 0x000007f6, + 0x000007f7, 0x00000ff5, 0x00000ff8, 0x00001ff4, 0x00001ff6, 0x00001ff8, + 0x00003ff8, 0x00003ff4, 0x0000fff0, 0x00007ff4, 0x0000fff6, 0x00007ff5, + 0x0003ffe2, 0x0007ffd9, 0x0007ffda, 0x0007ffdb, 0x0007ffdc, 0x0007ffdd, + 0x0007ffde, 0x0007ffd8, 0x0007ffd2, 0x0007ffd3, 0x0007ffd4, 0x0007ffd5, + 0x0007ffd6, 0x0007fff2, 0x0007ffdf, 0x0007ffe7, 0x0007ffe8, 0x0007ffe9, + 0x0007ffea, 0x0007ffeb, 0x0007ffe6, 0x0007ffe0, 0x0007ffe1, 0x0007ffe2, + 0x0007ffe3, 0x0007ffe4, 0x0007ffe5, 0x0007ffd7, 0x0007ffec, 0x0007fff4, + 0x0007fff3}; + +/* + table of (0.50000...1.00000) ^0.75 +*/ +const FIXP_QTD FDKaacEnc_mTab_3_4[MANT_SIZE] = { + QTC(0x4c1bf829), QTC(0x4c3880de), QTC(0x4c550603), QTC(0x4c71879c), + QTC(0x4c8e05aa), QTC(0x4caa8030), QTC(0x4cc6f72f), QTC(0x4ce36aab), + QTC(0x4cffdaa4), QTC(0x4d1c471d), QTC(0x4d38b019), QTC(0x4d55159a), + QTC(0x4d7177a1), QTC(0x4d8dd631), QTC(0x4daa314b), QTC(0x4dc688f3), + QTC(0x4de2dd2a), QTC(0x4dff2df2), QTC(0x4e1b7b4d), QTC(0x4e37c53d), + QTC(0x4e540bc5), QTC(0x4e704ee6), QTC(0x4e8c8ea3), QTC(0x4ea8cafd), + QTC(0x4ec503f7), QTC(0x4ee13992), QTC(0x4efd6bd0), QTC(0x4f199ab4), + QTC(0x4f35c640), QTC(0x4f51ee75), QTC(0x4f6e1356), QTC(0x4f8a34e4), + QTC(0x4fa65321), QTC(0x4fc26e10), QTC(0x4fde85b2), QTC(0x4ffa9a0a), + QTC(0x5016ab18), QTC(0x5032b8e0), QTC(0x504ec362), QTC(0x506acaa1), + QTC(0x5086cea0), QTC(0x50a2cf5e), QTC(0x50becce0), QTC(0x50dac725), + QTC(0x50f6be31), QTC(0x5112b205), QTC(0x512ea2a3), QTC(0x514a900d), + QTC(0x51667a45), QTC(0x5182614c), QTC(0x519e4524), QTC(0x51ba25cf), + QTC(0x51d60350), QTC(0x51f1dda7), QTC(0x520db4d6), QTC(0x522988e0), + QTC(0x524559c6), QTC(0x52612789), QTC(0x527cf22d), QTC(0x5298b9b1), + QTC(0x52b47e19), QTC(0x52d03f65), QTC(0x52ebfd98), QTC(0x5307b8b4), + QTC(0x532370b9), QTC(0x533f25aa), QTC(0x535ad789), QTC(0x53768656), + QTC(0x53923215), QTC(0x53addac6), QTC(0x53c9806b), QTC(0x53e52306), + QTC(0x5400c298), QTC(0x541c5f24), QTC(0x5437f8ab), QTC(0x54538f2e), + QTC(0x546f22af), QTC(0x548ab330), QTC(0x54a640b3), QTC(0x54c1cb38), + QTC(0x54dd52c2), QTC(0x54f8d753), QTC(0x551458eb), QTC(0x552fd78d), + QTC(0x554b5339), QTC(0x5566cbf3), QTC(0x558241bb), QTC(0x559db492), + QTC(0x55b9247b), QTC(0x55d49177), QTC(0x55effb87), QTC(0x560b62ad), + QTC(0x5626c6eb), QTC(0x56422842), QTC(0x565d86b4), QTC(0x5678e242), + QTC(0x56943aee), QTC(0x56af90b9), QTC(0x56cae3a4), QTC(0x56e633b2), + QTC(0x570180e4), QTC(0x571ccb3b), QTC(0x573812b8), QTC(0x5753575e), + QTC(0x576e992e), QTC(0x5789d829), QTC(0x57a51450), QTC(0x57c04da6), + QTC(0x57db842b), QTC(0x57f6b7e1), QTC(0x5811e8c9), QTC(0x582d16e6), + QTC(0x58484238), QTC(0x58636ac0), QTC(0x587e9081), QTC(0x5899b37c), + QTC(0x58b4d3b1), QTC(0x58cff123), QTC(0x58eb0bd3), QTC(0x590623c2), + QTC(0x592138f2), QTC(0x593c4b63), QTC(0x59575b19), QTC(0x59726812), + QTC(0x598d7253), QTC(0x59a879da), QTC(0x59c37eab), QTC(0x59de80c6), + QTC(0x59f9802d), QTC(0x5a147ce0), QTC(0x5a2f76e2), QTC(0x5a4a6e34), + QTC(0x5a6562d6), QTC(0x5a8054cb), QTC(0x5a9b4414), QTC(0x5ab630b2), + QTC(0x5ad11aa6), QTC(0x5aec01f1), QTC(0x5b06e696), QTC(0x5b21c895), + QTC(0x5b3ca7ef), QTC(0x5b5784a6), QTC(0x5b725ebc), QTC(0x5b8d3631), + QTC(0x5ba80b06), QTC(0x5bc2dd3e), QTC(0x5bddacd9), QTC(0x5bf879d8), + QTC(0x5c13443d), QTC(0x5c2e0c09), QTC(0x5c48d13e), QTC(0x5c6393dc), + QTC(0x5c7e53e5), QTC(0x5c99115a), QTC(0x5cb3cc3c), QTC(0x5cce848d), + QTC(0x5ce93a4e), QTC(0x5d03ed80), QTC(0x5d1e9e24), QTC(0x5d394c3b), + QTC(0x5d53f7c7), QTC(0x5d6ea0c9), QTC(0x5d894742), QTC(0x5da3eb33), + QTC(0x5dbe8c9e), QTC(0x5dd92b84), QTC(0x5df3c7e5), QTC(0x5e0e61c3), + QTC(0x5e28f920), QTC(0x5e438dfc), QTC(0x5e5e2059), QTC(0x5e78b037), + QTC(0x5e933d99), QTC(0x5eadc87e), QTC(0x5ec850e9), QTC(0x5ee2d6da), + QTC(0x5efd5a53), QTC(0x5f17db54), QTC(0x5f3259e0), QTC(0x5f4cd5f6), + QTC(0x5f674f99), QTC(0x5f81c6c8), QTC(0x5f9c3b87), QTC(0x5fb6add4), + QTC(0x5fd11db3), QTC(0x5feb8b23), QTC(0x6005f626), QTC(0x60205ebd), + QTC(0x603ac4e9), QTC(0x605528ac), QTC(0x606f8a05), QTC(0x6089e8f7), + QTC(0x60a44583), QTC(0x60be9fa9), QTC(0x60d8f76b), QTC(0x60f34cca), + QTC(0x610d9fc7), QTC(0x6127f062), QTC(0x61423e9e), QTC(0x615c8a7a), + QTC(0x6176d3f9), QTC(0x61911b1b), QTC(0x61ab5fe1), QTC(0x61c5a24d), + QTC(0x61dfe25f), QTC(0x61fa2018), QTC(0x62145b7a), QTC(0x622e9485), + QTC(0x6248cb3b), QTC(0x6262ff9d), QTC(0x627d31ab), QTC(0x62976167), + QTC(0x62b18ed1), QTC(0x62cbb9eb), QTC(0x62e5e2b6), QTC(0x63000933), + QTC(0x631a2d62), QTC(0x63344f45), QTC(0x634e6edd), QTC(0x63688c2b), + QTC(0x6382a730), QTC(0x639cbfec), QTC(0x63b6d661), QTC(0x63d0ea90), + QTC(0x63eafc7a), QTC(0x64050c1f), QTC(0x641f1982), QTC(0x643924a2), + QTC(0x64532d80), QTC(0x646d341f), QTC(0x6487387e), QTC(0x64a13a9e), + QTC(0x64bb3a81), QTC(0x64d53828), QTC(0x64ef3393), QTC(0x65092cc4), + QTC(0x652323bb), QTC(0x653d1879), QTC(0x65570b00), QTC(0x6570fb50), + QTC(0x658ae96b), QTC(0x65a4d550), QTC(0x65bebf01), QTC(0x65d8a680), + QTC(0x65f28bcc), QTC(0x660c6ee8), QTC(0x66264fd3), QTC(0x66402e8f), + QTC(0x665a0b1c), QTC(0x6673e57d), QTC(0x668dbdb0), QTC(0x66a793b8), + QTC(0x66c16795), QTC(0x66db3949), QTC(0x66f508d4), QTC(0x670ed636), + QTC(0x6728a172), QTC(0x67426a87), QTC(0x675c3177), QTC(0x6775f643), + QTC(0x678fb8eb), QTC(0x67a97971), QTC(0x67c337d5), QTC(0x67dcf418), + QTC(0x67f6ae3b), QTC(0x6810663f), QTC(0x682a1c25), QTC(0x6843cfed), + QTC(0x685d8199), QTC(0x68773129), QTC(0x6890de9f), QTC(0x68aa89fa), + QTC(0x68c4333d), QTC(0x68ddda67), QTC(0x68f77f7a), QTC(0x69112277), + QTC(0x692ac35e), QTC(0x69446230), QTC(0x695dfeee), QTC(0x6977999a), + QTC(0x69913232), QTC(0x69aac8ba), QTC(0x69c45d31), QTC(0x69ddef98), + QTC(0x69f77ff0), QTC(0x6a110e3a), QTC(0x6a2a9a77), QTC(0x6a4424a8), + QTC(0x6a5daccc), QTC(0x6a7732e6), QTC(0x6a90b6f6), QTC(0x6aaa38fd), + QTC(0x6ac3b8fb), QTC(0x6add36f2), QTC(0x6af6b2e2), QTC(0x6b102ccd), + QTC(0x6b29a4b2), QTC(0x6b431a92), QTC(0x6b5c8e6f), QTC(0x6b76004a), + QTC(0x6b8f7022), QTC(0x6ba8ddf9), QTC(0x6bc249d0), QTC(0x6bdbb3a7), + QTC(0x6bf51b80), QTC(0x6c0e815a), QTC(0x6c27e537), QTC(0x6c414718), + QTC(0x6c5aa6fd), QTC(0x6c7404e7), QTC(0x6c8d60d7), QTC(0x6ca6bace), + QTC(0x6cc012cc), QTC(0x6cd968d2), QTC(0x6cf2bce1), QTC(0x6d0c0ef9), + QTC(0x6d255f1d), QTC(0x6d3ead4b), QTC(0x6d57f985), QTC(0x6d7143cc), + QTC(0x6d8a8c21), QTC(0x6da3d283), QTC(0x6dbd16f5), QTC(0x6dd65976), + QTC(0x6def9a08), QTC(0x6e08d8ab), QTC(0x6e221560), QTC(0x6e3b5027), + QTC(0x6e548902), QTC(0x6e6dbff1), QTC(0x6e86f4f5), QTC(0x6ea0280e), + QTC(0x6eb9593e), QTC(0x6ed28885), QTC(0x6eebb5e3), QTC(0x6f04e15a), + QTC(0x6f1e0aea), QTC(0x6f373294), QTC(0x6f505859), QTC(0x6f697c39), + QTC(0x6f829e35), QTC(0x6f9bbe4e), QTC(0x6fb4dc85), QTC(0x6fcdf8d9), + QTC(0x6fe7134d), QTC(0x70002be0), QTC(0x70194293), QTC(0x70325767), + QTC(0x704b6a5d), QTC(0x70647b76), QTC(0x707d8ab1), QTC(0x70969811), + QTC(0x70afa394), QTC(0x70c8ad3d), QTC(0x70e1b50c), QTC(0x70fabb01), + QTC(0x7113bf1d), QTC(0x712cc161), QTC(0x7145c1ce), QTC(0x715ec064), + QTC(0x7177bd24), QTC(0x7190b80f), QTC(0x71a9b124), QTC(0x71c2a866), + QTC(0x71db9dd4), QTC(0x71f49170), QTC(0x720d8339), QTC(0x72267331), + QTC(0x723f6159), QTC(0x72584db0), QTC(0x72713838), QTC(0x728a20f1), + QTC(0x72a307db), QTC(0x72bbecf9), QTC(0x72d4d049), QTC(0x72edb1ce), + QTC(0x73069187), QTC(0x731f6f75), QTC(0x73384b98), QTC(0x735125f3), + QTC(0x7369fe84), QTC(0x7382d54d), QTC(0x739baa4e), QTC(0x73b47d89), + QTC(0x73cd4efd), QTC(0x73e61eab), QTC(0x73feec94), QTC(0x7417b8b8), + QTC(0x74308319), QTC(0x74494bb6), QTC(0x74621291), QTC(0x747ad7aa), + QTC(0x74939b02), QTC(0x74ac5c98), QTC(0x74c51c6f), QTC(0x74ddda86), + QTC(0x74f696de), QTC(0x750f5178), QTC(0x75280a54), QTC(0x7540c174), + QTC(0x755976d7), QTC(0x75722a7e), QTC(0x758adc69), QTC(0x75a38c9b), + QTC(0x75bc3b12), QTC(0x75d4e7cf), QTC(0x75ed92d4), QTC(0x76063c21), + QTC(0x761ee3b6), QTC(0x76378994), QTC(0x76502dbc), QTC(0x7668d02e), + QTC(0x768170eb), QTC(0x769a0ff3), QTC(0x76b2ad47), QTC(0x76cb48e7), + QTC(0x76e3e2d5), QTC(0x76fc7b10), QTC(0x7715119a), QTC(0x772da673), + QTC(0x7746399b), QTC(0x775ecb13), QTC(0x77775adc), QTC(0x778fe8f6), + QTC(0x77a87561), QTC(0x77c1001f), QTC(0x77d98930), QTC(0x77f21095), + QTC(0x780a964d), QTC(0x78231a5b), QTC(0x783b9cbd), QTC(0x78541d75), + QTC(0x786c9c84), QTC(0x788519e9), QTC(0x789d95a6), QTC(0x78b60fbb), + QTC(0x78ce8828), QTC(0x78e6feef), QTC(0x78ff740f), QTC(0x7917e78a), + QTC(0x7930595f), QTC(0x7948c990), QTC(0x7961381d), QTC(0x7979a506), + QTC(0x7992104c), QTC(0x79aa79f0), QTC(0x79c2e1f1), QTC(0x79db4852), + QTC(0x79f3ad11), QTC(0x7a0c1031), QTC(0x7a2471b0), QTC(0x7a3cd191), + QTC(0x7a552fd3), QTC(0x7a6d8c76), QTC(0x7a85e77d), QTC(0x7a9e40e6), + QTC(0x7ab698b2), QTC(0x7aceeee3), QTC(0x7ae74378), QTC(0x7aff9673), + QTC(0x7b17e7d2), QTC(0x7b303799), QTC(0x7b4885c5), QTC(0x7b60d259), + QTC(0x7b791d55), QTC(0x7b9166b9), QTC(0x7ba9ae86), QTC(0x7bc1f4bc), + QTC(0x7bda395c), QTC(0x7bf27c66), QTC(0x7c0abddb), QTC(0x7c22fdbb), + QTC(0x7c3b3c07), QTC(0x7c5378c0), QTC(0x7c6bb3e5), QTC(0x7c83ed78), + QTC(0x7c9c2579), QTC(0x7cb45be9), QTC(0x7ccc90c7), QTC(0x7ce4c414), + QTC(0x7cfcf5d2), QTC(0x7d152600), QTC(0x7d2d549f), QTC(0x7d4581b0), + QTC(0x7d5dad32), QTC(0x7d75d727), QTC(0x7d8dff8f), QTC(0x7da6266a), + QTC(0x7dbe4bba), QTC(0x7dd66f7d), QTC(0x7dee91b6), QTC(0x7e06b264), + QTC(0x7e1ed188), QTC(0x7e36ef22), QTC(0x7e4f0b34), QTC(0x7e6725bd), + QTC(0x7e7f3ebd), QTC(0x7e975636), QTC(0x7eaf6c28), QTC(0x7ec78093), + QTC(0x7edf9378), QTC(0x7ef7a4d7), QTC(0x7f0fb4b1), QTC(0x7f27c307), + QTC(0x7f3fcfd8), QTC(0x7f57db25), QTC(0x7f6fe4ef), QTC(0x7f87ed36), + QTC(0x7f9ff3fb), QTC(0x7fb7f93e), QTC(0x7fcffcff), QTC(0x7fe7ff40)}; + +/* + table of pow(2.0,0.25*q)/2.0, q[0..4) +*/ +const FIXP_QTD FDKaacEnc_quantTableQ[4] = {QTC(0x40000000), QTC(0x4c1bf7ff), + QTC(0x5a82797f), QTC(0x6ba27e7f)}; + +/* + table of pow(2.0,0.75*e)/8.0, e[0..4) +*/ +const FIXP_QTD FDKaacEnc_quantTableE[4] = {QTC(0x10000000), QTC(0x1ae89f99), + QTC(0x2d413ccd), QTC(0x4c1bf828)}; + +/* + table to count used number of bits +*/ +const SHORT FDKaacEnc_sideInfoTabLong[] = { + 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, + 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, + 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, 0x0009, + 0x0009, 0x0009, 0x0009, 0x0009, 0x000e, 0x000e, 0x000e, 0x000e, 0x000e, + 0x000e, 0x000e, 0x000e, 0x000e, 0x000e, 0x000e, 0x000e, 0x000e, 0x000e, + 0x000e, 0x000e, 0x000e, 0x000e, 0x000e, 0x000e, 0x000e}; + +const SHORT FDKaacEnc_sideInfoTabShort[] = { + 0x0007, 0x0007, 0x0007, 0x0007, 0x0007, 0x0007, 0x0007, 0x000a, + 0x000a, 0x000a, 0x000a, 0x000a, 0x000a, 0x000a, 0x000d, 0x000d}; + +/* + Psy Configuration constants +*/ + +const SFB_PARAM_LONG p_FDKaacEnc_8000_long_1024 = { + 40, {12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 16, + 16, 16, 16, 16, 16, 16, 20, 20, 20, 20, 24, 24, 24, 28, + 28, 32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 80}}; +const SFB_PARAM_SHORT p_FDKaacEnc_8000_short_128 = { + 15, {4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 20}}; + +const SFB_PARAM_LONG p_FDKaacEnc_11025_long_1024 = { + 43, {8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 12, 12, 12, 12, + 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24, 24, 28, + 28, 32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64, 64}}; +const SFB_PARAM_SHORT p_FDKaacEnc_11025_short_128 = { + 15, {4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20}}; + +const SFB_PARAM_LONG p_FDKaacEnc_12000_long_1024 = { + 43, {8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 12, 12, 12, 12, + 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24, 24, 28, + 28, 32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64, 64}}; +const SFB_PARAM_SHORT p_FDKaacEnc_12000_short_128 = { + 15, {4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20}}; + +const SFB_PARAM_LONG p_FDKaacEnc_16000_long_1024 = { + 43, {8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 12, 12, 12, 12, + 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24, 24, 28, + 28, 32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64, 64}}; +const SFB_PARAM_SHORT p_FDKaacEnc_16000_short_128 = { + 15, {4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 20}}; +const SFB_PARAM_LONG p_FDKaacEnc_22050_long_1024 = { + 47, {4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, + 8, 8, 8, 8, 8, 12, 12, 12, 12, 16, 16, 16, 20, 20, 24, 24, + 28, 28, 32, 36, 36, 40, 44, 48, 52, 52, 64, 64, 64, 64, 64}}; +const SFB_PARAM_SHORT p_FDKaacEnc_22050_short_128 = { + 15, {4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 20}}; +const SFB_PARAM_LONG p_FDKaacEnc_24000_long_1024 = { + 47, {4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, + 8, 8, 8, 8, 8, 12, 12, 12, 12, 16, 16, 16, 20, 20, 24, 24, + 28, 28, 32, 36, 36, 40, 44, 48, 52, 52, 64, 64, 64, 64, 64}}; +const SFB_PARAM_SHORT p_FDKaacEnc_24000_short_128 = { + 15, {4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 20}}; +const SFB_PARAM_LONG p_FDKaacEnc_32000_long_1024 = { + 51, {4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, + 12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28, 32, 32, 32, 32, 32, + 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32}}; +const SFB_PARAM_SHORT p_FDKaacEnc_32000_short_128 = { + 14, {4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16}}; +const SFB_PARAM_LONG p_FDKaacEnc_44100_long_1024 = { + 49, {4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, + 12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28, 32, 32, 32, 32, 32, + 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 96}}; +const SFB_PARAM_SHORT p_FDKaacEnc_44100_short_128 = { + 14, {4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16}}; +const SFB_PARAM_LONG p_FDKaacEnc_48000_long_1024 = { + 49, {4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, + 12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, 28, 32, 32, 32, 32, 32, + 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 96}}; +const SFB_PARAM_SHORT p_FDKaacEnc_48000_short_128 = { + 14, {4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 16}}; +const SFB_PARAM_LONG p_FDKaacEnc_64000_long_1024 = { + 47, {4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, + 8, 8, 12, 12, 12, 16, 16, 16, 20, 24, 24, 28, 36, 40, 40, 40, + 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40}}; +const SFB_PARAM_SHORT p_FDKaacEnc_64000_short_128 = { + 12, {4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36}}; +const SFB_PARAM_LONG p_FDKaacEnc_88200_long_1024 = { + 41, {4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, + 8, 8, 8, 8, 8, 12, 12, 12, 12, 12, 16, 16, 24, 28, + 36, 44, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64}}; +const SFB_PARAM_SHORT p_FDKaacEnc_88200_short_128 = { + 12, {4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36}}; +const SFB_PARAM_LONG p_FDKaacEnc_96000_long_1024 = { + 41, {4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, + 8, 8, 8, 8, 8, 12, 12, 12, 12, 12, 16, 16, 24, 28, + 36, 44, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64}}; +const SFB_PARAM_SHORT p_FDKaacEnc_96000_short_128 = { + 12, {4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 36}}; + +/* + TNS filter coefficients +*/ + +/* + 3 bit resolution +*/ +const FIXP_LPC FDKaacEnc_tnsEncCoeff3[8] = { + FX_DBL2FXCONST_LPC(0x81f1d201), FX_DBL2FXCONST_LPC(0x91261481), + FX_DBL2FXCONST_LPC(0xadb92301), FX_DBL2FXCONST_LPC(0xd438af00), + FX_DBL2FXCONST_LPC(0x00000000), FX_DBL2FXCONST_LPC(0x37898080), + FX_DBL2FXCONST_LPC(0x64130dff), FX_DBL2FXCONST_LPC(0x7cca6fff)}; +const FIXP_LPC FDKaacEnc_tnsCoeff3Borders[8] = { + FX_DBL2FXCONST_LPC(0x80000001) /*-4*/, + FX_DBL2FXCONST_LPC(0x87b826df) /*-3*/, + FX_DBL2FXCONST_LPC(0x9df24154) /*-2*/, + FX_DBL2FXCONST_LPC(0xbfffffe5) /*-1*/, + FX_DBL2FXCONST_LPC(0xe9c5e578) /* 0*/, + FX_DBL2FXCONST_LPC(0x1c7b90f0) /* 1*/, + FX_DBL2FXCONST_LPC(0x4fce83a9) /* 2*/, + FX_DBL2FXCONST_LPC(0x7352f2c3) /* 3*/ +}; + +/* + 4 bit resolution +*/ +const FIXP_LPC FDKaacEnc_tnsEncCoeff4[16] = { + FX_DBL2FXCONST_LPC(0x808bc881), FX_DBL2FXCONST_LPC(0x84e2e581), + FX_DBL2FXCONST_LPC(0x8d6b4a01), FX_DBL2FXCONST_LPC(0x99da9201), + FX_DBL2FXCONST_LPC(0xa9c45701), FX_DBL2FXCONST_LPC(0xbc9dde81), + FX_DBL2FXCONST_LPC(0xd1c2d500), FX_DBL2FXCONST_LPC(0xe87ae540), + FX_DBL2FXCONST_LPC(0x00000000), FX_DBL2FXCONST_LPC(0x1a9cd9c0), + FX_DBL2FXCONST_LPC(0x340ff240), FX_DBL2FXCONST_LPC(0x4b3c8bff), + FX_DBL2FXCONST_LPC(0x5f1f5e7f), FX_DBL2FXCONST_LPC(0x6ed9eb7f), + FX_DBL2FXCONST_LPC(0x79bc387f), FX_DBL2FXCONST_LPC(0x7f4c7e7f)}; +const FIXP_LPC FDKaacEnc_tnsCoeff4Borders[16] = { + FX_DBL2FXCONST_LPC(0x80000001) /*-8*/, + FX_DBL2FXCONST_LPC(0x822deff0) /*-7*/, + FX_DBL2FXCONST_LPC(0x88a4bfe6) /*-6*/, + FX_DBL2FXCONST_LPC(0x932c159d) /*-5*/, + FX_DBL2FXCONST_LPC(0xa16827c2) /*-4*/, + FX_DBL2FXCONST_LPC(0xb2dcde27) /*-3*/, + FX_DBL2FXCONST_LPC(0xc6f20b91) /*-2*/, + FX_DBL2FXCONST_LPC(0xdcf89c64) /*-1*/, + FX_DBL2FXCONST_LPC(0xf4308ce1) /* 0*/, + FX_DBL2FXCONST_LPC(0x0d613054) /* 1*/, + FX_DBL2FXCONST_LPC(0x278dde80) /* 2*/, + FX_DBL2FXCONST_LPC(0x4000001b) /* 3*/, + FX_DBL2FXCONST_LPC(0x55a6127b) /* 4*/, + FX_DBL2FXCONST_LPC(0x678dde8f) /* 5*/, + FX_DBL2FXCONST_LPC(0x74ef0ed7) /* 6*/, + FX_DBL2FXCONST_LPC(0x7d33f0da) /* 7*/ +}; +const FIXP_DBL FDKaacEnc_mTab_4_3Elc[512] = { + FL2FXCONST_DBL(0.3968502629920499), FL2FXCONST_DBL(0.3978840634868335), + FL2FXCONST_DBL(0.3989185359354711), FL2FXCONST_DBL(0.3999536794661432), + FL2FXCONST_DBL(0.4009894932098531), FL2FXCONST_DBL(0.4020259763004115), + FL2FXCONST_DBL(0.4030631278744227), FL2FXCONST_DBL(0.4041009470712695), + FL2FXCONST_DBL(0.4051394330330996), FL2FXCONST_DBL(0.4061785849048110), + FL2FXCONST_DBL(0.4072184018340380), FL2FXCONST_DBL(0.4082588829711372), + FL2FXCONST_DBL(0.4093000274691739), FL2FXCONST_DBL(0.4103418344839078), + FL2FXCONST_DBL(0.4113843031737798), FL2FXCONST_DBL(0.4124274326998980), + FL2FXCONST_DBL(0.4134712222260245), FL2FXCONST_DBL(0.4145156709185620), + FL2FXCONST_DBL(0.4155607779465400), FL2FXCONST_DBL(0.4166065424816022), + FL2FXCONST_DBL(0.4176529636979932), FL2FXCONST_DBL(0.4187000407725452), + FL2FXCONST_DBL(0.4197477728846652), FL2FXCONST_DBL(0.4207961592163222), + FL2FXCONST_DBL(0.4218451989520345), FL2FXCONST_DBL(0.4228948912788567), + FL2FXCONST_DBL(0.4239452353863673), FL2FXCONST_DBL(0.4249962304666564), + FL2FXCONST_DBL(0.4260478757143130), FL2FXCONST_DBL(0.4271001703264124), + FL2FXCONST_DBL(0.4281531135025046), FL2FXCONST_DBL(0.4292067044446017), + FL2FXCONST_DBL(0.4302609423571658), FL2FXCONST_DBL(0.4313158264470970), + FL2FXCONST_DBL(0.4323713559237216), FL2FXCONST_DBL(0.4334275299987803), + FL2FXCONST_DBL(0.4344843478864161), FL2FXCONST_DBL(0.4355418088031630), + FL2FXCONST_DBL(0.4365999119679339), FL2FXCONST_DBL(0.4376586566020096), + FL2FXCONST_DBL(0.4387180419290272), FL2FXCONST_DBL(0.4397780671749683), + FL2FXCONST_DBL(0.4408387315681480), FL2FXCONST_DBL(0.4419000343392039), + FL2FXCONST_DBL(0.4429619747210847), FL2FXCONST_DBL(0.4440245519490388), + FL2FXCONST_DBL(0.4450877652606038), FL2FXCONST_DBL(0.4461516138955953), + FL2FXCONST_DBL(0.4472160970960963), FL2FXCONST_DBL(0.4482812141064458), + FL2FXCONST_DBL(0.4493469641732286), FL2FXCONST_DBL(0.4504133465452648), + FL2FXCONST_DBL(0.4514803604735984), FL2FXCONST_DBL(0.4525480052114875), + FL2FXCONST_DBL(0.4536162800143939), FL2FXCONST_DBL(0.4546851841399719), + FL2FXCONST_DBL(0.4557547168480591), FL2FXCONST_DBL(0.4568248774006652), + FL2FXCONST_DBL(0.4578956650619623), FL2FXCONST_DBL(0.4589670790982746), + FL2FXCONST_DBL(0.4600391187780688), FL2FXCONST_DBL(0.4611117833719430), + FL2FXCONST_DBL(0.4621850721526184), FL2FXCONST_DBL(0.4632589843949278), + FL2FXCONST_DBL(0.4643335193758069), FL2FXCONST_DBL(0.4654086763742842), + FL2FXCONST_DBL(0.4664844546714713), FL2FXCONST_DBL(0.4675608535505532), + FL2FXCONST_DBL(0.4686378722967790), FL2FXCONST_DBL(0.4697155101974522), + FL2FXCONST_DBL(0.4707937665419216), FL2FXCONST_DBL(0.4718726406215713), + FL2FXCONST_DBL(0.4729521317298118), FL2FXCONST_DBL(0.4740322391620711), + FL2FXCONST_DBL(0.4751129622157845), FL2FXCONST_DBL(0.4761943001903867), + FL2FXCONST_DBL(0.4772762523873015), FL2FXCONST_DBL(0.4783588181099338), + FL2FXCONST_DBL(0.4794419966636599), FL2FXCONST_DBL(0.4805257873558190), + FL2FXCONST_DBL(0.4816101894957042), FL2FXCONST_DBL(0.4826952023945537), + FL2FXCONST_DBL(0.4837808253655421), FL2FXCONST_DBL(0.4848670577237714), + FL2FXCONST_DBL(0.4859538987862632), FL2FXCONST_DBL(0.4870413478719488), + FL2FXCONST_DBL(0.4881294043016621), FL2FXCONST_DBL(0.4892180673981298), + FL2FXCONST_DBL(0.4903073364859640), FL2FXCONST_DBL(0.4913972108916533), + FL2FXCONST_DBL(0.4924876899435545), FL2FXCONST_DBL(0.4935787729718844), + FL2FXCONST_DBL(0.4946704593087116), FL2FXCONST_DBL(0.4957627482879484), + FL2FXCONST_DBL(0.4968556392453423), FL2FXCONST_DBL(0.4979491315184684), + FL2FXCONST_DBL(0.4990432244467211), FL2FXCONST_DBL(0.5001379173713062), + FL2FXCONST_DBL(0.5012332096352328), FL2FXCONST_DBL(0.5023291005833056), + FL2FXCONST_DBL(0.5034255895621171), FL2FXCONST_DBL(0.5045226759200399), + FL2FXCONST_DBL(0.5056203590072181), FL2FXCONST_DBL(0.5067186381755611), + FL2FXCONST_DBL(0.5078175127787346), FL2FXCONST_DBL(0.5089169821721536), + FL2FXCONST_DBL(0.5100170457129749), FL2FXCONST_DBL(0.5111177027600893), + FL2FXCONST_DBL(0.5122189526741143), FL2FXCONST_DBL(0.5133207948173868), + FL2FXCONST_DBL(0.5144232285539552), FL2FXCONST_DBL(0.5155262532495726), + FL2FXCONST_DBL(0.5166298682716894), FL2FXCONST_DBL(0.5177340729894460), + FL2FXCONST_DBL(0.5188388667736652), FL2FXCONST_DBL(0.5199442489968457), + FL2FXCONST_DBL(0.5210502190331544), FL2FXCONST_DBL(0.5221567762584198), + FL2FXCONST_DBL(0.5232639200501247), FL2FXCONST_DBL(0.5243716497873989), + FL2FXCONST_DBL(0.5254799648510130), FL2FXCONST_DBL(0.5265888646233705), + FL2FXCONST_DBL(0.5276983484885021), FL2FXCONST_DBL(0.5288084158320574), + FL2FXCONST_DBL(0.5299190660412995), FL2FXCONST_DBL(0.5310302985050975), + FL2FXCONST_DBL(0.5321421126139198), FL2FXCONST_DBL(0.5332545077598274), + FL2FXCONST_DBL(0.5343674833364678), FL2FXCONST_DBL(0.5354810387390675), + FL2FXCONST_DBL(0.5365951733644262), FL2FXCONST_DBL(0.5377098866109097), + FL2FXCONST_DBL(0.5388251778784438), FL2FXCONST_DBL(0.5399410465685075), + FL2FXCONST_DBL(0.5410574920841272), FL2FXCONST_DBL(0.5421745138298695), + FL2FXCONST_DBL(0.5432921112118353), FL2FXCONST_DBL(0.5444102836376534), + FL2FXCONST_DBL(0.5455290305164744), FL2FXCONST_DBL(0.5466483512589642), + FL2FXCONST_DBL(0.5477682452772976), FL2FXCONST_DBL(0.5488887119851529), + FL2FXCONST_DBL(0.5500097507977050), FL2FXCONST_DBL(0.5511313611316194), + FL2FXCONST_DBL(0.5522535424050467), FL2FXCONST_DBL(0.5533762940376158), + FL2FXCONST_DBL(0.5544996154504284), FL2FXCONST_DBL(0.5556235060660528), + FL2FXCONST_DBL(0.5567479653085183), FL2FXCONST_DBL(0.5578729926033087), + FL2FXCONST_DBL(0.5589985873773569), FL2FXCONST_DBL(0.5601247490590389), + FL2FXCONST_DBL(0.5612514770781683), FL2FXCONST_DBL(0.5623787708659898), + FL2FXCONST_DBL(0.5635066298551742), FL2FXCONST_DBL(0.5646350534798125), + FL2FXCONST_DBL(0.5657640411754097), FL2FXCONST_DBL(0.5668935923788799), + FL2FXCONST_DBL(0.5680237065285404), FL2FXCONST_DBL(0.5691543830641059), + FL2FXCONST_DBL(0.5702856214266832), FL2FXCONST_DBL(0.5714174210587655), + FL2FXCONST_DBL(0.5725497814042271), FL2FXCONST_DBL(0.5736827019083177), + FL2FXCONST_DBL(0.5748161820176573), FL2FXCONST_DBL(0.5759502211802304), + FL2FXCONST_DBL(0.5770848188453810), FL2FXCONST_DBL(0.5782199744638067), + FL2FXCONST_DBL(0.5793556874875542), FL2FXCONST_DBL(0.5804919573700131), + FL2FXCONST_DBL(0.5816287835659116), FL2FXCONST_DBL(0.5827661655313104), + FL2FXCONST_DBL(0.5839041027235979), FL2FXCONST_DBL(0.5850425946014850), + FL2FXCONST_DBL(0.5861816406250000), FL2FXCONST_DBL(0.5873212402554834), + FL2FXCONST_DBL(0.5884613929555826), FL2FXCONST_DBL(0.5896020981892474), + FL2FXCONST_DBL(0.5907433554217242), FL2FXCONST_DBL(0.5918851641195517), + FL2FXCONST_DBL(0.5930275237505556), FL2FXCONST_DBL(0.5941704337838434), + FL2FXCONST_DBL(0.5953138936897999), FL2FXCONST_DBL(0.5964579029400819), + FL2FXCONST_DBL(0.5976024610076139), FL2FXCONST_DBL(0.5987475673665825), + FL2FXCONST_DBL(0.5998932214924321), FL2FXCONST_DBL(0.6010394228618597), + FL2FXCONST_DBL(0.6021861709528106), FL2FXCONST_DBL(0.6033334652444733), + FL2FXCONST_DBL(0.6044813052172748), FL2FXCONST_DBL(0.6056296903528761), + FL2FXCONST_DBL(0.6067786201341671), FL2FXCONST_DBL(0.6079280940452625), + FL2FXCONST_DBL(0.6090781115714966), FL2FXCONST_DBL(0.6102286721994192), + FL2FXCONST_DBL(0.6113797754167908), FL2FXCONST_DBL(0.6125314207125777), + FL2FXCONST_DBL(0.6136836075769482), FL2FXCONST_DBL(0.6148363355012674), + FL2FXCONST_DBL(0.6159896039780929), FL2FXCONST_DBL(0.6171434125011708), + FL2FXCONST_DBL(0.6182977605654305), FL2FXCONST_DBL(0.6194526476669808), + FL2FXCONST_DBL(0.6206080733031054), FL2FXCONST_DBL(0.6217640369722584), + FL2FXCONST_DBL(0.6229205381740598), FL2FXCONST_DBL(0.6240775764092919), + FL2FXCONST_DBL(0.6252351511798939), FL2FXCONST_DBL(0.6263932619889586), + FL2FXCONST_DBL(0.6275519083407275), FL2FXCONST_DBL(0.6287110897405869), + FL2FXCONST_DBL(0.6298708056950635), FL2FXCONST_DBL(0.6310310557118203), + FL2FXCONST_DBL(0.6321918392996523), FL2FXCONST_DBL(0.6333531559684823), + FL2FXCONST_DBL(0.6345150052293571), FL2FXCONST_DBL(0.6356773865944432), + FL2FXCONST_DBL(0.6368402995770224), FL2FXCONST_DBL(0.6380037436914881), + FL2FXCONST_DBL(0.6391677184533411), FL2FXCONST_DBL(0.6403322233791856), + FL2FXCONST_DBL(0.6414972579867254), FL2FXCONST_DBL(0.6426628217947594), + FL2FXCONST_DBL(0.6438289143231779), FL2FXCONST_DBL(0.6449955350929588), + FL2FXCONST_DBL(0.6461626836261636), FL2FXCONST_DBL(0.6473303594459330), + FL2FXCONST_DBL(0.6484985620764839), FL2FXCONST_DBL(0.6496672910431047), + FL2FXCONST_DBL(0.6508365458721518), FL2FXCONST_DBL(0.6520063260910459), + FL2FXCONST_DBL(0.6531766312282679), FL2FXCONST_DBL(0.6543474608133552), + FL2FXCONST_DBL(0.6555188143768979), FL2FXCONST_DBL(0.6566906914505349), + FL2FXCONST_DBL(0.6578630915669509), FL2FXCONST_DBL(0.6590360142598715), + FL2FXCONST_DBL(0.6602094590640603), FL2FXCONST_DBL(0.6613834255153149), + FL2FXCONST_DBL(0.6625579131504635), FL2FXCONST_DBL(0.6637329215073610), + FL2FXCONST_DBL(0.6649084501248851), FL2FXCONST_DBL(0.6660844985429335), + FL2FXCONST_DBL(0.6672610663024197), FL2FXCONST_DBL(0.6684381529452691), + FL2FXCONST_DBL(0.6696157580144163), FL2FXCONST_DBL(0.6707938810538011), + FL2FXCONST_DBL(0.6719725216083646), FL2FXCONST_DBL(0.6731516792240465), + FL2FXCONST_DBL(0.6743313534477807), FL2FXCONST_DBL(0.6755115438274927), + FL2FXCONST_DBL(0.6766922499120955), FL2FXCONST_DBL(0.6778734712514865), + FL2FXCONST_DBL(0.6790552073965435), FL2FXCONST_DBL(0.6802374578991223), + FL2FXCONST_DBL(0.6814202223120524), FL2FXCONST_DBL(0.6826035001891340), + FL2FXCONST_DBL(0.6837872910851345), FL2FXCONST_DBL(0.6849715945557853), + FL2FXCONST_DBL(0.6861564101577784), FL2FXCONST_DBL(0.6873417374487629), + FL2FXCONST_DBL(0.6885275759873420), FL2FXCONST_DBL(0.6897139253330697), + FL2FXCONST_DBL(0.6909007850464473), FL2FXCONST_DBL(0.6920881546889198), + FL2FXCONST_DBL(0.6932760338228737), FL2FXCONST_DBL(0.6944644220116332), + FL2FXCONST_DBL(0.6956533188194565), FL2FXCONST_DBL(0.6968427238115332), + FL2FXCONST_DBL(0.6980326365539813), FL2FXCONST_DBL(0.6992230566138435), + FL2FXCONST_DBL(0.7004139835590845), FL2FXCONST_DBL(0.7016054169585869), + FL2FXCONST_DBL(0.7027973563821499), FL2FXCONST_DBL(0.7039898014004843), + FL2FXCONST_DBL(0.7051827515852106), FL2FXCONST_DBL(0.7063762065088554), + FL2FXCONST_DBL(0.7075701657448483), FL2FXCONST_DBL(0.7087646288675196), + FL2FXCONST_DBL(0.7099595954520960), FL2FXCONST_DBL(0.7111550650746988), + FL2FXCONST_DBL(0.7123510373123402), FL2FXCONST_DBL(0.7135475117429202), + FL2FXCONST_DBL(0.7147444879452244), FL2FXCONST_DBL(0.7159419654989200), + FL2FXCONST_DBL(0.7171399439845538), FL2FXCONST_DBL(0.7183384229835486), + FL2FXCONST_DBL(0.7195374020782005), FL2FXCONST_DBL(0.7207368808516762), + FL2FXCONST_DBL(0.7219368588880097), FL2FXCONST_DBL(0.7231373357720997), + FL2FXCONST_DBL(0.7243383110897066), FL2FXCONST_DBL(0.7255397844274496), + FL2FXCONST_DBL(0.7267417553728043), FL2FXCONST_DBL(0.7279442235140992), + FL2FXCONST_DBL(0.7291471884405130), FL2FXCONST_DBL(0.7303506497420724), + FL2FXCONST_DBL(0.7315546070096487), FL2FXCONST_DBL(0.7327590598349553), + FL2FXCONST_DBL(0.7339640078105445), FL2FXCONST_DBL(0.7351694505298055), + FL2FXCONST_DBL(0.7363753875869610), FL2FXCONST_DBL(0.7375818185770647), + FL2FXCONST_DBL(0.7387887430959987), FL2FXCONST_DBL(0.7399961607404706), + FL2FXCONST_DBL(0.7412040711080108), FL2FXCONST_DBL(0.7424124737969701), + FL2FXCONST_DBL(0.7436213684065166), FL2FXCONST_DBL(0.7448307545366334), + FL2FXCONST_DBL(0.7460406317881158), FL2FXCONST_DBL(0.7472509997625686), + FL2FXCONST_DBL(0.7484618580624036), FL2FXCONST_DBL(0.7496732062908372), + FL2FXCONST_DBL(0.7508850440518872), FL2FXCONST_DBL(0.7520973709503704), + FL2FXCONST_DBL(0.7533101865919009), FL2FXCONST_DBL(0.7545234905828862), + FL2FXCONST_DBL(0.7557372825305252), FL2FXCONST_DBL(0.7569515620428062), + FL2FXCONST_DBL(0.7581663287285035), FL2FXCONST_DBL(0.7593815821971756), + FL2FXCONST_DBL(0.7605973220591619), FL2FXCONST_DBL(0.7618135479255810), + FL2FXCONST_DBL(0.7630302594083277), FL2FXCONST_DBL(0.7642474561200708), + FL2FXCONST_DBL(0.7654651376742505), FL2FXCONST_DBL(0.7666833036850760), + FL2FXCONST_DBL(0.7679019537675227), FL2FXCONST_DBL(0.7691210875373307), + FL2FXCONST_DBL(0.7703407046110011), FL2FXCONST_DBL(0.7715608046057948), + FL2FXCONST_DBL(0.7727813871397293), FL2FXCONST_DBL(0.7740024518315765), + FL2FXCONST_DBL(0.7752239983008605), FL2FXCONST_DBL(0.7764460261678551), + FL2FXCONST_DBL(0.7776685350535814), FL2FXCONST_DBL(0.7788915245798054), + FL2FXCONST_DBL(0.7801149943690360), FL2FXCONST_DBL(0.7813389440445223), + FL2FXCONST_DBL(0.7825633732302513), FL2FXCONST_DBL(0.7837882815509458), + FL2FXCONST_DBL(0.7850136686320621), FL2FXCONST_DBL(0.7862395340997874), + FL2FXCONST_DBL(0.7874658775810378), FL2FXCONST_DBL(0.7886926987034559), + FL2FXCONST_DBL(0.7899199970954088), FL2FXCONST_DBL(0.7911477723859853), + FL2FXCONST_DBL(0.7923760242049944), FL2FXCONST_DBL(0.7936047521829623), + FL2FXCONST_DBL(0.7948339559511308), FL2FXCONST_DBL(0.7960636351414546), + FL2FXCONST_DBL(0.7972937893865995), FL2FXCONST_DBL(0.7985244183199399), + FL2FXCONST_DBL(0.7997555215755570), FL2FXCONST_DBL(0.8009870987882359), + FL2FXCONST_DBL(0.8022191495934644), FL2FXCONST_DBL(0.8034516736274301), + FL2FXCONST_DBL(0.8046846705270185), FL2FXCONST_DBL(0.8059181399298110), + FL2FXCONST_DBL(0.8071520814740822), FL2FXCONST_DBL(0.8083864947987989), + FL2FXCONST_DBL(0.8096213795436166), FL2FXCONST_DBL(0.8108567353488784), + FL2FXCONST_DBL(0.8120925618556127), FL2FXCONST_DBL(0.8133288587055308), + FL2FXCONST_DBL(0.8145656255410253), FL2FXCONST_DBL(0.8158028620051674), + FL2FXCONST_DBL(0.8170405677417053), FL2FXCONST_DBL(0.8182787423950622), + FL2FXCONST_DBL(0.8195173856103341), FL2FXCONST_DBL(0.8207564970332875), + FL2FXCONST_DBL(0.8219960763103580), FL2FXCONST_DBL(0.8232361230886477), + FL2FXCONST_DBL(0.8244766370159234), FL2FXCONST_DBL(0.8257176177406150), + FL2FXCONST_DBL(0.8269590649118125), FL2FXCONST_DBL(0.8282009781792650), + FL2FXCONST_DBL(0.8294433571933784), FL2FXCONST_DBL(0.8306862016052132), + FL2FXCONST_DBL(0.8319295110664831), FL2FXCONST_DBL(0.8331732852295520), + FL2FXCONST_DBL(0.8344175237474336), FL2FXCONST_DBL(0.8356622262737878), + FL2FXCONST_DBL(0.8369073924629202), FL2FXCONST_DBL(0.8381530219697793), + FL2FXCONST_DBL(0.8393991144499545), FL2FXCONST_DBL(0.8406456695596752), + FL2FXCONST_DBL(0.8418926869558079), FL2FXCONST_DBL(0.8431401662958544), + FL2FXCONST_DBL(0.8443881072379507), FL2FXCONST_DBL(0.8456365094408642), + FL2FXCONST_DBL(0.8468853725639923), FL2FXCONST_DBL(0.8481346962673606), + FL2FXCONST_DBL(0.8493844802116208), FL2FXCONST_DBL(0.8506347240580492), + FL2FXCONST_DBL(0.8518854274685442), FL2FXCONST_DBL(0.8531365901056253), + FL2FXCONST_DBL(0.8543882116324307), FL2FXCONST_DBL(0.8556402917127157), + FL2FXCONST_DBL(0.8568928300108512), FL2FXCONST_DBL(0.8581458261918209), + FL2FXCONST_DBL(0.8593992799212207), FL2FXCONST_DBL(0.8606531908652563), + FL2FXCONST_DBL(0.8619075586907414), FL2FXCONST_DBL(0.8631623830650962), + FL2FXCONST_DBL(0.8644176636563452), FL2FXCONST_DBL(0.8656734001331161), + FL2FXCONST_DBL(0.8669295921646375), FL2FXCONST_DBL(0.8681862394207371), + FL2FXCONST_DBL(0.8694433415718407), FL2FXCONST_DBL(0.8707008982889695), + FL2FXCONST_DBL(0.8719589092437391), FL2FXCONST_DBL(0.8732173741083574), + FL2FXCONST_DBL(0.8744762925556232), FL2FXCONST_DBL(0.8757356642589241), + FL2FXCONST_DBL(0.8769954888922352), FL2FXCONST_DBL(0.8782557661301171), + FL2FXCONST_DBL(0.8795164956477146), FL2FXCONST_DBL(0.8807776771207545), + FL2FXCONST_DBL(0.8820393102255443), FL2FXCONST_DBL(0.8833013946389704), + FL2FXCONST_DBL(0.8845639300384969), FL2FXCONST_DBL(0.8858269161021629), + FL2FXCONST_DBL(0.8870903525085819), FL2FXCONST_DBL(0.8883542389369399), + FL2FXCONST_DBL(0.8896185750669933), FL2FXCONST_DBL(0.8908833605790678), + FL2FXCONST_DBL(0.8921485951540565), FL2FXCONST_DBL(0.8934142784734187), + FL2FXCONST_DBL(0.8946804102191776), FL2FXCONST_DBL(0.8959469900739191), + FL2FXCONST_DBL(0.8972140177207906), FL2FXCONST_DBL(0.8984814928434985), + FL2FXCONST_DBL(0.8997494151263077), FL2FXCONST_DBL(0.9010177842540390), + FL2FXCONST_DBL(0.9022865999120682), FL2FXCONST_DBL(0.9035558617863242), + FL2FXCONST_DBL(0.9048255695632878), FL2FXCONST_DBL(0.9060957229299895), + FL2FXCONST_DBL(0.9073663215740092), FL2FXCONST_DBL(0.9086373651834729), + FL2FXCONST_DBL(0.9099088534470528), FL2FXCONST_DBL(0.9111807860539647), + FL2FXCONST_DBL(0.9124531626939672), FL2FXCONST_DBL(0.9137259830573594), + FL2FXCONST_DBL(0.9149992468349805), FL2FXCONST_DBL(0.9162729537182071), + FL2FXCONST_DBL(0.9175471033989524), FL2FXCONST_DBL(0.9188216955696648), + FL2FXCONST_DBL(0.9200967299233258), FL2FXCONST_DBL(0.9213722061534494), + FL2FXCONST_DBL(0.9226481239540795), FL2FXCONST_DBL(0.9239244830197896), + FL2FXCONST_DBL(0.9252012830456805), FL2FXCONST_DBL(0.9264785237273793), + FL2FXCONST_DBL(0.9277562047610376), FL2FXCONST_DBL(0.9290343258433305), + FL2FXCONST_DBL(0.9303128866714547), FL2FXCONST_DBL(0.9315918869431275), + FL2FXCONST_DBL(0.9328713263565848), FL2FXCONST_DBL(0.9341512046105802), + FL2FXCONST_DBL(0.9354315214043836), FL2FXCONST_DBL(0.9367122764377792), + FL2FXCONST_DBL(0.9379934694110648), FL2FXCONST_DBL(0.9392751000250497), + FL2FXCONST_DBL(0.9405571679810542), FL2FXCONST_DBL(0.9418396729809072), + FL2FXCONST_DBL(0.9431226147269456), FL2FXCONST_DBL(0.9444059929220124), + FL2FXCONST_DBL(0.9456898072694558), FL2FXCONST_DBL(0.9469740574731275), + FL2FXCONST_DBL(0.9482587432373810), FL2FXCONST_DBL(0.9495438642670713), + FL2FXCONST_DBL(0.9508294202675522), FL2FXCONST_DBL(0.9521154109446763), + FL2FXCONST_DBL(0.9534018360047926), FL2FXCONST_DBL(0.9546886951547455), + FL2FXCONST_DBL(0.9559759881018738), FL2FXCONST_DBL(0.9572637145540087), + FL2FXCONST_DBL(0.9585518742194732), FL2FXCONST_DBL(0.9598404668070802), + FL2FXCONST_DBL(0.9611294920261317), FL2FXCONST_DBL(0.9624189495864168), + FL2FXCONST_DBL(0.9637088391982110), FL2FXCONST_DBL(0.9649991605722750), + FL2FXCONST_DBL(0.9662899134198524), FL2FXCONST_DBL(0.9675810974526697), + FL2FXCONST_DBL(0.9688727123829343), FL2FXCONST_DBL(0.9701647579233330), + FL2FXCONST_DBL(0.9714572337870316), FL2FXCONST_DBL(0.9727501396876727), + FL2FXCONST_DBL(0.9740434753393749), FL2FXCONST_DBL(0.9753372404567313), + FL2FXCONST_DBL(0.9766314347548087), FL2FXCONST_DBL(0.9779260579491460), + FL2FXCONST_DBL(0.9792211097557527), FL2FXCONST_DBL(0.9805165898911081), + FL2FXCONST_DBL(0.9818124980721600), FL2FXCONST_DBL(0.9831088340163232), + FL2FXCONST_DBL(0.9844055974414786), FL2FXCONST_DBL(0.9857027880659716), + FL2FXCONST_DBL(0.9870004056086111), FL2FXCONST_DBL(0.9882984497886684), + FL2FXCONST_DBL(0.9895969203258759), FL2FXCONST_DBL(0.9908958169404255), + FL2FXCONST_DBL(0.9921951393529680), FL2FXCONST_DBL(0.9934948872846116), + FL2FXCONST_DBL(0.9947950604569206), FL2FXCONST_DBL(0.9960956585919144), + FL2FXCONST_DBL(0.9973966814120665), FL2FXCONST_DBL(0.9986981286403025)}; + +const FIXP_DBL FDKaacEnc_specExpMantTableCombElc[4][14] = { + {FL2FXCONST_DBL(0.5000000000000000), FL2FXCONST_DBL(0.6299605249474366), + FL2FXCONST_DBL(0.7937005259840998), FL2FXCONST_DBL(0.5000000000000000), + FL2FXCONST_DBL(0.6299605249474366), FL2FXCONST_DBL(0.7937005259840998), + FL2FXCONST_DBL(0.5000000000000000), FL2FXCONST_DBL(0.6299605249474366), + FL2FXCONST_DBL(0.7937005259840998), FL2FXCONST_DBL(0.5000000000000000), + FL2FXCONST_DBL(0.6299605249474366), FL2FXCONST_DBL(0.7937005259840998), + FL2FXCONST_DBL(0.5000000000000000), FL2FXCONST_DBL(0.6299605249474366)}, + + {FL2FXCONST_DBL(0.5946035575013605), FL2FXCONST_DBL(0.7491535384383408), + FL2FXCONST_DBL(0.9438743126816935), FL2FXCONST_DBL(0.5946035575013605), + FL2FXCONST_DBL(0.7491535384383408), FL2FXCONST_DBL(0.9438743126816935), + FL2FXCONST_DBL(0.5946035575013605), FL2FXCONST_DBL(0.7491535384383408), + FL2FXCONST_DBL(0.9438743126816935), FL2FXCONST_DBL(0.5946035575013605), + FL2FXCONST_DBL(0.7491535384383408), FL2FXCONST_DBL(0.9438743126816935), + FL2FXCONST_DBL(0.5946035575013605), FL2FXCONST_DBL(0.7491535384383408)}, + + {FL2FXCONST_DBL(0.7071067811865476), FL2FXCONST_DBL(0.8908987181403393), + FL2FXCONST_DBL(0.5612310241546865), FL2FXCONST_DBL(0.7071067811865476), + FL2FXCONST_DBL(0.8908987181403393), FL2FXCONST_DBL(0.5612310241546865), + FL2FXCONST_DBL(0.7071067811865476), FL2FXCONST_DBL(0.8908987181403393), + FL2FXCONST_DBL(0.5612310241546865), FL2FXCONST_DBL(0.7071067811865476), + FL2FXCONST_DBL(0.8908987181403393), FL2FXCONST_DBL(0.5612310241546865), + FL2FXCONST_DBL(0.7071067811865476), FL2FXCONST_DBL(0.8908987181403393)}, + + {FL2FXCONST_DBL(0.8408964152537145), FL2FXCONST_DBL(0.5297315471796477), + FL2FXCONST_DBL(0.6674199270850172), FL2FXCONST_DBL(0.8408964152537145), + FL2FXCONST_DBL(0.5297315471796477), FL2FXCONST_DBL(0.6674199270850172), + FL2FXCONST_DBL(0.8408964152537145), FL2FXCONST_DBL(0.5297315471796477), + FL2FXCONST_DBL(0.6674199270850172), FL2FXCONST_DBL(0.8408964152537145), + FL2FXCONST_DBL(0.5297315471796477), FL2FXCONST_DBL(0.6674199270850172), + FL2FXCONST_DBL(0.8408964152537145), FL2FXCONST_DBL(0.5297315471796477)}}; + +const UCHAR FDKaacEnc_specExpTableComb[4][14] = { + {1, 2, 3, 5, 6, 7, 9, 10, 11, 13, 14, 15, 17, 18}, + {1, 2, 3, 5, 6, 7, 9, 10, 11, 13, 14, 15, 17, 18}, + {1, 2, 4, 5, 6, 8, 9, 10, 12, 13, 14, 16, 17, 18}, + {1, 3, 4, 5, 7, 8, 9, 11, 12, 13, 15, 16, 17, 19}}; + +#define WTS0 1 +#define WTS1 0 +#define WTS2 -2 + +const FIXP_WTB ELDAnalysis512[1536] = { + /* part 0 */ + WTC0(0xfac5a770), WTC0(0xfaafbab8), WTC0(0xfa996a40), WTC0(0xfa82bbd0), + WTC0(0xfa6bb538), WTC0(0xfa545c38), WTC0(0xfa3cb698), WTC0(0xfa24ca28), + WTC0(0xfa0c9ca8), WTC0(0xf9f433e8), WTC0(0xf9db9580), WTC0(0xf9c2c298), + WTC0(0xf9a9b800), WTC0(0xf9907250), WTC0(0xf976ee38), WTC0(0xf95d2b88), + WTC0(0xf9432d10), WTC0(0xf928f5c0), WTC0(0xf90e8868), WTC0(0xf8f3e400), + WTC0(0xf8d903a0), WTC0(0xf8bde238), WTC0(0xf8a27af0), WTC0(0xf886cde8), + WTC0(0xf86ae020), WTC0(0xf84eb6c0), WTC0(0xf83256f8), WTC0(0xf815c4b8), + WTC0(0xf7f902c0), WTC0(0xf7dc13b0), WTC0(0xf7befa60), WTC0(0xf7a1ba40), + WTC0(0xf78457c0), WTC0(0xf766d780), WTC0(0xf7493d90), WTC0(0xf72b8990), + WTC0(0xf70db5f0), WTC0(0xf6efbd30), WTC0(0xf6d19a20), WTC0(0xf6b352e0), + WTC0(0xf694f8c0), WTC0(0xf6769da0), WTC0(0xf6585310), WTC0(0xf63a28d0), + WTC0(0xf61c2c60), WTC0(0xf5fe6b10), WTC0(0xf5e0f250), WTC0(0xf5c3ceb0), + WTC0(0xf5a70be0), WTC0(0xf58ab5a0), WTC0(0xf56ed7b0), WTC0(0xf5537e40), + WTC0(0xf538b610), WTC0(0xf51e8bf0), WTC0(0xf5050c90), WTC0(0xf4ec4330), + WTC0(0xf4d439b0), WTC0(0xf4bcf9b0), WTC0(0xf4a68ce0), WTC0(0xf490fa80), + WTC0(0xf47c4760), WTC0(0xf4687830), WTC0(0xf4558f00), WTC0(0xf4434fc0), + WTC0(0xf4314070), WTC0(0xf41ee450), WTC0(0xf40bc130), WTC0(0xf3f799c0), + WTC0(0xf3e26d30), WTC0(0xf3cc3d70), WTC0(0xf3b50c80), WTC0(0xf39cdd60), + WTC0(0xf383b440), WTC0(0xf3699550), WTC0(0xf34e84c0), WTC0(0xf33286b0), + WTC0(0xf3159f10), WTC0(0xf2f7d1b0), WTC0(0xf2d92290), WTC0(0xf2b994d0), + WTC0(0xf2992ad0), WTC0(0xf277e6d0), WTC0(0xf255cb60), WTC0(0xf232dd00), + WTC0(0xf20f2240), WTC0(0xf1eaa1d0), WTC0(0xf1c56240), WTC0(0xf19f63d0), + WTC0(0xf178a0f0), WTC0(0xf15113a0), WTC0(0xf128b5c0), WTC0(0xf0ff7fd0), + WTC0(0xf0d56860), WTC0(0xf0aa6610), WTC0(0xf07e6fd0), WTC0(0xf0518190), + WTC0(0xf0239cd0), WTC0(0xeff4c320), WTC0(0xefc4f720), WTC0(0xef945080), + WTC0(0xef62fce0), WTC0(0xef312a40), WTC0(0xeeff05c0), WTC0(0xeecca2c0), + WTC0(0xee99faa0), WTC0(0xee6705a0), WTC0(0xee33bb60), WTC0(0xee000060), + WTC0(0xedcba660), WTC0(0xed967e80), WTC0(0xed605b80), WTC0(0xed293b40), + WTC0(0xecf146a0), WTC0(0xecb8a8a0), WTC0(0xec7f8bc0), WTC0(0xec461260), + WTC0(0xec0c5720), WTC0(0xebd27440), WTC0(0xeb988220), WTC0(0xeb5e7040), + WTC0(0xeb2404c0), WTC0(0xeae90440), WTC0(0xeaad33c0), WTC0(0xea7066c0), + WTC0(0xea327f60), WTC0(0xe9f36000), WTC0(0xe9b2ed60), WTC0(0xe9713920), + WTC0(0xe92e81e0), WTC0(0xe8eb08c0), WTC0(0xe8a70e60), WTC0(0xe862d8e0), + WTC0(0xe81eb340), WTC0(0xe7dae8a0), WTC0(0xe797c1a0), WTC0(0xe7554ca0), + WTC0(0xe7135dc0), WTC0(0xe6d1c6a0), WTC0(0xe6905720), WTC0(0xe64eb9c0), + WTC0(0xe60c7300), WTC0(0xe5c90600), WTC0(0xe583f920), WTC0(0xe53d1ce0), + WTC0(0xe4f48c80), WTC0(0xe4aa6640), WTC0(0xe45ecaa0), WTC0(0xe4120be0), + WTC0(0xe3c4ae60), WTC0(0xe3773860), WTC0(0xe32a2ea0), WTC0(0xe2ddeea0), + WTC0(0xe292af00), WTC0(0xe248a4a0), WTC0(0xe2000140), WTC0(0xe1b8b640), + WTC0(0xe1727440), WTC0(0xe12ce900), WTC0(0xe0e7c280), WTC0(0xe0a2b420), + WTC0(0xe05d76c0), WTC0(0xe017c360), WTC0(0xdfd15440), WTC0(0xdf8a0540), + WTC0(0xdf41d300), WTC0(0xdef8bb40), WTC0(0xdeaebd40), WTC0(0xde63e7c0), + WTC0(0xde185940), WTC0(0xddcc3180), WTC0(0xdd7f9000), WTC0(0xdd329e80), + WTC0(0xdce58e80), WTC0(0xdc989300), WTC0(0xdc4bde40), WTC0(0xdbff96c0), + WTC0(0xdbb3d780), WTC0(0xdb68bb80), WTC0(0xdb1e5c80), WTC0(0xdad4c380), + WTC0(0xda8be840), WTC0(0xda43c1c0), WTC0(0xd9fc4740), WTC0(0xd9b56640), + WTC0(0xd96f0440), WTC0(0xd9290600), WTC0(0xd8e35080), WTC0(0xd89dcd40), + WTC0(0xd8586b40), WTC0(0xd8131940), WTC0(0xd7cdc640), WTC0(0xd7886180), + WTC0(0xd742dc80), WTC0(0xd6fd2780), WTC0(0xd6b73400), WTC0(0xd670fd80), + WTC0(0xd62a8a40), WTC0(0xd5e3e080), WTC0(0xd59d0840), WTC0(0xd5562b80), + WTC0(0xd50f9540), WTC0(0xd4c992c0), WTC0(0xd4846f80), WTC0(0xd4405a80), + WTC0(0xd3fd6580), WTC0(0xd3bba140), WTC0(0xd37b1c80), WTC0(0xd33bb780), + WTC0(0xd2fd2400), WTC0(0xd2bf1240), WTC0(0xd2813300), WTC0(0xd2435ac0), + WTC0(0xd2057fc0), WTC0(0xd1c79a00), WTC0(0xd189a240), WTC0(0xd14b9dc0), + WTC0(0xd10d9e00), WTC0(0xd0cfb580), WTC0(0xd091f6c0), WTC0(0xd0548100), + WTC0(0xd0177f40), WTC0(0xcfdb1cc0), WTC0(0xcf9f84c0), WTC0(0xcf64d780), + WTC0(0xcf2b2b00), WTC0(0xcef29440), WTC0(0xcebb2640), WTC0(0xce84c000), + WTC0(0xce4f0bc0), WTC0(0xce19b200), WTC0(0xcde45d40), WTC0(0xcdaeedc0), + WTC0(0xcd7979c0), WTC0(0xcd4419c0), WTC0(0xcd0ee6c0), WTC0(0xccda0540), + WTC0(0xcca5a500), WTC0(0xcc71f640), WTC0(0xcc3f2800), WTC0(0xcc0d4300), + WTC0(0xcbdc2a00), WTC0(0xcbabbe80), WTC0(0xcb7be200), WTC0(0xcb4c8200), + WTC0(0xcb1d9800), WTC0(0xcaef1d40), WTC0(0xcac10bc0), WTC0(0xca936440), + WTC0(0xca662d00), WTC0(0xca396d40), WTC0(0xca0d2b80), WTC0(0xc9e16f80), + WTC0(0xc9b63f80), WTC0(0xc98ba2c0), WTC0(0xc961a000), WTC0(0xc9383ec0), + WTC0(0xc90a0440), WTC0(0xc8e0d280), WTC0(0xc8b73b80), WTC0(0xc88d4900), + WTC0(0xc86304c0), WTC0(0xc83878c0), WTC0(0xc80dae80), WTC0(0xc7e2afc0), + WTC0(0xc7b78640), WTC0(0xc78c3c40), WTC0(0xc760da80), WTC0(0xc7356640), + WTC0(0xc709de40), WTC0(0xc6de41c0), WTC0(0xc6b28fc0), WTC0(0xc686bd40), + WTC0(0xc65ab600), WTC0(0xc62e6580), WTC0(0xc601b880), WTC0(0xc5d4bac0), + WTC0(0xc5a79640), WTC0(0xc57a76c0), WTC0(0xc54d8780), WTC0(0xc520e840), + WTC0(0xc4f4acc0), WTC0(0xc4c8e880), WTC0(0xc49dad80), WTC0(0xc472e640), + WTC0(0xc44856c0), WTC0(0xc41dc140), WTC0(0xc3f2e940), WTC0(0xc3c7bc00), + WTC0(0xc39c4f00), WTC0(0xc370b9c0), WTC0(0xc34513c0), WTC0(0xc3197940), + WTC0(0xc2ee0a00), WTC0(0xc2c2e640), WTC0(0xc2982d80), WTC0(0xc26df5c0), + WTC0(0xc2444b00), WTC0(0xc21b3940), WTC0(0xc1f2cbc0), WTC0(0xc1cb05c0), + WTC0(0xc1a3e340), WTC0(0xc17d5f00), WTC0(0xc15773c0), WTC0(0xc1320940), + WTC0(0xc10cf480), WTC0(0xc0e80a00), WTC0(0xc0c31f00), WTC0(0xc09e2640), + WTC0(0xc0792ec0), WTC0(0xc0544940), WTC0(0xc02f86c0), WTC0(0xc00b04c0), + WTC0(0xbfe6ed01), WTC0(0xbfc36a01), WTC0(0xbfa0a581), WTC0(0xbf7eb581), + WTC0(0xbf5d9a81), WTC0(0xbf3d5501), WTC0(0xbf1de601), WTC0(0xbeff4801), + WTC0(0xbee17201), WTC0(0xbec45881), WTC0(0xbea7f301), WTC0(0xbe8c3781), + WTC0(0xbe712001), WTC0(0xbe56a381), WTC0(0xbe3cbc01), WTC0(0xbe236001), + WTC0(0xbe0a8581), WTC0(0xbdf22181), WTC0(0xbdda2a01), WTC0(0xbdc29a81), + WTC0(0xbdab7181), WTC0(0xbd94b001), WTC0(0xbd7e5581), WTC0(0xbd686681), + WTC0(0xbd52eb01), WTC0(0xbd3deb81), WTC0(0xbd297181), WTC0(0xbd158801), + WTC0(0xbd023f01), WTC0(0xbcefa601), WTC0(0xbcddcc81), WTC0(0xbcccbd01), + WTC0(0xbcbc7e01), WTC0(0xbcad1501), WTC0(0xbc9e8801), WTC0(0xbc90d481), + WTC0(0xbc83f201), WTC0(0xbc77d601), WTC0(0xbc6c7781), WTC0(0xbc61c401), + WTC0(0xbc57a301), WTC0(0xbc4dfb81), WTC0(0xbc44b481), WTC0(0xbc3bbc01), + WTC0(0xbc330781), WTC0(0xbc2a8c81), WTC0(0xbc224181), WTC0(0xbc1a2401), + WTC0(0xbc123b81), WTC0(0xbc0a8f01), WTC0(0xbc032601), WTC0(0xbbfc0f81), + WTC0(0xbbf56181), WTC0(0xbbef3301), WTC0(0xbbe99981), WTC0(0xbbe49d01), + WTC0(0xbbe03801), WTC0(0xbbdc6481), WTC0(0xbbd91b81), WTC0(0xbbd64d01), + WTC0(0xbbd3e101), WTC0(0xbbd1bd81), WTC0(0xbbcfca81), WTC0(0xbbce0601), + WTC0(0xbbcc8201), WTC0(0xbbcb5301), WTC0(0xbbca8d01), WTC0(0xbbca5081), + WTC0(0xbbcaca01), WTC0(0xbbcc2681), WTC0(0xbbce9181), WTC0(0xbbd21281), + WTC0(0xbbd68c81), WTC0(0xbbdbe201), WTC0(0xbbe1f401), WTC0(0xbbe89901), + WTC0(0xbbef9b81), WTC0(0xbbf6c601), WTC0(0xbbfde481), WTC0(0xbc04e381), + WTC0(0xbc0bcf81), WTC0(0xbc12b801), WTC0(0xbc19ab01), WTC0(0xbc20ae01), + WTC0(0xbc27bd81), WTC0(0xbc2ed681), WTC0(0xbc35f501), WTC0(0xbc3d1801), + WTC0(0xbc444081), WTC0(0xbc4b6e81), WTC0(0xbc52a381), WTC0(0xbc59df81), + WTC0(0xbc612301), WTC0(0xbc686e01), WTC0(0xbc6fc101), WTC0(0xbc771c01), + WTC0(0xbc7e7e01), WTC0(0xbc85e801), WTC0(0xbc8d5901), WTC0(0xbc94d201), + WTC0(0xbc9c5281), WTC0(0xbca3db01), WTC0(0xbcab6c01), WTC0(0xbcb30601), + WTC0(0xbcbaa801), WTC0(0xbcc25181), WTC0(0xbcca0301), WTC0(0xbcd1bb81), + WTC0(0xbcd97c81), WTC0(0xbce14601), WTC0(0xbce91801), WTC0(0xbcf0f381), + WTC0(0xbcf8d781), WTC0(0xbd00c381), WTC0(0xbd08b781), WTC0(0xbd10b381), + WTC0(0xbd18b781), WTC0(0xbd20c401), WTC0(0xbd28d981), WTC0(0xbd30f881), + WTC0(0xbd391f81), WTC0(0xbd414f01), WTC0(0xbd498601), WTC0(0xbd51c481), + WTC0(0xbd5a0b01), WTC0(0xbd625981), WTC0(0xbd6ab101), WTC0(0xbd731081), + WTC0(0xbd7b7781), WTC0(0xbd83e681), WTC0(0xbd8c5c01), WTC0(0xbd94d801), + WTC0(0xbd9d5b81), WTC0(0xbda5e601), WTC0(0xbdae7881), WTC0(0xbdb71201), + WTC0(0xbdbfb281), WTC0(0xbdc85981), WTC0(0xbdd10681), WTC0(0xbdd9b981), + WTC0(0xbde27201), WTC0(0xbdeb3101), WTC0(0xbdf3f701), WTC0(0xbdfcc301), + WTC0(0xbe059481), WTC0(0xbe0e6c01), WTC0(0xbe174781), WTC0(0xbe202801), + WTC0(0xbe290d01), WTC0(0xbe31f701), WTC0(0xbe3ae601), WTC0(0xbe43da81), + WTC0(0xbe4cd381), WTC0(0xbe55d001), WTC0(0xbe5ed081), WTC0(0xbe67d381), + WTC0(0xbe70da01), WTC0(0xbe79e481), WTC0(0xbe82f301), WTC0(0xbe8c0501), + WTC0(0xbe951a81), WTC0(0xbe9e3281), WTC0(0xbea74c81), WTC0(0xbeb06881), + WTC0(0xbeb98681), WTC0(0xbec2a781), WTC0(0xbecbca81), WTC0(0xbed4f081), + WTC0(0xbede1901), WTC0(0xbee74281), WTC0(0xbef06d01), WTC0(0xbef99901), + WTC0(0xbf02c581), WTC0(0xbf0bf381), WTC0(0xbf152381), WTC0(0xbf1e5501), + WTC0(0xbf278801), WTC0(0xbf30bb01), WTC0(0xbf39ee81), WTC0(0xbf432281), + WTC0(0xbf4c5681), WTC0(0xbf558b01), WTC0(0xbf5ec101), WTC0(0xbf67f801), + WTC0(0xbf712f01), WTC0(0xbf7a6681), WTC0(0xbf839d81), WTC0(0xbf8cd481), + WTC0(0xbf960b01), WTC0(0xbf9f4181), WTC0(0xbfa87901), WTC0(0xbfb1b101), + WTC0(0xbfbae981), WTC0(0xbfc42201), WTC0(0xbfcd5a01), WTC0(0xbfd69101), + WTC0(0xbfdfc781), WTC0(0xbfe8fc01), WTC0(0xbff22f81), WTC0(0xbffb6081), + /* part 1 */ + WTC1(0x80093e01), WTC1(0x801b9b01), WTC1(0x802df701), WTC1(0x80405101), + WTC1(0x8052a881), WTC1(0x8064fc81), WTC1(0x80774c81), WTC1(0x80899881), + WTC1(0x809bdf01), WTC1(0x80ae1f81), WTC1(0x80c05a01), WTC1(0x80d28d81), + WTC1(0x80e4bb81), WTC1(0x80f6e481), WTC1(0x81090981), WTC1(0x811b2981), + WTC1(0x812d4481), WTC1(0x813f5981), WTC1(0x81516701), WTC1(0x81636d81), + WTC1(0x81756d81), WTC1(0x81876781), WTC1(0x81995c01), WTC1(0x81ab4b01), + WTC1(0x81bd3401), WTC1(0x81cf1581), WTC1(0x81e0ee81), WTC1(0x81f2bf81), + WTC1(0x82048881), WTC1(0x82164a81), WTC1(0x82280581), WTC1(0x8239b981), + WTC1(0x824b6601), WTC1(0x825d0901), WTC1(0x826ea201), WTC1(0x82803101), + WTC1(0x8291b601), WTC1(0x82a33281), WTC1(0x82b4a601), WTC1(0x82c61101), + WTC1(0x82d77201), WTC1(0x82e8c801), WTC1(0x82fa1181), WTC1(0x830b4f81), + WTC1(0x831c8101), WTC1(0x832da781), WTC1(0x833ec381), WTC1(0x834fd481), + WTC1(0x8360d901), WTC1(0x8371d081), WTC1(0x8382ba01), WTC1(0x83939501), + WTC1(0x83a46181), WTC1(0x83b52101), WTC1(0x83c5d381), WTC1(0x83d67881), + WTC1(0x83e70f01), WTC1(0x83f79681), WTC1(0x84080d81), WTC1(0x84187401), + WTC1(0x8428ca01), WTC1(0x84391081), WTC1(0x84494881), WTC1(0x84597081), + WTC1(0x84698881), WTC1(0x84798f81), WTC1(0x84898481), WTC1(0x84996701), + WTC1(0x84a93801), WTC1(0x84b8f801), WTC1(0x84c8a701), WTC1(0x84d84601), + WTC1(0x84e7d381), WTC1(0x84f74e01), WTC1(0x8506b581), WTC1(0x85160981), + WTC1(0x85254a81), WTC1(0x85347901), WTC1(0x85439601), WTC1(0x8552a181), + WTC1(0x85619a01), WTC1(0x85707f81), WTC1(0x857f5101), WTC1(0x858e0e01), + WTC1(0x859cb781), WTC1(0x85ab4f01), WTC1(0x85b9d481), WTC1(0x85c84801), + WTC1(0x85d6a981), WTC1(0x85e4f801), WTC1(0x85f33281), WTC1(0x86015981), + WTC1(0x860f6e01), WTC1(0x861d7081), WTC1(0x862b6201), WTC1(0x86394301), + WTC1(0x86471281), WTC1(0x8654d001), WTC1(0x86627b01), WTC1(0x86701381), + WTC1(0x867d9a81), WTC1(0x868b1001), WTC1(0x86987581), WTC1(0x86a5ca81), + WTC1(0x86b30f01), WTC1(0x86c04381), WTC1(0x86cd6681), WTC1(0x86da7901), + WTC1(0x86e77b81), WTC1(0x86f46d81), WTC1(0x87014f81), WTC1(0x870e2301), + WTC1(0x871ae981), WTC1(0x8727a381), WTC1(0x87345381), WTC1(0x8740f681), + WTC1(0x874d8681), WTC1(0x8759fd01), WTC1(0x87665481), WTC1(0x87729701), + WTC1(0x877ede01), WTC1(0x878b4301), WTC1(0x8797dd81), WTC1(0x87a48b01), + WTC1(0x87b0ef01), WTC1(0x87bcab81), WTC1(0x87c76201), WTC1(0x87d0ca81), + WTC1(0x87fdd781), WTC1(0x881dd301), WTC1(0x88423301), WTC1(0x886a8a81), + WTC1(0x88962981), WTC1(0x88c45e81), WTC1(0x88f47901), WTC1(0x8925f101), + WTC1(0x89586901), WTC1(0x898b8301), WTC1(0x89bee581), WTC1(0x89f26101), + WTC1(0x8a25f301), WTC1(0x8a599a81), WTC1(0x8a8d5801), WTC1(0x8ac13381), + WTC1(0x8af53e81), WTC1(0x8b298b81), WTC1(0x8b5e2c81), WTC1(0x8b933001), + WTC1(0x8bc8a401), WTC1(0x8bfe9401), WTC1(0x8c350d01), WTC1(0x8c6c1b01), + WTC1(0x8ca3cb01), WTC1(0x8cdc2901), WTC1(0x8d154081), WTC1(0x8d4f1b01), + WTC1(0x8d89be81), WTC1(0x8dc53001), WTC1(0x8e017581), WTC1(0x8e3e9481), + WTC1(0x8e7c9301), WTC1(0x8ebb7581), WTC1(0x8efb4181), WTC1(0x8f3bfb01), + WTC1(0x8f7da401), WTC1(0x8fc03f01), WTC1(0x9003ce81), WTC1(0x90485401), + WTC1(0x908dd101), WTC1(0x90d44781), WTC1(0x911bb981), WTC1(0x91642781), + WTC1(0x91ad9281), WTC1(0x91f7f981), WTC1(0x92435d01), WTC1(0x928fbe01), + WTC1(0x92dd1b01), WTC1(0x932b7501), WTC1(0x937acb01), WTC1(0x93cb1c81), + WTC1(0x941c6901), WTC1(0x946eaf81), WTC1(0x94c1ee01), WTC1(0x95162381), + WTC1(0x956b4f81), WTC1(0x95c17081), WTC1(0x96188501), WTC1(0x96708b81), + WTC1(0x96c98381), WTC1(0x97236b01), WTC1(0x977e4181), WTC1(0x97da0481), + WTC1(0x9836b201), WTC1(0x98944901), WTC1(0x98f2c601), WTC1(0x99522801), + WTC1(0x99b26c81), WTC1(0x9a139101), WTC1(0x9a759301), WTC1(0x9ad87081), + WTC1(0x9b3c2801), WTC1(0x9ba0b701), WTC1(0x9c061b81), WTC1(0x9c6c5481), + WTC1(0x9cd35f81), WTC1(0x9d3b3b81), WTC1(0x9da3e601), WTC1(0x9e0d5e01), + WTC1(0x9e779f81), WTC1(0x9ee2a901), WTC1(0x9f4e7801), WTC1(0x9fbb0981), + WTC1(0xa0285d81), WTC1(0xa0967201), WTC1(0xa1054701), WTC1(0xa174da81), + WTC1(0xa1e52a81), WTC1(0xa2563501), WTC1(0xa2c7f801), WTC1(0xa33a7201), + WTC1(0xa3ada281), WTC1(0xa4218801), WTC1(0xa4962181), WTC1(0xa50b6e81), + WTC1(0xa5816e81), WTC1(0xa5f81f81), WTC1(0xa66f8201), WTC1(0xa6e79401), + WTC1(0xa7605601), WTC1(0xa7d9c681), WTC1(0xa853e501), WTC1(0xa8ceb201), + WTC1(0xa94a2c01), WTC1(0xa9c65401), WTC1(0xaa432981), WTC1(0xaac0ad01), + WTC1(0xab3edf01), WTC1(0xabbdc001), WTC1(0xac3d5001), WTC1(0xacbd9081), + WTC1(0xad3e8101), WTC1(0xadc02281), WTC1(0xae427481), WTC1(0xaec57801), + WTC1(0xaf492f01), WTC1(0xafcd9a81), WTC1(0xb052bc01), WTC1(0xb0d89401), + WTC1(0xb15f2381), WTC1(0xb1e66a01), WTC1(0xb26e6881), WTC1(0xb2f71f01), + WTC1(0xb3808d81), WTC1(0xb40ab501), WTC1(0xb4959501), WTC1(0xb5212e81), + WTC1(0x4a6cf67f), WTC1(0x49dffeff), WTC1(0x495265ff), WTC1(0x48c4277f), + WTC1(0x4835407f), WTC1(0x47a5aeff), WTC1(0x471570ff), WTC1(0x468484ff), + WTC1(0x45f2eaff), WTC1(0x4560a2ff), WTC1(0x44cdad7f), WTC1(0x443a0c7f), + WTC1(0x43a5c07f), WTC1(0x4310caff), WTC1(0x427b2bff), WTC1(0x41e4e3ff), + WTC1(0x414df2ff), WTC1(0x40b6557f), WTC1(0x401e06ff), WTC1(0x3f8503c0), + WTC1(0x3eeb4e00), WTC1(0x3e50ebc0), WTC1(0x3db5e680), WTC1(0x3d1a4680), + WTC1(0x3c7e10c0), WTC1(0x3be14cc0), WTC1(0x3b4402c0), WTC1(0x3aa63800), + WTC1(0x3a07e840), WTC1(0x39690880), WTC1(0x38c98700), WTC1(0x38295b40), + WTC1(0x37888a80), WTC1(0x36e71d40), WTC1(0x36451d80), WTC1(0x35a29400), + WTC1(0x34ff8800), WTC1(0x345c04c0), WTC1(0x33b81940), WTC1(0x3313d200), + WTC1(0x326f3800), WTC1(0x31ca5600), WTC1(0x31253840), WTC1(0x307fe8c0), + WTC1(0x2fda6e40), WTC1(0x2f34ce40), WTC1(0x2e8f0e40), WTC1(0x2de92ec0), + WTC1(0x2d432780), WTC1(0x2c9cea40), WTC1(0x2bf66300), WTC1(0x2b4f88c0), + WTC1(0x2aa864c0), WTC1(0x2a010240), WTC1(0x29596e40), WTC1(0x28b1ba80), + WTC1(0x2809ff40), WTC1(0x27625b80), WTC1(0x26baf580), WTC1(0x2613e7c0), + WTC1(0x256d3dc0), WTC1(0x24c70300), WTC1(0x24214380), WTC1(0x237c0800), + WTC1(0x22d75400), WTC1(0x22332a80), WTC1(0x218f8cc0), WTC1(0x20ec7e40), + WTC1(0x204a04c0), WTC1(0x1fa82540), WTC1(0x1f06e300), WTC1(0x1e664000), + WTC1(0x1dc63bc0), WTC1(0x1d26d3c0), WTC1(0x1c8803a0), WTC1(0x1be9cc40), + WTC1(0x1b4c34c0), WTC1(0x1aaf4480), WTC1(0x1a130260), WTC1(0x197774a0), + WTC1(0x18dca260), WTC1(0x184294e0), WTC1(0x17a95840), WTC1(0x1710fd80), + WTC1(0x16799ce0), WTC1(0x15e35340), WTC1(0x154e41a0), WTC1(0x14ba8360), + WTC1(0x14282be0), WTC1(0x13975100), WTC1(0x13080aa0), WTC1(0x127a6240), + WTC1(0x11ee50a0), WTC1(0x1163cc80), WTC1(0x10dacb20), WTC1(0x105333a0), + WTC1(0x0fccdb30), WTC1(0x0f478f40), WTC1(0x0ec31700), WTC1(0x0e3f4e80), + WTC1(0x0dbc27f0), WTC1(0x0d399000), WTC1(0x0cb76d00), WTC1(0x0c359d50), + WTC1(0x0bb3fd50), WTC1(0x0b326bd0), WTC1(0x0ab0ca80), WTC1(0x0a2f0dc0), + WTC1(0x09ad40c0), WTC1(0x092b7a90), WTC1(0x08a9db80), WTC1(0x08285c80), + WTC1(0x07a6c7b8), WTC1(0x0724e4e0), WTC1(0x06a27b80), WTC1(0x061f52f8), + WTC1(0x059b2ad0), WTC1(0x0515b568), WTC1(0x048ea058), WTC1(0x04066408), + WTC1(0x037e52d8), WTC1(0x02f7d3c8), WTC1(0x0274614c), WTC1(0x01f63008), + WTC1(0x0180403a), WTC1(0x0115c442), WTC1(0x00ba09e2), WTC1(0x006f077c), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + /* part 2 */ + WTC2(0xfff36be1), WTC2(0xffdafbc1), WTC2(0xffc28035), WTC2(0xffa9fe8a), + WTC2(0xff917c08), WTC2(0xff78fdfc), WTC2(0xff6089af), WTC2(0xff48246c), + WTC2(0xff2fd37f), WTC2(0xff179c31), WTC2(0xfeff83b6), WTC2(0xfee78d18), + WTC2(0xfecfb93e), WTC2(0xfeb808f2), WTC2(0xfea07d06), WTC2(0xfe8916b4), + WTC2(0xfe71d7a0), WTC2(0xfe5ac174), WTC2(0xfe43d5d6), WTC2(0xfe2d167e), + WTC2(0xfe16852e), WTC2(0xfe0023a6), WTC2(0xfde9f3f8), WTC2(0xfdd3ff7c), + WTC2(0xfdbe56c0), WTC2(0xfda90aa8), WTC2(0xfd942b78), WTC2(0xfd7fbb20), + WTC2(0xfd6bad50), WTC2(0xfd57f510), WTC2(0xfd44857c), WTC2(0xfd3153fc), + WTC2(0xfd1e5840), WTC2(0xfd0b8a0c), WTC2(0xfcf8e180), WTC2(0xfce65eec), + WTC2(0xfcd40ad0), WTC2(0xfcc1ee0c), WTC2(0xfcb011e8), WTC2(0xfc9e896c), + WTC2(0xfc8d716c), WTC2(0xfc7ce720), WTC2(0xfc6d072c), WTC2(0xfc5de09c), + WTC2(0xfc4f74e8), WTC2(0xfc41c4e8), WTC2(0xfc34d0dc), WTC2(0xfc288a68), + WTC2(0xfc1cd49c), WTC2(0xfc1191e0), WTC2(0xfc06a4d0), WTC2(0xfbfbf3e8), + WTC2(0xfbf16990), WTC2(0xfbe6f068), WTC2(0xfbdc7428), WTC2(0xfbd1fc68), + WTC2(0xfbc7ac50), WTC2(0xfbbda868), WTC2(0xfbb41500), WTC2(0xfbab1438), + WTC2(0xfba2c5f8), WTC2(0xfb9b4a00), WTC2(0xfb94bfa8), WTC2(0xfb8f3b48), + WTC2(0xfb8ac638), WTC2(0xfb876970), WTC2(0xfb852d20), WTC2(0xfb840ae0), + WTC2(0xfb83ed60), WTC2(0xfb84bec0), WTC2(0xfb866918), WTC2(0xfb88d4a8), + WTC2(0xfb8be810), WTC2(0xfb8f89d0), WTC2(0xfb93a080), WTC2(0xfb981418), + WTC2(0xfb9ccdf0), WTC2(0xfba1b770), WTC2(0xfba6bae0), WTC2(0xfbabd5c0), + WTC2(0xfbb118d8), WTC2(0xfbb695c0), WTC2(0xfbbc5e90), WTC2(0xfbc29030), + WTC2(0xfbc95268), WTC2(0xfbd0cd78), WTC2(0xfbd929c8), WTC2(0xfbe294d0), + WTC2(0xfbed4108), WTC2(0xfbf96118), WTC2(0xfc0726c8), WTC2(0xfc16b064), + WTC2(0xfc280890), WTC2(0xfc3b3920), WTC2(0xfc504a98), WTC2(0xfc67271c), + WTC2(0xfc7f9a74), WTC2(0xfc996f18), WTC2(0xfcb46eb8), WTC2(0xfcd050b0), + WTC2(0xfcecba24), WTC2(0xfd094f64), WTC2(0xfd25b720), WTC2(0xfd41ce40), + WTC2(0xfd5da7f8), WTC2(0xfd7959d8), WTC2(0xfd94fb74), WTC2(0xfdb0d3fc), + WTC2(0xfdcd5a34), WTC2(0xfdeb06e4), WTC2(0xfe0a5184), WTC2(0xfe2b92c4), + WTC2(0xfe4f0486), WTC2(0xfe74df54), WTC2(0xfe9d5886), WTC2(0xfec85b92), + WTC2(0xfef58a16), WTC2(0xff248275), WTC2(0xff54e401), WTC2(0xff866330), + WTC2(0xffb8c99b), WTC2(0xffebe1c9), WTC2(0x001f786a), WTC2(0x00538bf9), + WTC2(0x00884cbc), WTC2(0x00bded23), WTC2(0x00f49f54), WTC2(0x012c8ee4), + WTC2(0x0165e0d2), WTC2(0x01a0b9d6), WTC2(0x01dd3d80), WTC2(0x021b74d4), + WTC2(0x025b4e48), WTC2(0x029cb730), WTC2(0x02df9d0c), WTC2(0x0323f1a4), + WTC2(0x0369ab00), WTC2(0x03b0bf5c), WTC2(0x03f925a0), WTC2(0x0442e3d8), + WTC2(0x048e0f40), WTC2(0x04dabdb0), WTC2(0x05290430), WTC2(0x0578e428), + WTC2(0x05ca4b60), WTC2(0x061d26c0), WTC2(0x067163d8), WTC2(0x06c6ff10), + WTC2(0x071e03b0), WTC2(0x07767da0), WTC2(0x07d07918), WTC2(0x082c08e0), + WTC2(0x08894660), WTC2(0x08e84b70), WTC2(0x094930b0), WTC2(0x09abf8d0), + WTC2(0x0a109020), WTC2(0x0a76e210), WTC2(0x0adeda50), WTC2(0x0b486b80), + WTC2(0x0bb38f00), WTC2(0x0c203e80), WTC2(0x0c8e73e0), WTC2(0x0cfe2c30), + WTC2(0x0d6f6820), WTC2(0x0de22850), WTC2(0x0e566d90), WTC2(0x0ecc3dd0), + WTC2(0x0f43a3a0), WTC2(0x0fbca9f0), WTC2(0x10375b80), WTC2(0x10b3be20), + WTC2(0x1131d280), WTC2(0x11b19960), WTC2(0x123313a0), WTC2(0x12b64380), + WTC2(0x133b2d00), WTC2(0x13c1d440), WTC2(0x144a3d60), WTC2(0x14d46900), + WTC2(0x15605480), WTC2(0x15edfd20), WTC2(0x167d6040), WTC2(0x170e7e80), + WTC2(0x17a15b80), WTC2(0x1835fb00), WTC2(0x18cc60a0), WTC2(0x19648dc0), + WTC2(0x19fe80e0), WTC2(0x1a9a38a0), WTC2(0x1b37b3e0), WTC2(0x1bd6f400), + WTC2(0x1c77fd20), WTC2(0x1d1ad400), WTC2(0x1dbf7c80), WTC2(0x1e65f820), + WTC2(0x1f0e4540), WTC2(0x1fb861e0), WTC2(0x20644cc0), WTC2(0x21120640), + WTC2(0x21c19240), WTC2(0x2272f480), WTC2(0x23263000), WTC2(0x23db4580), + WTC2(0x24923340), WTC2(0x254af700), WTC2(0x26058e80), WTC2(0x26c1fa00), + WTC2(0x27803d00), WTC2(0x28405a40), WTC2(0x29025500), WTC2(0x29c62d40), + WTC2(0x2a8be0c0), WTC2(0x2b536cc0), WTC2(0x2c1ccf80), WTC2(0x2ce80840), + WTC2(0x2db519c0), WTC2(0x2e840600), WTC2(0x2f54cf80), WTC2(0x302775c0), + WTC2(0x30fbf640), WTC2(0x31d24e00), WTC2(0x32aa7a00), WTC2(0x338479c0), + WTC2(0x34604e40), WTC2(0x353df900), WTC2(0x361d7ac0), WTC2(0x36fed200), + WTC2(0x37e1fb40), WTC2(0x38c6f240), WTC2(0x39adb2c0), WTC2(0x3a963a00), + WTC2(0x3b808740), WTC2(0x3c6c9880), WTC2(0x3d5a6cc0), WTC2(0x3e4a0040), + WTC2(0x3f3b4bc0), WTC2(0x402e48ff), WTC2(0x4122f17f), WTC2(0x42193f7f), + WTC2(0x43112eff), WTC2(0x440abbff), WTC2(0x4505e2ff), WTC2(0x46029e7f), + WTC2(0x4700e9ff), WTC2(0x4800bfff), WTC2(0x49021bff), WTC2(0x4a050eff), + WTC2(0x4b09bc7f), WTC2(0x4c104aff), WTC2(0x4d18df7f), WTC2(0x4e23a07f), + WTC2(0x4f30b2ff), WTC2(0x50403c7f), WTC2(0x515262ff), WTC2(0x52674b7f), + WTC2(0x001678b2), WTC2(0x00061a3b), WTC2(0xfffb4622), WTC2(0xfff5ea94), + WTC2(0xfff5f5b9), WTC2(0xfffb55bd), WTC2(0x0005f8cb), WTC2(0x0015cd0c), + WTC2(0x002ac0ac), WTC2(0x0044c1d5), WTC2(0x0063beb2), WTC2(0x0087a56d), + WTC2(0x00b06431), WTC2(0x00dde929), WTC2(0x01102280), WTC2(0x0146fe5e), + WTC2(0x01826af2), WTC2(0x01c25662), WTC2(0x0206aedc), WTC2(0x024f6288), + WTC2(0x029c5f94), WTC2(0x02ed9424), WTC2(0x0342ee6c), WTC2(0x039c5c90), + WTC2(0x03f9ccbc), WTC2(0x045b2d18), WTC2(0x04c06bd8), WTC2(0x05297718), + WTC2(0x05963d10), WTC2(0x0606abe8), WTC2(0x067ab1c0), WTC2(0x06f23cd0), + WTC2(0x076d3b40), WTC2(0x07eb9b38), WTC2(0x086d4ae0), WTC2(0x08f23860), + WTC2(0x097a51f0), WTC2(0x0a0585b0), WTC2(0x0a93c1d0), WTC2(0x0b24f470), + WTC2(0x0bb90bc0), WTC2(0x0c4ff5f0), WTC2(0x0ce9a130), WTC2(0x0d85fb90), + WTC2(0x0e24f360), WTC2(0x0ec676b0), WTC2(0x0f6a73b0), WTC2(0x1010d880), + WTC2(0x10b99360), WTC2(0x11649280), WTC2(0x1211c400), WTC2(0x12c115e0), + WTC2(0x137276a0), WTC2(0x1425d420), WTC2(0x14db1ca0), WTC2(0x15923e60), + WTC2(0x164b2780), WTC2(0x1705c620), WTC2(0x17c20860), WTC2(0x187fdca0), + WTC2(0x193f30e0), WTC2(0x19fff340), WTC2(0x1ac21200), WTC2(0x1b857b40), + WTC2(0x1c4a1d40), WTC2(0x1d0fe600), WTC2(0x1dd6c3e0), WTC2(0x1e9ea4e0), + WTC2(0x1f677740), WTC2(0x20312940), WTC2(0x20fba8c0), WTC2(0x21c6e440), + WTC2(0x2292c9c0), WTC2(0x235f4780), WTC2(0x242c4b80), WTC2(0x24f9c400), + WTC2(0x25c79f40), WTC2(0x2695cb40), WTC2(0x27643680), WTC2(0x2832cec0), + WTC2(0x29018240), WTC2(0x29d03f80), WTC2(0x2a9ef480), WTC2(0x2b6d8f00), + WTC2(0x2c3bfdc0), WTC2(0x2d0a2ec0), WTC2(0x2dd81000), WTC2(0x2ea58fc0), + WTC2(0x2f729c40), WTC2(0x303f2380), WTC2(0x310b1400), WTC2(0x31d65b80), + WTC2(0x32a0e840), WTC2(0x336aa8c0), WTC2(0x34338ac0), WTC2(0x34fb7cc0), + WTC2(0x35c26cc0), WTC2(0x36884900), WTC2(0x374cff80), WTC2(0x38107e80), + WTC2(0x38d2b440), WTC2(0x39938ec0), WTC2(0x3a52fc40), WTC2(0x3b10eb00), + WTC2(0x3bcd4900), WTC2(0x3c880480), WTC2(0x3d410bc0), WTC2(0x3df84d00), + WTC2(0x3eadb600), WTC2(0x3f613540), WTC2(0x4012b8ff), WTC2(0x40c22eff), + WTC2(0x416f85ff), WTC2(0x421aab7f), WTC2(0x42c38e7f), WTC2(0x436a1c7f), + WTC2(0x440e437f), WTC2(0x44aff27f), WTC2(0x454f167f), WTC2(0x45eb9eff), + WTC2(0x468578ff), WTC2(0x471c937f), WTC2(0x47b0dc7f), WTC2(0x484241ff), + WTC2(0x48d0b1ff), WTC2(0x495c1a7f), WTC2(0x49e46a7f), WTC2(0x4a698f7f), + WTC2(0x4aeb77ff), WTC2(0x4b6a11ff), WTC2(0x4be54b7f), WTC2(0x4c5d12ff), + WTC2(0x4cd155ff), WTC2(0x4d4203ff), WTC2(0x4daf09ff), WTC2(0x4e18567f), + WTC2(0x4e7dd77f), WTC2(0x4edf7b7f), WTC2(0x4f3d307f), WTC2(0x4f96e47f), + WTC2(0x4fec85ff), WTC2(0x503e02ff), WTC2(0x508b497f), WTC2(0x50d447ff), + WTC2(0x5118ec7f), WTC2(0x515924ff), WTC2(0x5194dfff), WTC2(0x51cc0b7f), + WTC2(0x51fe95ff), WTC2(0x522c6cff), WTC2(0x52557eff), WTC2(0x5279b9ff), + WTC2(0x52990c7f), WTC2(0x52b364ff), WTC2(0x52c8b07f), WTC2(0x52d8ddff), + WTC2(0x52e3db7f), WTC2(0x52e996ff), WTC2(0x52e9ff7f), WTC2(0x52e501ff), + WTC2(0x52da8cff), WTC2(0x52ca8f7f), WTC2(0x52b4f67f), WTC2(0x5299b07f), + WTC2(0x5278ac7f), WTC2(0x5251d77f), WTC2(0x52251fff), WTC2(0x51f274ff), + WTC2(0x51b9c37f), WTC2(0x517af9ff), WTC2(0x5136077f), WTC2(0x50ead8ff), + WTC2(0x50995cff), WTC2(0x504181ff), WTC2(0x4fe335ff), WTC2(0x4f7e677f), + WTC2(0x4f1303ff), WTC2(0x4ea0f9ff), WTC2(0x4e2837ff), WTC2(0x4da8ab7f), + WTC2(0x4d2242ff), WTC2(0x4c94ecff), WTC2(0x4c0096ff), WTC2(0x4b652f7f), + WTC2(0x4ac2a4ff), WTC2(0x4a18e4ff), WTC2(0x4967ddff), WTC2(0x48af7e7f), + WTC2(0x47efb3ff), WTC2(0x47286cff), WTC2(0x4659ad7f), WTC2(0x45856f7f), + WTC2(0x44afa3ff), WTC2(0x43dc507f), WTC2(0x430f657f), WTC2(0x424ad47f), + WTC2(0x418e927f), WTC2(0x40da7bff), WTC2(0x402e6f7f), WTC2(0x3f8a3100), + WTC2(0x3eed6f40), WTC2(0x3e57d700), WTC2(0x3dc914c0), WTC2(0x3d40cc40), + WTC2(0x3cbe98c0), WTC2(0x3c421540), WTC2(0x3bcadbc0), WTC2(0x3b588880), + WTC2(0x3aeab780), WTC2(0x3a810540), WTC2(0x3a1b0e00), WTC2(0x39b86d00), + WTC2(0x3958bcc0), WTC2(0x38fb9700), WTC2(0x38a095c0), WTC2(0x38473d80), + WTC2(0x37eeff40), WTC2(0x37974b40), WTC2(0x373f9500), WTC2(0x36e7ae00), + WTC2(0x368fc4c0), WTC2(0x36380b80), WTC2(0x35e0b300), WTC2(0x3589c140), + WTC2(0x35331180), WTC2(0x34dc7c80), WTC2(0x3485dc80), WTC2(0x342f1600), + WTC2(0x33d81780), WTC2(0x3380d0c0), WTC2(0x33293100), WTC2(0x32d11800), + WTC2(0x32785780), WTC2(0x321ec0c0), WTC2(0x31c42680), WTC2(0x316885c0), + WTC2(0x310c0580), WTC2(0x30aecec0), WTC2(0x30510940), WTC2(0x2ff2b8c0), + WTC2(0x2f93bf40), WTC2(0x2f33fc00), WTC2(0x2ed350c0), WTC2(0x2e71ba80), + WTC2(0x2e0f5340), WTC2(0x2dac35c0), WTC2(0x2d487c80), WTC2(0x2ce431c0), + WTC2(0x2c7f4fc0), WTC2(0x2c19d080), WTC2(0x2bb3ad80), WTC2(0x2b4ce080), + WTC2(0x2ae56340), WTC2(0x2a7d2f80), WTC2(0x2a143f00), WTC2(0x29aa8b40)}; + +const FIXP_WTB ELDAnalysis480[1440] = { + WTC0(0xfacfbef0), WTC0(0xfab88c18), WTC0(0xfaa0e520), WTC0(0xfa88d110), + WTC0(0xfa7056e8), WTC0(0xfa577db0), WTC0(0xfa3e4c70), WTC0(0xfa24ca28), + WTC0(0xfa0afde0), WTC0(0xf9f0eea0), WTC0(0xf9d6a2c8), WTC0(0xf9bc1ab8), + WTC0(0xf9a15230), WTC0(0xf9864510), WTC0(0xf96af058), WTC0(0xf94f55c0), + WTC0(0xf93378e0), WTC0(0xf9175d80), WTC0(0xf8fb0468), WTC0(0xf8de68b8), + WTC0(0xf8c18438), WTC0(0xf8a450d8), WTC0(0xf886cde8), WTC0(0xf8690148), + WTC0(0xf84af148), WTC0(0xf82ca410), WTC0(0xf80e1e18), WTC0(0xf7ef62a0), + WTC0(0xf7d074e0), WTC0(0xf7b15870), WTC0(0xf7921240), WTC0(0xf772a7a0), + WTC0(0xf7531e50), WTC0(0xf7337820), WTC0(0xf713afd0), WTC0(0xf6f3bea0), + WTC0(0xf6d39dc0), WTC0(0xf6b352e0), WTC0(0xf692f280), WTC0(0xf6729250), + WTC0(0xf65247a0), WTC0(0xf63224c0), WTC0(0xf6123a00), WTC0(0xf5f297c0), + WTC0(0xf5d34dd0), WTC0(0xf5b46b10), WTC0(0xf595fd90), WTC0(0xf5781390), + WTC0(0xf55abba0), WTC0(0xf53e0510), WTC0(0xf521ff70), WTC0(0xf506ba30), + WTC0(0xf4ec4330), WTC0(0xf4d2a680), WTC0(0xf4b9efe0), WTC0(0xf4a22ac0), + WTC0(0xf48b5f70), WTC0(0xf4759310), WTC0(0xf460cde0), WTC0(0xf44cfcc0), + WTC0(0xf439aff0), WTC0(0xf4264e00), WTC0(0xf4123d90), WTC0(0xf3fd1370), + WTC0(0xf3e6be00), WTC0(0xf3cf41a0), WTC0(0xf3b6a030), WTC0(0xf39cdd60), + WTC0(0xf381fe00), WTC0(0xf3660760), WTC0(0xf348fe70), WTC0(0xf32ae820), + WTC0(0xf30bc940), WTC0(0xf2eba690), WTC0(0xf2ca8480), WTC0(0xf2a86670), + WTC0(0xf2854f40), WTC0(0xf2614190), WTC0(0xf23c41e0), WTC0(0xf21657a0), + WTC0(0xf1ef8ae0), WTC0(0xf1c7e3e0), WTC0(0xf19f63d0), WTC0(0xf1760450), + WTC0(0xf14bbdf0), WTC0(0xf1208960), WTC0(0xf0f45cd0), WTC0(0xf0c72ce0), + WTC0(0xf098ee00), WTC0(0xf06996f0), WTC0(0xf0392620), WTC0(0xf0079e10), + WTC0(0xefd4ffc0), WTC0(0xefa15ca0), WTC0(0xef6ce600), WTC0(0xef37d460), + WTC0(0xef025f80), WTC0(0xeecca2c0), WTC0(0xee969760), WTC0(0xee603440), + WTC0(0xee296d20), WTC0(0xedf21c00), WTC0(0xedba07e0), WTC0(0xed80f640), + WTC0(0xed46bf40), WTC0(0xed0b7b00), WTC0(0xeccf5fc0), WTC0(0xec92a120), + WTC0(0xec556d60), WTC0(0xec17e700), WTC0(0xebda2d40), WTC0(0xeb9c5fa0), + WTC0(0xeb5e7040), WTC0(0xeb201b20), WTC0(0xeae117c0), WTC0(0xeaa12000), + WTC0(0xea600180), WTC0(0xea1d9940), WTC0(0xe9d9c160), WTC0(0xe99468a0), + WTC0(0xe94dc040), WTC0(0xe9061940), WTC0(0xe8bdc140), WTC0(0xe8750ae0), + WTC0(0xe82c4fa0), WTC0(0xe7e3ea40), WTC0(0xe79c35e0), WTC0(0xe7554ca0), + WTC0(0xe70efc00), WTC0(0xe6c90c20), WTC0(0xe6833f00), WTC0(0xe63d2300), + WTC0(0xe5f620a0), WTC0(0xe5ad9dc0), WTC0(0xe5632080), WTC0(0xe5169da0), + WTC0(0xe4c83e60), WTC0(0xe4782400), WTC0(0xe4269840), WTC0(0xe3d42dc0), + WTC0(0xe38188c0), WTC0(0xe32f4be0), WTC0(0xe2ddeea0), WTC0(0xe28db520), + WTC0(0xe23ee000), WTC0(0xe1f1a580), WTC0(0xe1a5e3a0), WTC0(0xe15b35a0), + WTC0(0xe1113860), WTC0(0xe0c78a00), WTC0(0xe07dd0e0), WTC0(0xe033b7c0), + WTC0(0xdfe8e680), WTC0(0xdf9d1fc0), WTC0(0xdf5055c0), WTC0(0xdf0287c0), + WTC0(0xdeb3b340), WTC0(0xde63e7c0), WTC0(0xde134a00), WTC0(0xddc20000), + WTC0(0xdd703180), WTC0(0xdd1e1280), WTC0(0xdccbe080), WTC0(0xdc79d980), + WTC0(0xdc283600), WTC0(0xdbd71e00), WTC0(0xdb86b140), WTC0(0xdb3710c0), + WTC0(0xdae850c0), WTC0(0xda9a6bc0), WTC0(0xda4d5640), WTC0(0xda010640), + WTC0(0xd9b56640), WTC0(0xd96a5700), WTC0(0xd91fb700), WTC0(0xd8d56600), + WTC0(0xd88b4a40), WTC0(0xd8414f00), WTC0(0xd7f75f80), WTC0(0xd7ad6740), + WTC0(0xd76352c0), WTC0(0xd7191040), WTC0(0xd6ce8c80), WTC0(0xd683bd00), + WTC0(0xd638a5c0), WTC0(0xd5ed4f80), WTC0(0xd5a1c240), WTC0(0xd5562b80), + WTC0(0xd50ae500), WTC0(0xd4c04c80), WTC0(0xd476bb40), WTC0(0xd42e62c0), + WTC0(0xd3e75680), WTC0(0xd3a1ad00), WTC0(0xd35d6780), WTC0(0xd31a4300), + WTC0(0xd2d7dc00), WTC0(0xd295d080), WTC0(0xd253d8c0), WTC0(0xd211df40), + WTC0(0xd1cfdbc0), WTC0(0xd18dc480), WTC0(0xd14b9dc0), WTC0(0xd1097c80), + WTC0(0xd0c77700), WTC0(0xd085a500), WTC0(0xd0442f40), WTC0(0xd0034a80), + WTC0(0xcfc32c00), WTC0(0xcf840400), WTC0(0xcf45f400), WTC0(0xcf0913c0), + WTC0(0xcecd8000), WTC0(0xce932c80), WTC0(0xce59bf40), WTC0(0xce20cd40), + WTC0(0xcde7ec40), WTC0(0xcdaeedc0), WTC0(0xcd75ea00), WTC0(0xcd3cfec0), + WTC0(0xcd044b40), WTC0(0xcccbff00), WTC0(0xcc945480), WTC0(0xcc5d8780), + WTC0(0xcc27c3c0), WTC0(0xcbf2fc40), WTC0(0xcbbf0a00), WTC0(0xcb8bc7c0), + WTC0(0xcb591880), WTC0(0xcb26f0c0), WTC0(0xcaf54980), WTC0(0xcac41ac0), + WTC0(0xca936440), WTC0(0xca632d80), WTC0(0xca337f00), WTC0(0xca046180), + WTC0(0xc9d5dd40), WTC0(0xc9a7fa80), WTC0(0xc97ac200), WTC0(0xc94e3c00), + WTC0(0xc91d1840), WTC0(0xc8f15980), WTC0(0xc8c52340), WTC0(0xc8988100), + WTC0(0xc86b7f00), WTC0(0xc83e28c0), WTC0(0xc8108a80), WTC0(0xc7e2afc0), + WTC0(0xc7b4a480), WTC0(0xc7867480), WTC0(0xc7582b40), WTC0(0xc729cc80), + WTC0(0xc6fb5700), WTC0(0xc6ccca40), WTC0(0xc69e2180), WTC0(0xc66f49c0), + WTC0(0xc64029c0), WTC0(0xc610a740), WTC0(0xc5e0bfc0), WTC0(0xc5b09e80), + WTC0(0xc5807900), WTC0(0xc5508440), WTC0(0xc520e840), WTC0(0xc4f1bdc0), + WTC0(0xc4c31d00), WTC0(0xc4951780), WTC0(0xc4678a00), WTC0(0xc43a28c0), + WTC0(0xc40ca800), WTC0(0xc3deccc0), WTC0(0xc3b09940), WTC0(0xc3822c00), + WTC0(0xc353a0c0), WTC0(0xc3251740), WTC0(0xc2f6b500), WTC0(0xc2c8a140), + WTC0(0xc29b02c0), WTC0(0xc26df5c0), WTC0(0xc2418940), WTC0(0xc215cbc0), + WTC0(0xc1eaca00), WTC0(0xc1c08680), WTC0(0xc196fb00), WTC0(0xc16e22c0), + WTC0(0xc145f040), WTC0(0xc11e3a80), WTC0(0xc0f6cc00), WTC0(0xc0cf6ec0), + WTC0(0xc0a802c0), WTC0(0xc0809280), WTC0(0xc0593340), WTC0(0xc031f880), + WTC0(0xc00b04c0), WTC0(0xbfe48981), WTC0(0xbfbebb81), WTC0(0xbf99cb01), + WTC0(0xbf75cc81), WTC0(0xbf52c101), WTC0(0xbf30a901), WTC0(0xbf0f8301), + WTC0(0xbeef4601), WTC0(0xbecfe601), WTC0(0xbeb15701), WTC0(0xbe938c81), + WTC0(0xbe767e81), WTC0(0xbe5a2301), WTC0(0xbe3e7201), WTC0(0xbe236001), + WTC0(0xbe08e181), WTC0(0xbdeee981), WTC0(0xbdd56b81), WTC0(0xbdbc6381), + WTC0(0xbda3d081), WTC0(0xbd8bb281), WTC0(0xbd740b81), WTC0(0xbd5ce281), + WTC0(0xbd464281), WTC0(0xbd303581), WTC0(0xbd1ac801), WTC0(0xbd060c81), + WTC0(0xbcf21601), WTC0(0xbcdef701), WTC0(0xbcccbd01), WTC0(0xbcbb7001), + WTC0(0xbcab1781), WTC0(0xbc9bb901), WTC0(0xbc8d5101), WTC0(0xbc7fd301), + WTC0(0xbc733401), WTC0(0xbc676501), WTC0(0xbc5c4c81), WTC0(0xbc51cb01), + WTC0(0xbc47c281), WTC0(0xbc3e1981), WTC0(0xbc34c081), WTC0(0xbc2bab01), + WTC0(0xbc22cd81), WTC0(0xbc1a2401), WTC0(0xbc11b681), WTC0(0xbc098d81), + WTC0(0xbc01b381), WTC0(0xbbfa3c01), WTC0(0xbbf34281), WTC0(0xbbece281), + WTC0(0xbbe73201), WTC0(0xbbe23281), WTC0(0xbbdddb01), WTC0(0xbbda2501), + WTC0(0xbbd70201), WTC0(0xbbd45601), WTC0(0xbbd20301), WTC0(0xbbcfea81), + WTC0(0xbbce0601), WTC0(0xbbcc6b01), WTC0(0xbbcb3201), WTC0(0xbbca7481), + WTC0(0xbbca5d01), WTC0(0xbbcb2281), WTC0(0xbbccfc81), WTC0(0xbbd01301), + WTC0(0xbbd45881), WTC0(0xbbd9a781), WTC0(0xbbdfdb81), WTC0(0xbbe6c801), + WTC0(0xbbee2f81), WTC0(0xbbf5d181), WTC0(0xbbfd6c01), WTC0(0xbc04e381), + WTC0(0xbc0c4581), WTC0(0xbc13a481), WTC0(0xbc1b1081), WTC0(0xbc228f01), + WTC0(0xbc2a1a81), WTC0(0xbc31af01), WTC0(0xbc394901), WTC0(0xbc40e881), + WTC0(0xbc488e81), WTC0(0xbc503b81), WTC0(0xbc57f101), WTC0(0xbc5fae81), + WTC0(0xbc677501), WTC0(0xbc6f4401), WTC0(0xbc771c01), WTC0(0xbc7efc81), + WTC0(0xbc86e581), WTC0(0xbc8ed701), WTC0(0xbc96d101), WTC0(0xbc9ed481), + WTC0(0xbca6e101), WTC0(0xbcaef701), WTC0(0xbcb71701), WTC0(0xbcbf4001), + WTC0(0xbcc77181), WTC0(0xbccfac01), WTC0(0xbcd7ef01), WTC0(0xbce03b81), + WTC0(0xbce89281), WTC0(0xbcf0f381), WTC0(0xbcf95e81), WTC0(0xbd01d281), + WTC0(0xbd0a4f81), WTC0(0xbd12d581), WTC0(0xbd1b6501), WTC0(0xbd23ff01), + WTC0(0xbd2ca281), WTC0(0xbd355081), WTC0(0xbd3e0801), WTC0(0xbd46c801), + WTC0(0xbd4f9101), WTC0(0xbd586281), WTC0(0xbd613d81), WTC0(0xbd6a2201), + WTC0(0xbd731081), WTC0(0xbd7c0781), WTC0(0xbd850701), WTC0(0xbd8e0e01), + WTC0(0xbd971c81), WTC0(0xbda03381), WTC0(0xbda95301), WTC0(0xbdb27b01), + WTC0(0xbdbbab01), WTC0(0xbdc4e301), WTC0(0xbdce2181), WTC0(0xbdd76701), + WTC0(0xbde0b301), WTC0(0xbdea0681), WTC0(0xbdf36101), WTC0(0xbdfcc301), + WTC0(0xbe062b81), WTC0(0xbe0f9a01), WTC0(0xbe190d81), WTC0(0xbe228681), + WTC0(0xbe2c0501), WTC0(0xbe358901), WTC0(0xbe3f1381), WTC0(0xbe48a301), + WTC0(0xbe523781), WTC0(0xbe5bd001), WTC0(0xbe656c01), WTC0(0xbe6f0c01), + WTC0(0xbe78b001), WTC0(0xbe825801), WTC0(0xbe8c0501), WTC0(0xbe95b581), + WTC0(0xbe9f6901), WTC0(0xbea91f01), WTC0(0xbeb2d681), WTC0(0xbebc9181), + WTC0(0xbec64e81), WTC0(0xbed00f81), WTC0(0xbed9d281), WTC0(0xbee39801), + WTC0(0xbeed5f01), WTC0(0xbef72681), WTC0(0xbf00ef81), WTC0(0xbf0aba01), + WTC0(0xbf148681), WTC0(0xbf1e5501), WTC0(0xbf282501), WTC0(0xbf31f501), + WTC0(0xbf3bc601), WTC0(0xbf459681), WTC0(0xbf4f6801), WTC0(0xbf593a01), + WTC0(0xbf630d81), WTC0(0xbf6ce201), WTC0(0xbf76b701), WTC0(0xbf808b81), + WTC0(0xbf8a5f81), WTC0(0xbf943301), WTC0(0xbf9e0701), WTC0(0xbfa7dc01), + WTC0(0xbfb1b101), WTC0(0xbfbb8701), WTC0(0xbfc55c81), WTC0(0xbfcf3181), + WTC0(0xbfd90601), WTC0(0xbfe2d901), WTC0(0xbfecaa81), WTC0(0xbff67a01), + /* part 1 */ + WTC1(0x80130981), WTC1(0x80269f81), WTC1(0x803a3381), WTC1(0x804dc481), + WTC1(0x80615281), WTC1(0x8074dc01), WTC1(0x80886081), WTC1(0x809bdf01), + WTC1(0x80af5701), WTC1(0x80c2c781), WTC1(0x80d63101), WTC1(0x80e99401), + WTC1(0x80fcf181), WTC1(0x81104a01), WTC1(0x81239d81), WTC1(0x8136ea01), + WTC1(0x814a2f81), WTC1(0x815d6c01), WTC1(0x8170a181), WTC1(0x8183cf81), + WTC1(0x8196f781), WTC1(0x81aa1981), WTC1(0x81bd3401), WTC1(0x81d04681), + WTC1(0x81e34f81), WTC1(0x81f64f01), WTC1(0x82094581), WTC1(0x821c3401), + WTC1(0x822f1b01), WTC1(0x8241fa01), WTC1(0x8254cf01), WTC1(0x82679901), + WTC1(0x827a5801), WTC1(0x828d0b01), WTC1(0x829fb401), WTC1(0x82b25301), + WTC1(0x82c4e801), WTC1(0x82d77201), WTC1(0x82e9ef01), WTC1(0x82fc5f01), + WTC1(0x830ec081), WTC1(0x83211501), WTC1(0x83335c81), WTC1(0x83459881), + WTC1(0x8357c701), WTC1(0x8369e781), WTC1(0x837bf801), WTC1(0x838df801), + WTC1(0x839fe801), WTC1(0x83b1c881), WTC1(0x83c39a81), WTC1(0x83d55d01), + WTC1(0x83e70f01), WTC1(0x83f8b001), WTC1(0x840a3e81), WTC1(0x841bb981), + WTC1(0x842d2281), WTC1(0x843e7a81), WTC1(0x844fc081), WTC1(0x8460f581), + WTC1(0x84721701), WTC1(0x84832481), WTC1(0x84941d81), WTC1(0x84a50201), + WTC1(0x84b5d301), WTC1(0x84c69101), WTC1(0x84d73c01), WTC1(0x84e7d381), + WTC1(0x84f85581), WTC1(0x8508c181), WTC1(0x85191801), WTC1(0x85295881), + WTC1(0x85398481), WTC1(0x85499d01), WTC1(0x8559a081), WTC1(0x85698e81), + WTC1(0x85796601), WTC1(0x85892681), WTC1(0x8598d081), WTC1(0x85a86581), + WTC1(0x85b7e601), WTC1(0x85c75201), WTC1(0x85d6a981), WTC1(0x85e5eb81), + WTC1(0x85f51681), WTC1(0x86042c01), WTC1(0x86132c01), WTC1(0x86221801), + WTC1(0x8630f181), WTC1(0x863fb701), WTC1(0x864e6901), WTC1(0x865d0581), + WTC1(0x866b8d81), WTC1(0x867a0081), WTC1(0x86886001), WTC1(0x8696ad01), + WTC1(0x86a4e781), WTC1(0x86b30f01), WTC1(0x86c12401), WTC1(0x86cf2601), + WTC1(0x86dd1481), WTC1(0x86eaf081), WTC1(0x86f8ba81), WTC1(0x87067281), + WTC1(0x87141b01), WTC1(0x8721b481), WTC1(0x872f4201), WTC1(0x873cc201), + WTC1(0x874a2f01), WTC1(0x87578181), WTC1(0x8764b101), WTC1(0x8771c601), + WTC1(0x877ede01), WTC1(0x878c1881), WTC1(0x87998f01), WTC1(0x87a70e81), + WTC1(0x87b42481), WTC1(0x87c05e81), WTC1(0x87cb5101), WTC1(0x87d4ac81), + WTC1(0x87e73d81), WTC1(0x88124281), WTC1(0x88353501), WTC1(0x885f8481), + WTC1(0x888d3181), WTC1(0x88be1681), WTC1(0x88f13801), WTC1(0x8925f101), + WTC1(0x895bcd01), WTC1(0x89925a81), WTC1(0x89c92f81), WTC1(0x8a001f01), + WTC1(0x8a372881), WTC1(0x8a6e4a01), WTC1(0x8aa58681), WTC1(0x8adcee01), + WTC1(0x8b149701), WTC1(0x8b4c9701), WTC1(0x8b850281), WTC1(0x8bbde981), + WTC1(0x8bf75b01), WTC1(0x8c316681), WTC1(0x8c6c1b01), WTC1(0x8ca78781), + WTC1(0x8ce3ba81), WTC1(0x8d20c301), WTC1(0x8d5eaa01), WTC1(0x8d9d7781), + WTC1(0x8ddd3201), WTC1(0x8e1de001), WTC1(0x8e5f8881), WTC1(0x8ea23201), + WTC1(0x8ee5e301), WTC1(0x8f2aa101), WTC1(0x8f706f01), WTC1(0x8fb74f81), + WTC1(0x8fff4601), WTC1(0x90485401), WTC1(0x90927b81), WTC1(0x90ddc001), + WTC1(0x912a2201), WTC1(0x9177a301), WTC1(0x91c64301), WTC1(0x92160301), + WTC1(0x9266e281), WTC1(0x92b8e101), WTC1(0x930bff81), WTC1(0x93603d01), + WTC1(0x93b59901), WTC1(0x940c1281), WTC1(0x9463a881), WTC1(0x94bc5981), + WTC1(0x95162381), WTC1(0x95710601), WTC1(0x95ccff01), WTC1(0x962a0c81), + WTC1(0x96882e01), WTC1(0x96e76101), WTC1(0x9747a481), WTC1(0x97a8f681), + WTC1(0x980b5501), WTC1(0x986ebd81), WTC1(0x98d32d81), WTC1(0x9938a281), + WTC1(0x999f1981), WTC1(0x9a069001), WTC1(0x9a6f0381), WTC1(0x9ad87081), + WTC1(0x9b42d581), WTC1(0x9bae2f81), WTC1(0x9c1a7c81), WTC1(0x9c87ba81), + WTC1(0x9cf5e701), WTC1(0x9d650081), WTC1(0x9dd50481), WTC1(0x9e45f081), + WTC1(0x9eb7c101), WTC1(0x9f2a7281), WTC1(0x9f9e0301), WTC1(0xa0127081), + WTC1(0xa087b981), WTC1(0xa0fddd81), WTC1(0xa174da81), WTC1(0xa1ecae01), + WTC1(0xa2655581), WTC1(0xa2dece81), WTC1(0xa3591801), WTC1(0xa3d43001), + WTC1(0xa4501601), WTC1(0xa4ccc901), WTC1(0xa54a4701), WTC1(0xa5c89001), + WTC1(0xa647a301), WTC1(0xa6c77e01), WTC1(0xa7482101), WTC1(0xa7c98b01), + WTC1(0xa84bbb81), WTC1(0xa8ceb201), WTC1(0xa9526d81), WTC1(0xa9d6ef01), + WTC1(0xaa5c3601), WTC1(0xaae24301), WTC1(0xab691681), WTC1(0xabf0b181), + WTC1(0xac791401), WTC1(0xad023f01), WTC1(0xad8c3301), WTC1(0xae16f001), + WTC1(0xaea27681), WTC1(0xaf2ec901), WTC1(0xafbbe801), WTC1(0xb049d601), + WTC1(0xb0d89401), WTC1(0xb1682281), WTC1(0xb1f88181), WTC1(0xb289b181), + WTC1(0xb31bb301), WTC1(0xb3ae8601), WTC1(0xb4422b81), WTC1(0xb4d6a381), + WTC1(0x4a5a327f), WTC1(0x49c4adff), WTC1(0x492e637f), WTC1(0x48974f7f), + WTC1(0x47ff6d7f), WTC1(0x4766baff), WTC1(0x46cd35ff), WTC1(0x4632dd7f), + WTC1(0x4597b0ff), WTC1(0x44fbb1ff), WTC1(0x445eeaff), WTC1(0x43c165ff), + WTC1(0x4323227f), WTC1(0x4284277f), WTC1(0x41e48aff), WTC1(0x4144557f), + WTC1(0x40a3867f), WTC1(0x4001f5ff), WTC1(0x3f5f5d80), WTC1(0x3ebbad00), + WTC1(0x3e16ee40), WTC1(0x3d713d00), WTC1(0x3ccab700), WTC1(0x3c236500), + WTC1(0x3b7b5800), WTC1(0x3ad2ecc0), WTC1(0x3a2a6540), WTC1(0x3981b7c0), + WTC1(0x38d8ba00), WTC1(0x382f01c0), WTC1(0x37846240), WTC1(0x36d8eb00), + WTC1(0x362c9ec0), WTC1(0x357f7a00), WTC1(0x34d18340), WTC1(0x3422c900), + WTC1(0x33736c40), WTC1(0x32c39040), WTC1(0x32134280), WTC1(0x31629280), + WTC1(0x30b1a000), WTC1(0x30008380), WTC1(0x2f4f4240), WTC1(0x2e9df180), + WTC1(0x2decc780), WTC1(0x2d3bd640), WTC1(0x2c8b0cc0), WTC1(0x2bda3080), + WTC1(0x2b28ec80), WTC1(0x2a773500), WTC1(0x29c51b40), WTC1(0x291293c0), + WTC1(0x285f9280), WTC1(0x27ac35c0), WTC1(0x26f8ab40), WTC1(0x26454c00), + WTC1(0x25925600), WTC1(0x24dfd580), WTC1(0x242ddd40), WTC1(0x237c87c0), + WTC1(0x22cbe240), WTC1(0x221bef40), WTC1(0x216cb040), WTC1(0x20be2800), + WTC1(0x20105c80), WTC1(0x1f6352a0), WTC1(0x1eb71240), WTC1(0x1e0ba140), + WTC1(0x1d60fe40), WTC1(0x1cb723e0), WTC1(0x1c0e0300), WTC1(0x1b6596c0), + WTC1(0x1abde8a0), WTC1(0x1a16fbe0), WTC1(0x1970c680), WTC1(0x18cb4840), + WTC1(0x18268e20), WTC1(0x1782a0c0), WTC1(0x16df8960), WTC1(0x163d6300), + WTC1(0x159c52c0), WTC1(0x14fc87e0), WTC1(0x145e2c80), WTC1(0x13c15b60), + WTC1(0x13263240), WTC1(0x128cd9a0), WTC1(0x11f562a0), WTC1(0x115fc1c0), + WTC1(0x10cbf160), WTC1(0x1039f200), WTC1(0x0fa9a080), WTC1(0x0f1abd90), + WTC1(0x0e8d01d0), WTC1(0x0e003330), WTC1(0x0d743590), WTC1(0x0ce8ef40), + WTC1(0x0c5e1900), WTC1(0x0bd35d70), WTC1(0x0b488eb0), WTC1(0x0abd8410), + WTC1(0x0a320a00), WTC1(0x09a60e70), WTC1(0x0919ab00), WTC1(0x088d0de0), + WTC1(0x080065e0), WTC1(0x07739710), WTC1(0x06e65808), WTC1(0x06588348), + WTC1(0x05ca0ae0), WTC1(0x053aaaf8), WTC1(0x04a9faf0), WTC1(0x0417f698), + WTC1(0x03859ff4), WTC1(0x02f49be4), WTC1(0x0266b668), WTC1(0x01de554e), + WTC1(0x015f50ca), WTC1(0x00eb7e5d), WTC1(0x00904f24), WTC1(0x00212889), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), WTC1(0x00000000), + /* part 2 */ + WTC2(0xfffece02), WTC2(0xffe4c3df), WTC2(0xffcaaa55), WTC2(0xffb087d1), + WTC2(0xff9662bf), WTC2(0xff7c418b), WTC2(0xff622aa0), WTC2(0xff48246c), + WTC2(0xff2e355a), WTC2(0xff1463db), WTC2(0xfefab608), WTC2(0xfee12f0a), + WTC2(0xfec7cfd2), WTC2(0xfeae995a), WTC2(0xfe958cc4), WTC2(0xfe7cabce), + WTC2(0xfe63f882), WTC2(0xfe4b74e0), WTC2(0xfe3322f6), WTC2(0xfe1b04dc), + WTC2(0xfe031ccc), WTC2(0xfdeb6cf0), WTC2(0xfdd3ff7c), WTC2(0xfdbce834), + WTC2(0xfda63bb8), WTC2(0xfd900c68), WTC2(0xfd7a590c), WTC2(0xfd6511b4), + WTC2(0xfd5026c0), WTC2(0xfd3b8954), WTC2(0xfd272df0), WTC2(0xfd130adc), + WTC2(0xfcff15ac), WTC2(0xfceb4a68), WTC2(0xfcd7b110), WTC2(0xfcc454d0), + WTC2(0xfcb14064), WTC2(0xfc9e896c), WTC2(0xfc8c5264), WTC2(0xfc7abef0), + WTC2(0xfc69f078), WTC2(0xfc59f5e8), WTC2(0xfc4acfec), WTC2(0xfc3c8060), + WTC2(0xfc2f0264), WTC2(0xfc223b7c), WTC2(0xfc160714), WTC2(0xfc0a4150), + WTC2(0xfbfec920), WTC2(0xfbf38320), WTC2(0xfbe855d0), WTC2(0xfbdd2740), + WTC2(0xfbd1fc68), WTC2(0xfbc6fea0), WTC2(0xfbbc5a48), WTC2(0xfbb23b48), + WTC2(0xfba8ca78), WTC2(0xfba02e50), WTC2(0xfb988de0), WTC2(0xfb920b40), + WTC2(0xfb8cb870), WTC2(0xfb889f68), WTC2(0xfb85cbe8), WTC2(0xfb843dd0), + WTC2(0xfb83df78), WTC2(0xfb8495d0), WTC2(0xfb864660), WTC2(0xfb88d4a8), + WTC2(0xfb8c21e8), WTC2(0xfb900f28), WTC2(0xfb947dc0), WTC2(0xfb9950c0), + WTC2(0xfb9e6d08), WTC2(0xfba3b658), WTC2(0xfba91908), WTC2(0xfbae9e08), + WTC2(0xfbb45bd0), WTC2(0xfbba66f8), WTC2(0xfbc0dcf0), WTC2(0xfbc7ead8), + WTC2(0xfbcfc200), WTC2(0xfbd89330), WTC2(0xfbe294d0), WTC2(0xfbee03d0), + WTC2(0xfbfb1de8), WTC2(0xfc0a1da4), WTC2(0xfc1b22e0), WTC2(0xfc2e38f0), + WTC2(0xfc436d48), WTC2(0xfc5abf7c), WTC2(0xfc74024c), WTC2(0xfc8ef2e8), + WTC2(0xfcab51ac), WTC2(0xfcc8d024), WTC2(0xfce704f0), WTC2(0xfd0580cc), + WTC2(0xfd23d4d0), WTC2(0xfd41ce40), WTC2(0xfd5f81b0), WTC2(0xfd7d08f0), + WTC2(0xfd9a8560), WTC2(0xfdb85938), WTC2(0xfdd71798), WTC2(0xfdf753b8), + WTC2(0xfe1993ee), WTC2(0xfe3e30f8), WTC2(0xfe656cba), WTC2(0xfe8f8fdc), + WTC2(0xfebca8a4), WTC2(0xfeec590e), WTC2(0xff1e285c), WTC2(0xff51a0b7), + WTC2(0xff866330), WTC2(0xffbc2cbb), WTC2(0xfff2bbff), WTC2(0x0029d79d), + WTC2(0x00618a22), WTC2(0x009a1185), WTC2(0x00d3aa8c), WTC2(0x010e8ff6), + WTC2(0x014af29e), WTC2(0x0188fe56), WTC2(0x01c8e108), WTC2(0x020ab3c4), + WTC2(0x024e68a8), WTC2(0x0293e824), WTC2(0x02db1bc8), WTC2(0x0323f1a4), + WTC2(0x036e5d6c), WTC2(0x03ba5320), WTC2(0x0407c938), WTC2(0x0456cad0), + WTC2(0x04a77288), WTC2(0x04f9db88), WTC2(0x054e1888), WTC2(0x05a41ef0), + WTC2(0x05fbd6e0), WTC2(0x065528c0), WTC2(0x06b00838), WTC2(0x070c7ee0), + WTC2(0x076a9bb0), WTC2(0x07ca6d10), WTC2(0x082c08e0), WTC2(0x088f8da0), + WTC2(0x08f51ac0), WTC2(0x095ccc20), WTC2(0x09c69f70), WTC2(0x0a327b40), + WTC2(0x0aa046d0), WTC2(0x0b0febb0), WTC2(0x0b815dd0), WTC2(0x0bf49600), + WTC2(0x0c698c50), WTC2(0x0ce03ba0), WTC2(0x0d58a380), WTC2(0x0dd2c510), + WTC2(0x0e4ea110), WTC2(0x0ecc3dd0), WTC2(0x0f4ba800), WTC2(0x0fcced10), + WTC2(0x10501960), WTC2(0x10d532a0), WTC2(0x115c39c0), WTC2(0x11e52fa0), + WTC2(0x12701560), WTC2(0x12fcef20), WTC2(0x138bc200), WTC2(0x141c9300), + WTC2(0x14af64a0), WTC2(0x154434e0), WTC2(0x15db0020), WTC2(0x1673c360), + WTC2(0x170e7e80), WTC2(0x17ab35e0), WTC2(0x1849ee40), WTC2(0x18eaaba0), + WTC2(0x198d6f00), WTC2(0x1a3236a0), WTC2(0x1ad90080), WTC2(0x1b81cc60), + WTC2(0x1c2c9da0), WTC2(0x1cd97980), WTC2(0x1d8865c0), WTC2(0x1e396540), + WTC2(0x1eec7700), WTC2(0x1fa198c0), WTC2(0x2058c840), WTC2(0x21120640), + WTC2(0x21cd5700), WTC2(0x228abec0), WTC2(0x234a4180), WTC2(0x240bdf80), + WTC2(0x24cf95c0), WTC2(0x259561c0), WTC2(0x265d4200), WTC2(0x27273840), + WTC2(0x27f348c0), WTC2(0x28c17700), WTC2(0x2991c500), WTC2(0x2a643080), + WTC2(0x2b38b680), WTC2(0x2c0f53c0), WTC2(0x2ce80840), WTC2(0x2dc2d680), + WTC2(0x2e9fc100), WTC2(0x2f7ecac0), WTC2(0x305ff280), WTC2(0x314334c0), + WTC2(0x32288e00), WTC2(0x330ffb80), WTC2(0x33f97d80), WTC2(0x34e515c0), + WTC2(0x35d2c5c0), WTC2(0x36c28d00), WTC2(0x37b467c0), WTC2(0x38a85080), + WTC2(0x399e4240), WTC2(0x3a963a00), WTC2(0x3b903600), WTC2(0x3c8c3480), + WTC2(0x3d8a3380), WTC2(0x3e8a2dc0), WTC2(0x3f8c1b40), WTC2(0x408ff2ff), + WTC2(0x4195ae7f), WTC2(0x429d477f), WTC2(0x43a6b87f), WTC2(0x44b1fdff), + WTC2(0x45bf11ff), WTC2(0x46cdee7f), WTC2(0x47de8cff), WTC2(0x48f0e77f), + WTC2(0x4a050eff), WTC2(0x4b1b2dff), WTC2(0x4c3372ff), WTC2(0x4d4e0bff), + WTC2(0x4e6b257f), WTC2(0x4f8aedff), WTC2(0x50ad92ff), WTC2(0x51d341ff), + WTC2(0x002006a9), WTC2(0x000bfb36), WTC2(0xfffe45ac), WTC2(0xfff6d064), + WTC2(0xfff585bc), WTC2(0xfffa500d), WTC2(0x000519b4), WTC2(0x0015cd0c), + WTC2(0x002c5470), WTC2(0x00489a3b), WTC2(0x006a88c8), WTC2(0x00920a74), + WTC2(0x00bf0999), WTC2(0x00f17092), WTC2(0x012929bc), WTC2(0x01661f70), + WTC2(0x01a83c0c), WTC2(0x01ef69e8), WTC2(0x023b9364), WTC2(0x028ca2d4), + WTC2(0x02e2829c), WTC2(0x033d1d10), WTC2(0x039c5c90), WTC2(0x04002b78), + WTC2(0x04687418), WTC2(0x04d520e0), WTC2(0x05461c18), WTC2(0x05bb5020), + WTC2(0x0634a758), WTC2(0x06b20c20), WTC2(0x073368c8), WTC2(0x07b8a7b0), + WTC2(0x0841b340), WTC2(0x08ce75b0), WTC2(0x095ed980), WTC2(0x09f2c900), + WTC2(0x0a8a2e80), WTC2(0x0b24f470), WTC2(0x0bc30510), WTC2(0x0c644ad0), + WTC2(0x0d08b010), WTC2(0x0db01f10), WTC2(0x0e5a8250), WTC2(0x0f07c400), + WTC2(0x0fb7cea0), WTC2(0x106a8c80), WTC2(0x111fe800), WTC2(0x11d7cb60), + WTC2(0x12922120), WTC2(0x134ed3a0), WTC2(0x140dcd00), WTC2(0x14cef7e0), + WTC2(0x15923e60), WTC2(0x16578b00), WTC2(0x171ec820), WTC2(0x17e7e020), + WTC2(0x18b2bd20), WTC2(0x197f49c0), WTC2(0x1a4d7040), WTC2(0x1b1d1b00), + WTC2(0x1bee3460), WTC2(0x1cc0a6a0), WTC2(0x1d945c40), WTC2(0x1e693f80), + WTC2(0x1f3f3ac0), WTC2(0x20163880), WTC2(0x20ee22c0), WTC2(0x21c6e440), + WTC2(0x22a06740), WTC2(0x237a9600), WTC2(0x24555ac0), WTC2(0x2530a040), + WTC2(0x260c5080), WTC2(0x26e85600), WTC2(0x27c49b00), WTC2(0x28a10a00), + WTC2(0x297d8d80), WTC2(0x2a5a0f80), WTC2(0x2b367a80), WTC2(0x2c12b8c0), + WTC2(0x2ceeb500), WTC2(0x2dca5940), WTC2(0x2ea58fc0), WTC2(0x2f804340), + WTC2(0x305a5dc0), WTC2(0x3133ca00), WTC2(0x320c7200), WTC2(0x32e44000), + WTC2(0x33bb1ec0), WTC2(0x3490f880), WTC2(0x3565b7c0), WTC2(0x36394640), + WTC2(0x370b8f00), WTC2(0x37dc7c00), WTC2(0x38abf7c0), WTC2(0x3979ecc0), + WTC2(0x3a464500), WTC2(0x3b10eb00), WTC2(0x3bd9c940), WTC2(0x3ca0c9c0), + WTC2(0x3d65d740), WTC2(0x3e28dc00), WTC2(0x3ee9c240), WTC2(0x3fa87480), + WTC2(0x4064dcff), WTC2(0x411ee67f), WTC2(0x41d67a7f), WTC2(0x428b847f), + WTC2(0x433ded7f), WTC2(0x43eda0ff), WTC2(0x449a887f), WTC2(0x45448f7f), + WTC2(0x45eb9eff), WTC2(0x468fa1ff), WTC2(0x473082ff), WTC2(0x47ce2c7f), + WTC2(0x4868887f), WTC2(0x48ff80ff), WTC2(0x499300ff), WTC2(0x4a22f2ff), + WTC2(0x4aaf407f), WTC2(0x4b37d47f), WTC2(0x4bbc997f), WTC2(0x4c3d78ff), + WTC2(0x4cba5e7f), WTC2(0x4d33337f), WTC2(0x4da7e27f), WTC2(0x4e18567f), + WTC2(0x4e8478ff), WTC2(0x4eec347f), WTC2(0x4f4f737f), WTC2(0x4fae20ff), + WTC2(0x500825ff), WTC2(0x505d6dff), WTC2(0x50ade37f), WTC2(0x50f96f7f), + WTC2(0x513ffdff), WTC2(0x518177ff), WTC2(0x51bdc87f), WTC2(0x51f4d9ff), + WTC2(0x5226967f), WTC2(0x5252e87f), WTC2(0x5279b9ff), WTC2(0x529af5ff), + WTC2(0x52b6867f), WTC2(0x52cc55ff), WTC2(0x52dc4eff), WTC2(0x52e65aff), + WTC2(0x52ea657f), WTC2(0x52e857ff), WTC2(0x52e01d7f), WTC2(0x52d19fff), + WTC2(0x52bcc9ff), WTC2(0x52a1857f), WTC2(0x527fbd7f), WTC2(0x52575b7f), + WTC2(0x52284a7f), WTC2(0x51f274ff), WTC2(0x51b5c47f), WTC2(0x5172247f), + WTC2(0x51277dff), WTC2(0x50d5bc7f), WTC2(0x507cc9ff), WTC2(0x501c90ff), + WTC2(0x4fb4fb7f), WTC2(0x4f45f3ff), WTC2(0x4ecf64ff), WTC2(0x4e5138ff), + WTC2(0x4dcb597f), WTC2(0x4d3db1ff), WTC2(0x4ca82bff), WTC2(0x4c0ab27f), + WTC2(0x4b652f7f), WTC2(0x4ab78d7f), WTC2(0x4a01b67f), WTC2(0x4943957f), + WTC2(0x487d12ff), WTC2(0x47ae1f7f), WTC2(0x46d68f7f), WTC2(0x45f7187f), + WTC2(0x4513597f), WTC2(0x4430467f), WTC2(0x4352d2ff), WTC2(0x427e6bff), + WTC2(0x41b390ff), WTC2(0x40f2077f), WTC2(0x4039a87f), WTC2(0x3f8a3100), + WTC2(0x3ee33e00), WTC2(0x3e446ac0), WTC2(0x3dad5180), WTC2(0x3d1d7fc0), + WTC2(0x3c947b00), WTC2(0x3c11c7c0), WTC2(0x3b94ebc0), WTC2(0x3b1d6dc0), + WTC2(0x3aaad480), WTC2(0x3a3ca740), WTC2(0x39d26c40), WTC2(0x396ba8c0), + WTC2(0x3907e080), WTC2(0x38a69800), WTC2(0x38473d80), WTC2(0x37e923c0), + WTC2(0x378b9b80), WTC2(0x372e0380), WTC2(0x36d03a80), WTC2(0x36727f00), + WTC2(0x36150e40), WTC2(0x35b81540), WTC2(0x355b8000), WTC2(0x34ff1dc0), + WTC2(0x34a2bfc0), WTC2(0x34463e80), WTC2(0x33e982c0), WTC2(0x338c7880), + WTC2(0x332f0bc0), WTC2(0x32d11800), WTC2(0x327265c0), WTC2(0x3212bbc0), + WTC2(0x31b1e740), WTC2(0x314fef00), WTC2(0x30ed0540), WTC2(0x30895c80), + WTC2(0x30251880), WTC2(0x2fc02880), WTC2(0x2f5a6480), WTC2(0x2ef3a480), + WTC2(0x2e8bd640), WTC2(0x2e231100), WTC2(0x2db97680), WTC2(0x2d4f2700), + WTC2(0x2ce431c0), WTC2(0x2c789080), WTC2(0x2c0c3bc0), WTC2(0x2b9f2bc0), + WTC2(0x2b315940), WTC2(0x2ac2bc00), WTC2(0x2a534cc0), WTC2(0x29e303c0)}; + +const FIXP_WTB ELDAnalysis256[768] = { + WTC(0xfababde8), WTC(0xfa8e1e6a), WTC(0xfa6012a9), WTC(0xfa30c8dd), + WTC(0xfa006f4b), WTC(0xf9cf32c4), WTC(0xf99d1cc8), WTC(0xf96a148d), + WTC(0xf936184d), WTC(0xf9013d5b), WTC(0xf8cb7b67), WTC(0xf894ace0), + WTC(0xf85cd28e), WTC(0xf82413f8), WTC(0xf7ea90af), WTC(0xf7b05ee6), + WTC(0xf7759b0b), WTC(0xf73a671f), WTC(0xf6febea3), WTC(0xf6c27a0e), + WTC(0xf685ca33), WTC(0xf6493907), WTC(0xf60d437b), WTC(0xf5d2551f), + WTC(0xf598d273), WTC(0xf561199e), WTC(0xf52b8c6f), WTC(0xf4f8907d), + WTC(0xf4c87fdf), WTC(0xf49ba806), WTC(0xf4724286), WTC(0xf44c6127), + WTC(0xf4282435), WTC(0xf401ceae), WTC(0xf3d775a1), WTC(0xf3a91477), + WTC(0xf376c33f), WTC(0xf340a328), WTC(0xf306d4d6), WTC(0xf2c9775c), + WTC(0xf288a3ed), WTC(0xf2446e2a), WTC(0xf1fcfa45), WTC(0xf1b27b2d), + WTC(0xf164f3f4), WTC(0xf114365c), WTC(0xf0c00532), WTC(0xf06817a9), + WTC(0xf00c4ea4), WTC(0xefacbc7f), WTC(0xef4a205f), WTC(0xeee5dc33), + WTC(0xee808a0d), WTC(0xee19eeb2), WTC(0xedb12f6e), WTC(0xed44e8eb), + WTC(0xecd50a13), WTC(0xec62d8dd), WTC(0xebef68b2), WTC(0xeb7b805c), + WTC(0xeb069af4), WTC(0xea8eef1c), WTC(0xea131c86), WTC(0xe99234c6), + WTC(0xe90cd9c2), WTC(0xe884f65b), WTC(0xe7fcbd6d), WTC(0xe7767300), + WTC(0xe6f289d0), WTC(0xe66f958a), WTC(0xe5eae99f), WTC(0xe560c403), + WTC(0xe4cfaaa1), WTC(0xe43887dc), WTC(0xe39dedc4), WTC(0xe303f190), + WTC(0xe26d7f5d), WTC(0xe1dc34ff), WTC(0xe14f9ced), WTC(0xe0c53cd0), + WTC(0xe03ab085), WTC(0xdfadc948), WTC(0xdf1d640c), WTC(0xde896bb6), + WTC(0xddf256ad), WTC(0xdd591e3d), WTC(0xdcbf0aec), WTC(0xdc25ab0a), + WTC(0xdb8e334c), WTC(0xdaf97794), WTC(0xda67bed9), WTC(0xd9d8c524), + WTC(0xd94bfa62), WTC(0xd8c089b5), WTC(0xd835c151), WTC(0xd7ab1704), + WTC(0xd7200906), WTC(0xd69420dc), WTC(0xd6073c0d), WTC(0xd5799615), + WTC(0xd4ec7c87), WTC(0xd46241c9), WTC(0xd3dc5bde), WTC(0xd35b4a79), + WTC(0xd2de1032), WTC(0xd26246f5), WTC(0xd1e68ed2), WTC(0xd16aa0a4), + WTC(0xd0eea5d2), WTC(0xd073302b), WTC(0xcff93749), WTC(0xcf820f45), + WTC(0xcf0ebb30), WTC(0xce9fd702), WTC(0xce34596c), WTC(0xcdc9a803), + WTC(0xcd5ec5d6), WTC(0xccf468ec), WTC(0xcc8bb41e), WTC(0xcc2619cc), + WTC(0xcbc3e090), WTC(0xcb6422f5), WTC(0xcb064d2f), WTC(0xcaaa2a6d), + WTC(0xca4fbdc9), WTC(0xc9f73c43), WTC(0xc9a0dc9b), WTC(0xc94cdd02), + WTC(0xc8f578a4), WTC(0xc8a24d15), WTC(0xc84dc71f), WTC(0xc7f83516), + WTC(0xc7a1e4b9), WTC(0xc74b22b1), WTC(0xc6f41284), WTC(0xc69cabc1), + WTC(0xc644986d), WTC(0xc5eb4167), WTC(0xc5910312), WTC(0xc5372c7f), + WTC(0xc4deba2e), WTC(0xc4883eca), WTC(0xc43310f0), WTC(0xc3dd5c5a), + WTC(0xc3868802), WTC(0xc32f431d), WTC(0xc2d86c9e), WTC(0xc28300a6), + WTC(0xc22fae33), WTC(0xc1ded3f7), WTC(0xc1908d7d), WTC(0xc144b0ed), + WTC(0xc0fa7cee), WTC(0xc0b0a3b5), WTC(0xc066b8d3), WTC(0xc01d3b32), + WTC(0xbfd5161c), WTC(0xbf8f92af), WTC(0xbf4d5cea), WTC(0xbf0e7d5e), + WTC(0xbed2ce3a), WTC(0xbe9a0062), WTC(0xbe63cec2), WTC(0xbe2ffd2f), + WTC(0xbdfe4565), WTC(0xbdce5568), WTC(0xbda003df), WTC(0xbd735018), + WTC(0xbd485b2c), WTC(0xbd1f69bd), WTC(0xbcf8db7c), WTC(0xbcd52b0a), + WTC(0xbcb4ae4a), WTC(0xbc979382), WTC(0xbc7dcbab), WTC(0xbc6709dc), + WTC(0xbc52c1b1), WTC(0xbc402f2b), WTC(0xbc2ec37b), WTC(0xbc1e2cb3), + WTC(0xbc0e5d5f), WTC(0xbbff8f23), WTC(0xbbf238d2), WTC(0xbbe707d4), + WTC(0xbbde3c63), WTC(0xbbd7a658), WTC(0xbbd2c7f0), WTC(0xbbcee18b), + WTC(0xbbcbdebb), WTC(0xbbca5ab1), WTC(0xbbcb5622), WTC(0xbbd032e4), + WTC(0xbbd91d4d), WTC(0xbbe53757), WTC(0xbbf32f54), WTC(0xbc016781), + WTC(0xbc0f433a), WTC(0xbc1d2aa4), WTC(0xbc2b4912), WTC(0xbc3985df), + WTC(0xbc47d6b9), WTC(0xbc564099), WTC(0xbc64c78a), WTC(0xbc736d96), + WTC(0xbc823210), WTC(0xbc911484), WTC(0xbca015b8), WTC(0xbcaf37eb), + WTC(0xbcbe7bc3), WTC(0xbccdde4d), WTC(0xbcdd6037), WTC(0xbced049a), + WTC(0xbcfccc81), WTC(0xbd0cb482), WTC(0xbd1cbcaa), WTC(0xbd2ce7ea), + WTC(0xbd3d363b), WTC(0xbd4da445), WTC(0xbd5e312d), WTC(0xbd6edfd1), + WTC(0xbd7fae14), WTC(0xbd90991b), WTC(0xbda19fcf), WTC(0xbdb2c464), + WTC(0xbdc4053b), WTC(0xbdd55f4b), WTC(0xbde6d0a0), WTC(0xbdf85c51), + WTC(0xbe09ffa3), WTC(0xbe1bb724), WTC(0xbe2d8160), WTC(0xbe3f5f98), + WTC(0xbe515144), WTC(0xbe6351a9), WTC(0xbe755ebd), WTC(0xbe877b8e), + WTC(0xbe99a63d), WTC(0xbeabda45), WTC(0xbebe16b0), WTC(0xbed05d1c), + WTC(0xbee2ada9), WTC(0xbef502e2), WTC(0xbf075c40), WTC(0xbf19bc0b), + WTC(0xbf2c217f), WTC(0xbf3e887a), WTC(0xbf50f09d), WTC(0xbf635c77), + WTC(0xbf75cac0), WTC(0xbf883905), WTC(0xbf9aa62b), WTC(0xbfad14f1), + WTC(0xbfbf85c7), WTC(0xbfd1f592), WTC(0xbfe461fc), WTC(0xbff6c86a), + WTC(0x80126c8d), WTC(0x80372448), WTC(0x805bd2fd), WTC(0x80807315), + WTC(0x80a4fffa), WTC(0x80c9748d), WTC(0x80edd08b), WTC(0x81121a23), + WTC(0x81364fde), WTC(0x815a6b16), WTC(0x817e6b36), WTC(0x81a25433), + WTC(0x81c625c8), WTC(0x81e9d801), WTC(0x820d6a5c), WTC(0x8230e060), + WTC(0x825438c0), WTC(0x82776ac7), WTC(0x829a7555), WTC(0x82bd5ca3), + WTC(0x82e01e80), WTC(0x8302b200), WTC(0x83251590), WTC(0x83474d79), + WTC(0x8369566f), WTC(0x838b2957), WTC(0x83acc2d9), WTC(0x83ce27c1), + WTC(0x83ef54b9), WTC(0x841042d1), WTC(0x8430ef15), WTC(0x84515e84), + WTC(0x84718e32), WTC(0x84917804), WTC(0x84b11a25), WTC(0x84d0788d), + WTC(0x84ef9322), WTC(0x850e61ec), WTC(0x852ce400), WTC(0x854b1e0a), + WTC(0x85690f2c), WTC(0x8586b207), WTC(0x85a4057b), WTC(0x85c1107d), + WTC(0x85ddd335), WTC(0x85fa485e), WTC(0x86167172), WTC(0x8632549d), + WTC(0x864df388), WTC(0x8669497e), WTC(0x86845757), WTC(0x869f2218), + WTC(0x86b9ab5a), WTC(0x86d3f1bf), WTC(0x86edf68f), WTC(0x8707baf1), + WTC(0x872147e0), WTC(0x873aa6fc), WTC(0x8753c571), WTC(0x876c76e6), + WTC(0x87850ab7), WTC(0x879e373b), WTC(0x87b6ea37), WTC(0x87cc4188), + WTC(0x880d4300), WTC(0x8855e9ff), WTC(0x88acfca0), WTC(0x890d0f94), + WTC(0x8971e7d5), WTC(0x89d8a0c1), WTC(0x8a3fc425), WTC(0x8aa74105), + WTC(0x8b0f5b93), WTC(0x8b78a107), WTC(0x8be38bb3), WTC(0x8c508092), + WTC(0x8cbfe384), WTC(0x8d3214f1), WTC(0x8da75d21), WTC(0x8e1fe96c), + WTC(0x8e9be76a), WTC(0x8f1b806c), WTC(0x8f9ed314), WTC(0x9025f26a), + WTC(0x90b0ecea), WTC(0x913fd0eb), WTC(0x91d2a684), WTC(0x92696dea), + WTC(0x93042868), WTC(0x93a2d456), WTC(0x94456d20), WTC(0x94ebe9e5), + WTC(0x95964178), WTC(0x96446a05), WTC(0x96f65958), WTC(0x97ac059a), + WTC(0x98656089), WTC(0x99225a80), WTC(0x99e2e2e8), WTC(0x9aa6e666), + WTC(0x9b6e54b8), WTC(0x9c391d99), WTC(0x9d07338a), WTC(0x9dd8888d), + WTC(0x9ead0b5c), WTC(0x9f84a871), WTC(0xa05f4fb3), WTC(0xa13cf913), + WTC(0xa21d9891), WTC(0xa3011e27), WTC(0xa3e77eb4), WTC(0xa4d0b190), + WTC(0xa5bcb0d7), WTC(0xa6ab750c), WTC(0xa79cf884), WTC(0xa89135cb), + WTC(0xa9882a44), WTC(0xaa81d578), WTC(0xab7e39a6), WTC(0xac7d5a36), + WTC(0xad7f3ba5), WTC(0xae83dfed), WTC(0xaf8b4e16), WTC(0xb095911c), + WTC(0xb1a2afd1), WTC(0xb2b2ac9f), WTC(0xb3c58807), WTC(0xb4db4d5e), + WTC(0x4a268ead), WTC(0x490b5ba7), WTC(0x47ed8d30), WTC(0x46cd10c5), + WTC(0x45a9dcc1), WTC(0x4483f267), WTC(0x435b5aeb), WTC(0x42301d12), + WTC(0x41023a15), WTC(0x3fd19bf1), WTC(0x3e9e31e1), WTC(0x3d682986), + WTC(0x3c2fc001), WTC(0x3af52d8f), WTC(0x39b88b7d), WTC(0x38798642), + WTC(0x3737e6d3), WTC(0x35f3e98a), WTC(0x34add45c), WTC(0x33660083), + WTC(0x321ccf3a), WTC(0x30d2963e), WTC(0x2f87a28f), WTC(0x2e3c22cd), + WTC(0x2cf010e5), WTC(0x2ba2ffe5), WTC(0x2a54ba93), WTC(0x290596f5), + WTC(0x27b62806), WTC(0x266762b8), WTC(0x251a11b1), WTC(0x23ce94f9), + WTC(0x22852ddb), WTC(0x213df340), WTC(0x1ff90185), WTC(0x1eb67d94), + WTC(0x1d767485), WTC(0x1c38d477), WTC(0x1afda747), WTC(0x19c5248b), + WTC(0x188f8259), WTC(0x175d0d40), WTC(0x162e5320), WTC(0x150436cd), + WTC(0x13df8d3f), WTC(0x12c102f1), WTC(0x11a8dd65), WTC(0x1096d490), + WTC(0x0f8a1755), WTC(0x0e811dcd), WTC(0x0d7acb9a), WTC(0x0c767d00), + WTC(0x0b7334d9), WTC(0x0a6fef31), WTC(0x096c5a87), WTC(0x08691adb), + WTC(0x0765e395), WTC(0x06610309), WTC(0x0558a0d2), WTC(0x044a946c), + WTC(0x033acb52), WTC(0x0234706f), WTC(0x014939dc), WTC(0x00928577), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0xffe73593), WTC(0xffb63fcf), WTC(0xff853c1f), WTC(0xff5454d7), + WTC(0xff23b44b), WTC(0xfef38417), WTC(0xfec3dc9a), WTC(0xfe94c511), + WTC(0xfe664753), WTC(0xfe387086), WTC(0xfe0b4e63), WTC(0xfddef15c), + WTC(0xfdb3a3f6), WTC(0xfd89e611), WTC(0xfd61c750), WTC(0xfd3ae585), + WTC(0xfd14ec09), WTC(0xfcef9b06), WTC(0xfccaf509), WTC(0xfca74180), + WTC(0xfc8518a3), WTC(0xfc655c7a), WTC(0xfc488545), WTC(0xfc2e9998), + WTC(0xfc1726bb), WTC(0xfc01463f), WTC(0xfbec2c64), WTC(0xfbd735ce), + WTC(0xfbc29e8e), WTC(0xfbaf8042), WTC(0xfb9eeba0), WTC(0xfb91dc05), + WTC(0xfb88f420), WTC(0xfb8479eb), WTC(0xfb84398b), WTC(0xfb87884b), + WTC(0xfb8da8bf), WTC(0xfb95d020), WTC(0xfb9f3e49), WTC(0xfba9448a), + WTC(0xfbb3cf10), WTC(0xfbbf67e7), WTC(0xfbccf65d), WTC(0xfbddba58), + WTC(0xfbf31f46), WTC(0xfc0eb236), WTC(0xfc3164f0), WTC(0xfc5b8269), + WTC(0xfc8c5bcd), WTC(0xfcc248ee), WTC(0xfcfb056c), WTC(0xfd33cc26), + WTC(0xfd6b84ee), WTC(0xfda2d9e7), WTC(0xfddc03fb), WTC(0xfe1aaf57), + WTC(0xfe61a0af), WTC(0xfeb28df7), WTC(0xff0cd343), WTC(0xff6d8388), + WTC(0xffd24331), WTC(0x00396fe3), WTC(0x00a2fb3e), WTC(0x01107050), + WTC(0x01831900), WTC(0x01fc2377), WTC(0x027bd1fc), WTC(0x03019a2d), + WTC(0x038d0a88), WTC(0x041dd88f), WTC(0x04b43495), WTC(0x0550c1ef), + WTC(0x05f38bd6), WTC(0x069c0523), WTC(0x074a114e), WTC(0x07fe0ceb), + WTC(0x08b88e33), WTC(0x097a5965), WTC(0x0a438318), WTC(0x0b137046), + WTC(0x0be9b5ab), WTC(0x0cc61fa9), WTC(0x0da897b2), WTC(0x0e9123b3), + WTC(0x0f7ff200), WTC(0x10755696), WTC(0x11717f94), WTC(0x127474a0), + WTC(0x137e489d), WTC(0x148f1b02), WTC(0x15a6f15e), WTC(0x16c5b7c9), + WTC(0x17eb72b1), WTC(0x19183e51), WTC(0x1a4c2444), WTC(0x1b871b1c), + WTC(0x1cc92e92), WTC(0x1e127ffc), WTC(0x1f6319b9), WTC(0x20baef78), + WTC(0x221a0861), WTC(0x23807f94), WTC(0x24ee5a89), WTC(0x2663898d), + WTC(0x27e0101e), WTC(0x2964058d), WTC(0x2aef6bcf), WTC(0x2c8230fc), + WTC(0x2e1c545b), WTC(0x2fbde72b), WTC(0x3166e76f), WTC(0x33173f5d), + WTC(0x34cee8c3), WTC(0x368debe1), WTC(0x38543d4f), WTC(0x3a21bd94), + WTC(0x3bf6576f), WTC(0x3dd1ff07), WTC(0x3fb4948e), WTC(0x419de414), + WTC(0x438dc202), WTC(0x45840e7d), WTC(0x4780a435), WTC(0x4983609f), + WTC(0x4b8cc548), WTC(0x4d9df796), WTC(0x4fb81f46), WTC(0x51dc8690), + WTC(0x000d970d), WTC(0xfff7ea67), WTC(0xfff7fc3d), WTC(0x000d3de2), + WTC(0x003720ad), WTC(0x007515f1), WTC(0x00c68f04), WTC(0x012afd3b), + WTC(0x01a1d1ec), WTC(0x022a7e69), WTC(0x02c47408), WTC(0x036f2420), + WTC(0x042a0001), WTC(0x04f47905), WTC(0x05ce007e), WTC(0x06b607be), + WTC(0x07ac0028), WTC(0x08af5b01), WTC(0x09bf89a7), WTC(0x0adbfd6d), + WTC(0x0c042798), WTC(0x0d377997), WTC(0x0e7564b5), WTC(0x0fbd5a3a), + WTC(0x110ecb85), WTC(0x126929fb), WTC(0x13cbe6e6), WTC(0x15367376), + WTC(0x16a8413f), WTC(0x1820c15f), WTC(0x199f6568), WTC(0x1b239e6b), + WTC(0x1cacdde2), WTC(0x1e3a951a), WTC(0x1fcc356f), WTC(0x2161301f), + WTC(0x22f8f6b7), WTC(0x2492fa4a), WTC(0x262eac3f), WTC(0x27cb7e20), + WTC(0x2968e0c4), WTC(0x2b064625), WTC(0x2ca31f1a), WTC(0x2e3edd2a), + WTC(0x2fd8f19f), WTC(0x3170ce00), WTC(0x3305e32c), WTC(0x3497a2df), + WTC(0x36257e78), WTC(0x37aee70b), WTC(0x39334e05), WTC(0x3ab22498), + WTC(0x3c2adc2c), WTC(0x3d9ce645), WTC(0x3f07b3ef), WTC(0x406ab6ca), + WTC(0x41c56001), WTC(0x4317214a), WTC(0x445f6b34), WTC(0x459daf5d), + WTC(0x46d15f56), WTC(0x47f9ed71), WTC(0x4916d11f), WTC(0x4a275770), + WTC(0x4b2b2fff), WTC(0x4c219eae), WTC(0x4d0a20cb), WTC(0x4de4288e), + WTC(0x4eaf263d), WTC(0x4f6a8bb8), WTC(0x5015ca33), WTC(0x50b052dd), + WTC(0x51399757), WTC(0x51b108c6), WTC(0x5216190a), WTC(0x5268387c), + WTC(0x52a6d933), WTC(0x52d16c19), WTC(0x52e7628b), WTC(0x52e82ea3), + WTC(0x52d3407d), WTC(0x52a80a28), WTC(0x5265fd43), WTC(0x520c8a1d), + WTC(0x519b22c8), WTC(0x511138e0), WTC(0x506e3c82), WTC(0x4fb1a037), + WTC(0x4edad4e3), WTC(0x4de94c2d), WTC(0x4cdc76d8), WTC(0x4bb3c683), + WTC(0x4a6eacd2), WTC(0x490c9abe), WTC(0x478d04f1), WTC(0x45f00420), + WTC(0x4445673f), WTC(0x42ac0d2e), WTC(0x41338364), WTC(0x3fdb5b58), + WTC(0x3ea1c30f), WTC(0x3d842780), WTC(0x3c7fa763), WTC(0x3b911b96), + WTC(0x3ab560bf), WTC(0x39e95908), WTC(0x3929debb), WTC(0x3873bd4d), + WTC(0x37c31db2), WTC(0x3713a59c), WTC(0x3663deb2), WTC(0x35b52f23), + WTC(0x3507c61e), WTC(0x345a7f42), WTC(0x33ac7e0c), WTC(0x32fd366f), + WTC(0x324baa28), WTC(0x319674e9), WTC(0x30dd7e1a), WTC(0x3021f3e8), + WTC(0x2f63f903), WTC(0x2ea2a1aa), WTC(0x2dddd97b), WTC(0x2d166985), + WTC(0x2c4ca42f), WTC(0x2b805cca), WTC(0x2ab162aa), WTC(0x29df7b17), +}; + +const FIXP_WTB ELDAnalysis240[720] = { + WTC(0xfab9477b), WTC(0xfa899344), WTC(0xfa5845dd), WTC(0xfa259762), + WTC(0xf9f1c005), WTC(0xf9bcefe6), WTC(0xf9871e8b), WTC(0xf9503397), + WTC(0xf9183f47), WTC(0xf8df4eac), WTC(0xf8a53ba7), WTC(0xf869f0be), + WTC(0xf82d9759), WTC(0xf7f0593e), WTC(0xf7b2520a), WTC(0xf773a37c), + WTC(0xf73475ce), WTC(0xf6f4bedd), WTC(0xf6b455a8), WTC(0xf6739525), + WTC(0xf6332510), WTC(0xf5f3938b), WTC(0xf5b56073), WTC(0xf57900bd), + WTC(0xf53ee82d), WTC(0xf5079149), WTC(0xf4d36ffc), WTC(0xf4a2e526), + WTC(0xf4763d91), WTC(0xf44d9872), WTC(0xf426eaed), WTC(0xf3fdc161), + WTC(0xf3d001ff), WTC(0xf39dafcc), WTC(0xf366eb43), WTC(0xf32bdcdc), + WTC(0xf2ecab80), WTC(0xf2a97b34), WTC(0xf26265ae), WTC(0xf2178a6f), + WTC(0xf1c92458), WTC(0xf17752b9), WTC(0xf121e6ac), WTC(0xf0c89a63), + WTC(0xf06b15ef), WTC(0xf0092e86), WTC(0xefa2fd42), WTC(0xef397ebc), + WTC(0xeece51c6), WTC(0xee61e8b6), WTC(0xedf3d92e), WTC(0xed82c330), + WTC(0xed0d58bb), WTC(0xec94891b), WTC(0xec19d435), WTC(0xeb9e4e4e), + WTC(0xeb221000), WTC(0xeaa32422), WTC(0xea1fb440), WTC(0xe99695d2), + WTC(0xe90859ab), WTC(0xe8775114), WTC(0xe7e62b37), WTC(0xe7578147), + WTC(0xe6cb3ac1), WTC(0xe63f5696), WTC(0xe5afe916), WTC(0xe519090f), + WTC(0xe47aab0d), WTC(0xe3d6c2d0), WTC(0xe331dae7), WTC(0xe29031e1), + WTC(0xe1f40926), WTC(0xe15d87d2), WTC(0xe0c9d727), WTC(0xe0360ad5), + WTC(0xdf9f81af), WTC(0xdf04f9f9), WTC(0xde66697f), WTC(0xddc48ca1), + WTC(0xdd20a42a), WTC(0xdc7c6853), WTC(0xdbd9a476), WTC(0xdb398a8c), + WTC(0xda9cd7c2), WTC(0xda0365cf), WTC(0xd96cad85), WTC(0xd8d7b7a3), + WTC(0xd8439e8c), WTC(0xd7afb73d), WTC(0xd71b6347), WTC(0xd686149a), + WTC(0xd5efab2c), WTC(0xd558877e), WTC(0xd4c29dbc), WTC(0xd430a0aa), + WTC(0xd3a3d490), WTC(0xd31c588f), WTC(0xd297e075), WTC(0xd213ef33), + WTC(0xd18fd566), WTC(0xd10b8d3f), WTC(0xd087b250), WTC(0xd0054ef2), + WTC(0xcf85f94a), WTC(0xcf0af5f7), WTC(0xce94faf5), WTC(0xce229409), + WTC(0xcdb0b5f8), WTC(0xcd3ec554), WTC(0xcccdbf58), WTC(0xcc5f39d5), + WTC(0xcbf49ef5), WTC(0xcb8d5f73), WTC(0xcb28801c), WTC(0xcac5a265), + WTC(0xca64ad2e), WTC(0xca05d7fd), WTC(0xc9a96602), WTC(0xc94f9f79), + WTC(0xc8f2b954), WTC(0xc899e795), WTC(0xc83f94aa), WTC(0xc7e41f63), + WTC(0xc787e69f), WTC(0xc72b3fd0), WTC(0xc6ce3f0f), WTC(0xc670c175), + WTC(0xc61224cf), WTC(0xc5b21fec), WTC(0xc55202a4), WTC(0xc4f3353a), + WTC(0xc4968597), WTC(0xc43b93f2), WTC(0xc3e03d26), WTC(0xc383a011), + WTC(0xc3268aed), WTC(0xc2ca1039), WTC(0xc26f5bcc), WTC(0xc21726c9), + WTC(0xc1c1d5b2), WTC(0xc16f66ba), WTC(0xc11f76d9), WTC(0xc0d0a9f6), + WTC(0xc081cddb), WTC(0xc0333180), WTC(0xbfe5bb54), WTC(0xbf9aee90), + WTC(0xbf53d587), WTC(0xbf108855), WTC(0xbed0de05), WTC(0xbe9477d7), + WTC(0xbe5b030f), WTC(0xbe243642), WTC(0xbdefb72f), WTC(0xbdbd29df), + WTC(0xbd8c71ab), WTC(0xbd5d99cb), WTC(0xbd30e375), WTC(0xbd06afcc), + WTC(0xbcdf8c7f), WTC(0xbcbbf704), WTC(0xbc9c307e), WTC(0xbc803b86), + WTC(0xbc67c0c7), WTC(0xbc521d3d), WTC(0xbc3e6561), WTC(0xbc2bf2cb), + WTC(0xbc1a6872), WTC(0xbc09ce15), WTC(0xbbfa764f), WTC(0xbbed1356), + WTC(0xbbe257fa), WTC(0xbbda4099), WTC(0xbbd46a31), WTC(0xbbcffa76), + WTC(0xbbcc766d), WTC(0xbbca782f), WTC(0xbbcb16c7), WTC(0xbbcff77c), + WTC(0xbbd978e6), WTC(0xbbe68e5f), WTC(0xbbf593ed), WTC(0xbc04a834), + WTC(0xbc136941), WTC(0xbc2252c3), WTC(0xbc31723d), WTC(0xbc40ab92), + WTC(0xbc4ffe2d), WTC(0xbc5f7072), WTC(0xbc6f0520), WTC(0xbc7ebd23), + WTC(0xbc8e9746), WTC(0xbc9e942f), WTC(0xbcaeb633), WTC(0xbcbefe8b), + WTC(0xbccf69bb), WTC(0xbcdff92e), WTC(0xbcf0b04f), WTC(0xbd018ebd), + WTC(0xbd129192), WTC(0xbd23b9b8), WTC(0xbd350afb), WTC(0xbd46820e), + WTC(0xbd581bfc), WTC(0xbd69db11), WTC(0xbd7bbf57), WTC(0xbd8dc584), + WTC(0xbd9feaad), WTC(0xbdb231a4), WTC(0xbdc498ea), WTC(0xbdd71cd1), + WTC(0xbde9bb57), WTC(0xbdfc77d9), WTC(0xbe0f4e93), WTC(0xbe223ae5), + WTC(0xbe353cf5), WTC(0xbe485689), WTC(0xbe5b8329), WTC(0xbe6ebe88), + WTC(0xbe820afd), WTC(0xbe956811), WTC(0xbea8d109), WTC(0xbebc4352), + WTC(0xbecfc0fb), WTC(0xbee34a07), WTC(0xbef6d884), WTC(0xbf0a6bb1), + WTC(0xbf1e0685), WTC(0xbf31a685), WTC(0xbf45483c), WTC(0xbf58eb6b), + WTC(0xbf6c9376), WTC(0xbf803c90), WTC(0xbf93e4b9), WTC(0xbfa78d05), + WTC(0xbfbb3830), WTC(0xbfcee339), WTC(0xbfe28aa9), WTC(0xbff62b89), + WTC(0x8013a5f4), WTC(0x803acfd6), WTC(0x8061eec7), WTC(0x8088fc73), + WTC(0x80aff270), WTC(0x80d6cbe5), WTC(0x80fd8c2a), WTC(0x812437a8), + WTC(0x814ac94f), WTC(0x81713adc), WTC(0x81979098), WTC(0x81bdccb7), + WTC(0x81e3e738), WTC(0x8209dd04), WTC(0x822fb23a), WTC(0x825565bb), + WTC(0x827aed94), WTC(0x82a04909), WTC(0x82c57c85), WTC(0x82ea831c), + WTC(0x830f539d), WTC(0x8333eeba), WTC(0x8358585a), WTC(0x837c882e), + WTC(0x83a07742), WTC(0x83c428a5), WTC(0x83e79c4c), WTC(0x840aca65), + WTC(0x842dad81), WTC(0x84504ac0), WTC(0x84729fb1), WTC(0x8494a4f1), + WTC(0x84b65932), WTC(0x84d7c0f8), WTC(0x84f8d936), WTC(0x85199a59), + WTC(0x853a05a1), WTC(0x855a2023), WTC(0x8579e46e), WTC(0x85994d55), + WTC(0x85b86190), WTC(0x85d723e6), WTC(0x85f58fa9), WTC(0x8613a3ce), + WTC(0x863167b5), WTC(0x864eddfe), WTC(0x866c0138), WTC(0x8688d2e4), + WTC(0x86a55901), WTC(0x86c19497), WTC(0x86dd8390), WTC(0x86f9288f), + WTC(0x871487e0), WTC(0x872fadd0), WTC(0x874a9a1e), WTC(0x876519d0), + WTC(0x877f471e), WTC(0x8799fb36), WTC(0x87b48b97), WTC(0x87cba021), + WTC(0x880f67ae), WTC(0x885e0f91), WTC(0x88bc84cd), WTC(0x89244640), + WTC(0x8990a45d), WTC(0x89fe6766), WTC(0x8a6c9065), WTC(0x8adb31e6), + WTC(0x8b4ad5b3), WTC(0x8bbc2068), WTC(0x8c2f93ff), WTC(0x8ca5a922), + WTC(0x8d1ed72d), WTC(0x8d9b7ddb), WTC(0x8e1bd6cc), WTC(0x8ea01924), + WTC(0x8f287716), WTC(0x8fb5143e), WTC(0x9046074e), WTC(0x90db612b), + WTC(0x91753263), WTC(0x92138094), WTC(0x92b64cf3), WTC(0x935d96c9), + WTC(0x94095a56), WTC(0x94b98fd4), WTC(0x956e2a87), WTC(0x96271ff6), + WTC(0x96e46309), WTC(0x97a5e80d), WTC(0x986b9e55), WTC(0x993572af), + WTC(0x9a0350ce), WTC(0x9ad52154), WTC(0x9baad10f), WTC(0x9c844cdd), + WTC(0x9d618437), WTC(0x9e4265b2), WTC(0x9f26d9ad), WTC(0xa00ec9b0), + WTC(0xa0fa2916), WTC(0xa1e8ec20), WTC(0xa2daffa4), WTC(0xa3d05468), + WTC(0xa4c8e007), WTC(0xa5c49ae4), WTC(0xa6c37c24), WTC(0xa7c57d03), + WTC(0xa8ca9750), WTC(0xa9d2c7f2), WTC(0xaade0f6f), WTC(0xabec7177), + WTC(0xacfdf2b1), WTC(0xae129740), WTC(0xaf2a6321), WTC(0xb04563a6), + WTC(0xb163a2e6), WTC(0xb28524c4), WTC(0xb3a9eaf7), WTC(0xb4d1ff1b), + WTC(0x4a1d2880), WTC(0x48eee56e), WTC(0x47bda882), WTC(0x46895c79), + WTC(0x4551f8a1), WTC(0x4417817b), WTC(0x42da023d), WTC(0x419980ca), + WTC(0x4055f463), WTC(0x3f0f3b51), WTC(0x3dc56e18), WTC(0x3c78d943), + WTC(0x3b29bf3d), WTC(0x39d84ea0), WTC(0x3884337d), WTC(0x372d2371), + WTC(0x35d364ea), WTC(0x34774cef), WTC(0x33194d3a), WTC(0x31b9d586), + WTC(0x30594fcf), WTC(0x2ef80b63), WTC(0x2d9630d5), WTC(0x2c337c00), + WTC(0x2acf6a9e), WTC(0x296a3205), WTC(0x28046825), WTC(0x269f1752), + WTC(0x253b5314), WTC(0x23d9993f), WTC(0x227a3c77), WTC(0x211d59a0), + WTC(0x1fc314fd), WTC(0x1e6b9834), WTC(0x1d16eb58), WTC(0x1bc4f82e), + WTC(0x1a75e481), WTC(0x1929f389), WTC(0x17e16ee3), WTC(0x169cd758), + WTC(0x155d1ae5), WTC(0x14235182), WTC(0x12f051de), WTC(0x11c4993b), + WTC(0x109fdf4c), WTC(0x0f81351c), WTC(0x0e66c5e6), WTC(0x0d4f4b16), + WTC(0x0c39f013), WTC(0x0b25765d), WTC(0x0a10c51e), WTC(0x08fbee35), + WTC(0x07e7986f), WTC(0x06d25fe7), WTC(0x05ba1b52), WTC(0x049c33b7), + WTC(0x0379ceb9), WTC(0x025ee7c7), WTC(0x015edc1c), WTC(0x00978deb), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0xffe59474), WTC(0xffb158f8), WTC(0xff7d1275), WTC(0xff48f44c), + WTC(0xff1531e3), WTC(0xfee1faa9), WTC(0xfeaf626e), WTC(0xfe7d7227), + WTC(0xfe4c383f), WTC(0xfe1bc4ff), WTC(0xfdec297f), WTC(0xfdbd9f6c), + WTC(0xfd90bbf0), WTC(0xfd65ba73), WTC(0xfd3c2d32), WTC(0xfd13ab35), + WTC(0xfcebe811), WTC(0xfcc4eeae), WTC(0xfc9f1d64), WTC(0xfc7b48b0), + WTC(0xfc5a7282), WTC(0xfc3cef9b), WTC(0xfc229f4c), WTC(0xfc0a9e29), + WTC(0xfbf3dccb), WTC(0xfbdd80c3), WTC(0xfbc7556c), WTC(0xfbb289cf), + WTC(0xfba06f99), WTC(0xfb923aca), WTC(0xfb88bb57), WTC(0xfb84457f), + WTC(0xfb848c2e), WTC(0xfb88bd28), WTC(0xfb8feda1), WTC(0xfb9928eb), + WTC(0xfba38b9e), WTC(0xfbae711c), WTC(0xfbba3538), WTC(0xfbc7af9d), + WTC(0xfbd8485e), WTC(0xfbeda238), WTC(0xfc099de4), WTC(0xfc2d981f), + WTC(0xfc59fd0d), WTC(0xfc8e1583), WTC(0xfcc7e0d7), WTC(0xfd048d03), + WTC(0xfd40dfaf), WTC(0xfd7c1d35), WTC(0xfdb767cc), WTC(0xfdf64a9f), + WTC(0xfe3d0242), WTC(0xfe8e3316), WTC(0xfeead2b9), WTC(0xff4fff8c), + WTC(0xffba7b11), WTC(0x00281cae), WTC(0x009847d1), WTC(0x010cb653), + WTC(0x01870639), WTC(0x02089db5), WTC(0x0291b571), WTC(0x0321a4c1), + WTC(0x03b7eda0), WTC(0x04544c7f), WTC(0x04f74134), WTC(0x05a16810), + WTC(0x06525829), WTC(0x070994d4), WTC(0x07c767ba), WTC(0x088c69b8), + WTC(0x09598634), WTC(0x0a2f14ca), WTC(0x0b0c677b), WTC(0x0bf0f576), + WTC(0x0cdc7f73), WTC(0x0dceed54), WTC(0x0ec84a00), WTC(0x0fc8db86), + WTC(0x10d10278), WTC(0x11e0e05e), WTC(0x12f880cc), WTC(0x1418049b), + WTC(0x153f86b3), WTC(0x166ef5a3), WTC(0x17a64878), WTC(0x18e59dde), + WTC(0x1a2d088b), WTC(0x1b7c7e41), WTC(0x1cd40ab7), WTC(0x1e33d542), + WTC(0x1f9be7a5), WTC(0x210c344f), WTC(0x2284cb85), WTC(0x2405ca48), + WTC(0x258f2b7f), WTC(0x2720e063), WTC(0x28bafd49), WTC(0x2a5d950c), + WTC(0x2c0896e4), WTC(0x2dbbf7d4), WTC(0x2f77ca28), WTC(0x313c1273), + WTC(0x3308b7e4), WTC(0x34ddb0ec), WTC(0x36bb06b7), WTC(0x38a0a935), + WTC(0x3a8e7270), WTC(0x3c844ca9), WTC(0x3e82267e), WTC(0x4087ccfa), + WTC(0x42950352), WTC(0x44a99ce7), WTC(0x46c57093), WTC(0x48e84dbe), + WTC(0x4b127506), WTC(0x4d452d29), WTC(0x4f81e066), WTC(0x51ca11c4), + WTC(0x000c82e8), WTC(0xfff6f40c), WTC(0xfffa1260), WTC(0x001530bf), + WTC(0x0047a202), WTC(0x0090b903), WTC(0x00efc89f), WTC(0x016423af), + WTC(0x01ed1d0e), WTC(0x028a0796), WTC(0x033a3620), WTC(0x03fcfb89), + WTC(0x04d1aaaa), WTC(0x05b7965c), WTC(0x06ae1179), WTC(0x07b46ee8), + WTC(0x08ca0173), WTC(0x09ee1c00), WTC(0x0b201162), WTC(0x0c5f346e), + WTC(0x0daad808), WTC(0x0f024f17), WTC(0x1064ec4b), WTC(0x11d202c4), + WTC(0x1348e514), WTC(0x14c8e62f), WTC(0x1651590a), WTC(0x17e19051), + WTC(0x1978df27), WTC(0x1b169812), WTC(0x1cba0e15), WTC(0x1e629407), + WTC(0x200f7cd4), WTC(0x21c01b29), WTC(0x2373c228), WTC(0x2529c453), + WTC(0x26e174b9), WTC(0x289a262f), WTC(0x2a532bba), WTC(0x2c0bd7b2), + WTC(0x2dc37d92), WTC(0x2f796fce), WTC(0x312d017a), WTC(0x32dd8513), + WTC(0x348a4dde), WTC(0x3632aeb3), WTC(0x37d5fa29), WTC(0x39738334), + WTC(0x3b0a9c99), WTC(0x3c9a9926), WTC(0x3e22cc21), WTC(0x3fa287dc), + WTC(0x41191f89), WTC(0x4285e5fc), WTC(0x43e82e02), WTC(0x453f4a40), + WTC(0x468a8dd9), WTC(0x47c94c23), WTC(0x48fadc7c), WTC(0x4a1e75f9), + WTC(0x4b339ecf), WTC(0x4c3981b1), WTC(0x4d2f7cd3), WTC(0x4e14e381), + WTC(0x4ee90804), WTC(0x4fab3d6a), WTC(0x505ad6bd), WTC(0x50f726a3), + WTC(0x517f7fea), WTC(0x51f335fd), WTC(0x52519b0f), WTC(0x529a01f2), + WTC(0x52cbbe31), WTC(0x52e621d9), WTC(0x52e880aa), WTC(0x52d22c7a), + WTC(0x52a278a5), WTC(0x5258b880), WTC(0x51f43e1d), WTC(0x51745c38), + WTC(0x50d8669e), WTC(0x501faf0e), WTC(0x4f49897e), WTC(0x4e554804), + WTC(0x4d423d9e), WTC(0x4c0fbd8b), WTC(0x4abd1a4d), WTC(0x4949a698), + WTC(0x47b4b7f9), WTC(0x45fe2b6d), WTC(0x44375019), WTC(0x4284e96e), + WTC(0x40f7efa2), WTC(0x3f8f8b33), WTC(0x3e494311), WTC(0x3d21e35b), + WTC(0x3c15c621), WTC(0x3b2115f3), WTC(0x3a4008aa), WTC(0x396ed2a6), + WTC(0x38a99a1d), WTC(0x37ec1177), WTC(0x3730f154), WTC(0x36756c15), + WTC(0x35bafb0d), WTC(0x35020093), WTC(0x34492381), WTC(0x338f6226), + WTC(0x32d40a34), WTC(0x3215bd73), WTC(0x315302ce), WTC(0x308c7c41), + WTC(0x2fc3532f), WTC(0x2ef6de8f), WTC(0x2e265a7f), WTC(0x2d527bfd), + WTC(0x2c7bf035), WTC(0x2ba2975b), WTC(0x2ac63552), WTC(0x29e686ca), +}; + +const FIXP_WTB ELDAnalysis128[384] = { + WTC(0xfaa49e98), WTC(0xfa48929f), WTC(0xf9e7eb39), WTC(0xf983b829), + WTC(0xf91bc5cb), WTC(0xf8b0376f), WTC(0xf8408d62), WTC(0xf7cd8c1e), + WTC(0xf7580da3), WTC(0xf6e0b0dc), WTC(0xf667753c), WTC(0xf5efa4cf), + WTC(0xf57cb6de), WTC(0xf511b62b), WTC(0xf4b1a860), WTC(0xf45ee8f8), + WTC(0xf415710d), WTC(0xf3c0c4f3), WTC(0xf35c2af9), WTC(0xf2e89620), + WTC(0xf266f3cb), WTC(0xf1d819bf), WTC(0xf13cff2f), WTC(0xf09489d2), + WTC(0xefdcfa80), WTC(0xef182059), WTC(0xee4d6c60), WTC(0xed7b8da7), + WTC(0xec9c27b1), WTC(0xebb57d0d), WTC(0xeacb3918), WTC(0xe9d35591), + WTC(0xe8c9176e), WTC(0xe7b93e42), WTC(0xe6b10e47), WTC(0xe5a6b875), + WTC(0xe484c345), WTC(0xe3509f1b), WTC(0xe224254c), WTC(0xe10a4e18), + WTC(0xdff4a668), WTC(0xded3d881), WTC(0xdda5ed98), WTC(0xdc722d13), + WTC(0xdb437360), WTC(0xda1fefed), WTC(0xd90623b2), WTC(0xd7f070f5), + WTC(0xd6da361b), WTC(0xd5c0786f), WTC(0xd4a6e188), WTC(0xd39b37b4), + WTC(0xd2a01ff2), WTC(0xd1a8a05c), WTC(0xd0b0cf47), WTC(0xcfbd3527), + WTC(0xced6b8d7), WTC(0xcdff0a66), WTC(0xcd2978a4), WTC(0xcc587183), + WTC(0xcb93bfdb), WTC(0xcad80773), WTC(0xca233c2b), WTC(0xc9768b5e), + WTC(0xc8cc130c), WTC(0xc8231acd), WTC(0xc7768de4), WTC(0xc6c86bdf), + WTC(0xc6181aa1), WTC(0xc563f6ce), WTC(0xc4b33a2a), WTC(0xc4085fcf), + WTC(0xc35ae72e), WTC(0xc2ad7adf), WTC(0xc206ed94), WTC(0xc16a5744), + WTC(0xc0d59625), WTC(0xc041e21b), WTC(0xbfb1ee05), WTC(0xbf2d82ea), + WTC(0xbeb60fe9), WTC(0xbe499da8), WTC(0xbde61891), WTC(0xbd8975b6), + WTC(0xbd339d36), WTC(0xbce6a08b), WTC(0xbca5b2f9), WTC(0xbc721002), + WTC(0xbc494d41), WTC(0xbc266160), WTC(0xbc06d14f), WTC(0xbbec52c7), + WTC(0xbbdaaf79), WTC(0xbbd0be99), WTC(0xbbcae139), WTC(0xbbcd359c), + WTC(0xbbded5d3), WTC(0xbbfa58cf), WTC(0xbc162f9d), WTC(0xbc326534), + WTC(0xbc4f081c), WTC(0xbc6c1678), WTC(0xbc899f93), WTC(0xbca7a263), + WTC(0xbcc62954), WTC(0xbce52ddc), WTC(0xbd04bc7f), WTC(0xbd24cd8f), + WTC(0xbd456998), WTC(0xbd668428), WTC(0xbd88207c), WTC(0xbdaa2e4b), + WTC(0xbdccaf3e), WTC(0xbdef932c), WTC(0xbe12d936), WTC(0xbe366dd6), + WTC(0xbe5a4fd9), WTC(0xbe7e6b49), WTC(0xbea2bf3f), WTC(0xbec738b5), + WTC(0xbeebd791), WTC(0xbf108b49), WTC(0xbf3554aa), WTC(0xbf5a25cf), + WTC(0xbf7f020b), WTC(0xbfa3dd25), WTC(0xbfc8be1e), WTC(0xbfed95f6), + WTC(0x8024c933), WTC(0x806e24fb), WTC(0x80b73d96), WTC(0x80fff78c), + WTC(0x8148612f), WTC(0x8190626e), WTC(0x81d8030a), WTC(0x821f28e1), + WTC(0x8265d6be), WTC(0x82abed56), WTC(0x82f16e40), WTC(0x833636d2), + WTC(0x837a470a), WTC(0x83bd7be6), WTC(0x83ffd415), WTC(0x84412e66), + WTC(0x84818c4f), WTC(0x84c0d18c), WTC(0x84ff0429), WTC(0x853c09a5), + WTC(0x8577eacf), WTC(0x85b29402), WTC(0x85ec17bf), WTC(0x86246b50), + WTC(0x865ba7d7), WTC(0x8691c4c0), WTC(0x86c6d72a), WTC(0x86fae06c), + WTC(0x872dfcd1), WTC(0x87602c6a), WTC(0x87918a27), WTC(0x87c22ef8), + WTC(0x882f7c20), WTC(0x88dc38ab), WTC(0x89a52f47), WTC(0x8a737635), + WTC(0x8b43d08d), WTC(0x8c19be9f), WTC(0x8cf89ce7), WTC(0x8de337e9), + WTC(0x8edb3e02), WTC(0x8fe1e815), WTC(0x90f7e107), WTC(0x921d8b95), + WTC(0x9353008c), WTC(0x94982fd7), WTC(0x95ecdc6d), WTC(0x9750b87f), + WTC(0x98c36ad6), WTC(0x9a447674), WTC(0x9bd34e9c), WTC(0x9d6f76fa), + WTC(0x9f18780d), WTC(0xa0cdc487), WTC(0xa28eff8e), WTC(0xa45bbe4b), + WTC(0xa633baaf), WTC(0xa816bff0), WTC(0xaa04a90d), WTC(0xabfd7205), + WTC(0xae01356b), WTC(0xb0101477), WTC(0xb22a5244), WTC(0xb4500b02), + WTC(0x499946f3), WTC(0x475da5af), WTC(0x45173dce), WTC(0x42c610ad), + WTC(0x406a44b0), WTC(0x3e037ce8), WTC(0x3b92b806), WTC(0x39195cd0), + WTC(0x36962f17), WTC(0x340a1b28), WTC(0x3177cde6), WTC(0x2ee1f20d), + WTC(0x2c49b0d6), WTC(0x29ad3d74), WTC(0x270e9ec1), WTC(0x2474132f), + WTC(0x21e14a01), WTC(0x1f57704e), WTC(0x1cd758ce), WTC(0x1a610d48), + WTC(0x17f5db88), WTC(0x1598a188), WTC(0x134f7a40), WTC(0x111f1ca0), + WTC(0x0f053b83), WTC(0x0cf871db), WTC(0x0af19e60), WTC(0x08eaa56d), + WTC(0x06e3c473), WTC(0x04d25cc9), WTC(0x02b59b4d), WTC(0x00e5bc4c), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0xffcebf14), WTC(0xff6cc249), WTC(0xff0b8bda), WTC(0xfeac3e5f), + WTC(0xfe4f4638), WTC(0xfdf5052f), WTC(0xfd9e8cf4), WTC(0xfd4e34ea), + WTC(0xfd023142), WTC(0xfcb8f6f7), WTC(0xfc74e090), WTC(0xfc3b33e8), + WTC(0xfc0c1254), WTC(0xfbe1b2eb), WTC(0xfbb8ce73), WTC(0xfb97e511), + WTC(0xfb86273d), WTC(0xfb857a3e), WTC(0xfb918813), WTC(0xfba43692), + WTC(0xfbb96f24), WTC(0xfbd4dcc3), WTC(0xfc000cd5), WTC(0xfc458653), + WTC(0xfca6cd98), WTC(0xfd178c8e), WTC(0xfd872979), WTC(0xfdfa721e), + WTC(0xfe88c77c), WTC(0xff3c8b5c), WTC(0x00059ebc), WTC(0x00d91db0), + WTC(0x01bec555), WTC(0x02bdfb80), WTC(0x03d4c846), WTC(0x0501ad53), + WTC(0x064719b2), WTC(0x07a34a2b), WTC(0x0918814b), WTC(0x0aaaaa6b), + WTC(0x0c5728b9), WTC(0x0e1c1a0f), WTC(0x0ff9ccd1), WTC(0x11f21ffb), + WTC(0x1405d073), WTC(0x1635776b), WTC(0x1880f4e5), WTC(0x1ae8bdcb), + WTC(0x1d6cede3), WTC(0x200e1d93), WTC(0x22cc56fd), WTC(0x25a80858), + WTC(0x28a11bdd), WTC(0x2bb7e363), WTC(0x2eec2f0f), WTC(0x323e298d), + WTC(0x35ad7f0b), WTC(0x393a1989), WTC(0x3ce34a65), WTC(0x40a8683d), + WTC(0x44881c94), WTC(0x48813cb3), WTC(0x4c94523d), WTC(0x50c8eff0), + WTC(0x00000000), WTC(0x00000000), WTC(0x0053a1ea), WTC(0x00f67066), + WTC(0x01e3f61b), WTC(0x0317bdaf), WTC(0x048d51ca), WTC(0x06403d11), + WTC(0x082c0a34), WTC(0x0a4c43d2), WTC(0x0c9c748e), WTC(0x0f182718), + WTC(0x11bae616), WTC(0x14803c1b), WTC(0x1763b3e6), WTC(0x1a60d832), + WTC(0x1d73336b), WTC(0x20965068), WTC(0x23c5b9d6), WTC(0x26fcfa1a), + WTC(0x2a379c30), WTC(0x2d712a75), WTC(0x30a52fba), WTC(0x33cf36a9), + WTC(0x36eaca15), WTC(0x39f3743b), WTC(0x3ce4bfc2), WTC(0x3fba37b8), + WTC(0x426f6671), WTC(0x44ffd6e8), WTC(0x47671326), WTC(0x49a0b2d0), + WTC(0x4ba81be1), WTC(0x4d78fd1e), WTC(0x4f0ed503), WTC(0x50652e28), + WTC(0x51779351), WTC(0x52418fbe), WTC(0x52bead75), WTC(0x52ea76cc), + WTC(0x52c07793), WTC(0x523c3918), WTC(0x5159470e), WTC(0x50132b36), + WTC(0x4e657128), WTC(0x4c4ba31d), WTC(0x49c14b60), WTC(0x46c1e9e4), + WTC(0x4374e179), WTC(0x408377b1), WTC(0x3e0fa0aa), WTC(0x3c05d584), + WTC(0x3a4d9888), WTC(0x38cdd8fa), WTC(0x376b75c4), WTC(0x360c51a7), + WTC(0x34b12fdc), WTC(0x33550d9f), WTC(0x31f1955c), WTC(0x307ffcb2), + WTC(0x2f03c44d), WTC(0x2d7a6b86), WTC(0x2be6d3d4), WTC(0x2a48d219), +}; + +const FIXP_WTB ELDAnalysis120[360] = { + WTC(0xfaa1a40a), WTC(0xfa3f173d), WTC(0xf9d7760c), WTC(0xf96bcc12), + WTC(0xf8fbe82c), WTC(0xf887bb26), WTC(0xf80f131a), WTC(0xf7930d03), + WTC(0xf714aeaf), WTC(0xf693f5a1), WTC(0xf613385b), WTC(0xf596ef0e), + WTC(0xf522dcc4), WTC(0xf4bab1f6), WTC(0xf461700c), WTC(0xf412e1fe), + WTC(0xf3b769b1), WTC(0xf349eaca), WTC(0xf2cb9164), WTC(0xf23d6d7b), + WTC(0xf1a0ab28), WTC(0xf0f5c20f), WTC(0xf03aadd9), WTC(0xef6e8c66), + WTC(0xee984910), WTC(0xedbbcbf8), WTC(0xecd1435b), WTC(0xebdc1b77), + WTC(0xeae31338), WTC(0xe9dbea9a), WTC(0xe8c005ea), WTC(0xe79e7068), + WTC(0xe6856dce), WTC(0xe5657c79), WTC(0xe42928e9), WTC(0xe2e0756e), + WTC(0xe1a83cf5), WTC(0xe0801fab), WTC(0xdf52bfeb), WTC(0xde15d1b1), + WTC(0xdcce71ba), WTC(0xdb8931bf), WTC(0xda4fbbe8), WTC(0xd9220a98), + WTC(0xd7f9af0c), WTC(0xd6d0e1df), WTC(0xd5a41f15), WTC(0xd4790330), + WTC(0xd35f84b8), WTC(0xd255e881), WTC(0xd14daf00), WTC(0xd04638e2), + WTC(0xcf47dff7), WTC(0xce5b8850), WTC(0xcd77b1d6), WTC(0xcc960ec0), + WTC(0xcbc0a6a4), WTC(0xcaf6d4f7), WTC(0xca34fa7d), WTC(0xc97c26c0), + WTC(0xc8c6868a), WTC(0xc811f873), WTC(0xc7599db8), WTC(0xc69f9780), + WTC(0xc5e24043), WTC(0xc5226384), WTC(0xc468f547), WTC(0xc3b20be9), + WTC(0xc2f826f0), WTC(0xc242ea02), WTC(0xc19844ae), WTC(0xc0f80714), + WTC(0xc05a6da1), WTC(0xbfbfe6d3), WTC(0xbf31b5b4), WTC(0xbeb24843), + WTC(0xbe3f4cb4), WTC(0xbdd63599), WTC(0xbd74c6b2), WTC(0xbd1b7128), + WTC(0xbccd4b41), WTC(0xbc8dc08e), WTC(0xbc5ca2bd), WTC(0xbc3509f1), + WTC(0xbc11f904), WTC(0xbbf37848), WTC(0xbbddfb6b), WTC(0xbbd214ea), + WTC(0xbbcb3a11), WTC(0xbbcce581), WTC(0xbbdfa6ba), WTC(0xbbfd2fa5), + WTC(0xbc1ad516), WTC(0xbc390c16), WTC(0xbc57b323), WTC(0xbc76dd1f), + WTC(0xbc969172), WTC(0xbcb6d611), WTC(0xbcd7ac8e), WTC(0xbcf91af9), + WTC(0xbd1b20bd), WTC(0xbd3dc1f5), WTC(0xbd60f678), WTC(0xbd84bec7), + WTC(0xbda909c0), WTC(0xbdcdd76e), WTC(0xbdf3161c), WTC(0xbe18c1e5), + WTC(0xbe3ec6c5), WTC(0xbe651f19), WTC(0xbe8bb78b), WTC(0xbeb288e1), + WTC(0xbed9848b), WTC(0xbf00a16f), WTC(0xbf27d681), WTC(0xbf4f1958), + WTC(0xbf76680e), WTC(0xbf9db85c), WTC(0xbfc50e09), WTC(0xbfec5bdf), + WTC(0x80273bdb), WTC(0x807577de), WTC(0x80c3630d), WTC(0x8110e4cb), + WTC(0x815e05da), WTC(0x81aab20f), WTC(0x81f6e68e), WTC(0x824290d0), + WTC(0x828da04c), WTC(0x82d8061f), WTC(0x8321a740), WTC(0x836a77f9), + WTC(0x83b2574f), WTC(0x83f93cc1), WTC(0x843f04a6), WTC(0x8483acd1), + WTC(0x84c71639), WTC(0x850944da), WTC(0x854a1d3a), WTC(0x8589a437), + WTC(0x85c7ccf9), WTC(0x8604a42f), WTC(0x86402cfb), WTC(0x867a740d), + WTC(0x86b37ff2), WTC(0x86eb5f6f), WTC(0x87222109), WTC(0x8757eb5f), + WTC(0x878c8341), WTC(0x87c0be51), WTC(0x88345fca), WTC(0x88ef97fb), + WTC(0x89c778c5), WTC(0x8aa3cc20), WTC(0x8b833d56), WTC(0x8c6a42a9), + WTC(0x8d5cb787), WTC(0x8e5d7787), WTC(0x8f6e3c5b), WTC(0x90902640), + WTC(0x91c3ca28), WTC(0x9309622d), WTC(0x9460e7d0), WTC(0x95ca1ad2), + WTC(0x97449dff), WTC(0x98d00612), WTC(0x9a6bbc19), WTC(0x9c17164d), + WTC(0x9dd180fc), WTC(0x9f9a6304), WTC(0xa1711f56), WTC(0xa355425e), + WTC(0xa5465846), WTC(0xa744190f), WTC(0xa94e4ca9), WTC(0xab64dcf0), + WTC(0xad87e068), WTC(0xafb77bfa), WTC(0xb1f3fb2b), WTC(0xb43d885c), + WTC(0x49866451), WTC(0x4723e56b), WTC(0x44b51e8d), WTC(0x423a2180), + WTC(0x3fb2fe0f), WTC(0x3d1f78f2), WTC(0x3a8153b7), WTC(0x37d906ff), + WTC(0x35259e7e), WTC(0x3269ba18), WTC(0x2fa8c1ea), WTC(0x2ce4fbab), + WTC(0x2a1cebd5), WTC(0x27519ac1), WTC(0x248a2de9), WTC(0x21cb7a79), + WTC(0x1f16fc13), WTC(0x1c6d9a50), WTC(0x19cf83c1), WTC(0x173e9943), + WTC(0x14bf6a71), WTC(0x12598f74), WTC(0x100fe27f), WTC(0x0ddab46a), + WTC(0x0bafab9d), WTC(0x09864fc7), WTC(0x075d3ac8), WTC(0x052c0fc2), + WTC(0x02ea842b), WTC(0x00f21d19), WTC(0x00000000), WTC(0x00000000), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), WTC(0x00000000), + WTC(0xffcb7b4d), WTC(0xff62fb20), WTC(0xfefb82e2), WTC(0xfe965482), + WTC(0xfe33e4bf), WTC(0xfdd4b9b5), WTC(0xfd7b04ec), WTC(0xfd27cfe4), + WTC(0xfcd84cfd), WTC(0xfc8ce1d4), WTC(0xfc4b45e1), WTC(0xfc1666da), + WTC(0xfbe8af26), WTC(0xfbbcad9e), WTC(0xfb98c6ad), WTC(0xfb85dd87), + WTC(0xfb86355b), WTC(0xfb94589b), WTC(0xfba8ed8a), WTC(0xfbc0a735), + WTC(0xfbe23f8d), WTC(0xfc1a92c4), WTC(0xfc73315f), WTC(0xfce611a1), + WTC(0xfd5e94d9), WTC(0xfdd61cab), WTC(0xfe6427fd), WTC(0xff1c92da), + WTC(0xfff10542), WTC(0x00d1d58a), WTC(0x01c6daab), WTC(0x02d8dbc3), + WTC(0x040557a6), WTC(0x054b6f91), WTC(0x06ad2b2b), WTC(0x0828f4c5), + WTC(0x09c348d1), WTC(0x0b7dcb56), WTC(0x0d54d98e), WTC(0x0f47a599), + WTC(0x1157fa1b), WTC(0x138743ae), WTC(0x15d6423f), WTC(0x1844f0ae), + WTC(0x1ad3c273), WTC(0x1d82e65c), WTC(0x205306e1), WTC(0x23443d5c), + WTC(0x2656fae8), WTC(0x298b3a5a), WTC(0x2ce13a7d), WTC(0x3058e0e3), + WTC(0x33f22950), WTC(0x37acd0b2), WTC(0x3b885e6b), WTC(0x3f840412), + WTC(0x439e65ee), WTC(0x47d6014e), WTC(0x4c2aa857), WTC(0x50a46876), + WTC(0xfffe9b02), WTC(0x0004ac61), WTC(0x0069639d), WTC(0x0127578b), + WTC(0x02391efe), WTC(0x039950cc), WTC(0x054283bf), WTC(0x072f4eba), + WTC(0x095a488c), WTC(0x0bbe0802), WTC(0x0e5523f9), WTC(0x111a332e), + WTC(0x1407cca2), WTC(0x171886f5), WTC(0x1a46f927), WTC(0x1d8db9e1), + WTC(0x20e7600d), WTC(0x244e82a3), WTC(0x27bdb846), WTC(0x2b2f97a8), + WTC(0x2e9eb7ea), WTC(0x3205afd1), WTC(0x355f161c), WTC(0x38a581d7), + WTC(0x3bd3894a), WTC(0x3ee3c398), WTC(0x41d0c7ae), WTC(0x44952cb8), + WTC(0x472b8856), WTC(0x498e7eee), WTC(0x4bb88245), WTC(0x4da44d9f), + WTC(0x4f4c6b7d), WTC(0x50ab7298), WTC(0x51bbfa11), WTC(0x527898fb), + WTC(0x52dbe5f2), WTC(0x52e07778), WTC(0x5280e51f), WTC(0x51b7c4f1), + WTC(0x507fadc7), WTC(0x4ed336dd), WTC(0x4cacf749), WTC(0x4a078534), + WTC(0x46dd6dcc), WTC(0x43599e22), WTC(0x403f48da), WTC(0x3db1ed89), + WTC(0x3b98bd24), WTC(0x39d5b003), WTC(0x384a3292), WTC(0x36d328ff), + WTC(0x355e63a6), WTC(0x33ec69d8), WTC(0x32755ebd), WTC(0x30f01fce), + WTC(0x2f5d9646), WTC(0x2dbcc615), WTC(0x2c0fa145), WTC(0x2a56ce53), +}; diff --git a/fdk-aac/libAACenc/src/aacEnc_rom.h b/fdk-aac/libAACenc/src/aacEnc_rom.h new file mode 100644 index 0000000..fd50cab --- /dev/null +++ b/fdk-aac/libAACenc/src/aacEnc_rom.h @@ -0,0 +1,217 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M. Lohwasser, M. Gayer + + Description: + +*******************************************************************************/ + +/*! + \file + \brief Memory layout + \author Markus Lohwasser +*/ + +#ifndef AACENC_ROM_H +#define AACENC_ROM_H + +#include "common_fix.h" + +#include "psy_const.h" +#include "psy_configuration.h" +#include "FDK_tools_rom.h" +#include "FDK_lpc.h" + +/* + Huffman Tables +*/ +extern const ULONG FDKaacEnc_huff_ltab1_2[3][3][3][3]; +extern const ULONG FDKaacEnc_huff_ltab3_4[3][3][3][3]; +extern const ULONG FDKaacEnc_huff_ltab5_6[9][9]; +extern const ULONG FDKaacEnc_huff_ltab7_8[8][8]; +extern const ULONG FDKaacEnc_huff_ltab9_10[13][13]; +extern const UCHAR FDKaacEnc_huff_ltab11[17][17]; +extern const UCHAR FDKaacEnc_huff_ltabscf[121]; +extern const USHORT FDKaacEnc_huff_ctab1[3][3][3][3]; +extern const USHORT FDKaacEnc_huff_ctab2[3][3][3][3]; +extern const USHORT FDKaacEnc_huff_ctab3[3][3][3][3]; +extern const USHORT FDKaacEnc_huff_ctab4[3][3][3][3]; +extern const USHORT FDKaacEnc_huff_ctab5[9][9]; +extern const USHORT FDKaacEnc_huff_ctab6[9][9]; +extern const USHORT FDKaacEnc_huff_ctab7[8][8]; +extern const USHORT FDKaacEnc_huff_ctab8[8][8]; +extern const USHORT FDKaacEnc_huff_ctab9[13][13]; +extern const USHORT FDKaacEnc_huff_ctab10[13][13]; +extern const USHORT FDKaacEnc_huff_ctab11[21][17]; +extern const ULONG FDKaacEnc_huff_ctabscf[121]; + +/* + quantizer +*/ +#define MANT_DIGITS 9 +#define MANT_SIZE (1 << MANT_DIGITS) + +#if defined(ARCH_PREFER_MULT_32x16) +#define FIXP_QTD FIXP_SGL +#define QTC FX_DBL2FXCONST_SGL +#else +#define FIXP_QTD FIXP_DBL +#define QTC +#endif + +extern const FIXP_QTD FDKaacEnc_mTab_3_4[MANT_SIZE]; +extern const FIXP_QTD FDKaacEnc_quantTableQ[4]; +extern const FIXP_QTD FDKaacEnc_quantTableE[4]; + +extern const FIXP_DBL FDKaacEnc_mTab_4_3Elc[512]; +extern const FIXP_DBL FDKaacEnc_specExpMantTableCombElc[4][14]; +extern const UCHAR FDKaacEnc_specExpTableComb[4][14]; + +/* + table to count used number of bits +*/ +extern const SHORT FDKaacEnc_sideInfoTabLong[]; +extern const SHORT FDKaacEnc_sideInfoTabShort[]; + +/* + Psy Configuration constants +*/ +extern const SFB_PARAM_LONG p_FDKaacEnc_8000_long_1024; +extern const SFB_PARAM_SHORT p_FDKaacEnc_8000_short_128; +extern const SFB_PARAM_LONG p_FDKaacEnc_11025_long_1024; +extern const SFB_PARAM_SHORT p_FDKaacEnc_11025_short_128; +extern const SFB_PARAM_LONG p_FDKaacEnc_12000_long_1024; +extern const SFB_PARAM_SHORT p_FDKaacEnc_12000_short_128; +extern const SFB_PARAM_LONG p_FDKaacEnc_16000_long_1024; +extern const SFB_PARAM_SHORT p_FDKaacEnc_16000_short_128; +extern const SFB_PARAM_LONG p_FDKaacEnc_22050_long_1024; +extern const SFB_PARAM_SHORT p_FDKaacEnc_22050_short_128; +extern const SFB_PARAM_LONG p_FDKaacEnc_24000_long_1024; +extern const SFB_PARAM_SHORT p_FDKaacEnc_24000_short_128; +extern const SFB_PARAM_LONG p_FDKaacEnc_32000_long_1024; +extern const SFB_PARAM_SHORT p_FDKaacEnc_32000_short_128; +extern const SFB_PARAM_LONG p_FDKaacEnc_44100_long_1024; +extern const SFB_PARAM_SHORT p_FDKaacEnc_44100_short_128; +extern const SFB_PARAM_LONG p_FDKaacEnc_48000_long_1024; +extern const SFB_PARAM_SHORT p_FDKaacEnc_48000_short_128; +extern const SFB_PARAM_LONG p_FDKaacEnc_64000_long_1024; +extern const SFB_PARAM_SHORT p_FDKaacEnc_64000_short_128; +extern const SFB_PARAM_LONG p_FDKaacEnc_88200_long_1024; +extern const SFB_PARAM_SHORT p_FDKaacEnc_88200_short_128; +extern const SFB_PARAM_LONG p_FDKaacEnc_96000_long_1024; +extern const SFB_PARAM_SHORT p_FDKaacEnc_96000_short_128; + +/* + TNS filter coefficients +*/ +extern const FIXP_LPC FDKaacEnc_tnsEncCoeff3[8]; +extern const FIXP_LPC FDKaacEnc_tnsCoeff3Borders[8]; +extern const FIXP_LPC FDKaacEnc_tnsEncCoeff4[16]; +extern const FIXP_LPC FDKaacEnc_tnsCoeff4Borders[16]; + +#define WTC0 WTC +#define WTC1 WTC +#define WTC2 WTC + +extern const FIXP_WTB ELDAnalysis512[1536]; +extern const FIXP_WTB ELDAnalysis480[1440]; +extern const FIXP_WTB ELDAnalysis256[768]; +extern const FIXP_WTB ELDAnalysis240[720]; +extern const FIXP_WTB ELDAnalysis128[384]; +extern const FIXP_WTB ELDAnalysis120[360]; + +#endif /* #ifndef AACENC_ROM_H */ diff --git a/fdk-aac/libAACenc/src/aacenc.cpp b/fdk-aac/libAACenc/src/aacenc.cpp new file mode 100644 index 0000000..372df31 --- /dev/null +++ b/fdk-aac/libAACenc/src/aacenc.cpp @@ -0,0 +1,1057 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M. Schug / A. Groeschel + + Description: fast aac coder functions + +*******************************************************************************/ + +#include "aacenc.h" + +#include "bitenc.h" +#include "interface.h" +#include "psy_configuration.h" +#include "psy_main.h" +#include "qc_main.h" +#include "bandwidth.h" +#include "channel_map.h" +#include "tns_func.h" +#include "aacEnc_ram.h" + +#include "genericStds.h" + +#define BITRES_MAX_LD 4000 +#define BITRES_MIN_LD 500 +#define BITRATE_MAX_LD 70000 /* Max assumed bitrate for bitres calculation */ +#define BITRATE_MIN_LD 12000 /* Min assumed bitrate for bitres calculation */ + +INT FDKaacEnc_CalcBitsPerFrame(const INT bitRate, const INT frameLength, + const INT samplingRate) { + int shift = 0; + while ((frameLength & ~((1 << (shift + 1)) - 1)) == frameLength && + (samplingRate & ~((1 << (shift + 1)) - 1)) == samplingRate) { + shift++; + } + + return (bitRate * (frameLength >> shift)) / (samplingRate >> shift); +} + +INT FDKaacEnc_CalcBitrate(const INT bitsPerFrame, const INT frameLength, + const INT samplingRate) { + int shift = 0; + while ((frameLength & ~((1 << (shift + 1)) - 1)) == frameLength && + (samplingRate & ~((1 << (shift + 1)) - 1)) == samplingRate) { + shift++; + } + + return (bitsPerFrame * (samplingRate >> shift)) / (frameLength >> shift); +} + +static AAC_ENCODER_ERROR FDKaacEnc_InitCheckAncillary( + INT bitRate, INT framelength, INT ancillaryRate, INT *ancillaryBitsPerFrame, + INT sampleRate); + +INT FDKaacEnc_LimitBitrate(HANDLE_TRANSPORTENC hTpEnc, AUDIO_OBJECT_TYPE aot, + INT coreSamplingRate, INT frameLength, INT nChannels, + INT nChannelsEff, INT bitRate, INT averageBits, + INT *pAverageBitsPerFrame, + AACENC_BITRATE_MODE bitrateMode, INT nSubFrames) { + INT transportBits, prevBitRate, averageBitsPerFrame, minBitrate = 0, iter = 0; + INT minBitsPerFrame = 40 * nChannels; + if (isLowDelay(aot)) { + minBitrate = 8000 * nChannelsEff; + } + + do { + prevBitRate = bitRate; + averageBitsPerFrame = + FDKaacEnc_CalcBitsPerFrame(bitRate, frameLength, coreSamplingRate) / + nSubFrames; + + if (pAverageBitsPerFrame != NULL) { + *pAverageBitsPerFrame = averageBitsPerFrame; + } + + if (hTpEnc != NULL) { + transportBits = transportEnc_GetStaticBits(hTpEnc, averageBitsPerFrame); + } else { + /* Assume some worst case */ + transportBits = 208; + } + + bitRate = fMax(bitRate, + fMax(minBitrate, + FDKaacEnc_CalcBitrate((minBitsPerFrame + transportBits), + frameLength, coreSamplingRate))); + FDK_ASSERT(bitRate >= 0); + + bitRate = fMin(bitRate, FDKaacEnc_CalcBitrate( + (nChannelsEff * MIN_BUFSIZE_PER_EFF_CHAN), + frameLength, coreSamplingRate)); + FDK_ASSERT(bitRate >= 0); + + } while (prevBitRate != bitRate && iter++ < 3); + + //fprintf(stderr, "FDKaacEnc_LimitBitrate(): bitRate=%d\n", bitRate); + return bitRate; +} + +typedef struct { + AACENC_BITRATE_MODE bitrateMode; + int chanBitrate[2]; /* mono/stereo settings */ +} CONFIG_TAB_ENTRY_VBR; + +static const CONFIG_TAB_ENTRY_VBR configTabVBR[] = { + {AACENC_BR_MODE_CBR, {0, 0}}, + {AACENC_BR_MODE_VBR_1, {32000, 20000}}, + {AACENC_BR_MODE_VBR_2, {40000, 32000}}, + {AACENC_BR_MODE_VBR_3, {56000, 48000}}, + {AACENC_BR_MODE_VBR_4, {72000, 64000}}, + {AACENC_BR_MODE_VBR_5, {112000, 96000}}}; + +/*----------------------------------------------------------------------------- + + functionname: FDKaacEnc_GetVBRBitrate + description: Get VBR bitrate from vbr quality + input params: int vbrQuality (VBR0, VBR1, VBR2) + channelMode + returns: vbr bitrate + + ------------------------------------------------------------------------------*/ +INT FDKaacEnc_GetVBRBitrate(AACENC_BITRATE_MODE bitrateMode, + CHANNEL_MODE channelMode) { + INT bitrate = 0; + INT monoStereoMode = 0; /* default mono */ + + if (FDKaacEnc_GetMonoStereoMode(channelMode) == EL_MODE_STEREO) { + monoStereoMode = 1; + } + + switch (bitrateMode) { + case AACENC_BR_MODE_VBR_1: + case AACENC_BR_MODE_VBR_2: + case AACENC_BR_MODE_VBR_3: + case AACENC_BR_MODE_VBR_4: + case AACENC_BR_MODE_VBR_5: + bitrate = configTabVBR[bitrateMode].chanBitrate[monoStereoMode]; + break; + case AACENC_BR_MODE_INVALID: + case AACENC_BR_MODE_CBR: + case AACENC_BR_MODE_SFR: + case AACENC_BR_MODE_FF: + default: + bitrate = 0; + break; + } + + /* convert channel bitrate to overall bitrate*/ + bitrate *= FDKaacEnc_GetChannelModeConfiguration(channelMode)->nChannelsEff; + + return bitrate; +} + +/** + * \brief Convert encoder bitreservoir value for transport library. + * + * \param hAacEnc Encoder handle + * + * \return Corrected bitreservoir level used in transport library. + */ +static INT FDKaacEnc_EncBitresToTpBitres(const HANDLE_AAC_ENC hAacEnc) { + INT transportBitreservoir = 0; + + switch (hAacEnc->bitrateMode) { + case AACENC_BR_MODE_CBR: + transportBitreservoir = + hAacEnc->qcKernel->bitResTot; /* encoder bitreservoir level */ + break; + case AACENC_BR_MODE_VBR_1: + case AACENC_BR_MODE_VBR_2: + case AACENC_BR_MODE_VBR_3: + case AACENC_BR_MODE_VBR_4: + case AACENC_BR_MODE_VBR_5: + transportBitreservoir = FDK_INT_MAX; /* signal variable bitrate */ + break; + case AACENC_BR_MODE_SFR: + transportBitreservoir = 0; /* super framing and fixed framing */ + break; /* without bitreservoir signaling */ + default: + case AACENC_BR_MODE_INVALID: + transportBitreservoir = 0; /* invalid configuration*/ + } + + if (hAacEnc->config->audioMuxVersion == 2) { + transportBitreservoir = + MIN_BUFSIZE_PER_EFF_CHAN * hAacEnc->channelMapping.nChannelsEff; + } + + return transportBitreservoir; +} + +INT FDKaacEnc_GetBitReservoirState(const HANDLE_AAC_ENC hAacEncoder) { + return FDKaacEnc_EncBitresToTpBitres(hAacEncoder); +} + +/*----------------------------------------------------------------------------- + + functionname: FDKaacEnc_AacInitDefaultConfig + description: gives reasonable default configuration + returns: --- + + ------------------------------------------------------------------------------*/ +void FDKaacEnc_AacInitDefaultConfig(AACENC_CONFIG *config) { + /* make the preinitialization of the structs flexible */ + FDKmemclear(config, sizeof(AACENC_CONFIG)); + + /* default ancillary */ + config->anc_Rate = 0; /* no ancillary data */ + config->ancDataBitRate = 0; /* no additional consumed bitrate */ + + /* default configurations */ + config->bitRate = -1; /* bitrate must be set*/ + config->averageBits = + -1; /* instead of bitrate/s we can configure bits/superframe */ + config->bitrateMode = + AACENC_BR_MODE_CBR; /* set bitrate mode to constant bitrate */ + config->bandWidth = 0; /* get bandwidth from table */ + config->useTns = TNS_ENABLE_MASK; /* tns enabled completly */ + config->usePns = + 1; /* depending on channelBitrate this might be set to 0 later */ + config->useIS = 1; /* Intensity Stereo Configuration */ + config->useMS = 1; /* MS Stereo tool */ + config->framelength = -1; /* Framesize not configured */ + config->syntaxFlags = 0; /* default syntax with no specialities */ + config->epConfig = -1; /* no ER syntax -> no additional error protection */ + config->nSubFrames = 1; /* default, no sub frames */ + config->channelOrder = CH_ORDER_MPEG; /* Use MPEG channel ordering. */ + config->channelMode = MODE_UNKNOWN; + config->minBitsPerFrame = -1; /* minum number of bits in each AU */ + config->maxBitsPerFrame = -1; /* minum number of bits in each AU */ + config->audioMuxVersion = -1; /* audio mux version not configured */ + config->downscaleFactor = + 1; /* downscale factor for ELD reduced delay mode, 1 is normal ELD */ +} + +/*--------------------------------------------------------------------------- + + functionname: FDKaacEnc_Open + description: allocate and initialize a new encoder instance + returns: error code + + ---------------------------------------------------------------------------*/ +AAC_ENCODER_ERROR FDKaacEnc_Open(HANDLE_AAC_ENC *phAacEnc, const INT nElements, + const INT nChannels, const INT nSubFrames) { + AAC_ENCODER_ERROR ErrorStatus; + AAC_ENC *hAacEnc = NULL; + UCHAR *dynamicRAM = NULL; + + if (phAacEnc == NULL) { + return AAC_ENC_INVALID_HANDLE; + } + + /* allocate encoder structure */ + hAacEnc = GetRam_aacEnc_AacEncoder(); + if (hAacEnc == NULL) { + ErrorStatus = AAC_ENC_NO_MEMORY; + goto bail; + } + FDKmemclear(hAacEnc, sizeof(AAC_ENC)); + + if (NULL == (hAacEnc->dynamic_RAM = GetAACdynamic_RAM())) { + ErrorStatus = AAC_ENC_NO_MEMORY; + goto bail; + } + dynamicRAM = (UCHAR *)hAacEnc->dynamic_RAM; + + /* allocate the Psy aud Psy Out structure */ + ErrorStatus = + FDKaacEnc_PsyNew(&hAacEnc->psyKernel, nElements, nChannels, dynamicRAM); + if (ErrorStatus != AAC_ENC_OK) goto bail; + + ErrorStatus = FDKaacEnc_PsyOutNew(hAacEnc->psyOut, nElements, nChannels, + nSubFrames, dynamicRAM); + if (ErrorStatus != AAC_ENC_OK) goto bail; + + /* allocate the Q&C Out structure */ + ErrorStatus = FDKaacEnc_QCOutNew(hAacEnc->qcOut, nElements, nChannels, + nSubFrames, dynamicRAM); + if (ErrorStatus != AAC_ENC_OK) goto bail; + + /* allocate the Q&C kernel */ + ErrorStatus = FDKaacEnc_QCNew(&hAacEnc->qcKernel, nElements, dynamicRAM); + if (ErrorStatus != AAC_ENC_OK) goto bail; + + hAacEnc->maxChannels = nChannels; + hAacEnc->maxElements = nElements; + hAacEnc->maxFrames = nSubFrames; + +bail: + *phAacEnc = hAacEnc; + return ErrorStatus; +} + +AAC_ENCODER_ERROR FDKaacEnc_Initialize( + HANDLE_AAC_ENC hAacEnc, + AACENC_CONFIG *config, /* pre-initialized config struct */ + HANDLE_TRANSPORTENC hTpEnc, ULONG initFlags) { + AAC_ENCODER_ERROR ErrorStatus; + INT psyBitrate, tnsMask; // INT profile = 1; + CHANNEL_MAPPING *cm = NULL; + + INT mbfac_e, qbw; + FIXP_DBL mbfac, bw_ratio; + QC_INIT qcInit; + INT averageBitsPerFrame = 0; + int bitresMin = 0; /* the bitreservoir is always big for AAC-LC */ + const CHANNEL_MODE prevChannelMode = hAacEnc->encoderMode; + + if (config == NULL) return AAC_ENC_INVALID_HANDLE; + + /******************* sanity checks *******************/ + + /* check config structure */ + if (config->nChannels < 1 || config->nChannels > (8)) { + return AAC_ENC_UNSUPPORTED_CHANNELCONFIG; + } + + /* check sample rate */ + switch (config->sampleRate) { + case 8000: + case 11025: + case 12000: + case 16000: + case 22050: + case 24000: + case 32000: + case 44100: + case 48000: + case 64000: + case 88200: + case 96000: + break; + default: + return AAC_ENC_UNSUPPORTED_SAMPLINGRATE; + } + + /* bitrate has to be set */ + if (config->bitRate == -1) { + return AAC_ENC_UNSUPPORTED_BITRATE; + } + + INT superframe_size = 110*8*(config->bitRate/8000); + INT frames_per_superframe = 6; + INT staticBits = 0; + if((config->syntaxFlags & AC_DAB) && hTpEnc) { + staticBits = transportEnc_GetStaticBits(hTpEnc, 0); + switch(config->sampleRate) { + case 48000: + frames_per_superframe=6; + break; + case 32000: + frames_per_superframe=4; + break; + case 24000: + frames_per_superframe=3; + break; + case 16000: + frames_per_superframe=2; + break; + } + + //config->nSubFrames = frames_per_superframe; + //fprintf(stderr, "DAB+ superframe size=%d\n", superframe_size); + config->bitRate = (superframe_size - 16*(frames_per_superframe-1) - staticBits) * 1000/120; + //fprintf(stderr, "DAB+ tuned bitrate=%d\n", config->bitRate); + config->maxBitsPerFrame = (superframe_size - 16*(frames_per_superframe-1) - staticBits) / frames_per_superframe; + config->maxBitsPerFrame += 7; /*padding*/ + //config->bitreservoir=(superframe_size - 16*(frames_per_superframe-1) - staticBits - 2*8)/frames_per_superframe; + //fprintf(stderr, "DAB+ tuned maxBitsPerFrame=%d\n", (superframe_size - 16*(frames_per_superframe-1) - staticBits)/frames_per_superframe); + } + + /* check bit rate */ + + if (FDKaacEnc_LimitBitrate( + hTpEnc, config->audioObjectType, config->sampleRate, + config->framelength, config->nChannels, + FDKaacEnc_GetChannelModeConfiguration(config->channelMode) + ->nChannelsEff, + config->bitRate, config->averageBits, &averageBitsPerFrame, + config->bitrateMode, config->nSubFrames) != config->bitRate && + !(AACENC_BR_MODE_IS_VBR(config->bitrateMode))) { + return AAC_ENC_UNSUPPORTED_BITRATE; + } + + if (config->syntaxFlags & AC_ER_VCB11) { + return AAC_ENC_UNSUPPORTED_ER_FORMAT; + } + if (config->syntaxFlags & AC_ER_HCR) { + return AAC_ENC_UNSUPPORTED_ER_FORMAT; + } + + /* check frame length */ + switch (config->framelength) { + case 1024: + case 960: + if (isLowDelay(config->audioObjectType)) { + return AAC_ENC_INVALID_FRAME_LENGTH; + } + break; + case 128: + case 256: + case 512: + case 120: + case 240: + case 480: + if (!isLowDelay(config->audioObjectType)) { + return AAC_ENC_INVALID_FRAME_LENGTH; + } + break; + default: + return AAC_ENC_INVALID_FRAME_LENGTH; + } + + if (config->anc_Rate != 0) { + ErrorStatus = FDKaacEnc_InitCheckAncillary( + config->bitRate, config->framelength, config->anc_Rate, + &hAacEnc->ancillaryBitsPerFrame, config->sampleRate); + if (ErrorStatus != AAC_ENC_OK) goto bail; + + /* update estimated consumed bitrate */ + config->ancDataBitRate += + FDKaacEnc_CalcBitrate(hAacEnc->ancillaryBitsPerFrame, + config->framelength, config->sampleRate); + } + + /* maximal allowed DSE bytes in frame */ + config->maxAncBytesPerAU = + fMin((256), fMax(0, FDKaacEnc_CalcBitsPerFrame( + (config->bitRate - (config->nChannels * 8000)), + config->framelength, config->sampleRate) >> + 3)); + + /* bind config to hAacEnc->config */ + hAacEnc->config = config; + + /* set hAacEnc->bitrateMode */ + hAacEnc->bitrateMode = config->bitrateMode; + + hAacEnc->encoderMode = config->channelMode; + + ErrorStatus = FDKaacEnc_InitChannelMapping( + hAacEnc->encoderMode, config->channelOrder, &hAacEnc->channelMapping); + if (ErrorStatus != AAC_ENC_OK) goto bail; + + cm = &hAacEnc->channelMapping; + + ErrorStatus = FDKaacEnc_DetermineBandWidth( + config->bandWidth, config->bitRate - config->ancDataBitRate, + hAacEnc->bitrateMode, config->sampleRate, config->framelength, cm, + hAacEnc->encoderMode, &hAacEnc->config->bandWidth); + if (ErrorStatus != AAC_ENC_OK) goto bail; + + hAacEnc->bandwidth90dB = (INT)hAacEnc->config->bandWidth; + + tnsMask = config->useTns ? TNS_ENABLE_MASK : 0x0; + psyBitrate = config->bitRate - config->ancDataBitRate; + + if ((hAacEnc->encoderMode != prevChannelMode) || (initFlags != 0)) { + /* Reinitialize psych states in case of channel configuration change ore if + * full reset requested. */ + ErrorStatus = FDKaacEnc_psyInit(hAacEnc->psyKernel, hAacEnc->psyOut, + hAacEnc->maxFrames, hAacEnc->maxChannels, + config->audioObjectType, cm); + if (ErrorStatus != AAC_ENC_OK) goto bail; + } + + ErrorStatus = FDKaacEnc_psyMainInit( + hAacEnc->psyKernel, config->audioObjectType, cm, config->sampleRate, + config->framelength, psyBitrate, tnsMask, hAacEnc->bandwidth90dB, + config->usePns, config->useIS, config->useMS, config->syntaxFlags, + initFlags); + if (ErrorStatus != AAC_ENC_OK) goto bail; + + ErrorStatus = FDKaacEnc_QCOutInit(hAacEnc->qcOut, hAacEnc->maxFrames, cm); + if (ErrorStatus != AAC_ENC_OK) goto bail; + + qcInit.channelMapping = &hAacEnc->channelMapping; + qcInit.sceCpe = 0; + + if (AACENC_BR_MODE_IS_VBR(config->bitrateMode)) { + qcInit.averageBits = (averageBitsPerFrame + 7) & ~7; + qcInit.bitRes = MIN_BUFSIZE_PER_EFF_CHAN * cm->nChannelsEff; + qcInit.maxBits = MIN_BUFSIZE_PER_EFF_CHAN * cm->nChannelsEff; + qcInit.maxBits = (config->maxBitsPerFrame != -1) + ? fixMin(qcInit.maxBits, config->maxBitsPerFrame) + : qcInit.maxBits; + qcInit.maxBits = fixMax(qcInit.maxBits, (averageBitsPerFrame + 7) & ~7); + qcInit.minBits = + (config->minBitsPerFrame != -1) ? config->minBitsPerFrame : 0; + qcInit.minBits = fixMin(qcInit.minBits, averageBitsPerFrame & ~7); + } else { + INT bitreservoir = -1; /* default bitrservoir size*/ + if (isLowDelay(config->audioObjectType)) { + INT brPerChannel = config->bitRate / config->nChannels; + brPerChannel = fMin(BITRATE_MAX_LD, fMax(BITRATE_MIN_LD, brPerChannel)); + + /* bitreservoir = + * (maxBitRes-minBitRes)/(maxBitRate-minBitrate)*(bitRate-minBitrate)+minBitRes; + */ + FIXP_DBL slope = fDivNorm( + (brPerChannel - BITRATE_MIN_LD), + BITRATE_MAX_LD - BITRATE_MIN_LD); /* calc slope for interpolation */ + bitreservoir = fMultI(slope, (INT)(BITRES_MAX_LD - BITRES_MIN_LD)) + + BITRES_MIN_LD; /* interpolate */ + bitreservoir = bitreservoir & ~7; /* align to bytes */ + bitresMin = BITRES_MIN_LD; + } + + int maxBitres; + qcInit.averageBits = (averageBitsPerFrame + 7) & ~7; + maxBitres = + (MIN_BUFSIZE_PER_EFF_CHAN * cm->nChannelsEff) - qcInit.averageBits; + qcInit.bitRes = + (bitreservoir != -1) ? fMin(bitreservoir, maxBitres) : maxBitres; + + qcInit.maxBits = fixMin(MIN_BUFSIZE_PER_EFF_CHAN * cm->nChannelsEff, + ((averageBitsPerFrame + 7) & ~7) + qcInit.bitRes); + qcInit.maxBits = (config->maxBitsPerFrame != -1) + ? fixMin(qcInit.maxBits, config->maxBitsPerFrame) + : qcInit.maxBits; + qcInit.maxBits = + fixMin(MIN_BUFSIZE_PER_EFF_CHAN * cm->nChannelsEff, + fixMax(qcInit.maxBits, (averageBitsPerFrame + 7 + 8) & ~7)); + + qcInit.minBits = fixMax( + 0, ((averageBitsPerFrame - 1) & ~7) - qcInit.bitRes - + transportEnc_GetStaticBits( + hTpEnc, ((averageBitsPerFrame + 7) & ~7) + qcInit.bitRes)); + qcInit.minBits = (config->minBitsPerFrame != -1) + ? fixMax(qcInit.minBits, config->minBitsPerFrame) + : qcInit.minBits; + qcInit.minBits = fixMin( + qcInit.minBits, (averageBitsPerFrame - + transportEnc_GetStaticBits(hTpEnc, qcInit.maxBits)) & + ~7); + } + + qcInit.sampleRate = config->sampleRate; + qcInit.isLowDelay = isLowDelay(config->audioObjectType) ? 1 : 0; + qcInit.nSubFrames = config->nSubFrames; + qcInit.padding.paddingRest = config->sampleRate; + + if (qcInit.bitRes >= bitresMin * config->nChannels) { + qcInit.bitResMode = AACENC_BR_MODE_FULL; /* full bitreservoir */ + } else if (qcInit.bitRes > 0) { + qcInit.bitResMode = AACENC_BR_MODE_REDUCED; /* reduced bitreservoir */ + } else { + qcInit.bitResMode = AACENC_BR_MODE_DISABLED; /* disabled bitreservoir */ + } + + /* Configure bitrate distribution strategy. */ + switch (config->channelMode) { + case MODE_1_2: + case MODE_1_2_1: + case MODE_1_2_2: + case MODE_1_2_2_1: + case MODE_6_1: + case MODE_1_2_2_2_1: + case MODE_7_1_BACK: + case MODE_7_1_TOP_FRONT: + case MODE_7_1_REAR_SURROUND: + case MODE_7_1_FRONT_CENTER: + qcInit.bitDistributionMode = 0; /* over all elements bitrate estimation */ + break; + case MODE_1: + case MODE_2: + default: /* all non mpeg defined channel modes */ + qcInit.bitDistributionMode = 1; /* element-wise bit bitrate estimation */ + } /* config->channelMode */ + + /* Calc meanPe: qcInit.meanPe = 10.0f * FRAME_LEN_LONG * + * hAacEnc->bandwidth90dB/(config->sampleRate/2.0f); */ + bw_ratio = + fDivNorm((FIXP_DBL)(10 * config->framelength * hAacEnc->bandwidth90dB), + (FIXP_DBL)(config->sampleRate), &qbw); + qcInit.meanPe = + fMax((INT)scaleValue(bw_ratio, qbw + 1 - (DFRACT_BITS - 1)), 1); + + /* Calc maxBitFac, scale it to 24 bit accuracy */ + mbfac = fDivNorm(qcInit.maxBits, qcInit.averageBits / qcInit.nSubFrames, + &mbfac_e); + qcInit.maxBitFac = scaleValue(mbfac, -(DFRACT_BITS - 1 - 24 - mbfac_e)); + + switch (config->bitrateMode) { + case AACENC_BR_MODE_CBR: + qcInit.bitrateMode = QCDATA_BR_MODE_CBR; + break; + case AACENC_BR_MODE_VBR_1: + qcInit.bitrateMode = QCDATA_BR_MODE_VBR_1; + break; + case AACENC_BR_MODE_VBR_2: + qcInit.bitrateMode = QCDATA_BR_MODE_VBR_2; + break; + case AACENC_BR_MODE_VBR_3: + qcInit.bitrateMode = QCDATA_BR_MODE_VBR_3; + break; + case AACENC_BR_MODE_VBR_4: + qcInit.bitrateMode = QCDATA_BR_MODE_VBR_4; + break; + case AACENC_BR_MODE_VBR_5: + qcInit.bitrateMode = QCDATA_BR_MODE_VBR_5; + break; + case AACENC_BR_MODE_SFR: + qcInit.bitrateMode = QCDATA_BR_MODE_SFR; + break; + case AACENC_BR_MODE_FF: + qcInit.bitrateMode = QCDATA_BR_MODE_FF; + break; + default: + ErrorStatus = AAC_ENC_UNSUPPORTED_BITRATE_MODE; + goto bail; + } + + qcInit.invQuant = (config->useRequant) ? 2 : 0; + + /* maxIterations should be set to the maximum number of requantization + * iterations that are allowed before the crash recovery functionality is + * activated. This setting should be adjusted to the processing power + * available, i.e. to the processing power headroom in one frame that is still + * left after normal encoding without requantization. Please note that if + * activated this functionality is used most likely only in cases where the + * encoder is operating beyond recommended settings, i.e. the audio quality is + * suboptimal anyway. Activating the crash recovery does not further reduce + * audio quality significantly in these cases. */ + if (isLowDelay(config->audioObjectType)) { + qcInit.maxIterations = 2; + } else { + qcInit.maxIterations = 5; + } + + qcInit.bitrate = config->bitRate - config->ancDataBitRate; + + qcInit.staticBits = transportEnc_GetStaticBits( + hTpEnc, qcInit.averageBits / qcInit.nSubFrames); + + ErrorStatus = FDKaacEnc_QCInit(hAacEnc->qcKernel, &qcInit, initFlags); + if (ErrorStatus != AAC_ENC_OK) goto bail; + + /* Map virtual aot's to intern aot used in bitstream writer. */ + switch (hAacEnc->config->audioObjectType) { + case AOT_MP2_AAC_LC: + hAacEnc->aot = AOT_AAC_LC; + break; + case AOT_MP2_SBR: + hAacEnc->aot = AOT_SBR; + break; + default: + hAacEnc->aot = hAacEnc->config->audioObjectType; + } + + /* common things */ + + return AAC_ENC_OK; + +bail: + + return ErrorStatus; +} + +/*--------------------------------------------------------------------------- + + functionname: FDKaacEnc_EncodeFrame + description: encodes one frame + returns: error code + + ---------------------------------------------------------------------------*/ +AAC_ENCODER_ERROR FDKaacEnc_EncodeFrame( + HANDLE_AAC_ENC hAacEnc, /* encoder handle */ + HANDLE_TRANSPORTENC hTpEnc, INT_PCM *RESTRICT inputBuffer, + const UINT inputBufferBufSize, INT *nOutBytes, + AACENC_EXT_PAYLOAD extPayload[MAX_TOTAL_EXT_PAYLOADS]) { + AAC_ENCODER_ERROR ErrorStatus; + int el, n, c = 0; + UCHAR extPayloadUsed[MAX_TOTAL_EXT_PAYLOADS]; + + CHANNEL_MAPPING *cm = &hAacEnc->channelMapping; + + PSY_OUT *psyOut = hAacEnc->psyOut[c]; + QC_OUT *qcOut = hAacEnc->qcOut[c]; + + FDKmemclear(extPayloadUsed, MAX_TOTAL_EXT_PAYLOADS * sizeof(UCHAR)); + + qcOut->elementExtBits = 0; /* sum up all extended bit of each element */ + qcOut->staticBits = 0; /* sum up side info bits of each element */ + qcOut->totalNoRedPe = 0; /* sum up PE */ + + /* advance psychoacoustics */ + for (el = 0; el < cm->nElements; el++) { + ELEMENT_INFO elInfo = cm->elInfo[el]; + + if ((elInfo.elType == ID_SCE) || (elInfo.elType == ID_CPE) || + (elInfo.elType == ID_LFE)) { + int ch; + + /* update pointer!*/ + for (ch = 0; ch < elInfo.nChannelsInEl; ch++) { + PSY_OUT_CHANNEL *psyOutChan = + psyOut->psyOutElement[el]->psyOutChannel[ch]; + QC_OUT_CHANNEL *qcOutChan = qcOut->qcElement[el]->qcOutChannel[ch]; + + psyOutChan->mdctSpectrum = qcOutChan->mdctSpectrum; + psyOutChan->sfbSpreadEnergy = qcOutChan->sfbSpreadEnergy; + psyOutChan->sfbEnergy = qcOutChan->sfbEnergy; + psyOutChan->sfbEnergyLdData = qcOutChan->sfbEnergyLdData; + psyOutChan->sfbMinSnrLdData = qcOutChan->sfbMinSnrLdData; + psyOutChan->sfbThresholdLdData = qcOutChan->sfbThresholdLdData; + } + + ErrorStatus = FDKaacEnc_psyMain( + elInfo.nChannelsInEl, hAacEnc->psyKernel->psyElement[el], + hAacEnc->psyKernel->psyDynamic, hAacEnc->psyKernel->psyConf, + psyOut->psyOutElement[el], inputBuffer, inputBufferBufSize, + cm->elInfo[el].ChannelIndex, cm->nChannels); + + if (ErrorStatus != AAC_ENC_OK) return ErrorStatus; + + /* FormFactor, Pe and staticBitDemand calculation */ + ErrorStatus = FDKaacEnc_QCMainPrepare( + &elInfo, hAacEnc->qcKernel->hAdjThr->adjThrStateElem[el], + psyOut->psyOutElement[el], qcOut->qcElement[el], hAacEnc->aot, + hAacEnc->config->syntaxFlags, hAacEnc->config->epConfig); + + if (ErrorStatus != AAC_ENC_OK) return ErrorStatus; + + /*-------------------------------------------- */ + + qcOut->qcElement[el]->extBitsUsed = 0; + qcOut->qcElement[el]->nExtensions = 0; + /* reset extension payload */ + FDKmemclear(&qcOut->qcElement[el]->extension, + (1) * sizeof(QC_OUT_EXTENSION)); + + for (n = 0; n < MAX_TOTAL_EXT_PAYLOADS; n++) { + if (!extPayloadUsed[n] && (extPayload[n].associatedChElement == el) && + (extPayload[n].dataSize > 0) && (extPayload[n].pData != NULL)) { + int idx = qcOut->qcElement[el]->nExtensions++; + + qcOut->qcElement[el]->extension[idx].type = + extPayload[n].dataType; /* Perform a sanity check on the type? */ + qcOut->qcElement[el]->extension[idx].nPayloadBits = + extPayload[n].dataSize; + qcOut->qcElement[el]->extension[idx].pPayload = extPayload[n].pData; + /* Now ask the bitstream encoder how many bits we need to encode the + * data with the current bitstream syntax: */ + qcOut->qcElement[el]->extBitsUsed += FDKaacEnc_writeExtensionData( + NULL, &qcOut->qcElement[el]->extension[idx], 0, 0, + hAacEnc->config->syntaxFlags, hAacEnc->aot, + hAacEnc->config->epConfig); + extPayloadUsed[n] = 1; + } + } + + /* sum up extension and static bits for all channel elements */ + qcOut->elementExtBits += qcOut->qcElement[el]->extBitsUsed; + qcOut->staticBits += qcOut->qcElement[el]->staticBitsUsed; + + /* sum up pe */ + qcOut->totalNoRedPe += qcOut->qcElement[el]->peData.pe; + } + } + + qcOut->nExtensions = 0; + qcOut->globalExtBits = 0; + + /* reset extension payload */ + FDKmemclear(&qcOut->extension, (2 + 2) * sizeof(QC_OUT_EXTENSION)); + + /* Add extension payload not assigned to an channel element + (Ancillary data is the only supported type up to now) */ + for (n = 0; n < MAX_TOTAL_EXT_PAYLOADS; n++) { + if (!extPayloadUsed[n] && (extPayload[n].associatedChElement == -1) && + (extPayload[n].pData != NULL)) { + UINT payloadBits = 0; + + if (extPayload[n].dataType == EXT_DATA_ELEMENT) { + if (hAacEnc->ancillaryBitsPerFrame) { + /* granted frame dse bitrate */ + payloadBits = hAacEnc->ancillaryBitsPerFrame; + } else { + /* write anc data if bitrate constraint fulfilled */ + if ((extPayload[n].dataSize >> 3) <= + hAacEnc->config->maxAncBytesPerAU) { + payloadBits = extPayload[n].dataSize; + } + } + payloadBits = fixMin(extPayload[n].dataSize, payloadBits); + } else { + payloadBits = extPayload[n].dataSize; + } + + if (payloadBits > 0) { + int idx = qcOut->nExtensions++; + + qcOut->extension[idx].type = + extPayload[n].dataType; /* Perform a sanity check on the type? */ + qcOut->extension[idx].nPayloadBits = payloadBits; + qcOut->extension[idx].pPayload = extPayload[n].pData; + /* Now ask the bitstream encoder how many bits we need to encode the + * data with the current bitstream syntax: */ + qcOut->globalExtBits += FDKaacEnc_writeExtensionData( + NULL, &qcOut->extension[idx], 0, 0, hAacEnc->config->syntaxFlags, + hAacEnc->aot, hAacEnc->config->epConfig); + if (extPayload[n].dataType == EXT_DATA_ELEMENT) { + /* substract the processed bits */ + extPayload[n].dataSize -= payloadBits; + } + extPayloadUsed[n] = 1; + } + } + } + + if (!(hAacEnc->config->syntaxFlags & (AC_SCALABLE | AC_ER))) { + qcOut->globalExtBits += EL_ID_BITS; /* add bits for ID_END */ + } + + /* build bitstream all nSubFrames */ + { + INT totalBits = 0; /* Total AU bits */ + ; + INT avgTotalBits = 0; + + /*-------------------------------------------- */ + /* Get average total bits */ + /*-------------------------------------------- */ + { + /* frame wise bitrate adaption */ + FDKaacEnc_AdjustBitrate( + hAacEnc->qcKernel, cm, &avgTotalBits, hAacEnc->config->bitRate, + hAacEnc->config->sampleRate, hAacEnc->config->framelength); + + /* adjust super frame bitrate */ + avgTotalBits *= hAacEnc->config->nSubFrames; + } + + /* Make first estimate of transport header overhead. + Take maximum possible frame size into account to prevent bitreservoir + underrun. */ + hAacEnc->qcKernel->globHdrBits = transportEnc_GetStaticBits( + hTpEnc, avgTotalBits + hAacEnc->qcKernel->bitResTot); + + /*-------------------------------------------- */ + /*-------------------------------------------- */ + /*-------------------------------------------- */ + + ErrorStatus = FDKaacEnc_QCMain( + hAacEnc->qcKernel, hAacEnc->psyOut, hAacEnc->qcOut, avgTotalBits, cm, + hAacEnc->aot, hAacEnc->config->syntaxFlags, hAacEnc->config->epConfig); + + if (ErrorStatus != AAC_ENC_OK) return ErrorStatus; + /*-------------------------------------------- */ + + /*-------------------------------------------- */ + ErrorStatus = FDKaacEnc_updateFillBits( + cm, hAacEnc->qcKernel, hAacEnc->qcKernel->elementBits, hAacEnc->qcOut); + if (ErrorStatus != AAC_ENC_OK) return ErrorStatus; + + /*-------------------------------------------- */ + ErrorStatus = FDKaacEnc_FinalizeBitConsumption( + cm, hAacEnc->qcKernel, qcOut, qcOut->qcElement, hTpEnc, hAacEnc->aot, + hAacEnc->config->syntaxFlags, hAacEnc->config->epConfig); + if (ErrorStatus != AAC_ENC_OK) return ErrorStatus; + /*-------------------------------------------- */ + totalBits += qcOut->totalBits; + + /*-------------------------------------------- */ + FDKaacEnc_updateBitres(cm, hAacEnc->qcKernel, hAacEnc->qcOut); + + /*-------------------------------------------- */ + + /* for ( all sub frames ) ... */ + /* write bitstream header */ + if (TRANSPORTENC_OK != + transportEnc_WriteAccessUnit(hTpEnc, totalBits, + FDKaacEnc_EncBitresToTpBitres(hAacEnc), + cm->nChannelsEff)) { + return AAC_ENC_UNKNOWN; + } + + /* write bitstream */ + ErrorStatus = FDKaacEnc_WriteBitstream( + hTpEnc, cm, qcOut, psyOut, hAacEnc->qcKernel, hAacEnc->aot, + hAacEnc->config->syntaxFlags, hAacEnc->config->epConfig); + + if (ErrorStatus != AAC_ENC_OK) return ErrorStatus; + + /* transportEnc_EndAccessUnit() is being called inside + * FDKaacEnc_WriteBitstream() */ + if (TRANSPORTENC_OK != transportEnc_GetFrame(hTpEnc, nOutBytes)) { + return AAC_ENC_UNKNOWN; + } + + } /* -end- if (curFrame==hAacEnc->qcKernel->nSubFrames) */ + + /*-------------------------------------------- */ + return AAC_ENC_OK; +} + +/*--------------------------------------------------------------------------- + + functionname:FDKaacEnc_Close + description: delete encoder instance + returns: + + ---------------------------------------------------------------------------*/ + +void FDKaacEnc_Close(HANDLE_AAC_ENC *phAacEnc) /* encoder handle */ +{ + if (*phAacEnc == NULL) { + return; + } + AAC_ENC *hAacEnc = (AAC_ENC *)*phAacEnc; + + if (hAacEnc->dynamic_RAM != NULL) FreeAACdynamic_RAM(&hAacEnc->dynamic_RAM); + + FDKaacEnc_PsyClose(&hAacEnc->psyKernel, hAacEnc->psyOut); + + FDKaacEnc_QCClose(&hAacEnc->qcKernel, hAacEnc->qcOut); + + FreeRam_aacEnc_AacEncoder(phAacEnc); +} + +/* The following functions are in this source file only for convenience and */ +/* need not be visible outside of a possible encoder library. */ + +/* basic defines for ancillary data */ +#define MAX_ANCRATE 19200 /* ancillary rate >= 19200 isn't valid */ + +/*--------------------------------------------------------------------------- + + functionname: FDKaacEnc_InitCheckAncillary + description: initialize and check ancillary data struct + return: if success or NULL if error + + ---------------------------------------------------------------------------*/ +static AAC_ENCODER_ERROR FDKaacEnc_InitCheckAncillary( + INT bitRate, INT framelength, INT ancillaryRate, INT *ancillaryBitsPerFrame, + INT sampleRate) { + /* don't use negative ancillary rates */ + if (ancillaryRate < -1) return AAC_ENC_UNSUPPORTED_ANC_BITRATE; + + /* check if ancillary rate is ok */ + if ((ancillaryRate != (-1)) && (ancillaryRate != 0)) { + /* ancRate <= 15% of bitrate && ancRate < 19200 */ + if ((ancillaryRate >= MAX_ANCRATE) || + ((ancillaryRate * 20) > (bitRate * 3))) { + return AAC_ENC_UNSUPPORTED_ANC_BITRATE; + } + } else if (ancillaryRate == -1) { + /* if no special ancRate is requested but a ancillary file is + stated, then generate a ancillary rate matching to the bitrate */ + if (bitRate >= (MAX_ANCRATE * 10)) { + /* ancillary rate is 19199 */ + ancillaryRate = (MAX_ANCRATE - 1); + } else { /* 10% of bitrate */ + ancillaryRate = bitRate / 10; + } + } + + /* make ancillaryBitsPerFrame byte align */ + *ancillaryBitsPerFrame = + FDKaacEnc_CalcBitsPerFrame(ancillaryRate, framelength, sampleRate) & ~0x7; + + return AAC_ENC_OK; +} diff --git a/fdk-aac/libAACenc/src/aacenc.h b/fdk-aac/libAACenc/src/aacenc.h new file mode 100644 index 0000000..291ea54 --- /dev/null +++ b/fdk-aac/libAACenc/src/aacenc.h @@ -0,0 +1,394 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M. Schug / A. Groeschel + + Description: fast aac coder interface library functions + +*******************************************************************************/ + +#ifndef AACENC_H +#define AACENC_H + +#include "common_fix.h" +#include "FDK_audio.h" + +#include "tpenc_lib.h" + +#include "sbr_encoder.h" + +#define MIN_BUFSIZE_PER_EFF_CHAN 6144 + +#ifdef __cplusplus +extern "C" { +#endif + +/* + * AAC-LC error codes. + */ +typedef enum { + AAC_ENC_OK = 0x0000, /*!< All fine. */ + + AAC_ENC_UNKNOWN = 0x0002, /*!< Error condition is of unknown reason, or from + another module. */ + + /* initialization errors */ + aac_enc_init_error_start = 0x2000, + AAC_ENC_INVALID_HANDLE = 0x2020, /*!< The handle passed to the function call + was invalid (probably NULL). */ + AAC_ENC_INVALID_FRAME_LENGTH = + 0x2080, /*!< Invalid frame length (must be 1024 or 960). */ + AAC_ENC_INVALID_N_CHANNELS = + 0x20e0, /*!< Invalid amount of audio input channels. */ + AAC_ENC_INVALID_SFB_TABLE = 0x2140, /*!< Internal encoder error. */ + + AAC_ENC_UNSUPPORTED_AOT = + 0x3000, /*!< The Audio Object Type (AOT) is not supported. */ + AAC_ENC_UNSUPPORTED_FILTERBANK = + 0x3010, /*!< Filterbank type is not supported. */ + AAC_ENC_UNSUPPORTED_BITRATE = + 0x3020, /*!< The chosen bitrate is not supported. */ + AAC_ENC_UNSUPPORTED_BITRATE_MODE = + 0x3028, /*!< Unsupported bit rate mode (CBR or VBR). */ + AAC_ENC_UNSUPPORTED_ANC_BITRATE = + 0x3040, /*!< Unsupported ancillay bitrate. */ + AAC_ENC_UNSUPPORTED_ANC_MODE = 0x3060, + AAC_ENC_UNSUPPORTED_TRANSPORT_TYPE = + 0x3080, /*!< The bitstream format is not supported. */ + AAC_ENC_UNSUPPORTED_ER_FORMAT = + 0x30a0, /*!< The error resilience tool format is not supported. */ + AAC_ENC_UNSUPPORTED_EPCONFIG = + 0x30c0, /*!< The error protection format is not supported. */ + AAC_ENC_UNSUPPORTED_CHANNELCONFIG = + 0x30e0, /*!< The channel configuration (either number or arrangement) is + not supported. */ + AAC_ENC_UNSUPPORTED_SAMPLINGRATE = + 0x3100, /*!< Sample rate of audio input is not supported. */ + AAC_ENC_NO_MEMORY = 0x3120, /*!< Could not allocate memory. */ + AAC_ENC_PE_INIT_TABLE_NOT_FOUND = 0x3140, /*!< Internal encoder error. */ + + aac_enc_init_error_end, + + /* encode errors */ + aac_enc_error_start = 0x4000, + AAC_ENC_QUANT_ERROR = 0x4020, /*!< Too many bits used in quantization. */ + AAC_ENC_WRITTEN_BITS_ERROR = + 0x4040, /*!< Unexpected number of written bits, differs to + calculated number of bits. */ + AAC_ENC_PNS_TABLE_ERROR = 0x4060, /*!< PNS level out of range. */ + AAC_ENC_GLOBAL_GAIN_TOO_HIGH = 0x4080, /*!< Internal quantizer error. */ + AAC_ENC_BITRES_TOO_LOW = 0x40a0, /*!< Too few bits in bit reservoir. */ + AAC_ENC_BITRES_TOO_HIGH = 0x40a1, /*!< Too many bits in bit reservoir. */ + AAC_ENC_INVALID_CHANNEL_BITRATE = 0x4100, + AAC_ENC_INVALID_ELEMENTINFO_TYPE = 0x4120, /*!< Internal encoder error. */ + + AAC_ENC_WRITE_SCAL_ERROR = 0x41e0, /*!< Error writing scalefacData. */ + AAC_ENC_WRITE_SEC_ERROR = 0x4200, /*!< Error writing sectionData. */ + AAC_ENC_WRITE_SPEC_ERROR = 0x4220, /*!< Error writing spectralData. */ + aac_enc_error_end + +} AAC_ENCODER_ERROR; +/*-------------------------- defines --------------------------------------*/ + +#define ANC_DATA_BUFFERSIZE 1024 /* ancBuffer size */ + +#define MAX_TOTAL_EXT_PAYLOADS ((((8)) * (1)) + (2 + 2)) + +typedef enum { + AACENC_BR_MODE_INVALID = -1, /*!< Invalid bitrate mode. */ + AACENC_BR_MODE_CBR = 0, /*!< Constant bitrate mode. */ + AACENC_BR_MODE_VBR_1 = 1, /*!< Variable bitrate mode, very low bitrate. */ + AACENC_BR_MODE_VBR_2 = 2, /*!< Variable bitrate mode, low bitrate. */ + AACENC_BR_MODE_VBR_3 = 3, /*!< Variable bitrate mode, medium bitrate. */ + AACENC_BR_MODE_VBR_4 = 4, /*!< Variable bitrate mode, high bitrate. */ + AACENC_BR_MODE_VBR_5 = 5, /*!< Variable bitrate mode, very high bitrate. */ + AACENC_BR_MODE_FF = 6, /*!< Fixed frame mode. */ + AACENC_BR_MODE_SFR = 7 /*!< Superframe mode. */ + +} AACENC_BITRATE_MODE; + +#define AACENC_BR_MODE_IS_VBR(brMode) ((brMode >= 1) && (brMode <= 5)) + +typedef enum { + + CH_ORDER_MPEG = + 0, /*!< MPEG channel ordering (e. g. 5.1: C, L, R, SL, SR, LFE) */ + CH_ORDER_WAV, /*!< WAV fileformat channel ordering (e. g. 5.1: L, R, C, LFE, + SL, SR) */ + CH_ORDER_WG4 /*!< WG4 fileformat channel ordering (e. g. 5.1: L, R, SL, SR, C, LFE) */ + +} CHANNEL_ORDER; + +/*-------------------- structure definitions ------------------------------*/ + +struct AACENC_CONFIG { + INT sampleRate; /* encoder sample rate */ + INT bitRate; /* encoder bit rate in bits/sec */ + INT ancDataBitRate; /* additional bits consumed by anc data or sbr have to be + consiedered while configuration */ + + INT nSubFrames; /* number of frames in super frame (not ADTS/LATM subframes !) + */ + AUDIO_OBJECT_TYPE audioObjectType; /* Audio Object Type */ + + INT averageBits; /* encoder bit rate in bits/superframe */ + AACENC_BITRATE_MODE bitrateMode; /* encoder bitrate mode (CBR/VBR) */ + INT nChannels; /* number of channels to process */ + CHANNEL_ORDER channelOrder; /* input Channel ordering scheme. */ + INT bandWidth; /* targeted audio bandwidth in Hz */ + CHANNEL_MODE channelMode; /* encoder channel mode configuration */ + INT framelength; /* used frame size */ + + UINT syntaxFlags; /* bitstreams syntax configuration */ + SCHAR epConfig; /* error protection configuration */ + + INT anc_Rate; /* ancillary rate, 0 (disabled), -1 (default) else desired rate + */ + UINT maxAncBytesPerAU; + INT minBitsPerFrame; /* minimum number of bits in AU */ + INT maxBitsPerFrame; /* maximum number of bits in AU */ + + INT audioMuxVersion; /* audio mux version in loas/latm transport format */ + + UINT sbrRatio; /* sbr sampling rate ratio: dual- or single-rate */ + + UCHAR useTns; /* flag: use temporal noise shaping */ + UCHAR usePns; /* flag: use perceptual noise substitution */ + UCHAR useIS; /* flag: use intensity coding */ + UCHAR useMS; /* flag: use ms stereo tool */ + + UCHAR useRequant; /* flag: use afterburner */ + + UINT downscaleFactor; +}; + +typedef struct { + UCHAR *pData; /* pointer to extension payload data */ + UINT dataSize; /* extension payload data size in bits */ + EXT_PAYLOAD_TYPE dataType; /* extension payload data type */ + INT associatedChElement; /* number of the channel element the data is assigned + to */ +} AACENC_EXT_PAYLOAD; + +typedef struct AAC_ENC *HANDLE_AAC_ENC; + +/** + * \brief Calculate framesize in bits for given bit rate, frame length and + * sampling rate. + * + * \param bitRate Ttarget bitrate in bits per second. + * \param frameLength Number of audio samples in one frame. + * \param samplingRate Sampling rate in Hz. + * + * \return Framesize in bits per frame. + */ +INT FDKaacEnc_CalcBitsPerFrame(const INT bitRate, const INT frameLength, + const INT samplingRate); + +/** + * \brief Calculate bitrate in bits per second for given framesize, frame length + * and sampling rate. + * + * \param bitsPerFrame Framesize in bits per frame + * \param frameLength Number of audio samples in one frame. + * \param samplingRate Sampling rate in Hz. + * + * \return Bitrate in bits per second. + */ +INT FDKaacEnc_CalcBitrate(const INT bitsPerFrame, const INT frameLength, + const INT samplingRate); + +/** + * \brief Limit given bit rate to a valid value + * \param hTpEnc transport encoder handle + * \param aot audio object type + * \param coreSamplingRate the sample rate to be used for the AAC encoder + * \param frameLength the frameLength to be used for the AAC encoder + * \param nChannels number of total channels + * \param nChannelsEff number of effective channels + * \param bitRate the initial bit rate value for which the closest valid bit + * rate value is searched for + * \param averageBits average bits per frame for fixed framing. Set to -1 if not + * available. + * \param optional pointer where the current bits per frame are stored into. + * \param bitrateMode the current bit rate mode + * \param nSubFrames number of sub frames for super framing (not transport + * frames). + * \return a valid bit rate value as close as possible or identical to bitRate + */ +INT FDKaacEnc_LimitBitrate(HANDLE_TRANSPORTENC hTpEnc, AUDIO_OBJECT_TYPE aot, + INT coreSamplingRate, INT frameLength, INT nChannels, + INT nChannelsEff, INT bitRate, INT averageBits, + INT *pAverageBitsPerFrame, + AACENC_BITRATE_MODE bitrateMode, INT nSubFrames); + +/** + * \brief Get current state of the bit reservoir + * \param hAacEncoder encoder handle + * \return bit reservoir state in bits + */ +INT FDKaacEnc_GetBitReservoirState(const HANDLE_AAC_ENC hAacEncoder); + +/*----------------------------------------------------------------------------- + + functionname: FDKaacEnc_GetVBRBitrate + description: Get VBR bitrate from vbr quality + input params: int vbrQuality (VBR0, VBR1, VBR2) + channelMode + returns: vbr bitrate + +------------------------------------------------------------------------------*/ +INT FDKaacEnc_GetVBRBitrate(AACENC_BITRATE_MODE bitrateMode, + CHANNEL_MODE channelMode); + +/*----------------------------------------------------------------------------- + + functionname: FDKaacEnc_AacInitDefaultConfig + description: gives reasonable default configuration + returns: --- + + ------------------------------------------------------------------------------*/ +void FDKaacEnc_AacInitDefaultConfig(AACENC_CONFIG *config); + +/*--------------------------------------------------------------------------- + + functionname:FDKaacEnc_Open + description: allocate and initialize a new encoder instance + returns: 0 if success + + ---------------------------------------------------------------------------*/ +AAC_ENCODER_ERROR FDKaacEnc_Open( + HANDLE_AAC_ENC + *phAacEnc, /* pointer to an encoder handle, initialized on return */ + const INT nElements, /* number of maximal elements in instance to support */ + const INT nChannels, /* number of maximal channels in instance to support */ + const INT nSubFrames); /* support superframing in instance */ + +AAC_ENCODER_ERROR FDKaacEnc_Initialize( + HANDLE_AAC_ENC + hAacEncoder, /* pointer to an encoder handle, initialized on return */ + AACENC_CONFIG *config, /* pre-initialized config struct */ + HANDLE_TRANSPORTENC hTpEnc, ULONG initFlags); + +/*--------------------------------------------------------------------------- + + functionname: FDKaacEnc_EncodeFrame + description: encode one frame + returns: 0 if success + + ---------------------------------------------------------------------------*/ + +AAC_ENCODER_ERROR FDKaacEnc_EncodeFrame( + HANDLE_AAC_ENC hAacEnc, /* encoder handle */ + HANDLE_TRANSPORTENC hTpEnc, INT_PCM *inputBuffer, + const UINT inputBufferBufSize, INT *numOutBytes, + AACENC_EXT_PAYLOAD extPayload[MAX_TOTAL_EXT_PAYLOADS]); + +/*--------------------------------------------------------------------------- + + functionname:FDKaacEnc_Close + description: delete encoder instance + returns: + + ---------------------------------------------------------------------------*/ + +void FDKaacEnc_Close(HANDLE_AAC_ENC *phAacEnc); /* encoder handle */ + +#ifdef __cplusplus +} +#endif + +#endif /* AACENC_H */ diff --git a/fdk-aac/libAACenc/src/aacenc_lib.cpp b/fdk-aac/libAACenc/src/aacenc_lib.cpp new file mode 100644 index 0000000..aaa6c74 --- /dev/null +++ b/fdk-aac/libAACenc/src/aacenc_lib.cpp @@ -0,0 +1,2575 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M. Lohwasser + + Description: FDK HE-AAC Encoder interface library functions + +*******************************************************************************/ +#include <stdio.h> +#include "aacenc_lib.h" +#include "FDK_audio.h" +#include "aacenc.h" + +#include "aacEnc_ram.h" +#include "FDK_core.h" /* FDK_tools versioning info */ + +/* Encoder library info */ +#define AACENCODER_LIB_VL0 4 +#define AACENCODER_LIB_VL1 0 +#define AACENCODER_LIB_VL2 0 +#define AACENCODER_LIB_TITLE "AAC Encoder" +#ifdef __ANDROID__ +#define AACENCODER_LIB_BUILD_DATE "" +#define AACENCODER_LIB_BUILD_TIME "" +#else +#define AACENCODER_LIB_BUILD_DATE __DATE__ +#define AACENCODER_LIB_BUILD_TIME __TIME__ +#endif + +#include "pcm_utils.h" + +#include "sbr_encoder.h" +#include "../src/sbrenc_ram.h" +#include "channel_map.h" + +#include "psy_const.h" +#include "bitenc.h" + +#include "tpenc_lib.h" + +#include "metadata_main.h" +#include "mps_main.h" +#include "sacenc_lib.h" + +#define SBL(fl) \ + (fl / \ + 8) /*!< Short block length (hardcoded to 8 short blocks per long block) */ +#define BSLA(fl) \ + (4 * SBL(fl) + SBL(fl) / 2) /*!< AAC block switching look-ahead */ +#define DELAY_AAC(fl) (fl + BSLA(fl)) /*!< MDCT + blockswitching */ +#define DELAY_AACLD(fl) (fl) /*!< MDCT delay (no framing delay included) */ +#define DELAY_AACELD(fl) \ + ((fl) / 2) /*!< ELD FB delay (no framing delay included) */ + +#define MAX_DS_DELAY (100) /*!< Maximum downsampler delay in SBR. */ +#define INPUTBUFFER_SIZE \ + (2 * (1024) + MAX_DS_DELAY + 1537) /*!< Audio input samples + downsampler \ + delay + sbr/aac delay compensation */ + +#define DEFAULT_HEADER_PERIOD_REPETITION_RATE \ + 10 /*!< Default header repetition rate used in transport library and for SBR \ + header. */ + +//////////////////////////////////////////////////////////////////////////////////// +/** + * Flags to characterize encoder modules to be supported in present instance. + */ +enum { + ENC_MODE_FLAG_AAC = 0x0001, + ENC_MODE_FLAG_SBR = 0x0002, + ENC_MODE_FLAG_PS = 0x0004, + ENC_MODE_FLAG_SAC = 0x0008, + ENC_MODE_FLAG_META = 0x0010 +}; + +//////////////////////////////////////////////////////////////////////////////////// +typedef struct { + AUDIO_OBJECT_TYPE userAOT; /*!< Audio Object Type. */ + UINT userSamplerate; /*!< Sampling frequency. */ + UINT nChannels; /*!< will be set via channelMode. */ + CHANNEL_MODE userChannelMode; + UINT userBitrate; + UINT userBitrateMode; + UINT userBandwidth; + UINT userAfterburner; + UINT userFramelength; + UINT userAncDataRate; + UINT userPeakBitrate; + + UCHAR userTns; /*!< Use TNS coding. */ + UCHAR userPns; /*!< Use PNS coding. */ + UCHAR userIntensity; /*!< Use Intensity coding. */ + + TRANSPORT_TYPE userTpType; /*!< Transport type */ + UCHAR userTpSignaling; /*!< Extension AOT signaling mode. */ + UCHAR userTpNsubFrames; /*!< Number of sub frames in a transport frame for + LOAS/LATM or ADTS (default 1). */ + UCHAR userTpAmxv; /*!< AudioMuxVersion to be used for LATM (default 0). */ + UCHAR userTpProtection; + UCHAR userTpHeaderPeriod; /*!< Parameter used to configure LATM/LOAS SMC rate. + Moreover this parameters is used to configure + repetition rate of PCE in raw_data_block. */ + + UCHAR userErTools; /*!< Use VCB11, HCR and/or RVLC ER tool. */ + UINT userPceAdditions; /*!< Configure additional bits in PCE. */ + + UCHAR userMetaDataMode; /*!< Meta data library configuration. */ + + UCHAR userSbrEnabled; /*!< Enable SBR for ELD. */ + UINT userSbrRatio; /*!< SBR sampling rate ratio. Dual- or single-rate. */ + + UINT userDownscaleFactor; + +} USER_PARAM; + +/** + * SBR extenxion payload struct provides buffers to be filled in SBR encoder + * library. + */ +typedef struct { + UCHAR data[(1)][(8)][MAX_PAYLOAD_SIZE]; /*!< extension payload data buffer */ + UINT dataSize[(1)][(8)]; /*!< extension payload data size in bits */ +} SBRENC_EXT_PAYLOAD; + +//////////////////////////////////////////////////////////////////////////////////// + +/**************************************************************************** + Structure Definitions +****************************************************************************/ + +typedef struct AACENC_CONFIG *HANDLE_AACENC_CONFIG; + +struct AACENCODER { + USER_PARAM extParam; + CODER_CONFIG coderConfig; + + /* AAC */ + AACENC_CONFIG aacConfig; + HANDLE_AAC_ENC hAacEnc; + + /* SBR */ + HANDLE_SBR_ENCODER hEnvEnc; /* SBR encoder */ + SBRENC_EXT_PAYLOAD *pSbrPayload; /* SBR extension payload */ + + /* Meta Data */ + HANDLE_FDK_METADATA_ENCODER hMetadataEnc; + INT metaDataAllowed; /* Signal whether chosen configuration allows metadata. + Necessary for delay compensation. Metadata mode is a + separate parameter. */ + + HANDLE_MPS_ENCODER hMpsEnc; + + /* Transport */ + HANDLE_TRANSPORTENC hTpEnc; + + INT_PCM + *inputBuffer; /* Internal input buffer. Input source for AAC encoder */ + UCHAR *outBuffer; /* Internal bitstream buffer */ + + INT inputBufferSize; /* Size of internal input buffer */ + INT inputBufferSizePerChannel; /* Size of internal input buffer per channel */ + INT outBufferInBytes; /* Size of internal bitstream buffer*/ + + INT inputBufferOffset; /* Where to write new input samples. */ + + INT nSamplesToRead; /* number of input samples neeeded for encoding one frame + */ + INT nSamplesRead; /* number of input samples already in input buffer */ + INT nZerosAppended; /* appended zeros at end of file*/ + INT nDelay; /* codec delay */ + INT nDelayCore; /* codec delay, w/o the SBR decoder delay */ + + AACENC_EXT_PAYLOAD extPayload[MAX_TOTAL_EXT_PAYLOADS]; + + ULONG InitFlags; /* internal status to treggier re-initialization */ + + /* Memory allocation info. */ + INT nMaxAacElements; + INT nMaxAacChannels; + INT nMaxSbrElements; + INT nMaxSbrChannels; + + UINT encoder_modis; + + /* Capability flags */ + UINT CAPF_tpEnc; +}; + +typedef struct { + /* input */ + ULONG nChannels; /*!< Number of audio channels. */ + ULONG samplingRate; /*!< Encoder output sampling rate. */ + ULONG bitrateRange; /*!< Lower bitrate range for config entry. */ + + /* output*/ + UCHAR sbrMode; /*!< 0: ELD sbr off, + 1: ELD with downsampled sbr, + 2: ELD with dualrate sbr. */ + CHANNEL_MODE chMode; /*!< Channel mode. */ + +} ELD_SBR_CONFIGURATOR; + +/** + * \brief This table defines ELD/SBR default configurations. + */ +static const ELD_SBR_CONFIGURATOR eldSbrAutoConfigTab[] = { + {1, 48000, 0, 2, MODE_1}, {1, 48000, 64000, 0, MODE_1}, + + {1, 44100, 0, 2, MODE_1}, {1, 44100, 64000, 0, MODE_1}, + + {1, 32000, 0, 2, MODE_1}, {1, 32000, 28000, 1, MODE_1}, + {1, 32000, 56000, 0, MODE_1}, + + {1, 24000, 0, 1, MODE_1}, {1, 24000, 40000, 0, MODE_1}, + + {1, 16000, 0, 1, MODE_1}, {1, 16000, 28000, 0, MODE_1}, + + {1, 15999, 0, 0, MODE_1}, + + {2, 48000, 0, 2, MODE_2}, {2, 48000, 44000, 2, MODE_2}, + {2, 48000, 128000, 0, MODE_2}, + + {2, 44100, 0, 2, MODE_2}, {2, 44100, 44000, 2, MODE_2}, + {2, 44100, 128000, 0, MODE_2}, + + {2, 32000, 0, 2, MODE_2}, {2, 32000, 32000, 2, MODE_2}, + {2, 32000, 68000, 1, MODE_2}, {2, 32000, 96000, 0, MODE_2}, + + {2, 24000, 0, 1, MODE_2}, {2, 24000, 48000, 1, MODE_2}, + {2, 24000, 80000, 0, MODE_2}, + + {2, 16000, 0, 1, MODE_2}, {2, 16000, 32000, 1, MODE_2}, + {2, 16000, 64000, 0, MODE_2}, + + {2, 15999, 0, 0, MODE_2} + +}; + +/* + * \brief Configure SBR for ELD configuration. + * + * This function finds default SBR configuration for ELD based on number of + * channels, sampling rate and bitrate. + * + * \param nChannels Number of audio channels. + * \param samplingRate Audio signal sampling rate. + * \param bitrate Encoder bitrate. + * + * \return - pointer to eld sbr configuration. + * - NULL, on failure. + */ +static const ELD_SBR_CONFIGURATOR *eldSbrConfigurator(const ULONG nChannels, + const ULONG samplingRate, + const ULONG bitrate) { + int i; + const ELD_SBR_CONFIGURATOR *pSetup = NULL; + + for (i = 0; + i < (int)(sizeof(eldSbrAutoConfigTab) / sizeof(ELD_SBR_CONFIGURATOR)); + i++) { + if ((nChannels == eldSbrAutoConfigTab[i].nChannels) && + (samplingRate <= eldSbrAutoConfigTab[i].samplingRate) && + (bitrate >= eldSbrAutoConfigTab[i].bitrateRange)) { + pSetup = &eldSbrAutoConfigTab[i]; + } + } + + return pSetup; +} + +static inline INT isSbrActive(const HANDLE_AACENC_CONFIG hAacConfig) { + INT sbrUsed = 0; + + /* Note: Even if implicit signalling was selected, The AOT itself here is not + * AOT_AAC_LC */ + if ((hAacConfig->audioObjectType == AOT_SBR) || + (hAacConfig->audioObjectType == AOT_PS) || + (hAacConfig->audioObjectType == AOT_MP2_SBR) || + (hAacConfig->audioObjectType == AOT_DABPLUS_SBR) || + (hAacConfig->audioObjectType == AOT_DABPLUS_PS)) { + sbrUsed = 1; + } + if (hAacConfig->audioObjectType == AOT_ER_AAC_ELD && + (hAacConfig->syntaxFlags & AC_SBR_PRESENT)) { + sbrUsed = 1; + } + + return (sbrUsed); +} + +static inline INT isPsActive(const AUDIO_OBJECT_TYPE audioObjectType) { + INT psUsed = 0; + + if ((audioObjectType == AOT_PS) || + (audioObjectType == AOT_DABPLUS_PS)) { + psUsed = 1; + } + + return (psUsed); +} + +static CHANNEL_MODE GetCoreChannelMode( + const CHANNEL_MODE channelMode, const AUDIO_OBJECT_TYPE audioObjectType) { + CHANNEL_MODE mappedChannelMode = channelMode; + if ((isPsActive(audioObjectType) && (channelMode == MODE_2)) || + (channelMode == MODE_212)) { + mappedChannelMode = MODE_1; + } + return mappedChannelMode; +} + +static SBR_PS_SIGNALING getSbrSignalingMode( + const AUDIO_OBJECT_TYPE audioObjectType, const TRANSPORT_TYPE transportType, + const UCHAR transportSignaling, const UINT sbrRatio) + +{ + SBR_PS_SIGNALING sbrSignaling; + + if (transportType == TT_UNKNOWN || sbrRatio == 0) { + sbrSignaling = SIG_UNKNOWN; /* Needed parameters have not been set */ + return sbrSignaling; + } else { + sbrSignaling = + SIG_EXPLICIT_HIERARCHICAL; /* default: explicit hierarchical signaling + */ + } + + if ((audioObjectType == AOT_AAC_LC) || (audioObjectType == AOT_SBR) || + (audioObjectType == AOT_PS) || (audioObjectType == AOT_MP2_AAC_LC) || + (audioObjectType == AOT_MP2_SBR) || + (audioObjectType == AOT_DABPLUS_SBR) || + (audioObjectType == AOT_DABPLUS_PS)) { + switch (transportType) { + case TT_MP4_ADIF: + case TT_MP4_ADTS: + sbrSignaling = SIG_IMPLICIT; /* For MPEG-2 transport types, only + implicit signaling is possible */ + break; + + case TT_MP4_RAW: + case TT_MP4_LATM_MCP1: + case TT_MP4_LATM_MCP0: + case TT_MP4_LOAS: + default: + if (transportSignaling == 0xFF) { + /* Defaults */ + sbrSignaling = SIG_EXPLICIT_HIERARCHICAL; + } else { + /* User set parameters */ + /* Attention: Backward compatible explicit signaling does only work + * with AMV1 for LATM/LOAS */ + sbrSignaling = (SBR_PS_SIGNALING)transportSignaling; + } + break; + } + } + + return sbrSignaling; +} + +/**************************************************************************** + Allocate Encoder +****************************************************************************/ + +H_ALLOC_MEM(_AacEncoder, AACENCODER) +C_ALLOC_MEM(_AacEncoder, struct AACENCODER, 1) + +/* + * Map Encoder specific config structures to CODER_CONFIG. + */ +static void FDKaacEnc_MapConfig(CODER_CONFIG *const cc, + const USER_PARAM *const extCfg, + const SBR_PS_SIGNALING sbrSignaling, + const HANDLE_AACENC_CONFIG hAacConfig) { + AUDIO_OBJECT_TYPE transport_AOT = AOT_NULL_OBJECT; + FDKmemclear(cc, sizeof(CODER_CONFIG)); + + cc->flags = 0; + + cc->samplesPerFrame = hAacConfig->framelength; + cc->samplingRate = hAacConfig->sampleRate; + cc->extSamplingRate = extCfg->userSamplerate; + + /* Map virtual aot to transport aot. */ + switch (hAacConfig->audioObjectType) { + case AOT_MP2_AAC_LC: + case AOT_DABPLUS_AAC_LC: + transport_AOT = AOT_AAC_LC; + break; + case AOT_MP2_SBR: + case AOT_DABPLUS_SBR: + transport_AOT = AOT_SBR; + cc->flags |= CC_SBR; + break; + case AOT_DABPLUS_PS: + transport_AOT = AOT_PS; + cc->flags |= CC_SBR; + break; + default: + transport_AOT = hAacConfig->audioObjectType; + } + + if (hAacConfig->audioObjectType == AOT_ER_AAC_ELD) { + cc->flags |= (hAacConfig->syntaxFlags & AC_SBR_PRESENT) ? CC_SBR : 0; + cc->flags |= (hAacConfig->syntaxFlags & AC_LD_MPS) ? CC_SAC : 0; + } + + /* transport type is usually AAC-LC. */ + if ((transport_AOT == AOT_SBR) || (transport_AOT == AOT_PS)) { + cc->aot = AOT_AAC_LC; + } else { + cc->aot = transport_AOT; + } + + /* Configure extension aot. */ + if (sbrSignaling == SIG_IMPLICIT) { + cc->extAOT = AOT_NULL_OBJECT; /* implicit */ + } else { + if ((sbrSignaling == SIG_EXPLICIT_BW_COMPATIBLE) && + ((transport_AOT == AOT_SBR) || (transport_AOT == AOT_PS))) { + cc->extAOT = AOT_SBR; /* explicit backward compatible */ + } else { + cc->extAOT = transport_AOT; /* explicit hierarchical */ + } + } + + if ((transport_AOT == AOT_SBR) || (transport_AOT == AOT_PS)) { + cc->sbrPresent = 1; + if (transport_AOT == AOT_PS) { + cc->psPresent = 1; + } + } + cc->sbrSignaling = sbrSignaling; + + if (hAacConfig->downscaleFactor > 1) { + cc->downscaleSamplingRate = cc->samplingRate; + cc->samplingRate *= hAacConfig->downscaleFactor; + cc->extSamplingRate *= hAacConfig->downscaleFactor; + } + + cc->bitRate = hAacConfig->bitRate; + cc->noChannels = hAacConfig->nChannels; + cc->flags |= CC_IS_BASELAYER; + cc->channelMode = hAacConfig->channelMode; + +if (extCfg->userTpType == TT_DABPLUS && hAacConfig->nSubFrames==1) { + switch(hAacConfig->sampleRate) { + case 48000: + cc->nSubFrames=6; + break; + case 32000: + cc->nSubFrames=4; + break; + case 24000: + cc->nSubFrames=3; + break; + case 16000: + cc->nSubFrames=2; + break; + } + //fprintf(stderr, "hAacConfig->nSubFrames=%d hAacConfig->sampleRate=%d\n", hAacConfig->nSubFrames, hAacConfig->sampleRate); + } else { + cc->nSubFrames = (hAacConfig->nSubFrames > 1 && extCfg->userTpNsubFrames == 1) + ? hAacConfig->nSubFrames + : extCfg->userTpNsubFrames; + } + + cc->flags |= (extCfg->userTpProtection) ? CC_PROTECTION : 0; + + if (extCfg->userTpHeaderPeriod != 0xFF) { + cc->headerPeriod = extCfg->userTpHeaderPeriod; + } else { /* auto-mode */ + switch (extCfg->userTpType) { + case TT_MP4_ADTS: + case TT_MP4_LOAS: + case TT_MP4_LATM_MCP1: + cc->headerPeriod = DEFAULT_HEADER_PERIOD_REPETITION_RATE; + break; + default: + cc->headerPeriod = 0; + } + } + + /* Mpeg-4 signaling for transport library. */ + switch (hAacConfig->audioObjectType) { + case AOT_MP2_AAC_LC: + case AOT_MP2_SBR: + cc->flags &= ~CC_MPEG_ID; /* Required for ADTS. */ + cc->extAOT = AOT_NULL_OBJECT; + break; + default: + cc->flags |= CC_MPEG_ID; + } + + /* ER-tools signaling. */ + cc->flags |= (hAacConfig->syntaxFlags & AC_ER_VCB11) ? CC_VCB11 : 0; + cc->flags |= (hAacConfig->syntaxFlags & AC_ER_HCR) ? CC_HCR : 0; + cc->flags |= (hAacConfig->syntaxFlags & AC_ER_RVLC) ? CC_RVLC : 0; + + /* Matrix mixdown coefficient configuration. */ + if ((extCfg->userPceAdditions & 0x1) && (hAacConfig->epConfig == -1) && + ((cc->channelMode == MODE_1_2_2) || (cc->channelMode == MODE_1_2_2_1))) { + cc->matrixMixdownA = ((extCfg->userPceAdditions >> 1) & 0x3) + 1; + cc->flags |= (extCfg->userPceAdditions >> 3) & 0x1 ? CC_PSEUDO_SURROUND : 0; + } else { + cc->matrixMixdownA = 0; + } + + cc->channelConfigZero = 0; +} + +/* + * Validate prefilled pointers within buffer descriptor. + * + * \param pBufDesc Pointer to buffer descriptor + + * \return - AACENC_OK, all fine. + * - AACENC_INVALID_HANDLE, on missing pointer initializiation. + * - AACENC_UNSUPPORTED_PARAMETER, on incorrect buffer descriptor + initialization. + */ +static AACENC_ERROR validateBufDesc(const AACENC_BufDesc *pBufDesc) { + AACENC_ERROR err = AACENC_OK; + + if (pBufDesc != NULL) { + int i; + if ((pBufDesc->bufferIdentifiers == NULL) || (pBufDesc->bufSizes == NULL) || + (pBufDesc->bufElSizes == NULL) || (pBufDesc->bufs == NULL)) { + err = AACENC_UNSUPPORTED_PARAMETER; + goto bail; + } + for (i = 0; i < pBufDesc->numBufs; i++) { + if (pBufDesc->bufs[i] == NULL) { + err = AACENC_UNSUPPORTED_PARAMETER; + goto bail; + } + } + } else { + err = AACENC_INVALID_HANDLE; + } +bail: + return err; +} + +/* + * Examine buffer descriptor regarding choosen identifier. + * + * \param pBufDesc Pointer to buffer descriptor + * \param identifier Buffer identifier to look for. + + * \return - Buffer descriptor index. + * -1, if there is no entry available. + */ +static INT getBufDescIdx(const AACENC_BufDesc *pBufDesc, + const AACENC_BufferIdentifier identifier) { + INT i, idx = -1; + + if (pBufDesc != NULL) { + for (i = 0; i < pBufDesc->numBufs; i++) { + if ((AACENC_BufferIdentifier)pBufDesc->bufferIdentifiers[i] == + identifier) { + idx = i; + break; + } + } + } + return idx; +} + +/**************************************************************************** + Function Declarations +****************************************************************************/ + +AAC_ENCODER_ERROR aacEncDefaultConfig(HANDLE_AACENC_CONFIG hAacConfig, + USER_PARAM *config) { + /* make reasonable default settings */ + FDKaacEnc_AacInitDefaultConfig(hAacConfig); + + /* clear configuration structure and copy default settings */ + FDKmemclear(config, sizeof(USER_PARAM)); + + /* copy encoder configuration settings */ + config->nChannels = hAacConfig->nChannels; + config->userAOT = hAacConfig->audioObjectType = AOT_AAC_LC; + config->userSamplerate = hAacConfig->sampleRate; + config->userChannelMode = hAacConfig->channelMode; + config->userBitrate = hAacConfig->bitRate; + config->userBitrateMode = hAacConfig->bitrateMode; + config->userPeakBitrate = (UINT)-1; + config->userBandwidth = hAacConfig->bandWidth; + config->userTns = hAacConfig->useTns; + config->userPns = hAacConfig->usePns; + config->userIntensity = hAacConfig->useIS; + config->userAfterburner = hAacConfig->useRequant; + config->userFramelength = (UINT)-1; + + config->userDownscaleFactor = 1; + + /* initialize transport parameters */ + config->userTpType = TT_UNKNOWN; + config->userTpAmxv = 0; + config->userTpSignaling = 0xFF; /* choose signaling automatically */ + config->userTpNsubFrames = 1; + config->userTpProtection = 0; /* not crc protected*/ + config->userTpHeaderPeriod = 0xFF; /* header period in auto mode */ + config->userPceAdditions = 0; /* no matrix mixdown coefficient */ + config->userMetaDataMode = 0; /* do not embed any meta data info */ + + config->userAncDataRate = 0; + + /* SBR rate is set to 0 here, which means it should be set automatically + in FDKaacEnc_AdjustEncSettings() if the user did not set a rate + expilicitely. */ + config->userSbrRatio = 0; + + /* SBR enable set to -1 means to inquire ELD audio configurator for reasonable + * configuration. */ + config->userSbrEnabled = (UCHAR)-1; + + return AAC_ENC_OK; +} + +static void aacEncDistributeSbrBits(CHANNEL_MAPPING *channelMapping, + SBR_ELEMENT_INFO *sbrElInfo, INT bitRate) { + INT codebits = bitRate; + int el; + + /* Copy Element info */ + for (el = 0; el < channelMapping->nElements; el++) { + sbrElInfo[el].ChannelIndex[0] = channelMapping->elInfo[el].ChannelIndex[0]; + sbrElInfo[el].ChannelIndex[1] = channelMapping->elInfo[el].ChannelIndex[1]; + sbrElInfo[el].elType = channelMapping->elInfo[el].elType; + sbrElInfo[el].bitRate = + fMultIfloor(channelMapping->elInfo[el].relativeBits, bitRate); + sbrElInfo[el].instanceTag = channelMapping->elInfo[el].instanceTag; + sbrElInfo[el].nChannelsInEl = channelMapping->elInfo[el].nChannelsInEl; + sbrElInfo[el].fParametricStereo = 0; + sbrElInfo[el].fDualMono = 0; + + codebits -= sbrElInfo[el].bitRate; + } + sbrElInfo[0].bitRate += codebits; +} + +static INT aacEncoder_LimitBitrate(const HANDLE_TRANSPORTENC hTpEnc, + const INT samplingRate, + const INT frameLength, const INT nChannels, + const CHANNEL_MODE channelMode, INT bitRate, + const INT nSubFrames, const INT sbrActive, + const INT sbrDownSampleRate, + const UINT syntaxFlags, + const AUDIO_OBJECT_TYPE aot) { + INT coreSamplingRate; + CHANNEL_MAPPING cm; + + FDKaacEnc_InitChannelMapping(channelMode, CH_ORDER_MPEG, &cm); + + if (sbrActive) { + coreSamplingRate = + samplingRate >> + (sbrEncoder_IsSingleRatePossible(aot) ? (sbrDownSampleRate - 1) : 1); + } else { + coreSamplingRate = samplingRate; + } + + /* Limit bit rate in respect to the core coder */ + bitRate = FDKaacEnc_LimitBitrate(hTpEnc, aot, coreSamplingRate, frameLength, + nChannels, cm.nChannelsEff, bitRate, -1, + NULL, AACENC_BR_MODE_INVALID, nSubFrames); + + /* Limit bit rate in respect to available SBR modes if active */ + if (sbrActive) { + int numIterations = 0; + INT initialBitrate, adjustedBitrate; + adjustedBitrate = bitRate; + + /* Find total bitrate which provides valid configuration for each SBR + * element. */ + do { + int e; + SBR_ELEMENT_INFO sbrElInfo[((8))]; + FDK_ASSERT(cm.nElements <= ((8))); + + initialBitrate = adjustedBitrate; + + /* Get bit rate for each SBR element */ + aacEncDistributeSbrBits(&cm, sbrElInfo, initialBitrate); + + for (e = 0; e < cm.nElements; e++) { + INT sbrElementBitRateIn, sbrBitRateOut; + + if (cm.elInfo[e].elType != ID_SCE && cm.elInfo[e].elType != ID_CPE) { + continue; + } + sbrElementBitRateIn = sbrElInfo[e].bitRate; + + sbrBitRateOut = sbrEncoder_LimitBitRate(sbrElementBitRateIn, + cm.elInfo[e].nChannelsInEl, + coreSamplingRate, aot); + + if (sbrBitRateOut == 0) { + return 0; + } + + /* If bitrates don't match, distribution and limiting needs to be + determined again. Abort element loop and restart with adapted + bitrate. */ + if (sbrElementBitRateIn != sbrBitRateOut) { + if (sbrElementBitRateIn < sbrBitRateOut) { + adjustedBitrate = fMax(initialBitrate, + (INT)fDivNorm((FIXP_DBL)(sbrBitRateOut + 8), + cm.elInfo[e].relativeBits)); + break; + } + + if (sbrElementBitRateIn > sbrBitRateOut) { + adjustedBitrate = fMin(initialBitrate, + (INT)fDivNorm((FIXP_DBL)(sbrBitRateOut - 8), + cm.elInfo[e].relativeBits)); + break; + } + + } /* sbrElementBitRateIn != sbrBitRateOut */ + + } /* elements */ + + numIterations++; /* restrict iteration to worst case of num elements */ + + } while ((initialBitrate != adjustedBitrate) && + (numIterations <= cm.nElements)); + + /* Unequal bitrates mean that no reasonable bitrate configuration found. */ + bitRate = (initialBitrate == adjustedBitrate) ? adjustedBitrate : 0; + } + + /* Limit bit rate in respect to available MPS modes if active */ + if ((aot == AOT_ER_AAC_ELD) && (syntaxFlags & AC_LD_MPS) && + (channelMode == MODE_1)) { + bitRate = FDK_MpegsEnc_GetClosestBitRate( + aot, MODE_212, samplingRate, (sbrActive) ? sbrDownSampleRate : 0, + bitRate); + } + + //fprintf(stderr, "aacEncoder_LimitBitrate(): bitRate=%d\n", bitRate); + return bitRate; +} + +/* + * \brief Get CBR bitrate + * + * \hAacConfig Internal encoder config + * \return Bitrate + */ +static INT FDKaacEnc_GetCBRBitrate(const HANDLE_AACENC_CONFIG hAacConfig, + const INT userSbrRatio) { + INT bitrate = FDKaacEnc_GetChannelModeConfiguration(hAacConfig->channelMode) + ->nChannelsEff * + hAacConfig->sampleRate; + + if (isPsActive(hAacConfig->audioObjectType)) { + bitrate = 1 * bitrate; /* 0.5 bit per sample */ + } else if (isSbrActive(hAacConfig)) { + if ((userSbrRatio == 2) || + ((userSbrRatio == 0) && + (hAacConfig->audioObjectType != AOT_ER_AAC_ELD))) { + bitrate = (bitrate + (bitrate >> 2)) >> 1; /* 0.625 bits per sample */ + } + if ((userSbrRatio == 1) || + ((userSbrRatio == 0) && + (hAacConfig->audioObjectType == AOT_ER_AAC_ELD))) { + bitrate = (bitrate + (bitrate >> 3)); /* 1.125 bits per sample */ + } + } else { + bitrate = bitrate + (bitrate >> 1); /* 1.5 bits per sample */ + } + + return bitrate; +} + +/* + * \brief Consistency check of given USER_PARAM struct and + * copy back configuration from public struct into internal + * encoder configuration struct. + * + * \hAacEncoder Internal encoder config which is to be updated + * \param config User provided config (public struct) + * \return returns always AAC_ENC_OK + */ +static AACENC_ERROR FDKaacEnc_AdjustEncSettings(HANDLE_AACENCODER hAacEncoder, + USER_PARAM *config) { + AACENC_ERROR err = AACENC_OK; + + /* Get struct pointers. */ + HANDLE_AACENC_CONFIG hAacConfig = &hAacEncoder->aacConfig; + + /* Encoder settings update. */ + hAacConfig->sampleRate = config->userSamplerate; + if (config->userDownscaleFactor > 1) { + hAacConfig->useTns = 0; + hAacConfig->usePns = 0; + hAacConfig->useIS = 0; + } else { + hAacConfig->useTns = config->userTns; + hAacConfig->usePns = config->userPns; + hAacConfig->useIS = config->userIntensity; + } + + hAacConfig->audioObjectType = config->userAOT; + hAacConfig->channelMode = + GetCoreChannelMode(config->userChannelMode, hAacConfig->audioObjectType); + hAacConfig->nChannels = + FDKaacEnc_GetChannelModeConfiguration(hAacConfig->channelMode)->nChannels; + hAacConfig->bitrateMode = (AACENC_BITRATE_MODE)config->userBitrateMode; + hAacConfig->bandWidth = config->userBandwidth; + hAacConfig->useRequant = config->userAfterburner; + + hAacConfig->anc_Rate = config->userAncDataRate; + hAacConfig->syntaxFlags = 0; + hAacConfig->epConfig = -1; + + if (hAacConfig->audioObjectType != AOT_ER_AAC_ELD && + config->userDownscaleFactor > 1) { + return AACENC_INVALID_CONFIG; /* downscaling only allowed for AOT_ER_AAC_ELD + */ + } + if (config->userDownscaleFactor > 1 && config->userSbrEnabled == 1) { + return AACENC_INVALID_CONFIG; /* downscaling only allowed for AOT_ER_AAC_ELD + w/o SBR */ + } + if (config->userDownscaleFactor > 1 && config->userChannelMode == 128) { + return AACENC_INVALID_CONFIG; /* disallow downscaling for AAC-ELDv2 */ + } + + if (config->userTpType == TT_MP4_LATM_MCP1 || + config->userTpType == TT_MP4_LATM_MCP0 || + config->userTpType == TT_MP4_LOAS) { + hAacConfig->audioMuxVersion = config->userTpAmxv; + } else { + hAacConfig->audioMuxVersion = -1; + } + + /* Adapt internal AOT when necessary. */ + switch (config->userAOT) { + case AOT_MP2_AAC_LC: + case AOT_MP2_SBR: + hAacConfig->usePns = 0; + FDK_FALLTHROUGH; + case AOT_AAC_LC: + case AOT_SBR: + case AOT_PS: + config->userTpType = + (config->userTpType != TT_UNKNOWN) ? config->userTpType : TT_MP4_ADTS; + hAacConfig->framelength = (config->userFramelength != (UINT)-1) + ? config->userFramelength + : 1024; + if (hAacConfig->framelength != 1024 && hAacConfig->framelength != 960) { + return AACENC_INVALID_CONFIG; + } + break; + + case AOT_DABPLUS_SBR: + case AOT_DABPLUS_PS: + hAacConfig->syntaxFlags |= ((config->userSbrEnabled) ? AC_SBR_PRESENT : 0); + case AOT_DABPLUS_AAC_LC: + config->userTpType = (config->userTpType!=TT_UNKNOWN) ? config->userTpType : TT_DABPLUS; + hAacConfig->framelength = (config->userFramelength!=(UINT)-1) ? config->userFramelength : 960; + if (hAacConfig->framelength != 960) { + return AACENC_INVALID_CONFIG; + } + config->userTpSignaling=2; + if(config->userTpType == TT_DABPLUS) + hAacConfig->syntaxFlags |= AC_DAB; + break; + + case AOT_ER_AAC_LD: + hAacConfig->epConfig = 0; + hAacConfig->syntaxFlags |= AC_ER | AC_LD; + hAacConfig->syntaxFlags |= + ((config->userErTools & 0x1) ? AC_ER_VCB11 : 0); + hAacConfig->syntaxFlags |= ((config->userErTools & 0x2) ? AC_ER_HCR : 0); + hAacConfig->syntaxFlags |= ((config->userErTools & 0x4) ? AC_ER_RVLC : 0); + config->userTpType = + (config->userTpType != TT_UNKNOWN) ? config->userTpType : TT_MP4_LOAS; + hAacConfig->framelength = + (config->userFramelength != (UINT)-1) ? config->userFramelength : 512; + if (hAacConfig->framelength != 512 && hAacConfig->framelength != 480) { + return AACENC_INVALID_CONFIG; + } + break; + case AOT_ER_AAC_ELD: + hAacConfig->epConfig = 0; + hAacConfig->syntaxFlags |= AC_ER | AC_ELD; + hAacConfig->syntaxFlags |= + ((config->userErTools & 0x1) ? AC_ER_VCB11 : 0); + hAacConfig->syntaxFlags |= ((config->userErTools & 0x2) ? AC_ER_HCR : 0); + hAacConfig->syntaxFlags |= ((config->userErTools & 0x4) ? AC_ER_RVLC : 0); + hAacConfig->syntaxFlags |= + ((config->userSbrEnabled == 1) ? AC_SBR_PRESENT : 0); + hAacConfig->syntaxFlags |= + ((config->userChannelMode == MODE_212) ? AC_LD_MPS : 0); + config->userTpType = + (config->userTpType != TT_UNKNOWN) ? config->userTpType : TT_MP4_LOAS; + hAacConfig->framelength = + (config->userFramelength != (UINT)-1) ? config->userFramelength : 512; + + hAacConfig->downscaleFactor = config->userDownscaleFactor; + + switch (config->userDownscaleFactor) { + case 1: + break; + case 2: + case 4: + hAacConfig->syntaxFlags |= AC_ELD_DOWNSCALE; + break; + default: + return AACENC_INVALID_CONFIG; + } + + if (hAacConfig->framelength != 512 && hAacConfig->framelength != 480 && + hAacConfig->framelength != 256 && hAacConfig->framelength != 240 && + hAacConfig->framelength != 128 && hAacConfig->framelength != 120) { + return AACENC_INVALID_CONFIG; + } + break; + default: + break; + } + + /* Initialize SBR parameters */ + if ((config->userSbrRatio == 0) && (isSbrActive(hAacConfig))) { + /* Automatic SBR ratio configuration + * - downsampled SBR for ELD + * - otherwise always dualrate SBR + */ + if (hAacConfig->audioObjectType == AOT_ER_AAC_ELD) { + hAacConfig->sbrRatio = ((hAacConfig->syntaxFlags & AC_LD_MPS) && + (hAacConfig->sampleRate >= 27713)) + ? 2 + : 1; + } else { + hAacConfig->sbrRatio = 2; + } + } else { + /* SBR ratio has been set by the user, so use it. */ + hAacConfig->sbrRatio = isSbrActive(hAacConfig) ? config->userSbrRatio : 0; + } + + /* Set default bitrate */ + hAacConfig->bitRate = config->userBitrate; + + switch (hAacConfig->bitrateMode) { + case AACENC_BR_MODE_CBR: + /* Set default bitrate if no external bitrate declared. */ + if (config->userBitrate == (UINT)-1) { + hAacConfig->bitRate = + FDKaacEnc_GetCBRBitrate(hAacConfig, config->userSbrRatio); + } + hAacConfig->averageBits = -1; + break; + case AACENC_BR_MODE_VBR_1: + case AACENC_BR_MODE_VBR_2: + case AACENC_BR_MODE_VBR_3: + case AACENC_BR_MODE_VBR_4: + case AACENC_BR_MODE_VBR_5: + /* Get bitrate in VBR configuration */ + /* In VBR mode; SBR-modul depends on bitrate, core encoder on bitrateMode. + */ + hAacConfig->bitRate = FDKaacEnc_GetVBRBitrate(hAacConfig->bitrateMode, + hAacConfig->channelMode); + break; + default: + return AACENC_INVALID_CONFIG; + } + + /* set bitreservoir size */ + switch (hAacConfig->bitrateMode) { + case AACENC_BR_MODE_VBR_1: + case AACENC_BR_MODE_VBR_2: + case AACENC_BR_MODE_VBR_3: + case AACENC_BR_MODE_VBR_4: + case AACENC_BR_MODE_VBR_5: + case AACENC_BR_MODE_CBR: + if ((INT)config->userPeakBitrate != -1) { + hAacConfig->maxBitsPerFrame = + (FDKaacEnc_CalcBitsPerFrame( + fMax(hAacConfig->bitRate, (INT)config->userPeakBitrate), + hAacConfig->framelength, hAacConfig->sampleRate) + + 7) & + ~7; + } else { + hAacConfig->maxBitsPerFrame = -1; + } + if (hAacConfig->audioMuxVersion == 2) { + hAacConfig->minBitsPerFrame = + fMin(32 * 8, FDKaacEnc_CalcBitsPerFrame(hAacConfig->bitRate, + hAacConfig->framelength, + hAacConfig->sampleRate)) & + ~7; + } + break; + default: + return AACENC_INVALID_CONFIG; + } + + /* Max bits per frame limitation depending on transport format. */ + if ((config->userTpNsubFrames > 1)) { + int maxFrameLength = 8 * hAacEncoder->outBufferInBytes; + switch (config->userTpType) { + case TT_MP4_LOAS: + maxFrameLength = + fMin(maxFrameLength, 8 * (1 << 13)) / config->userTpNsubFrames; + break; + case TT_MP4_ADTS: + maxFrameLength = fMin(maxFrameLength, 8 * ((1 << 13) - 1)) / + config->userTpNsubFrames; + break; + default: + maxFrameLength = -1; + } + if (maxFrameLength != -1) { + if (hAacConfig->maxBitsPerFrame > maxFrameLength) { + return AACENC_INVALID_CONFIG; + } else if (hAacConfig->maxBitsPerFrame == -1) { + hAacConfig->maxBitsPerFrame = maxFrameLength; + } + } + } + + if ((hAacConfig->audioObjectType == AOT_ER_AAC_ELD) && + !(hAacConfig->syntaxFlags & AC_ELD_DOWNSCALE) && + (config->userSbrEnabled == (UCHAR)-1) && (config->userSbrRatio == 0) && + ((hAacConfig->syntaxFlags & AC_LD_MPS) == 0)) { + const ELD_SBR_CONFIGURATOR *pConfig = NULL; + + if (NULL != + (pConfig = eldSbrConfigurator( + FDKaacEnc_GetChannelModeConfiguration(hAacConfig->channelMode) + ->nChannels, + hAacConfig->sampleRate, hAacConfig->bitRate))) { + hAacConfig->syntaxFlags |= (pConfig->sbrMode == 0) ? 0 : AC_SBR_PRESENT; + hAacConfig->syntaxFlags |= (pConfig->chMode == MODE_212) ? AC_LD_MPS : 0; + hAacConfig->channelMode = + GetCoreChannelMode(pConfig->chMode, hAacConfig->audioObjectType); + hAacConfig->nChannels = + FDKaacEnc_GetChannelModeConfiguration(hAacConfig->channelMode) + ->nChannels; + hAacConfig->sbrRatio = + (pConfig->sbrMode == 0) ? 0 : (pConfig->sbrMode == 1) ? 1 : 2; + } + } + + { + UCHAR tpSignaling = + getSbrSignalingMode(hAacConfig->audioObjectType, config->userTpType, + config->userTpSignaling, hAacConfig->sbrRatio); + + if ((hAacConfig->audioObjectType == AOT_AAC_LC || + hAacConfig->audioObjectType == AOT_SBR || + hAacConfig->audioObjectType == AOT_PS) && + (config->userTpType == TT_MP4_LATM_MCP1 || + config->userTpType == TT_MP4_LATM_MCP0 || + config->userTpType == TT_MP4_LOAS) && + (tpSignaling == 1) && (config->userTpAmxv == 0)) { + /* For backward compatible explicit signaling, AMV1 has to be active */ + return AACENC_INVALID_CONFIG; + } + + if ((hAacConfig->audioObjectType == AOT_AAC_LC || + hAacConfig->audioObjectType == AOT_SBR || + hAacConfig->audioObjectType == AOT_PS) && + (tpSignaling == 0) && (hAacConfig->sbrRatio == 1)) { + /* Downsampled SBR has to be signaled explicitely (for transmission of SBR + * sampling fequency) */ + return AACENC_INVALID_CONFIG; + } + } + + switch (hAacConfig->bitrateMode) { + case AACENC_BR_MODE_CBR: + case AACENC_BR_MODE_VBR_1: + case AACENC_BR_MODE_VBR_2: + case AACENC_BR_MODE_VBR_3: + case AACENC_BR_MODE_VBR_4: + case AACENC_BR_MODE_VBR_5: + /* We need the frame length to call aacEncoder_LimitBitrate() */ + if (0 >= (hAacConfig->bitRate = aacEncoder_LimitBitrate( + NULL, hAacConfig->sampleRate, hAacConfig->framelength, + hAacConfig->nChannels, hAacConfig->channelMode, + hAacConfig->bitRate, hAacConfig->nSubFrames, + isSbrActive(hAacConfig), hAacConfig->sbrRatio, + hAacConfig->syntaxFlags, hAacConfig->audioObjectType))) { + return AACENC_INVALID_CONFIG; + } + break; + default: + break; + } + +#if 0 // TODO Is this still needed? + /* We need the frame length to call aacEncoder_LimitBitrate() */ + hAacConfig->bitRate = aacEncoder_LimitBitrate( + NULL, + hAacConfig->sampleRate, + hAacConfig->framelength, + hAacConfig->nChannels, + hAacConfig->channelMode, + hAacConfig->bitRate, + hAacConfig->nSubFrames, + isSbrActive(hAacConfig), + hAacConfig->sbrRatio, + hAacConfig->audioObjectType + ); +#endif + +/* Configure PNS */ + if (AACENC_BR_MODE_IS_VBR(hAacConfig->bitrateMode) /* VBR without PNS. */ + || (hAacConfig->useTns == 0)) /* TNS required. */ + { + hAacConfig->usePns = 0; + } + + if (hAacConfig->epConfig >= 0) { + hAacConfig->syntaxFlags |= AC_ER; + if (((INT)hAacConfig->channelMode < 1) || + ((INT)hAacConfig->channelMode > 14)) { + return AACENC_INVALID_CONFIG; /* Channel config 0 not supported. */ + } + } + + if ((hAacConfig->syntaxFlags & AC_LD_MPS) == 0) { + if (FDKaacEnc_DetermineEncoderMode(&hAacConfig->channelMode, + hAacConfig->nChannels) != AAC_ENC_OK) { + return AACENC_INVALID_CONFIG; /* nChannels doesn't match chMode, this is + just a check-up */ + } + } + + if ((hAacConfig->nChannels > hAacEncoder->nMaxAacChannels) || + ((FDKaacEnc_GetChannelModeConfiguration(hAacConfig->channelMode) + ->nChannelsEff > hAacEncoder->nMaxSbrChannels) && + isSbrActive(hAacConfig))) { + return AACENC_INVALID_CONFIG; /* not enough channels allocated */ + } + + /* Meta data restriction. */ + switch (hAacConfig->audioObjectType) { + /* Allow metadata support */ + case AOT_AAC_LC: + case AOT_SBR: + case AOT_PS: + case AOT_MP2_AAC_LC: + case AOT_MP2_SBR: + hAacEncoder->metaDataAllowed = 1; + if (!((((INT)hAacConfig->channelMode >= 1) && + ((INT)hAacConfig->channelMode <= 14)) || + (MODE_7_1_REAR_SURROUND == hAacConfig->channelMode) || + (MODE_7_1_FRONT_CENTER == hAacConfig->channelMode))) { + config->userMetaDataMode = 0; + } + break; + /* Prohibit metadata support */ + default: + hAacEncoder->metaDataAllowed = 0; + } + + return err; +} + +static INT aacenc_SbrCallback(void *self, HANDLE_FDK_BITSTREAM hBs, + const INT sampleRateIn, const INT sampleRateOut, + const INT samplesPerFrame, + const AUDIO_OBJECT_TYPE coreCodec, + const MP4_ELEMENT_ID elementID, + const INT elementIndex, const UCHAR harmonicSbr, + const UCHAR stereoConfigIndex, + const UCHAR configMode, UCHAR *configChanged, + const INT downscaleFactor) { + HANDLE_AACENCODER hAacEncoder = (HANDLE_AACENCODER)self; + + sbrEncoder_GetHeader(hAacEncoder->hEnvEnc, hBs, elementIndex, 0); + + return 0; +} + +INT aacenc_SscCallback(void *self, HANDLE_FDK_BITSTREAM hBs, + const AUDIO_OBJECT_TYPE coreCodec, + const INT samplingRate, const INT frameSize, + const INT stereoConfigIndex, + const INT coreSbrFrameLengthIndex, const INT configBytes, + const UCHAR configMode, UCHAR *configChanged) { + HANDLE_AACENCODER hAacEncoder = (HANDLE_AACENCODER)self; + + return (FDK_MpegsEnc_WriteSpatialSpecificConfig(hAacEncoder->hMpsEnc, hBs)); +} + +static AACENC_ERROR aacEncInit(HANDLE_AACENCODER hAacEncoder, ULONG InitFlags, + USER_PARAM *config) { + AACENC_ERROR err = AACENC_OK; + + INT aacBufferOffset = 0; + HANDLE_SBR_ENCODER *hSbrEncoder = &hAacEncoder->hEnvEnc; + HANDLE_AACENC_CONFIG hAacConfig = &hAacEncoder->aacConfig; + + hAacEncoder->nZerosAppended = 0; /* count appended zeros */ + + INT frameLength = hAacConfig->framelength; + + if ((InitFlags & AACENC_INIT_CONFIG)) { + CHANNEL_MODE prevChMode = hAacConfig->channelMode; + + /* Verify settings and update: config -> heAacEncoder */ + if ((err = FDKaacEnc_AdjustEncSettings(hAacEncoder, config)) != AACENC_OK) { + return err; + } + frameLength = hAacConfig->framelength; /* adapt temporal framelength */ + + /* Seamless channel reconfiguration in sbr not fully implemented */ + if ((prevChMode != hAacConfig->channelMode) && isSbrActive(hAacConfig)) { + InitFlags |= AACENC_INIT_STATES; + } + } + + /* Clear input buffer */ + if (InitFlags == AACENC_INIT_ALL) { + FDKmemclear(hAacEncoder->inputBuffer, + sizeof(INT_PCM) * hAacEncoder->inputBufferSize); + } + + if ((InitFlags & AACENC_INIT_CONFIG)) { + aacBufferOffset = 0; + switch (hAacConfig->audioObjectType) { + case AOT_ER_AAC_LD: + hAacEncoder->nDelay = DELAY_AACLD(hAacConfig->framelength); + break; + case AOT_ER_AAC_ELD: + hAacEncoder->nDelay = DELAY_AACELD(hAacConfig->framelength); + break; + default: + hAacEncoder->nDelay = + DELAY_AAC(hAacConfig->framelength); /* AAC encoder delay */ + } + + hAacConfig->ancDataBitRate = 0; + } + + if ((NULL != hAacEncoder->hEnvEnc) && isSbrActive(hAacConfig) && + ((InitFlags & AACENC_INIT_CONFIG) || (InitFlags & AACENC_INIT_STATES))) { + INT sbrError; + UINT initFlag = 0; + SBR_ELEMENT_INFO sbrElInfo[(8)]; + CHANNEL_MAPPING channelMapping; + CHANNEL_MODE channelMode = isPsActive(hAacConfig->audioObjectType) + ? config->userChannelMode + : hAacConfig->channelMode; + INT numChannels = isPsActive(hAacConfig->audioObjectType) + ? config->nChannels + : hAacConfig->nChannels; + + if (FDKaacEnc_InitChannelMapping(channelMode, hAacConfig->channelOrder, + &channelMapping) != AAC_ENC_OK) { + return AACENC_INIT_ERROR; + } + + /* Check return value and if the SBR encoder can handle enough elements */ + if (channelMapping.nElements > (8)) { + return AACENC_INIT_ERROR; + } + + aacEncDistributeSbrBits(&channelMapping, sbrElInfo, hAacConfig->bitRate); + + initFlag += (InitFlags & AACENC_INIT_STATES) ? 1 : 0; + + /* Let the SBR encoder take a look at the configuration and change if + * required. */ + sbrError = sbrEncoder_Init( + *hSbrEncoder, sbrElInfo, channelMapping.nElements, + hAacEncoder->inputBuffer, hAacEncoder->inputBufferSizePerChannel, + &hAacConfig->bandWidth, &aacBufferOffset, &numChannels, + hAacConfig->syntaxFlags, &hAacConfig->sampleRate, &hAacConfig->sbrRatio, + &frameLength, hAacConfig->audioObjectType, &hAacEncoder->nDelay, + (hAacConfig->audioObjectType == AOT_ER_AAC_ELD) ? 1 : TRANS_FAC, + (config->userTpHeaderPeriod != 0xFF) + ? config->userTpHeaderPeriod + : DEFAULT_HEADER_PERIOD_REPETITION_RATE, + initFlag); + + /* Suppress AOT reconfiguration and check error status. */ + if ((sbrError) || (numChannels != hAacConfig->nChannels)) { + return AACENC_INIT_SBR_ERROR; + } + + if (numChannels == 1) { + hAacConfig->channelMode = MODE_1; + } + + /* Never use PNS if SBR is active */ + if (hAacConfig->usePns) { + hAacConfig->usePns = 0; + } + + /* estimated bitrate consumed by SBR or PS */ + hAacConfig->ancDataBitRate = sbrEncoder_GetEstimateBitrate(*hSbrEncoder); + + } /* sbr initialization */ + + if ((hAacEncoder->hMpsEnc != NULL) && (hAacConfig->syntaxFlags & AC_LD_MPS)) { + int coreCoderDelay = DELAY_AACELD(hAacConfig->framelength); + + if (isSbrActive(hAacConfig)) { + coreCoderDelay = hAacConfig->sbrRatio * coreCoderDelay + + sbrEncoder_GetInputDataDelay(*hSbrEncoder); + } + + if (MPS_ENCODER_OK != + FDK_MpegsEnc_Init(hAacEncoder->hMpsEnc, hAacConfig->audioObjectType, + config->userSamplerate, hAacConfig->bitRate, + isSbrActive(hAacConfig) ? hAacConfig->sbrRatio : 0, + frameLength, /* for dual rate sbr this value is + already multiplied by 2 */ + hAacEncoder->inputBufferSizePerChannel, + coreCoderDelay)) { + return AACENC_INIT_MPS_ERROR; + } + } + hAacEncoder->nDelay = + fMax(FDK_MpegsEnc_GetDelay(hAacEncoder->hMpsEnc), hAacEncoder->nDelay); + + /* + * Initialize Transport - Module. + */ + if ((InitFlags & AACENC_INIT_TRANSPORT)) { + UINT flags = 0; + + FDKaacEnc_MapConfig( + &hAacEncoder->coderConfig, config, + getSbrSignalingMode(hAacConfig->audioObjectType, config->userTpType, + config->userTpSignaling, hAacConfig->sbrRatio), + hAacConfig); + + /* create flags for transport encoder */ + if (config->userTpAmxv != 0) { + flags |= TP_FLAG_LATM_AMV; + } + /* Clear output buffer */ + FDKmemclear(hAacEncoder->outBuffer, + hAacEncoder->outBufferInBytes * sizeof(UCHAR)); + + /* Initialize Bitstream encoder */ + if (transportEnc_Init(hAacEncoder->hTpEnc, hAacEncoder->outBuffer, + hAacEncoder->outBufferInBytes, config->userTpType, + &hAacEncoder->coderConfig, flags) != 0) { + return AACENC_INIT_TP_ERROR; + } + + } /* transport initialization */ + + /* + * Initialize AAC - Core. + */ + if ((InitFlags & AACENC_INIT_CONFIG) || (InitFlags & AACENC_INIT_STATES)) { + if (FDKaacEnc_Initialize( + hAacEncoder->hAacEnc, hAacConfig, hAacEncoder->hTpEnc, + (InitFlags & AACENC_INIT_STATES) ? 1 : 0) != AAC_ENC_OK) { + return AACENC_INIT_AAC_ERROR; + } + + } /* aac initialization */ + + /* + * Initialize Meta Data - Encoder. + */ + if (hAacEncoder->hMetadataEnc && (hAacEncoder->metaDataAllowed != 0) && + ((InitFlags & AACENC_INIT_CONFIG) || (InitFlags & AACENC_INIT_STATES))) { + INT inputDataDelay = DELAY_AAC(hAacConfig->framelength); + + if (isSbrActive(hAacConfig) && hSbrEncoder != NULL) { + inputDataDelay = hAacConfig->sbrRatio * inputDataDelay + + sbrEncoder_GetInputDataDelay(*hSbrEncoder); + } + + if (FDK_MetadataEnc_Init(hAacEncoder->hMetadataEnc, + ((InitFlags & AACENC_INIT_STATES) ? 1 : 0), + config->userMetaDataMode, inputDataDelay, + frameLength, config->userSamplerate, + config->nChannels, config->userChannelMode, + hAacConfig->channelOrder) != 0) { + return AACENC_INIT_META_ERROR; + } + + hAacEncoder->nDelay += FDK_MetadataEnc_GetDelay(hAacEncoder->hMetadataEnc); + } + + /* Get custom delay, i.e. the codec delay w/o the decoder's SBR- or MPS delay + */ + if ((hAacEncoder->hMpsEnc != NULL) && (hAacConfig->syntaxFlags & AC_LD_MPS)) { + hAacEncoder->nDelayCore = + hAacEncoder->nDelay - + fMax(0, FDK_MpegsEnc_GetDecDelay(hAacEncoder->hMpsEnc)); + } else if (isSbrActive(hAacConfig) && hSbrEncoder != NULL) { + hAacEncoder->nDelayCore = + hAacEncoder->nDelay - + fMax(0, sbrEncoder_GetSbrDecDelay(hAacEncoder->hEnvEnc)); + } else { + hAacEncoder->nDelayCore = hAacEncoder->nDelay; + } + + /* + * Update pointer to working buffer. + */ + if ((InitFlags & AACENC_INIT_CONFIG)) { + hAacEncoder->inputBufferOffset = aacBufferOffset; + + hAacEncoder->nSamplesToRead = frameLength * config->nChannels; + + } /* parameter changed */ + + return AACENC_OK; +} + +AACENC_ERROR aacEncOpen(HANDLE_AACENCODER *phAacEncoder, const UINT encModules, + const UINT maxChannels) { + AACENC_ERROR err = AACENC_OK; + HANDLE_AACENCODER hAacEncoder = NULL; + + if (phAacEncoder == NULL) { + err = AACENC_INVALID_HANDLE; + goto bail; + } + + /* allocate memory */ + hAacEncoder = Get_AacEncoder(); + + if (hAacEncoder == NULL) { + err = AACENC_MEMORY_ERROR; + goto bail; + } + + FDKmemclear(hAacEncoder, sizeof(AACENCODER)); + + /* Specify encoder modules to be allocated. */ + if (encModules == 0) { + C_ALLOC_SCRATCH_START(_pLibInfo, LIB_INFO, FDK_MODULE_LAST) + LIB_INFO(*pLibInfo) + [FDK_MODULE_LAST] = (LIB_INFO(*)[FDK_MODULE_LAST])_pLibInfo; + FDKinitLibInfo(*pLibInfo); + aacEncGetLibInfo(*pLibInfo); + + hAacEncoder->encoder_modis = ENC_MODE_FLAG_AAC; + if (FDKlibInfo_getCapabilities(*pLibInfo, FDK_SBRENC) & CAPF_SBR_HQ) { + hAacEncoder->encoder_modis |= ENC_MODE_FLAG_SBR; + } + if (FDKlibInfo_getCapabilities(*pLibInfo, FDK_SBRENC) & CAPF_SBR_PS_MPEG) { + hAacEncoder->encoder_modis |= ENC_MODE_FLAG_PS; + } + if (FDKlibInfo_getCapabilities(*pLibInfo, FDK_AACENC) & CAPF_AAC_DRC) { + hAacEncoder->encoder_modis |= ENC_MODE_FLAG_META; + } + hAacEncoder->encoder_modis |= ENC_MODE_FLAG_SAC; + + C_ALLOC_SCRATCH_END(_pLibInfo, LIB_INFO, FDK_MODULE_LAST) + } else { + hAacEncoder->encoder_modis = encModules; + } + + /* Determine max channel configuration. */ + if (maxChannels == 0) { + hAacEncoder->nMaxAacChannels = (8); + hAacEncoder->nMaxSbrChannels = (8); + } else { + hAacEncoder->nMaxAacChannels = (maxChannels & 0x00FF); + if ((hAacEncoder->encoder_modis & ENC_MODE_FLAG_SBR)) { + hAacEncoder->nMaxSbrChannels = (maxChannels & 0xFF00) + ? (maxChannels >> 8) + : hAacEncoder->nMaxAacChannels; + } + + if ((hAacEncoder->nMaxAacChannels > (8)) || + (hAacEncoder->nMaxSbrChannels > (8))) { + err = AACENC_INVALID_CONFIG; + goto bail; + } + } /* maxChannels==0 */ + + /* Max number of elements could be tuned any more. */ + hAacEncoder->nMaxAacElements = fixMin(((8)), hAacEncoder->nMaxAacChannels); + hAacEncoder->nMaxSbrElements = fixMin((8), hAacEncoder->nMaxSbrChannels); + + /* In case of memory overlay, allocate memory out of libraries */ + + if (hAacEncoder->encoder_modis & (ENC_MODE_FLAG_SBR | ENC_MODE_FLAG_PS)) + hAacEncoder->inputBufferSizePerChannel = INPUTBUFFER_SIZE; + else + hAacEncoder->inputBufferSizePerChannel = (1024); + + hAacEncoder->inputBufferSize = + hAacEncoder->nMaxAacChannels * hAacEncoder->inputBufferSizePerChannel; + + if (NULL == (hAacEncoder->inputBuffer = (INT_PCM *)FDKcalloc( + hAacEncoder->inputBufferSize, sizeof(INT_PCM)))) { + err = AACENC_MEMORY_ERROR; + goto bail; + } + + /* Open SBR Encoder */ + if (hAacEncoder->encoder_modis & ENC_MODE_FLAG_SBR) { + if (sbrEncoder_Open( + &hAacEncoder->hEnvEnc, hAacEncoder->nMaxSbrElements, + hAacEncoder->nMaxSbrChannels, + (hAacEncoder->encoder_modis & ENC_MODE_FLAG_PS) ? 1 : 0)) { + err = AACENC_MEMORY_ERROR; + goto bail; + } + + if (NULL == (hAacEncoder->pSbrPayload = (SBRENC_EXT_PAYLOAD *)FDKcalloc( + 1, sizeof(SBRENC_EXT_PAYLOAD)))) { + err = AACENC_MEMORY_ERROR; + goto bail; + } + } /* (encoder_modis&ENC_MODE_FLAG_SBR) */ + + /* Open Aac Encoder */ + if (FDKaacEnc_Open(&hAacEncoder->hAacEnc, hAacEncoder->nMaxAacElements, + hAacEncoder->nMaxAacChannels, (1)) != AAC_ENC_OK) { + err = AACENC_MEMORY_ERROR; + goto bail; + } + + /* Bitstream output buffer */ + hAacEncoder->outBufferInBytes = + 1 << (DFRACT_BITS - CntLeadingZeros(fixMax( + 1, ((1) * hAacEncoder->nMaxAacChannels * 6144) >> + 2))); /* buffer has to be 2^n */ + if (NULL == (hAacEncoder->outBuffer = (UCHAR *)FDKcalloc( + hAacEncoder->outBufferInBytes, sizeof(UCHAR)))) { + err = AACENC_MEMORY_ERROR; + goto bail; + } + + /* Open Meta Data Encoder */ + if (hAacEncoder->encoder_modis & ENC_MODE_FLAG_META) { + if (FDK_MetadataEnc_Open(&hAacEncoder->hMetadataEnc, + (UINT)hAacEncoder->nMaxAacChannels)) { + err = AACENC_MEMORY_ERROR; + goto bail; + } + } /* (encoder_modis&ENC_MODE_FLAG_META) */ + + /* Open MPEG Surround Encoder */ + if (hAacEncoder->encoder_modis & ENC_MODE_FLAG_SAC) { + if (MPS_ENCODER_OK != FDK_MpegsEnc_Open(&hAacEncoder->hMpsEnc)) { + err = AACENC_MEMORY_ERROR; + goto bail; + } + } /* (hAacEncoder->encoder_modis&ENC_MODE_FLAG_SAC) */ + + /* Open Transport Encoder */ + if (transportEnc_Open(&hAacEncoder->hTpEnc) != 0) { + err = AACENC_MEMORY_ERROR; + goto bail; + } else { + C_ALLOC_SCRATCH_START(_pLibInfo, LIB_INFO, FDK_MODULE_LAST) + + LIB_INFO(*pLibInfo) + [FDK_MODULE_LAST] = (LIB_INFO(*)[FDK_MODULE_LAST])_pLibInfo; + + FDKinitLibInfo(*pLibInfo); + transportEnc_GetLibInfo(*pLibInfo); + + /* Get capabilty flag for transport encoder. */ + hAacEncoder->CAPF_tpEnc = FDKlibInfo_getCapabilities(*pLibInfo, FDK_TPENC); + + C_ALLOC_SCRATCH_END(_pLibInfo, LIB_INFO, FDK_MODULE_LAST) + } + if (transportEnc_RegisterSbrCallback(hAacEncoder->hTpEnc, aacenc_SbrCallback, + hAacEncoder) != 0) { + err = AACENC_INIT_TP_ERROR; + goto bail; + } + if (transportEnc_RegisterSscCallback(hAacEncoder->hTpEnc, aacenc_SscCallback, + hAacEncoder) != 0) { + err = AACENC_INIT_TP_ERROR; + goto bail; + } + + /* Initialize encoder instance with default parameters. */ + aacEncDefaultConfig(&hAacEncoder->aacConfig, &hAacEncoder->extParam); + + /* Initialize headerPeriod in coderConfig for aacEncoder_GetParam(). */ + hAacEncoder->coderConfig.headerPeriod = + hAacEncoder->extParam.userTpHeaderPeriod; + + /* All encoder modules have to be initialized */ + hAacEncoder->InitFlags = AACENC_INIT_ALL; + + /* Return encoder instance */ + *phAacEncoder = hAacEncoder; + + return err; + +bail: + aacEncClose(&hAacEncoder); + + return err; +} + +AACENC_ERROR aacEncClose(HANDLE_AACENCODER *phAacEncoder) { + AACENC_ERROR err = AACENC_OK; + + if (phAacEncoder == NULL) { + err = AACENC_INVALID_HANDLE; + goto bail; + } + + if (*phAacEncoder != NULL) { + HANDLE_AACENCODER hAacEncoder = *phAacEncoder; + + if (hAacEncoder->inputBuffer != NULL) { + FDKfree(hAacEncoder->inputBuffer); + hAacEncoder->inputBuffer = NULL; + } + if (hAacEncoder->outBuffer != NULL) { + FDKfree(hAacEncoder->outBuffer); + hAacEncoder->outBuffer = NULL; + } + + if (hAacEncoder->hEnvEnc) { + sbrEncoder_Close(&hAacEncoder->hEnvEnc); + } + if (hAacEncoder->pSbrPayload != NULL) { + FDKfree(hAacEncoder->pSbrPayload); + hAacEncoder->pSbrPayload = NULL; + } + if (hAacEncoder->hAacEnc) { + FDKaacEnc_Close(&hAacEncoder->hAacEnc); + } + + transportEnc_Close(&hAacEncoder->hTpEnc); + + if (hAacEncoder->hMetadataEnc) { + FDK_MetadataEnc_Close(&hAacEncoder->hMetadataEnc); + } + if (hAacEncoder->hMpsEnc) { + FDK_MpegsEnc_Close(&hAacEncoder->hMpsEnc); + } + + Free_AacEncoder(phAacEncoder); + } + +bail: + return err; +} + +AACENC_ERROR aacEncEncode(const HANDLE_AACENCODER hAacEncoder, + const AACENC_BufDesc *inBufDesc, + const AACENC_BufDesc *outBufDesc, + const AACENC_InArgs *inargs, + AACENC_OutArgs *outargs) { + AACENC_ERROR err = AACENC_OK; + INT i, nBsBytes = 0; + INT outBytes[(1)]; + int nExtensions = 0; + int ancDataExtIdx = -1; + + /* deal with valid encoder handle */ + if (hAacEncoder == NULL) { + err = AACENC_INVALID_HANDLE; + goto bail; + } + + /* + * Adjust user settings and trigger reinitialization. + */ + if (hAacEncoder->InitFlags != 0) { + err = + aacEncInit(hAacEncoder, hAacEncoder->InitFlags, &hAacEncoder->extParam); + + if (err != AACENC_OK) { + /* keep init flags alive! */ + goto bail; + } + hAacEncoder->InitFlags = AACENC_INIT_NONE; + } + + if (outargs != NULL) { + FDKmemclear(outargs, sizeof(AACENC_OutArgs)); + } + + if (outBufDesc != NULL) { + for (i = 0; i < outBufDesc->numBufs; i++) { + if (outBufDesc->bufs[i] != NULL) { + FDKmemclear(outBufDesc->bufs[i], outBufDesc->bufSizes[i]); + } + } + } + + /* + * If only encoder handle given, independent (re)initialization can be + * triggered. + */ + if ((inBufDesc == NULL) && (outBufDesc == NULL) && (inargs == NULL) && + (outargs == NULL)) { + goto bail; + } + + /* check if buffer descriptors are filled out properly. */ + if ((inargs == NULL) || (outargs == NULL) || + ((AACENC_OK != validateBufDesc(inBufDesc)) && + (inargs->numInSamples > 0)) || + (AACENC_OK != validateBufDesc(outBufDesc))) { + err = AACENC_UNSUPPORTED_PARAMETER; + goto bail; + } + + /* reset buffer wich signals number of valid bytes in output bitstream buffer + */ + FDKmemclear(outBytes, hAacEncoder->aacConfig.nSubFrames * sizeof(INT)); + + /* + * Manage incoming audio samples. + */ + if ((inBufDesc != NULL) && (inargs->numInSamples > 0) && + (getBufDescIdx(inBufDesc, IN_AUDIO_DATA) != -1)) { + /* Fetch data until nSamplesToRead reached */ + INT idx = getBufDescIdx(inBufDesc, IN_AUDIO_DATA); + INT newSamples = + fixMax(0, fixMin(inargs->numInSamples, hAacEncoder->nSamplesToRead - + hAacEncoder->nSamplesRead)); + INT_PCM *pIn = + hAacEncoder->inputBuffer + + (hAacEncoder->inputBufferOffset + hAacEncoder->nSamplesRead) / + hAacEncoder->aacConfig.nChannels; + + /* Copy new input samples to internal buffer */ + if (inBufDesc->bufElSizes[idx] == (INT)sizeof(INT_PCM)) { + FDK_deinterleave((INT_PCM *)inBufDesc->bufs[idx], pIn, + hAacEncoder->extParam.nChannels, + newSamples / hAacEncoder->extParam.nChannels, + hAacEncoder->inputBufferSizePerChannel); + } else if (inBufDesc->bufElSizes[idx] > (INT)sizeof(INT_PCM)) { + FDK_deinterleave((LONG *)inBufDesc->bufs[idx], pIn, + hAacEncoder->extParam.nChannels, + newSamples / hAacEncoder->extParam.nChannels, + hAacEncoder->inputBufferSizePerChannel); + } else { + FDK_deinterleave((SHORT *)inBufDesc->bufs[idx], pIn, + hAacEncoder->extParam.nChannels, + newSamples / hAacEncoder->extParam.nChannels, + hAacEncoder->inputBufferSizePerChannel); + } + hAacEncoder->nSamplesRead += newSamples; + + /* Number of fetched input buffer samples. */ + outargs->numInSamples = newSamples; + } + + /* input buffer completely filled ? */ + if (hAacEncoder->nSamplesRead < hAacEncoder->nSamplesToRead) { + /* - eof reached and flushing enabled, or + - return to main and wait for further incoming audio samples */ + if (inargs->numInSamples == -1) { + if ((hAacEncoder->nZerosAppended < hAacEncoder->nDelay)) { + int nZeros = (hAacEncoder->nSamplesToRead - hAacEncoder->nSamplesRead) / + hAacEncoder->extParam.nChannels; + + FDK_ASSERT(nZeros >= 0); + + /* clear out until end-of-buffer */ + if (nZeros) { + for (i = 0; i < (int)hAacEncoder->extParam.nChannels; i++) { + FDKmemclear(hAacEncoder->inputBuffer + + i * hAacEncoder->inputBufferSizePerChannel + + (hAacEncoder->inputBufferOffset + + hAacEncoder->nSamplesRead) / + hAacEncoder->extParam.nChannels, + sizeof(INT_PCM) * nZeros); + } + hAacEncoder->nZerosAppended += nZeros; + hAacEncoder->nSamplesRead = hAacEncoder->nSamplesToRead; + } + } else { /* flushing completed */ + err = AACENC_ENCODE_EOF; /* eof reached */ + goto bail; + } + } else { /* inargs->numInSamples!= -1 */ + goto bail; /* not enough samples in input buffer and no flushing enabled + */ + } + } + + /* init payload */ + FDKmemclear(hAacEncoder->extPayload, + sizeof(AACENC_EXT_PAYLOAD) * MAX_TOTAL_EXT_PAYLOADS); + for (i = 0; i < MAX_TOTAL_EXT_PAYLOADS; i++) { + hAacEncoder->extPayload[i].associatedChElement = -1; + } + if (hAacEncoder->pSbrPayload != NULL) { + FDKmemclear(hAacEncoder->pSbrPayload, sizeof(*hAacEncoder->pSbrPayload)); + } + + /* + * Calculate Meta Data info. + */ + if ((hAacEncoder->hMetadataEnc != NULL) && + (hAacEncoder->metaDataAllowed != 0)) { + const AACENC_MetaData *pMetaData = NULL; + AACENC_EXT_PAYLOAD *pMetaDataExtPayload = NULL; + UINT nMetaDataExtensions = 0; + INT matrix_mixdown_idx = 0; + + /* New meta data info available ? */ + if (getBufDescIdx(inBufDesc, IN_METADATA_SETUP) != -1) { + pMetaData = + (AACENC_MetaData *) + inBufDesc->bufs[getBufDescIdx(inBufDesc, IN_METADATA_SETUP)]; + } + + FDK_MetadataEnc_Process( + hAacEncoder->hMetadataEnc, + hAacEncoder->inputBuffer + hAacEncoder->inputBufferOffset / + hAacEncoder->coderConfig.noChannels, + hAacEncoder->inputBufferSizePerChannel, hAacEncoder->nSamplesRead, + pMetaData, &pMetaDataExtPayload, &nMetaDataExtensions, + &matrix_mixdown_idx); + + for (i = 0; i < (INT)nMetaDataExtensions; + i++) { /* Get meta data extension payload. */ + hAacEncoder->extPayload[nExtensions++] = pMetaDataExtPayload[i]; + } + + if ((matrix_mixdown_idx != -1) && + ((hAacEncoder->extParam.userChannelMode == MODE_1_2_2) || + (hAacEncoder->extParam.userChannelMode == MODE_1_2_2_1))) { + /* Set matrix mixdown coefficient. */ + UINT pceValue = (UINT)((0 << 3) | ((matrix_mixdown_idx & 0x3) << 1) | 1); + if (hAacEncoder->extParam.userPceAdditions != pceValue) { + hAacEncoder->extParam.userPceAdditions = pceValue; + hAacEncoder->InitFlags |= AACENC_INIT_TRANSPORT; + } + } + } + + /* + * Encode MPS data. + */ + if ((hAacEncoder->hMpsEnc != NULL) && + (hAacEncoder->aacConfig.syntaxFlags & AC_LD_MPS)) { + AACENC_EXT_PAYLOAD mpsExtensionPayload; + FDKmemclear(&mpsExtensionPayload, sizeof(AACENC_EXT_PAYLOAD)); + + if (MPS_ENCODER_OK != + FDK_MpegsEnc_Process( + hAacEncoder->hMpsEnc, + hAacEncoder->inputBuffer + hAacEncoder->inputBufferOffset / + hAacEncoder->coderConfig.noChannels, + hAacEncoder->nSamplesRead, &mpsExtensionPayload)) { + err = AACENC_ENCODE_ERROR; + goto bail; + } + + if ((mpsExtensionPayload.pData != NULL) && + ((mpsExtensionPayload.dataSize != 0))) { + hAacEncoder->extPayload[nExtensions++] = mpsExtensionPayload; + } + } + + if ((NULL != hAacEncoder->hEnvEnc) && (NULL != hAacEncoder->pSbrPayload) && + isSbrActive(&hAacEncoder->aacConfig)) { + INT nPayload = 0; + + /* + * Encode SBR data. + */ + if (sbrEncoder_EncodeFrame(hAacEncoder->hEnvEnc, hAacEncoder->inputBuffer, + hAacEncoder->inputBufferSizePerChannel, + hAacEncoder->pSbrPayload->dataSize[nPayload], + hAacEncoder->pSbrPayload->data[nPayload])) { + err = AACENC_ENCODE_ERROR; + goto bail; + } else { + /* Add SBR extension payload */ + for (i = 0; i < (8); i++) { + if (hAacEncoder->pSbrPayload->dataSize[nPayload][i] > 0) { + hAacEncoder->extPayload[nExtensions].pData = + hAacEncoder->pSbrPayload->data[nPayload][i]; + { + hAacEncoder->extPayload[nExtensions].dataSize = + hAacEncoder->pSbrPayload->dataSize[nPayload][i]; + hAacEncoder->extPayload[nExtensions].associatedChElement = i; + } + hAacEncoder->extPayload[nExtensions].dataType = + EXT_SBR_DATA; /* Once SBR Encoder supports SBR CRC set + EXT_SBR_DATA_CRC */ + nExtensions++; /* or EXT_SBR_DATA according to configuration. */ + FDK_ASSERT(nExtensions <= MAX_TOTAL_EXT_PAYLOADS); + } + } + nPayload++; + } + } /* sbrEnabled */ + + if ((inargs->numAncBytes > 0) && + (getBufDescIdx(inBufDesc, IN_ANCILLRY_DATA) != -1)) { + INT idx = getBufDescIdx(inBufDesc, IN_ANCILLRY_DATA); + hAacEncoder->extPayload[nExtensions].dataSize = inargs->numAncBytes * 8; + hAacEncoder->extPayload[nExtensions].pData = (UCHAR *)inBufDesc->bufs[idx]; + hAacEncoder->extPayload[nExtensions].dataType = EXT_DATA_ELEMENT; + hAacEncoder->extPayload[nExtensions].associatedChElement = -1; + ancDataExtIdx = nExtensions; /* store index */ + nExtensions++; + } + + /* + * Encode AAC - Core. + */ + if (FDKaacEnc_EncodeFrame(hAacEncoder->hAacEnc, hAacEncoder->hTpEnc, + hAacEncoder->inputBuffer, + hAacEncoder->inputBufferSizePerChannel, outBytes, + hAacEncoder->extPayload) != AAC_ENC_OK) { + err = AACENC_ENCODE_ERROR; + goto bail; + } + + if (ancDataExtIdx >= 0) { + outargs->numAncBytes = + inargs->numAncBytes - + (hAacEncoder->extPayload[ancDataExtIdx].dataSize >> 3); + } + + /* samples exhausted */ + hAacEncoder->nSamplesRead -= hAacEncoder->nSamplesToRead; + + /* + * Delay balancing buffer handling + */ + if (isSbrActive(&hAacEncoder->aacConfig)) { + sbrEncoder_UpdateBuffers(hAacEncoder->hEnvEnc, hAacEncoder->inputBuffer, + hAacEncoder->inputBufferSizePerChannel); + } + + /* + * Make bitstream public + */ + if ((outBufDesc != NULL) && (outBufDesc->numBufs >= 1)) { + INT bsIdx = getBufDescIdx(outBufDesc, OUT_BITSTREAM_DATA); + INT auIdx = getBufDescIdx(outBufDesc, OUT_AU_SIZES); + + for (i = 0, nBsBytes = 0; i < hAacEncoder->aacConfig.nSubFrames; i++) { + nBsBytes += outBytes[i]; + + if (auIdx != -1) { + ((INT *)outBufDesc->bufs[auIdx])[i] = outBytes[i]; + } + } + + if ((bsIdx != -1) && (outBufDesc->bufSizes[bsIdx] >= nBsBytes)) { + FDKmemcpy(outBufDesc->bufs[bsIdx], hAacEncoder->outBuffer, + sizeof(UCHAR) * nBsBytes); + outargs->numOutBytes = nBsBytes; + outargs->bitResState = + FDKaacEnc_GetBitReservoirState(hAacEncoder->hAacEnc); + } else { + /* output buffer too small, can't write valid bitstream */ + err = AACENC_ENCODE_ERROR; + goto bail; + } + } + +bail: + if (err == AACENC_ENCODE_ERROR) { + /* All encoder modules have to be initialized */ + hAacEncoder->InitFlags = AACENC_INIT_ALL; + } + + return err; +} + +static AAC_ENCODER_ERROR aacEncGetConf(HANDLE_AACENCODER hAacEncoder, + UINT *size, UCHAR *confBuffer) { + FDK_BITSTREAM tmpConf; + UINT confType; + UCHAR buf[64]; + int err; + + /* Init bit buffer */ + FDKinitBitStream(&tmpConf, buf, 64, 0, BS_WRITER); + + /* write conf in tmp buffer */ + err = transportEnc_GetConf(hAacEncoder->hTpEnc, &hAacEncoder->coderConfig, + &tmpConf, &confType); + + /* copy data to outbuffer: length in bytes */ + FDKbyteAlign(&tmpConf, 0); + + /* Check buffer size */ + if (FDKgetValidBits(&tmpConf) > ((*size) << 3)) return AAC_ENC_UNKNOWN; + + FDKfetchBuffer(&tmpConf, confBuffer, size); + + if (err != 0) + return AAC_ENC_UNKNOWN; + else + return AAC_ENC_OK; +} + +AACENC_ERROR aacEncGetLibInfo(LIB_INFO *info) { + int i = 0; + + if (info == NULL) { + return AACENC_INVALID_HANDLE; + } + + FDK_toolsGetLibInfo(info); + transportEnc_GetLibInfo(info); + sbrEncoder_GetLibInfo(info); + FDK_MpegsEnc_GetLibInfo(info); + + /* search for next free tab */ + for (i = 0; i < FDK_MODULE_LAST; i++) { + if (info[i].module_id == FDK_NONE) break; + } + if (i == FDK_MODULE_LAST) { + return AACENC_INIT_ERROR; + } + + info[i].module_id = FDK_AACENC; + info[i].build_date = AACENCODER_LIB_BUILD_DATE; + info[i].build_time = AACENCODER_LIB_BUILD_TIME; + info[i].title = AACENCODER_LIB_TITLE; + info[i].version = + LIB_VERSION(AACENCODER_LIB_VL0, AACENCODER_LIB_VL1, AACENCODER_LIB_VL2); + ; + LIB_VERSION_STRING(&info[i]); + + /* Capability flags */ + info[i].flags = 0 | CAPF_AAC_1024 | CAPF_AAC_LC | CAPF_AAC_960| CAPF_AAC_512 | + CAPF_AAC_480 | CAPF_AAC_DRC | CAPF_AAC_ELD_DOWNSCALE; + /* End of flags */ + + return AACENC_OK; +} + +AACENC_ERROR aacEncoder_SetParam(const HANDLE_AACENCODER hAacEncoder, + const AACENC_PARAM param, const UINT value) { + AACENC_ERROR err = AACENC_OK; + USER_PARAM *settings = &hAacEncoder->extParam; + + /* check encoder handle */ + if (hAacEncoder == NULL) { + err = AACENC_INVALID_HANDLE; + goto bail; + } + + /* apply param value */ + switch (param) { + case AACENC_AOT: + if (settings->userAOT != (AUDIO_OBJECT_TYPE)value) { + /* check if AOT matches the allocated modules */ + switch (value) { + case AOT_PS: + case AOT_DABPLUS_PS: + if (!(hAacEncoder->encoder_modis & (ENC_MODE_FLAG_PS))) { + err = AACENC_INVALID_CONFIG; + goto bail; + } + FDK_FALLTHROUGH; + case AOT_SBR: + case AOT_MP2_SBR: + case AOT_DABPLUS_SBR: + if (!(hAacEncoder->encoder_modis & (ENC_MODE_FLAG_SBR))) { + err = AACENC_INVALID_CONFIG; + goto bail; + } + FDK_FALLTHROUGH; + case AOT_AAC_LC: + case AOT_MP2_AAC_LC: + case AOT_DABPLUS_AAC_LC: + case AOT_ER_AAC_LD: + case AOT_ER_AAC_ELD: + if (!(hAacEncoder->encoder_modis & (ENC_MODE_FLAG_AAC))) { + err = AACENC_INVALID_CONFIG; + goto bail; + } + break; + default: + err = AACENC_INVALID_CONFIG; + goto bail; + } /* switch value */ + settings->userAOT = (AUDIO_OBJECT_TYPE)value; + hAacEncoder->InitFlags |= + AACENC_INIT_CONFIG | AACENC_INIT_STATES | AACENC_INIT_TRANSPORT; + } + break; + case AACENC_BITRATE: + if (settings->userBitrate != value) { + settings->userBitrate = value; + hAacEncoder->InitFlags |= AACENC_INIT_CONFIG | AACENC_INIT_TRANSPORT; + } + break; + case AACENC_BITRATEMODE: + if (settings->userBitrateMode != value) { + switch (value) { + case 0: + case 1: + case 2: + case 3: + case 4: + case 5: + case 7: + case 8: + settings->userBitrateMode = value; + hAacEncoder->InitFlags |= + AACENC_INIT_CONFIG | AACENC_INIT_TRANSPORT; + break; + default: + err = AACENC_INVALID_CONFIG; + break; + } /* switch value */ + } + break; + case AACENC_SAMPLERATE: + if (settings->userSamplerate != value) { + if (!((value == 8000) || (value == 11025) || (value == 12000) || + (value == 16000) || (value == 22050) || (value == 24000) || + (value == 32000) || (value == 44100) || (value == 48000) || + (value == 64000) || (value == 88200) || (value == 96000))) { + err = AACENC_INVALID_CONFIG; + break; + } + settings->userSamplerate = value; + hAacEncoder->nSamplesRead = 0; /* reset internal inputbuffer */ + hAacEncoder->InitFlags |= + AACENC_INIT_CONFIG | AACENC_INIT_STATES | AACENC_INIT_TRANSPORT; + } + break; + case AACENC_CHANNELMODE: + if (settings->userChannelMode != (CHANNEL_MODE)value) { + if (((CHANNEL_MODE)value == MODE_212) && + (NULL != hAacEncoder->hMpsEnc)) { + settings->userChannelMode = (CHANNEL_MODE)value; + settings->nChannels = 2; + } else { + const CHANNEL_MODE_CONFIG_TAB *pConfig = + FDKaacEnc_GetChannelModeConfiguration((CHANNEL_MODE)value); + if (pConfig == NULL) { + err = AACENC_INVALID_CONFIG; + break; + } + if ((pConfig->nElements > hAacEncoder->nMaxAacElements) || + (pConfig->nChannelsEff > hAacEncoder->nMaxAacChannels)) { + err = AACENC_INVALID_CONFIG; + break; + } + + settings->userChannelMode = (CHANNEL_MODE)value; + settings->nChannels = pConfig->nChannels; + } + hAacEncoder->nSamplesRead = 0; /* reset internal inputbuffer */ + hAacEncoder->InitFlags |= AACENC_INIT_CONFIG | AACENC_INIT_TRANSPORT; + if (!((value >= 1) && (value <= 6))) { + hAacEncoder->InitFlags |= AACENC_INIT_STATES; + } + } + break; + case AACENC_BANDWIDTH: + if (settings->userBandwidth != value) { + settings->userBandwidth = value; + hAacEncoder->InitFlags |= AACENC_INIT_CONFIG; + } + break; + case AACENC_CHANNELORDER: + if (hAacEncoder->aacConfig.channelOrder != (CHANNEL_ORDER)value) { + if (!((value == 0) || (value == 1) || (value == 2))) { + err = AACENC_INVALID_CONFIG; + break; + } + hAacEncoder->aacConfig.channelOrder = (CHANNEL_ORDER)value; + hAacEncoder->nSamplesRead = 0; /* reset internal inputbuffer */ + hAacEncoder->InitFlags |= + AACENC_INIT_CONFIG | AACENC_INIT_STATES | AACENC_INIT_TRANSPORT; + } + break; + case AACENC_AFTERBURNER: + if (settings->userAfterburner != value) { + if (!((value == 0) || (value == 1))) { + err = AACENC_INVALID_CONFIG; + break; + } + settings->userAfterburner = value; + hAacEncoder->InitFlags |= AACENC_INIT_CONFIG; + } + break; + case AACENC_GRANULE_LENGTH: + if (settings->userFramelength != value) { + switch (value) { + case 1024: + case 960: + case 512: + case 480: + case 256: + case 240: + case 128: + case 120: + if ((value << 1) == 480 || (value << 1) == 512) { + settings->userDownscaleFactor = 2; + } else if ((value << 2) == 480 || (value << 2) == 512) { + settings->userDownscaleFactor = 4; + } + settings->userFramelength = value; + hAacEncoder->InitFlags |= + AACENC_INIT_CONFIG | AACENC_INIT_TRANSPORT; + break; + default: + err = AACENC_INVALID_CONFIG; + break; + } + } + break; + case AACENC_SBR_RATIO: + if (settings->userSbrRatio != value) { + if (!((value == 0) || (value == 1) || (value == 2))) { + err = AACENC_INVALID_CONFIG; + break; + } + settings->userSbrRatio = value; + hAacEncoder->InitFlags |= + AACENC_INIT_CONFIG | AACENC_INIT_STATES | AACENC_INIT_TRANSPORT; + } + break; + case AACENC_SBR_MODE: + if ((settings->userSbrEnabled != value) && + (NULL != hAacEncoder->hEnvEnc)) { + settings->userSbrEnabled = value; + hAacEncoder->InitFlags |= + AACENC_INIT_CONFIG | AACENC_INIT_STATES | AACENC_INIT_TRANSPORT; + } + break; + case AACENC_TRANSMUX: + if (settings->userTpType != (TRANSPORT_TYPE)value) { + TRANSPORT_TYPE type = (TRANSPORT_TYPE)value; + UINT flags = hAacEncoder->CAPF_tpEnc; + + if (!(((type == TT_MP4_ADIF) && (flags & CAPF_ADIF)) || + ((type == TT_MP4_ADTS) && (flags & CAPF_ADTS)) || + ((type == TT_MP4_LATM_MCP0) && + ((flags & CAPF_LATM) && (flags & CAPF_RAWPACKETS))) || + ((type == TT_MP4_LATM_MCP1) && + ((flags & CAPF_LATM) && (flags & CAPF_RAWPACKETS))) || + ((type == TT_MP4_LOAS) && (flags & CAPF_LOAS)) || + ((type == TT_MP4_RAW) && (flags & CAPF_RAWPACKETS)) || + ((type == TT_DABPLUS) && ((flags & CAPF_DAB_AAC) && (flags & CAPF_RAWPACKETS))) )) { + err = AACENC_INVALID_CONFIG; + break; + } + settings->userTpType = (TRANSPORT_TYPE)value; + hAacEncoder->InitFlags |= AACENC_INIT_TRANSPORT; + } + break; + case AACENC_SIGNALING_MODE: + if (settings->userTpSignaling != value) { + if (!((value == 0) || (value == 1) || (value == 2))) { + err = AACENC_INVALID_CONFIG; + break; + } + settings->userTpSignaling = value; + hAacEncoder->InitFlags |= AACENC_INIT_TRANSPORT; + } + break; + case AACENC_PROTECTION: + if (settings->userTpProtection != value) { + if (!((value == 0) || (value == 1))) { + err = AACENC_INVALID_CONFIG; + break; + } + settings->userTpProtection = value; + hAacEncoder->InitFlags |= AACENC_INIT_TRANSPORT; + } + break; + case AACENC_HEADER_PERIOD: + if (settings->userTpHeaderPeriod != value) { + if (!(((INT)value >= 0) && (value <= 255))) { + err = AACENC_INVALID_CONFIG; + break; + } + settings->userTpHeaderPeriod = value; + hAacEncoder->InitFlags |= AACENC_INIT_TRANSPORT; + } + break; + case AACENC_AUDIOMUXVER: + if (settings->userTpAmxv != value) { + if (!((value == 0) || (value == 1) || (value == 2))) { + err = AACENC_INVALID_CONFIG; + break; + } + settings->userTpAmxv = value; + hAacEncoder->InitFlags |= AACENC_INIT_TRANSPORT; + } + break; + case AACENC_TPSUBFRAMES: + if (settings->userTpNsubFrames != value) { + if (!((value >= 1) && (value <= 6))) { + err = AACENC_INVALID_CONFIG; + break; + } + settings->userTpNsubFrames = value; + hAacEncoder->InitFlags |= AACENC_INIT_TRANSPORT; + } + break; + case AACENC_ANCILLARY_BITRATE: + if (settings->userAncDataRate != value) { + settings->userAncDataRate = value; + } + break; + case AACENC_CONTROL_STATE: + if (hAacEncoder->InitFlags != value) { + if (value & AACENC_RESET_INBUFFER) { + hAacEncoder->nSamplesRead = 0; + } + hAacEncoder->InitFlags = value; + } + break; + case AACENC_METADATA_MODE: + if ((UINT)settings->userMetaDataMode != value) { + if (!(((INT)value >= 0) && ((INT)value <= 3))) { + err = AACENC_INVALID_CONFIG; + break; + } + settings->userMetaDataMode = value; + hAacEncoder->InitFlags |= AACENC_INIT_CONFIG; + } + break; + case AACENC_PEAK_BITRATE: + if (settings->userPeakBitrate != value) { + settings->userPeakBitrate = value; + hAacEncoder->InitFlags |= AACENC_INIT_CONFIG | AACENC_INIT_TRANSPORT; + } + break; + default: + err = AACENC_UNSUPPORTED_PARAMETER; + break; + } /* switch(param) */ + +bail: + return err; +} + +UINT aacEncoder_GetParam(const HANDLE_AACENCODER hAacEncoder, + const AACENC_PARAM param) { + UINT value = 0; + USER_PARAM *settings = &hAacEncoder->extParam; + + /* check encoder handle */ + if (hAacEncoder == NULL) { + goto bail; + } + + /* apply param value */ + switch (param) { + case AACENC_AOT: + value = (UINT)hAacEncoder->aacConfig.audioObjectType; + break; + case AACENC_BITRATE: + switch (hAacEncoder->aacConfig.bitrateMode) { + case AACENC_BR_MODE_CBR: + value = (UINT)hAacEncoder->aacConfig.bitRate; + break; + default: + value = (UINT)-1; + } + break; + case AACENC_BITRATEMODE: + value = (UINT)((hAacEncoder->aacConfig.bitrateMode != AACENC_BR_MODE_FF) + ? hAacEncoder->aacConfig.bitrateMode + : AACENC_BR_MODE_CBR); + break; + case AACENC_SAMPLERATE: + value = (UINT)hAacEncoder->coderConfig.extSamplingRate; + break; + case AACENC_CHANNELMODE: + if ((MODE_1 == hAacEncoder->aacConfig.channelMode) && + (hAacEncoder->aacConfig.syntaxFlags & AC_LD_MPS)) { + value = MODE_212; + } else { + value = (UINT)hAacEncoder->aacConfig.channelMode; + } + break; + case AACENC_BANDWIDTH: + value = (UINT)hAacEncoder->aacConfig.bandWidth; + break; + case AACENC_CHANNELORDER: + value = (UINT)hAacEncoder->aacConfig.channelOrder; + break; + case AACENC_AFTERBURNER: + value = (UINT)hAacEncoder->aacConfig.useRequant; + break; + case AACENC_GRANULE_LENGTH: + value = (UINT)hAacEncoder->aacConfig.framelength; + break; + case AACENC_SBR_RATIO: + value = isSbrActive(&hAacEncoder->aacConfig) + ? hAacEncoder->aacConfig.sbrRatio + : 0; + break; + case AACENC_SBR_MODE: + value = + (UINT)(hAacEncoder->aacConfig.syntaxFlags & AC_SBR_PRESENT) ? 1 : 0; + break; + case AACENC_TRANSMUX: + value = (UINT)settings->userTpType; + break; + case AACENC_SIGNALING_MODE: + value = (UINT)getSbrSignalingMode( + hAacEncoder->aacConfig.audioObjectType, settings->userTpType, + settings->userTpSignaling, hAacEncoder->aacConfig.sbrRatio); + break; + case AACENC_PROTECTION: + value = (UINT)settings->userTpProtection; + break; + case AACENC_HEADER_PERIOD: + value = (UINT)hAacEncoder->coderConfig.headerPeriod; + break; + case AACENC_AUDIOMUXVER: + value = (UINT)hAacEncoder->aacConfig.audioMuxVersion; + break; + case AACENC_TPSUBFRAMES: + value = (UINT)settings->userTpNsubFrames; + break; + case AACENC_ANCILLARY_BITRATE: + value = (UINT)hAacEncoder->aacConfig.anc_Rate; + break; + case AACENC_CONTROL_STATE: + value = (UINT)hAacEncoder->InitFlags; + break; + case AACENC_METADATA_MODE: + value = (hAacEncoder->metaDataAllowed == 0) + ? 0 + : (UINT)settings->userMetaDataMode; + break; + case AACENC_PEAK_BITRATE: + value = (UINT)-1; /* peak bitrate parameter is meaningless */ + if (((INT)hAacEncoder->extParam.userPeakBitrate != -1)) { + value = + (UINT)(fMax((INT)hAacEncoder->extParam.userPeakBitrate, + hAacEncoder->aacConfig + .bitRate)); /* peak bitrate parameter is in use */ + } + break; + + default: + // err = MPS_INVALID_PARAMETER; + break; + } /* switch(param) */ + +bail: + return value; +} + +AACENC_ERROR aacEncInfo(const HANDLE_AACENCODER hAacEncoder, + AACENC_InfoStruct *pInfo) { + AACENC_ERROR err = AACENC_OK; + + FDKmemclear(pInfo, sizeof(AACENC_InfoStruct)); + pInfo->confSize = 64; /* pre-initialize */ + + pInfo->maxOutBufBytes = ((hAacEncoder->nMaxAacChannels * 6144) + 7) >> 3; + pInfo->maxAncBytes = hAacEncoder->aacConfig.maxAncBytesPerAU; + pInfo->inBufFillLevel = + hAacEncoder->nSamplesRead / hAacEncoder->extParam.nChannels; + pInfo->inputChannels = hAacEncoder->extParam.nChannels; + pInfo->frameLength = + hAacEncoder->nSamplesToRead / hAacEncoder->extParam.nChannels; + pInfo->nDelay = hAacEncoder->nDelay; + pInfo->nDelayCore = hAacEncoder->nDelayCore; + + /* Get encoder configuration */ + if (aacEncGetConf(hAacEncoder, &pInfo->confSize, &pInfo->confBuf[0]) != + AAC_ENC_OK) { + err = AACENC_INIT_ERROR; + goto bail; + } +bail: + return err; +} diff --git a/fdk-aac/libAACenc/src/aacenc_pns.cpp b/fdk-aac/libAACenc/src/aacenc_pns.cpp new file mode 100644 index 0000000..f0571d6 --- /dev/null +++ b/fdk-aac/libAACenc/src/aacenc_pns.cpp @@ -0,0 +1,541 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M. Lohwasser + + Description: pns.c + +*******************************************************************************/ + +#include "aacenc_pns.h" + +#include "psy_data.h" +#include "pnsparam.h" +#include "noisedet.h" +#include "bit_cnt.h" +#include "interface.h" + +/* minCorrelationEnergy = (1.0e-10f)^2 ~ 2^-67 = 2^-47 * 2^-20 */ +static const FIXP_DBL minCorrelationEnergy = + FL2FXCONST_DBL(0.0); /* FL2FXCONST_DBL((float)FDKpow(2.0,-47)); */ +/* noiseCorrelationThresh = 0.6^2 */ +static const FIXP_DBL noiseCorrelationThresh = FL2FXCONST_DBL(0.36); + +static void FDKaacEnc_FDKaacEnc_noiseDetection( + PNS_CONFIG *pnsConf, PNS_DATA *pnsData, const INT sfbActive, + const INT *sfbOffset, INT tnsOrder, INT tnsPredictionGain, INT tnsActive, + FIXP_DBL *mdctSpectrum, INT *sfbMaxScaleSpec, FIXP_SGL *sfbtonality); + +static void FDKaacEnc_CalcNoiseNrgs(const INT sfbActive, INT *pnsFlag, + FIXP_DBL *sfbEnergyLdData, INT *noiseNrg); + +/***************************************************************************** + + functionname: initPnsConfiguration + description: fill pnsConf with pns parameters + returns: error status + input: PNS Config struct (modified) + bitrate, samplerate, usePns, + number of sfb's, pointer to sfb offset + output: error code + +*****************************************************************************/ + +AAC_ENCODER_ERROR FDKaacEnc_InitPnsConfiguration( + PNS_CONFIG *pnsConf, INT bitRate, INT sampleRate, INT usePns, INT sfbCnt, + const INT *sfbOffset, const INT numChan, const INT isLC) { + AAC_ENCODER_ERROR ErrorStatus; + + /* init noise detection */ + ErrorStatus = FDKaacEnc_GetPnsParam(&pnsConf->np, bitRate, sampleRate, sfbCnt, + sfbOffset, &usePns, numChan, isLC); + if (ErrorStatus != AAC_ENC_OK) return ErrorStatus; + + pnsConf->minCorrelationEnergy = minCorrelationEnergy; + pnsConf->noiseCorrelationThresh = noiseCorrelationThresh; + + pnsConf->usePns = usePns; + + return AAC_ENC_OK; +} + +/***************************************************************************** + + functionname: FDKaacEnc_PnsDetect + description: do decision, if PNS shall used or not + returns: + input: pns config structure + pns data structure (modified), + lastWindowSequence (long or short blocks) + sfbActive + pointer to Sfb Energy, Threshold, Offset + pointer to mdct Spectrum + length of each group + pointer to tonality calculated in chaosmeasure + tns order and prediction gain + calculated noiseNrg at active PNS + output: pnsFlag in pns data structure + +*****************************************************************************/ +void FDKaacEnc_PnsDetect(PNS_CONFIG *pnsConf, PNS_DATA *pnsData, + const INT lastWindowSequence, const INT sfbActive, + const INT maxSfbPerGroup, FIXP_DBL *sfbThresholdLdData, + const INT *sfbOffset, FIXP_DBL *mdctSpectrum, + INT *sfbMaxScaleSpec, FIXP_SGL *sfbtonality, + INT tnsOrder, INT tnsPredictionGain, INT tnsActive, + FIXP_DBL *sfbEnergyLdData, INT *noiseNrg) + +{ + int sfb; + int startNoiseSfb; + + /* Reset pns info. */ + FDKmemclear(pnsData->pnsFlag, sizeof(pnsData->pnsFlag)); + for (sfb = 0; sfb < MAX_GROUPED_SFB; sfb++) { + noiseNrg[sfb] = NO_NOISE_PNS; + } + + /* Disable PNS and skip detection in certain cases. */ + if (pnsConf->usePns == 0) { + return; + } else { + /* AAC - LC core encoder */ + if ((pnsConf->np.detectionAlgorithmFlags & IS_LOW_COMPLEXITY) && + (lastWindowSequence == SHORT_WINDOW)) { + return; + } + /* AAC - (E)LD core encoder */ + if (!(pnsConf->np.detectionAlgorithmFlags & IS_LOW_COMPLEXITY) && + (pnsConf->np.detectionAlgorithmFlags & JUST_LONG_WINDOW) && + (lastWindowSequence != LONG_WINDOW)) { + return; + } + } + + /* + call noise detection + */ + FDKaacEnc_FDKaacEnc_noiseDetection( + pnsConf, pnsData, sfbActive, sfbOffset, tnsOrder, tnsPredictionGain, + tnsActive, mdctSpectrum, sfbMaxScaleSpec, sfbtonality); + + /* set startNoiseSfb (long) */ + startNoiseSfb = pnsConf->np.startSfb; + + /* Set noise substitution status */ + for (sfb = 0; sfb < sfbActive; sfb++) { + /* No PNS below startNoiseSfb */ + if (sfb < startNoiseSfb) { + pnsData->pnsFlag[sfb] = 0; + continue; + } + + /* + do noise substitution if + fuzzy measure is high enough + sfb freq > minimum sfb freq + signal in coder band is not masked + */ + + if ((pnsData->noiseFuzzyMeasure[sfb] > FL2FXCONST_SGL(0.5)) && + ((sfbThresholdLdData[sfb] + + FL2FXCONST_DBL(0.5849625f / + 64.0f)) /* thr * 1.5 = thrLd +ld(1.5)/64 */ + < sfbEnergyLdData[sfb])) { + /* + mark in psyout flag array that we will code + this band with PNS + */ + pnsData->pnsFlag[sfb] = 1; /* PNS_ON */ + } else { + pnsData->pnsFlag[sfb] = 0; /* PNS_OFF */ + } + + /* no PNS if LTP is active */ + } + + /* avoid PNS holes */ + if ((pnsData->noiseFuzzyMeasure[0] > FL2FXCONST_SGL(0.5f)) && + (pnsData->pnsFlag[1])) { + pnsData->pnsFlag[0] = 1; + } + + for (sfb = 1; sfb < maxSfbPerGroup - 1; sfb++) { + if ((pnsData->noiseFuzzyMeasure[sfb] > pnsConf->np.gapFillThr) && + (pnsData->pnsFlag[sfb - 1]) && (pnsData->pnsFlag[sfb + 1])) { + pnsData->pnsFlag[sfb] = 1; + } + } + + if (maxSfbPerGroup > 0) { + /* avoid PNS hole */ + if ((pnsData->noiseFuzzyMeasure[maxSfbPerGroup - 1] > + pnsConf->np.gapFillThr) && + (pnsData->pnsFlag[maxSfbPerGroup - 2])) { + pnsData->pnsFlag[maxSfbPerGroup - 1] = 1; + } + /* avoid single PNS band */ + if (pnsData->pnsFlag[maxSfbPerGroup - 2] == 0) { + pnsData->pnsFlag[maxSfbPerGroup - 1] = 0; + } + } + + /* avoid single PNS bands */ + if (pnsData->pnsFlag[1] == 0) { + pnsData->pnsFlag[0] = 0; + } + + for (sfb = 1; sfb < maxSfbPerGroup - 1; sfb++) { + if ((pnsData->pnsFlag[sfb - 1] == 0) && (pnsData->pnsFlag[sfb + 1] == 0)) { + pnsData->pnsFlag[sfb] = 0; + } + } + + /* + calculate noiseNrg's + */ + FDKaacEnc_CalcNoiseNrgs(sfbActive, pnsData->pnsFlag, sfbEnergyLdData, + noiseNrg); +} + +/***************************************************************************** + + functionname:FDKaacEnc_FDKaacEnc_noiseDetection + description: wrapper for noisedet.c + returns: + input: pns config structure + pns data structure (modified), + sfbActive + tns order and prediction gain + pointer to mdct Spectrumand Sfb Energy + pointer to Sfb tonality + output: noiseFuzzyMeasure in structure pnsData + flags tonal / nontonal + +*****************************************************************************/ +static void FDKaacEnc_FDKaacEnc_noiseDetection( + PNS_CONFIG *pnsConf, PNS_DATA *pnsData, const INT sfbActive, + const INT *sfbOffset, int tnsOrder, INT tnsPredictionGain, INT tnsActive, + FIXP_DBL *mdctSpectrum, INT *sfbMaxScaleSpec, FIXP_SGL *sfbtonality) { + INT condition = TRUE; + if (!(pnsConf->np.detectionAlgorithmFlags & IS_LOW_COMPLEXITY)) { + condition = (tnsOrder > 3); + } + /* + no PNS if heavy TNS activity + clear pnsData->noiseFuzzyMeasure + */ + if ((pnsConf->np.detectionAlgorithmFlags & USE_TNS_GAIN_THR) && + (tnsPredictionGain >= pnsConf->np.tnsGainThreshold) && condition && + !((pnsConf->np.detectionAlgorithmFlags & USE_TNS_PNS) && + (tnsPredictionGain >= pnsConf->np.tnsPNSGainThreshold) && + (tnsActive))) { + /* clear all noiseFuzzyMeasure */ + FDKmemclear(pnsData->noiseFuzzyMeasure, sfbActive * sizeof(FIXP_SGL)); + } else { + /* + call noise detection, output in pnsData->noiseFuzzyMeasure, + use real mdct spectral data + */ + FDKaacEnc_noiseDetect(mdctSpectrum, sfbMaxScaleSpec, sfbActive, sfbOffset, + pnsData->noiseFuzzyMeasure, &pnsConf->np, + sfbtonality); + } +} + +/***************************************************************************** + + functionname:FDKaacEnc_CalcNoiseNrgs + description: Calculate the NoiseNrg's + returns: + input: sfbActive + if pnsFlag calculate NoiseNrg + pointer to sfbEnergy and groupLen + pointer to noiseNrg (modified) + output: noiseNrg's in pnsFlaged sfb's + +*****************************************************************************/ + +static void FDKaacEnc_CalcNoiseNrgs(const INT sfbActive, INT *RESTRICT pnsFlag, + FIXP_DBL *RESTRICT sfbEnergyLdData, + INT *RESTRICT noiseNrg) { + int sfb; + INT tmp = (-LOG_NORM_PCM) << 2; + + for (sfb = 0; sfb < sfbActive; sfb++) { + if (pnsFlag[sfb]) { + INT nrg = (-sfbEnergyLdData[sfb] + FL2FXCONST_DBL(0.5f / 64.0f)) >> + (DFRACT_BITS - 1 - 7); + noiseNrg[sfb] = tmp - nrg; + } + } +} + +/***************************************************************************** + + functionname:FDKaacEnc_CodePnsChannel + description: Execute pns decission + returns: + input: sfbActive + pns config structure + use PNS if pnsFlag + pointer to Sfb Energy, noiseNrg, Threshold + output: set sfbThreshold high to code pe with 0, + noiseNrg marks flag for pns coding + +*****************************************************************************/ + +void FDKaacEnc_CodePnsChannel(const INT sfbActive, PNS_CONFIG *pnsConf, + INT *RESTRICT pnsFlag, + FIXP_DBL *RESTRICT sfbEnergyLdData, + INT *RESTRICT noiseNrg, + FIXP_DBL *RESTRICT sfbThresholdLdData) { + INT sfb; + INT lastiNoiseEnergy = 0; + INT firstPNSband = 1; /* TRUE for first PNS-coded band */ + + /* no PNS */ + if (!pnsConf->usePns) { + for (sfb = 0; sfb < sfbActive; sfb++) { + /* no PNS coding */ + noiseNrg[sfb] = NO_NOISE_PNS; + } + return; + } + + /* code PNS */ + for (sfb = 0; sfb < sfbActive; sfb++) { + if (pnsFlag[sfb]) { + /* high sfbThreshold causes pe = 0 */ + if (noiseNrg[sfb] != NO_NOISE_PNS) + sfbThresholdLdData[sfb] = + sfbEnergyLdData[sfb] + FL2FXCONST_DBL(1.0f / LD_DATA_SCALING); + + /* set noiseNrg in valid region */ + if (!firstPNSband) { + INT deltaiNoiseEnergy = noiseNrg[sfb] - lastiNoiseEnergy; + + if (deltaiNoiseEnergy > CODE_BOOK_PNS_LAV) + noiseNrg[sfb] -= deltaiNoiseEnergy - CODE_BOOK_PNS_LAV; + else if (deltaiNoiseEnergy < -CODE_BOOK_PNS_LAV) + noiseNrg[sfb] -= deltaiNoiseEnergy + CODE_BOOK_PNS_LAV; + } else { + firstPNSband = 0; + } + lastiNoiseEnergy = noiseNrg[sfb]; + } else { + /* no PNS coding */ + noiseNrg[sfb] = NO_NOISE_PNS; + } + } +} + +/***************************************************************************** + + functionname:FDKaacEnc_PreProcessPnsChannelPair + description: Calculate the correlation of noise in a channel pair + + returns: + input: sfbActive + pointer to sfb energies left, right and mid channel + pns config structure + pns data structure left and right (modified) + + output: noiseEnergyCorrelation in pns data structure + +*****************************************************************************/ + +void FDKaacEnc_PreProcessPnsChannelPair( + const INT sfbActive, FIXP_DBL *RESTRICT sfbEnergyLeft, + FIXP_DBL *RESTRICT sfbEnergyRight, FIXP_DBL *RESTRICT sfbEnergyLeftLD, + FIXP_DBL *RESTRICT sfbEnergyRightLD, FIXP_DBL *RESTRICT sfbEnergyMid, + PNS_CONFIG *RESTRICT pnsConf, PNS_DATA *pnsDataLeft, + PNS_DATA *pnsDataRight) { + INT sfb; + FIXP_DBL ccf; + + if (!pnsConf->usePns) return; + + FIXP_DBL *RESTRICT pNoiseEnergyCorrelationL = + pnsDataLeft->noiseEnergyCorrelation; + FIXP_DBL *RESTRICT pNoiseEnergyCorrelationR = + pnsDataRight->noiseEnergyCorrelation; + + for (sfb = 0; sfb < sfbActive; sfb++) { + FIXP_DBL quot = (sfbEnergyLeftLD[sfb] >> 1) + (sfbEnergyRightLD[sfb] >> 1); + + if (quot < FL2FXCONST_DBL(-32.0f / (float)LD_DATA_SCALING)) + ccf = FL2FXCONST_DBL(0.0f); + else { + FIXP_DBL accu = + sfbEnergyMid[sfb] - + (((sfbEnergyLeft[sfb] >> 1) + (sfbEnergyRight[sfb] >> 1)) >> 1); + INT sign = (accu < FL2FXCONST_DBL(0.0f)) ? 1 : 0; + accu = fixp_abs(accu); + + ccf = CalcLdData(accu) + + FL2FXCONST_DBL((float)1.0f / (float)LD_DATA_SCALING) - + quot; /* ld(accu*2) = ld(accu) + 1 */ + ccf = (ccf >= FL2FXCONST_DBL(0.0)) + ? ((FIXP_DBL)MAXVAL_DBL) + : (sign) ? -CalcInvLdData(ccf) : CalcInvLdData(ccf); + } + + pNoiseEnergyCorrelationL[sfb] = ccf; + pNoiseEnergyCorrelationR[sfb] = ccf; + } +} + +/***************************************************************************** + + functionname:FDKaacEnc_PostProcessPnsChannelPair + description: if PNS used at left and right channel, + use msMask to flag correlation + returns: + input: sfbActive + pns config structure + pns data structure left and right (modified) + pointer to msMask, flags correlation by pns coding (modified) + Digest of MS coding + output: pnsFlag in pns data structure, + msFlag in msMask (flags correlation) + +*****************************************************************************/ + +void FDKaacEnc_PostProcessPnsChannelPair(const INT sfbActive, + PNS_CONFIG *pnsConf, + PNS_DATA *pnsDataLeft, + PNS_DATA *pnsDataRight, + INT *RESTRICT msMask, INT *msDigest) { + INT sfb; + + if (!pnsConf->usePns) return; + + for (sfb = 0; sfb < sfbActive; sfb++) { + /* + MS post processing + */ + if (msMask[sfb]) { + if ((pnsDataLeft->pnsFlag[sfb]) && (pnsDataRight->pnsFlag[sfb])) { + /* AAC only: Standard */ + /* do this to avoid ms flags in layers that should not have it */ + if (pnsDataLeft->noiseEnergyCorrelation[sfb] <= + pnsConf->noiseCorrelationThresh) { + msMask[sfb] = 0; + *msDigest = MS_SOME; + } + } else { + /* + No PNS coding + */ + pnsDataLeft->pnsFlag[sfb] = 0; + pnsDataRight->pnsFlag[sfb] = 0; + } + } + + /* + Use MS flag to signal noise correlation if + pns is active in both channels + */ + if ((pnsDataLeft->pnsFlag[sfb]) && (pnsDataRight->pnsFlag[sfb])) { + if (pnsDataLeft->noiseEnergyCorrelation[sfb] > + pnsConf->noiseCorrelationThresh) { + msMask[sfb] = 1; + *msDigest = MS_SOME; + } + } + } +} diff --git a/fdk-aac/libAACenc/src/aacenc_pns.h b/fdk-aac/libAACenc/src/aacenc_pns.h new file mode 100644 index 0000000..4938fcf --- /dev/null +++ b/fdk-aac/libAACenc/src/aacenc_pns.h @@ -0,0 +1,124 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M. Lohwasser + + Description: pns.h + +*******************************************************************************/ + +#ifndef AACENC_PNS_H +#define AACENC_PNS_H + +#include "common_fix.h" +#include "pnsparam.h" + +#define NO_NOISE_PNS FDK_INT_MIN + +typedef struct { + NOISEPARAMS np; + FIXP_DBL minCorrelationEnergy; + FIXP_DBL noiseCorrelationThresh; + INT usePns; +} PNS_CONFIG; + +typedef struct { + FIXP_SGL noiseFuzzyMeasure[MAX_GROUPED_SFB]; + FIXP_DBL noiseEnergyCorrelation[MAX_GROUPED_SFB]; + INT pnsFlag[MAX_GROUPED_SFB]; +} PNS_DATA; + +#endif /* AACENC_PNS_H */ diff --git a/fdk-aac/libAACenc/src/aacenc_tns.cpp b/fdk-aac/libAACenc/src/aacenc_tns.cpp new file mode 100644 index 0000000..3436150 --- /dev/null +++ b/fdk-aac/libAACenc/src/aacenc_tns.cpp @@ -0,0 +1,1210 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): Alex Groeschel, Tobias Chalupka + + Description: Temporal noise shaping + +*******************************************************************************/ + +#include "aacenc_tns.h" +#include "psy_const.h" +#include "psy_configuration.h" +#include "tns_func.h" +#include "aacEnc_rom.h" +#include "aacenc_tns.h" +#include "FDK_lpc.h" + +#define FILTER_DIRECTION 0 /* 0 = up, 1 = down */ + +static const FIXP_DBL acfWindowLong[12 + 3 + 1] = { + 0x7fffffff, 0x7fb80000, 0x7ee00000, 0x7d780000, 0x7b800000, 0x78f80000, + 0x75e00000, 0x72380000, 0x6e000000, 0x69380000, 0x63e00000, 0x5df80000, + 0x57800000, 0x50780000, 0x48e00000, 0x40b80000}; + +static const FIXP_DBL acfWindowShort[4 + 3 + 1] = { + 0x7fffffff, 0x7e000000, 0x78000000, 0x6e000000, + 0x60000000, 0x4e000000, 0x38000000, 0x1e000000}; + +typedef struct { + INT bitRateFrom[2]; /* noneSbr=0, useSbr=1 */ + INT bitRateTo[2]; /* noneSbr=0, useSbr=1 */ + TNS_PARAMETER_TABULATED paramTab[2]; /* mono=0, stereo=1 */ + +} TNS_INFO_TAB; + +#define TNS_TIMERES_SCALE (1) +#define FL2_TIMERES_FIX(a) (FL2FXCONST_DBL(a / (float)(1 << TNS_TIMERES_SCALE))) + +static const TNS_INFO_TAB tnsInfoTab[] = { + {{16000, 13500}, + {32000, 28000}, + {{{1, 1}, + {1437, 1500}, + {1400, 600}, + {12, 12}, + {FILTER_DIRECTION, FILTER_DIRECTION}, + {3, 1}, + {FL2_TIMERES_FIX(0.4f), FL2_TIMERES_FIX(1.2f)}, + 1}, + {{1, 1}, + {1437, 1500}, + {1400, 600}, + {12, 12}, + {FILTER_DIRECTION, FILTER_DIRECTION}, + {3, 1}, + {FL2_TIMERES_FIX(0.4f), FL2_TIMERES_FIX(1.2f)}, + 1}}}, + {{32001, 28001}, + {60000, 52000}, + {{{1, 1}, + {1437, 1500}, + {1400, 600}, + {12, 10}, + {FILTER_DIRECTION, FILTER_DIRECTION}, + {3, 1}, + {FL2_TIMERES_FIX(0.4f), FL2_TIMERES_FIX(1.0f)}, + 1}, + {{1, 1}, + {1437, 1500}, + {1400, 600}, + {12, 10}, + {FILTER_DIRECTION, FILTER_DIRECTION}, + {3, 1}, + {FL2_TIMERES_FIX(0.4f), FL2_TIMERES_FIX(1.0f)}, + 1}}}, + {{60001, 52001}, + {384000, 384000}, + {{{1, 1}, + {1437, 1500}, + {1400, 600}, + {12, 8}, + {FILTER_DIRECTION, FILTER_DIRECTION}, + {3, 1}, + {FL2_TIMERES_FIX(0.4f), FL2_TIMERES_FIX(1.0f)}, + 1}, + {{1, 1}, + {1437, 1500}, + {1400, 600}, + {12, 8}, + {FILTER_DIRECTION, FILTER_DIRECTION}, + {3, 1}, + {FL2_TIMERES_FIX(0.4f), FL2_TIMERES_FIX(1.0f)}, + 1}}}}; + +typedef struct { + INT samplingRate; + SCHAR maxBands[2]; /* long=0; short=1 */ + +} TNS_MAX_TAB_ENTRY; + +static const TNS_MAX_TAB_ENTRY tnsMaxBandsTab1024[] = { + {96000, {31, 9}}, {88200, {31, 9}}, {64000, {34, 10}}, {48000, {40, 14}}, + {44100, {42, 14}}, {32000, {51, 14}}, {24000, {46, 14}}, {22050, {46, 14}}, + {16000, {42, 14}}, {12000, {42, 14}}, {11025, {42, 14}}, {8000, {39, 14}}}; + +static const TNS_MAX_TAB_ENTRY tnsMaxBandsTab960[] = +{ + { 96000, { 31, 9}}, + { 88200, { 31, 9}}, + { 64000, { 34, 10}}, + { 48000, { 49, 14}}, + { 44100, { 49, 14}}, + { 32000, { 49, 14}}, + { 24000, { 46, 15}}, + { 22050, { 46, 14}}, + { 16000, { 46, 15}}, + { 12000, { 42, 15}}, + { 11025, { 42, 15}}, + { 8000, { 40, 15}} +}; + +static const TNS_MAX_TAB_ENTRY tnsMaxBandsTab120[] = { + {48000, {12, -1}}, /* 48000 */ + {44100, {12, -1}}, /* 44100 */ + {32000, {15, -1}}, /* 32000 */ + {24000, {15, -1}}, /* 24000 */ + {22050, {15, -1}} /* 22050 */ +}; + +static const TNS_MAX_TAB_ENTRY tnsMaxBandsTab128[] = { + {48000, {12, -1}}, /* 48000 */ + {44100, {12, -1}}, /* 44100 */ + {32000, {15, -1}}, /* 32000 */ + {24000, {15, -1}}, /* 24000 */ + {22050, {15, -1}} /* 22050 */ +}; + +static const TNS_MAX_TAB_ENTRY tnsMaxBandsTab240[] = { + {96000, {22, -1}}, /* 96000 */ + {48000, {22, -1}}, /* 48000 */ + {44100, {22, -1}}, /* 44100 */ + {32000, {21, -1}}, /* 32000 */ + {24000, {21, -1}}, /* 24000 */ + {22050, {21, -1}} /* 22050 */ +}; + +static const TNS_MAX_TAB_ENTRY tnsMaxBandsTab256[] = { + {96000, {25, -1}}, /* 96000 */ + {48000, {25, -1}}, /* 48000 */ + {44100, {25, -1}}, /* 44100 */ + {32000, {24, -1}}, /* 32000 */ + {24000, {24, -1}}, /* 24000 */ + {22050, {24, -1}} /* 22050 */ +}; +static const TNS_MAX_TAB_ENTRY tnsMaxBandsTab480[] = {{48000, {31, -1}}, + {44100, {32, -1}}, + {32000, {37, -1}}, + {24000, {30, -1}}, + {22050, {30, -1}}}; + +static const TNS_MAX_TAB_ENTRY tnsMaxBandsTab512[] = {{48000, {31, -1}}, + {44100, {32, -1}}, + {32000, {37, -1}}, + {24000, {31, -1}}, + {22050, {31, -1}}}; + +static void FDKaacEnc_Parcor2Index(const FIXP_LPC *parcor, INT *RESTRICT index, + const INT order, const INT bitsPerCoeff); + +static void FDKaacEnc_Index2Parcor(const INT *index, FIXP_LPC *RESTRICT parcor, + const INT order, const INT bitsPerCoeff); + +static void FDKaacEnc_CalcGaussWindow(FIXP_DBL *win, const int winSize, + const INT samplingRate, + const INT transformResolution, + const FIXP_DBL timeResolution, + const INT timeResolution_e); + +static const TNS_PARAMETER_TABULATED *FDKaacEnc_GetTnsParam(const INT bitRate, + const INT channels, + const INT sbrLd) { + int i; + const TNS_PARAMETER_TABULATED *tnsConfigTab = NULL; + + for (i = 0; i < (int)(sizeof(tnsInfoTab) / sizeof(TNS_INFO_TAB)); i++) { + if ((bitRate >= tnsInfoTab[i].bitRateFrom[sbrLd ? 1 : 0]) && + bitRate <= tnsInfoTab[i].bitRateTo[sbrLd ? 1 : 0]) { + tnsConfigTab = &tnsInfoTab[i].paramTab[(channels == 1) ? 0 : 1]; + } + } + + return tnsConfigTab; +} + +static INT getTnsMaxBands(const INT sampleRate, const INT granuleLength, + const INT isShortBlock) { + int i; + INT numBands = -1; + const TNS_MAX_TAB_ENTRY *pMaxBandsTab = NULL; + int maxBandsTabSize = 0; + + switch (granuleLength) { + case 960: + pMaxBandsTab = tnsMaxBandsTab960; + maxBandsTabSize = sizeof(tnsMaxBandsTab960) / sizeof(TNS_MAX_TAB_ENTRY); + break; + case 1024: + pMaxBandsTab = tnsMaxBandsTab1024; + maxBandsTabSize = sizeof(tnsMaxBandsTab1024) / sizeof(TNS_MAX_TAB_ENTRY); + break; + case 120: + pMaxBandsTab = tnsMaxBandsTab120; + maxBandsTabSize = sizeof(tnsMaxBandsTab120) / sizeof(TNS_MAX_TAB_ENTRY); + break; + case 128: + pMaxBandsTab = tnsMaxBandsTab128; + maxBandsTabSize = sizeof(tnsMaxBandsTab128) / sizeof(TNS_MAX_TAB_ENTRY); + break; + case 240: + pMaxBandsTab = tnsMaxBandsTab240; + maxBandsTabSize = sizeof(tnsMaxBandsTab240) / sizeof(TNS_MAX_TAB_ENTRY); + break; + case 256: + pMaxBandsTab = tnsMaxBandsTab256; + maxBandsTabSize = sizeof(tnsMaxBandsTab256) / sizeof(TNS_MAX_TAB_ENTRY); + break; + case 480: + pMaxBandsTab = tnsMaxBandsTab480; + maxBandsTabSize = sizeof(tnsMaxBandsTab480) / sizeof(TNS_MAX_TAB_ENTRY); + break; + case 512: + pMaxBandsTab = tnsMaxBandsTab512; + maxBandsTabSize = sizeof(tnsMaxBandsTab512) / sizeof(TNS_MAX_TAB_ENTRY); + break; + default: + numBands = -1; + } + + if (pMaxBandsTab != NULL) { + for (i = 0; i < maxBandsTabSize; i++) { + numBands = pMaxBandsTab[i].maxBands[(!isShortBlock) ? 0 : 1]; + if (sampleRate >= pMaxBandsTab[i].samplingRate) { + break; + } + } + } + + return numBands; +} + +/***************************************************************************/ +/*! + \brief FDKaacEnc_FreqToBandWidthRounding + + Returns index of nearest band border + + \param frequency + \param sampling frequency + \param total number of bands + \param pointer to table of band borders + + \return band border +****************************************************************************/ + +INT FDKaacEnc_FreqToBandWidthRounding(const INT freq, const INT fs, + const INT numOfBands, + const INT *bandStartOffset) { + INT lineNumber, band; + + /* assert(freq >= 0); */ + lineNumber = (freq * bandStartOffset[numOfBands] * 4 / fs + 1) / 2; + + /* freq > fs/2 */ + if (lineNumber >= bandStartOffset[numOfBands]) return numOfBands; + + /* find band the line number lies in */ + for (band = 0; band < numOfBands; band++) { + if (bandStartOffset[band + 1] > lineNumber) break; + } + + /* round to nearest band border */ + if (lineNumber - bandStartOffset[band] > + bandStartOffset[band + 1] - lineNumber) { + band++; + } + + return (band); +} + +/***************************************************************************** + + functionname: FDKaacEnc_InitTnsConfiguration + description: fill TNS_CONFIG structure with sensible content + returns: + input: bitrate, samplerate, number of channels, + blocktype (long or short), + TNS Config struct (modified), + psy config struct, + tns active flag + output: + +*****************************************************************************/ +AAC_ENCODER_ERROR FDKaacEnc_InitTnsConfiguration( + INT bitRate, INT sampleRate, INT channels, INT blockType, INT granuleLength, + INT isLowDelay, INT ldSbrPresent, TNS_CONFIG *tC, PSY_CONFIGURATION *pC, + INT active, INT useTnsPeak) { + int i; + // float acfTimeRes = (blockType == SHORT_WINDOW) ? 0.125f : 0.046875f; + + if (channels <= 0) return (AAC_ENCODER_ERROR)1; + + tC->isLowDelay = isLowDelay; + + /* initialize TNS filter flag, order, and coefficient resolution (in bits per + * coeff) */ + tC->tnsActive = (active) ? TRUE : FALSE; + tC->maxOrder = (blockType == SHORT_WINDOW) ? 5 : 12; /* maximum: 7, 20 */ + if (bitRate < 16000) tC->maxOrder -= 2; + tC->coefRes = (blockType == SHORT_WINDOW) ? 3 : 4; + + /* LPC stop line: highest MDCT line to be coded, but do not go beyond + * TNS_MAX_BANDS! */ + tC->lpcStopBand = getTnsMaxBands(sampleRate, granuleLength, + (blockType == SHORT_WINDOW) ? 1 : 0); + + if (tC->lpcStopBand < 0) { + return (AAC_ENCODER_ERROR)1; + } + + tC->lpcStopBand = fMin(tC->lpcStopBand, pC->sfbActive); + tC->lpcStopLine = pC->sfbOffset[tC->lpcStopBand]; + + switch (granuleLength) { + case 960: + case 1024: + /* TNS start line: skip lower MDCT lines to prevent artifacts due to + * filter mismatch */ + if (blockType == SHORT_WINDOW) { + tC->lpcStartBand[LOFILT] = 0; + } else { + tC->lpcStartBand[LOFILT] = + (sampleRate < 9391) ? 2 : ((sampleRate < 18783) ? 4 : 8); + } + tC->lpcStartLine[LOFILT] = pC->sfbOffset[tC->lpcStartBand[LOFILT]]; + + i = tC->lpcStopBand; + while (pC->sfbOffset[i] > + (tC->lpcStartLine[LOFILT] + + (tC->lpcStopLine - tC->lpcStartLine[LOFILT]) / 4)) + i--; + tC->lpcStartBand[HIFILT] = i; + tC->lpcStartLine[HIFILT] = pC->sfbOffset[i]; + + tC->confTab.threshOn[HIFILT] = 1437; + tC->confTab.threshOn[LOFILT] = 1500; + + tC->confTab.tnsLimitOrder[HIFILT] = tC->maxOrder; + tC->confTab.tnsLimitOrder[LOFILT] = fMax(0, tC->maxOrder - 7); + + tC->confTab.tnsFilterDirection[HIFILT] = FILTER_DIRECTION; + tC->confTab.tnsFilterDirection[LOFILT] = FILTER_DIRECTION; + + tC->confTab.acfSplit[HIFILT] = + -1; /* signal Merged4to2QuartersAutoCorrelation in + FDKaacEnc_MergedAutoCorrelation*/ + tC->confTab.acfSplit[LOFILT] = + -1; /* signal Merged4to2QuartersAutoCorrelation in + FDKaacEnc_MergedAutoCorrelation */ + + tC->confTab.filterEnabled[HIFILT] = 1; + tC->confTab.filterEnabled[LOFILT] = 1; + tC->confTab.seperateFiltersAllowed = 1; + + /* compute autocorrelation window based on maximum filter order for given + * block type */ + /* for (i = 0; i <= tC->maxOrder + 3; i++) { + float acfWinTemp = acfTimeRes * i; + acfWindow[i] = FL2FXCONST_DBL(1.0f - acfWinTemp * acfWinTemp); + } + */ + if (blockType == SHORT_WINDOW) { + FDKmemcpy(tC->acfWindow[HIFILT], acfWindowShort, + fMin((LONG)sizeof(acfWindowShort), + (LONG)sizeof(tC->acfWindow[HIFILT]))); + FDKmemcpy(tC->acfWindow[LOFILT], acfWindowShort, + fMin((LONG)sizeof(acfWindowShort), + (LONG)sizeof(tC->acfWindow[HIFILT]))); + } else { + FDKmemcpy(tC->acfWindow[HIFILT], acfWindowLong, + fMin((LONG)sizeof(acfWindowLong), + (LONG)sizeof(tC->acfWindow[HIFILT]))); + FDKmemcpy(tC->acfWindow[LOFILT], acfWindowLong, + fMin((LONG)sizeof(acfWindowLong), + (LONG)sizeof(tC->acfWindow[HIFILT]))); + } + break; + case 480: + case 512: { + const TNS_PARAMETER_TABULATED *pCfg = + FDKaacEnc_GetTnsParam(bitRate, channels, ldSbrPresent); + if (pCfg != NULL) { + FDKmemcpy(&(tC->confTab), pCfg, sizeof(tC->confTab)); + + tC->lpcStartBand[HIFILT] = FDKaacEnc_FreqToBandWidthRounding( + pCfg->filterStartFreq[HIFILT], sampleRate, pC->sfbCnt, + pC->sfbOffset); + tC->lpcStartLine[HIFILT] = pC->sfbOffset[tC->lpcStartBand[HIFILT]]; + tC->lpcStartBand[LOFILT] = FDKaacEnc_FreqToBandWidthRounding( + pCfg->filterStartFreq[LOFILT], sampleRate, pC->sfbCnt, + pC->sfbOffset); + tC->lpcStartLine[LOFILT] = pC->sfbOffset[tC->lpcStartBand[LOFILT]]; + + FDKaacEnc_CalcGaussWindow( + tC->acfWindow[HIFILT], tC->maxOrder + 1, sampleRate, granuleLength, + pCfg->tnsTimeResolution[HIFILT], TNS_TIMERES_SCALE); + FDKaacEnc_CalcGaussWindow( + tC->acfWindow[LOFILT], tC->maxOrder + 1, sampleRate, granuleLength, + pCfg->tnsTimeResolution[LOFILT], TNS_TIMERES_SCALE); + } else { + tC->tnsActive = + FALSE; /* no configuration available, disable tns tool */ + } + } break; + default: + tC->tnsActive = FALSE; /* no configuration available, disable tns tool */ + } + + return AAC_ENC_OK; +} + +/***************************************************************************/ +/*! + \brief FDKaacEnc_ScaleUpSpectrum + + Scales up spectrum lines in a given frequency section + + \param scaled spectrum + \param original spectrum + \param frequency line to start scaling + \param frequency line to enc scaling + + \return scale factor + +****************************************************************************/ +static inline INT FDKaacEnc_ScaleUpSpectrum(FIXP_DBL *dest, const FIXP_DBL *src, + const INT startLine, + const INT stopLine) { + INT i, scale; + + FIXP_DBL maxVal = FL2FXCONST_DBL(0.f); + + /* Get highest value in given spectrum */ + for (i = startLine; i < stopLine; i++) { + maxVal = fixMax(maxVal, fixp_abs(src[i])); + } + scale = CountLeadingBits(maxVal); + + /* Scale spectrum according to highest value */ + for (i = startLine; i < stopLine; i++) { + dest[i] = src[i] << scale; + } + + return scale; +} + +/***************************************************************************/ +/*! + \brief FDKaacEnc_CalcAutoCorrValue + + Calculate autocorellation value for one lag + + \param pointer to spectrum + \param start line + \param stop line + \param lag to be calculated + \param scaling of the lag + +****************************************************************************/ +static inline FIXP_DBL FDKaacEnc_CalcAutoCorrValue(const FIXP_DBL *spectrum, + const INT startLine, + const INT stopLine, + const INT lag, + const INT scale) { + int i; + FIXP_DBL result = FL2FXCONST_DBL(0.f); + + /* This versions allows to save memory accesses, when computing pow2 */ + /* It is of interest for ARM, XTENSA without parallel memory access */ + if (lag == 0) { + for (i = startLine; i < stopLine; i++) { + result += (fPow2(spectrum[i]) >> scale); + } + } else { + for (i = startLine; i < (stopLine - lag); i++) { + result += (fMult(spectrum[i], spectrum[i + lag]) >> scale); + } + } + + return result; +} + +/***************************************************************************/ +/*! + \brief FDKaacEnc_AutoCorrNormFac + + Autocorrelation function for 1st and 2nd half of the spectrum + + \param pointer to spectrum + \param pointer to autocorrelation window + \param filter start line + +****************************************************************************/ +static inline FIXP_DBL FDKaacEnc_AutoCorrNormFac(const FIXP_DBL value, + const INT scale, INT *sc) { +#define HLM_MIN_NRG 0.0000000037252902984619140625f /* 2^-28 */ +#define MAX_INV_NRGFAC (1.f / HLM_MIN_NRG) + + FIXP_DBL retValue; + FIXP_DBL A, B; + + if (scale >= 0) { + A = value; + B = FL2FXCONST_DBL(HLM_MIN_NRG) >> fixMin(DFRACT_BITS - 1, scale); + } else { + A = value >> fixMin(DFRACT_BITS - 1, (-scale)); + B = FL2FXCONST_DBL(HLM_MIN_NRG); + } + + if (A > B) { + int shift = 0; + FIXP_DBL tmp = invSqrtNorm2(value, &shift); + + retValue = fMult(tmp, tmp); + *sc += (2 * shift); + } else { + /* MAX_INV_NRGFAC*FDKpow(2,-28) = 1/2^-28 * 2^-28 = 1.0 */ + retValue = + /*FL2FXCONST_DBL(MAX_INV_NRGFAC*FDKpow(2,-28))*/ (FIXP_DBL)MAXVAL_DBL; + *sc += scale + 28; + } + + return retValue; +} + +static void FDKaacEnc_MergedAutoCorrelation( + const FIXP_DBL *spectrum, const INT isLowDelay, + const FIXP_DBL acfWindow[MAX_NUM_OF_FILTERS][TNS_MAX_ORDER + 3 + 1], + const INT lpcStartLine[MAX_NUM_OF_FILTERS], const INT lpcStopLine, + const INT maxOrder, const INT acfSplit[MAX_NUM_OF_FILTERS], FIXP_DBL *_rxx1, + FIXP_DBL *_rxx2) { + int i, idx0, idx1, idx2, idx3, idx4, lag; + FIXP_DBL rxx1_0, rxx2_0, rxx3_0, rxx4_0; + + /* buffer for temporal spectrum */ + C_ALLOC_SCRATCH_START(pSpectrum, FIXP_DBL, (1024)) + + /* MDCT line indices separating the 1st, 2nd, 3rd, and 4th analysis quarters + */ + if ((acfSplit[LOFILT] == -1) || (acfSplit[HIFILT] == -1)) { + /* autocorrelation function for 1st, 2nd, 3rd, and 4th quarter of the + * spectrum */ + idx0 = lpcStartLine[LOFILT]; + i = lpcStopLine - lpcStartLine[LOFILT]; + idx1 = idx0 + i / 4; + idx2 = idx0 + i / 2; + idx3 = idx0 + i * 3 / 4; + idx4 = lpcStopLine; + } else { + FDK_ASSERT(acfSplit[LOFILT] == 1); + FDK_ASSERT(acfSplit[HIFILT] == 3); + i = (lpcStopLine - lpcStartLine[HIFILT]) / 3; + idx0 = lpcStartLine[LOFILT]; + idx1 = lpcStartLine[HIFILT]; + idx2 = idx1 + i; + idx3 = idx2 + i; + idx4 = lpcStopLine; + } + + /* copy spectrum to temporal buffer and scale up as much as possible */ + INT sc1 = FDKaacEnc_ScaleUpSpectrum(pSpectrum, spectrum, idx0, idx1); + INT sc2 = FDKaacEnc_ScaleUpSpectrum(pSpectrum, spectrum, idx1, idx2); + INT sc3 = FDKaacEnc_ScaleUpSpectrum(pSpectrum, spectrum, idx2, idx3); + INT sc4 = FDKaacEnc_ScaleUpSpectrum(pSpectrum, spectrum, idx3, idx4); + + /* get scaling values for summation */ + INT nsc1, nsc2, nsc3, nsc4; + for (nsc1 = 1; (1 << nsc1) < (idx1 - idx0); nsc1++) + ; + for (nsc2 = 1; (1 << nsc2) < (idx2 - idx1); nsc2++) + ; + for (nsc3 = 1; (1 << nsc3) < (idx3 - idx2); nsc3++) + ; + for (nsc4 = 1; (1 << nsc4) < (idx4 - idx3); nsc4++) + ; + + /* compute autocorrelation value at lag zero, i. e. energy, for each quarter + */ + rxx1_0 = FDKaacEnc_CalcAutoCorrValue(pSpectrum, idx0, idx1, 0, nsc1); + rxx2_0 = FDKaacEnc_CalcAutoCorrValue(pSpectrum, idx1, idx2, 0, nsc2); + rxx3_0 = FDKaacEnc_CalcAutoCorrValue(pSpectrum, idx2, idx3, 0, nsc3); + rxx4_0 = FDKaacEnc_CalcAutoCorrValue(pSpectrum, idx3, idx4, 0, nsc4); + + /* compute energy normalization factors, i. e. 1/energy (saves some divisions) + */ + if (rxx1_0 != FL2FXCONST_DBL(0.f)) { + INT sc_fac1 = -1; + FIXP_DBL fac1 = + FDKaacEnc_AutoCorrNormFac(rxx1_0, ((-2 * sc1) + nsc1), &sc_fac1); + _rxx1[0] = scaleValue(fMult(rxx1_0, fac1), sc_fac1); + + if (isLowDelay) { + for (lag = 1; lag <= maxOrder; lag++) { + /* compute energy-normalized and windowed autocorrelation values at this + * lag */ + FIXP_DBL x1 = + FDKaacEnc_CalcAutoCorrValue(pSpectrum, idx0, idx1, lag, nsc1); + _rxx1[lag] = + fMult(scaleValue(fMult(x1, fac1), sc_fac1), acfWindow[LOFILT][lag]); + } + } else { + for (lag = 1; lag <= maxOrder; lag++) { + if ((3 * lag) <= maxOrder + 3) { + FIXP_DBL x1 = + FDKaacEnc_CalcAutoCorrValue(pSpectrum, idx0, idx1, lag, nsc1); + _rxx1[lag] = fMult(scaleValue(fMult(x1, fac1), sc_fac1), + acfWindow[LOFILT][3 * lag]); + } + } + } + } + + /* auto corr over upper 3/4 of spectrum */ + if (!((rxx2_0 == FL2FXCONST_DBL(0.f)) && (rxx3_0 == FL2FXCONST_DBL(0.f)) && + (rxx4_0 == FL2FXCONST_DBL(0.f)))) { + FIXP_DBL fac2, fac3, fac4; + fac2 = fac3 = fac4 = FL2FXCONST_DBL(0.f); + INT sc_fac2, sc_fac3, sc_fac4; + sc_fac2 = sc_fac3 = sc_fac4 = 0; + + if (rxx2_0 != FL2FXCONST_DBL(0.f)) { + fac2 = FDKaacEnc_AutoCorrNormFac(rxx2_0, ((-2 * sc2) + nsc2), &sc_fac2); + sc_fac2 -= 2; + } + if (rxx3_0 != FL2FXCONST_DBL(0.f)) { + fac3 = FDKaacEnc_AutoCorrNormFac(rxx3_0, ((-2 * sc3) + nsc3), &sc_fac3); + sc_fac3 -= 2; + } + if (rxx4_0 != FL2FXCONST_DBL(0.f)) { + fac4 = FDKaacEnc_AutoCorrNormFac(rxx4_0, ((-2 * sc4) + nsc4), &sc_fac4); + sc_fac4 -= 2; + } + + _rxx2[0] = scaleValue(fMult(rxx2_0, fac2), sc_fac2) + + scaleValue(fMult(rxx3_0, fac3), sc_fac3) + + scaleValue(fMult(rxx4_0, fac4), sc_fac4); + + for (lag = 1; lag <= maxOrder; lag++) { + /* merge quarters 2, 3, 4 into one autocorrelation; quarter 1 stays + * separate */ + FIXP_DBL x2 = scaleValue(fMult(FDKaacEnc_CalcAutoCorrValue( + pSpectrum, idx1, idx2, lag, nsc2), + fac2), + sc_fac2) + + scaleValue(fMult(FDKaacEnc_CalcAutoCorrValue( + pSpectrum, idx2, idx3, lag, nsc3), + fac3), + sc_fac3) + + scaleValue(fMult(FDKaacEnc_CalcAutoCorrValue( + pSpectrum, idx3, idx4, lag, nsc4), + fac4), + sc_fac4); + + _rxx2[lag] = fMult(x2, acfWindow[HIFILT][lag]); + } + } + + C_ALLOC_SCRATCH_END(pSpectrum, FIXP_DBL, (1024)) +} + +/***************************************************************************** + functionname: FDKaacEnc_TnsDetect + description: do decision, if TNS shall be used or not + returns: + input: tns data structure (modified), + tns config structure, + scalefactor size and table, + spectrum, + subblock num, blocktype, + sfb-wise energy. + +*****************************************************************************/ +INT FDKaacEnc_TnsDetect(TNS_DATA *tnsData, const TNS_CONFIG *tC, + TNS_INFO *tnsInfo, INT sfbCnt, const FIXP_DBL *spectrum, + INT subBlockNumber, INT blockType) { + /* autocorrelation function for 1st, 2nd, 3rd, and 4th quarter of the + * spectrum. */ + FIXP_DBL rxx1[TNS_MAX_ORDER + 1]; /* higher part */ + FIXP_DBL rxx2[TNS_MAX_ORDER + 1]; /* lower part */ + FIXP_LPC parcor_tmp[TNS_MAX_ORDER]; + + int i; + + FDKmemclear(rxx1, sizeof(rxx1)); + FDKmemclear(rxx2, sizeof(rxx2)); + + TNS_SUBBLOCK_INFO *tsbi = + (blockType == SHORT_WINDOW) + ? &tnsData->dataRaw.Short.subBlockInfo[subBlockNumber] + : &tnsData->dataRaw.Long.subBlockInfo; + + tnsData->filtersMerged = FALSE; + + tsbi->tnsActive[HIFILT] = FALSE; + tsbi->predictionGain[HIFILT] = 1000; + tsbi->tnsActive[LOFILT] = FALSE; + tsbi->predictionGain[LOFILT] = 1000; + + tnsInfo->numOfFilters[subBlockNumber] = 0; + tnsInfo->coefRes[subBlockNumber] = tC->coefRes; + for (i = 0; i < tC->maxOrder; i++) { + tnsInfo->coef[subBlockNumber][HIFILT][i] = + tnsInfo->coef[subBlockNumber][LOFILT][i] = 0; + } + + tnsInfo->length[subBlockNumber][HIFILT] = + tnsInfo->length[subBlockNumber][LOFILT] = 0; + tnsInfo->order[subBlockNumber][HIFILT] = + tnsInfo->order[subBlockNumber][LOFILT] = 0; + + if ((tC->tnsActive) && (tC->maxOrder > 0)) { + int sumSqrCoef; + + FDKaacEnc_MergedAutoCorrelation( + spectrum, tC->isLowDelay, tC->acfWindow, tC->lpcStartLine, + tC->lpcStopLine, tC->maxOrder, tC->confTab.acfSplit, rxx1, rxx2); + + /* compute higher TNS filter coefficients in lattice form (ParCor) with + * LeRoux-Gueguen/Schur algorithm */ + { + FIXP_DBL predictionGain_m; + INT predictionGain_e; + + CLpc_AutoToParcor(rxx2, 0, parcor_tmp, tC->confTab.tnsLimitOrder[HIFILT], + &predictionGain_m, &predictionGain_e); + tsbi->predictionGain[HIFILT] = + (INT)fMultNorm(predictionGain_m, predictionGain_e, 1000, 31, 31); + } + + /* non-linear quantization of TNS lattice coefficients with given resolution + */ + FDKaacEnc_Parcor2Index(parcor_tmp, tnsInfo->coef[subBlockNumber][HIFILT], + tC->confTab.tnsLimitOrder[HIFILT], tC->coefRes); + + /* reduce filter order by truncating trailing zeros, compute sum(abs(coefs)) + */ + for (i = tC->confTab.tnsLimitOrder[HIFILT] - 1; i >= 0; i--) { + if (tnsInfo->coef[subBlockNumber][HIFILT][i] != 0) { + break; + } + } + + tnsInfo->order[subBlockNumber][HIFILT] = i + 1; + + sumSqrCoef = 0; + for (; i >= 0; i--) { + sumSqrCoef += tnsInfo->coef[subBlockNumber][HIFILT][i] * + tnsInfo->coef[subBlockNumber][HIFILT][i]; + } + + tnsInfo->direction[subBlockNumber][HIFILT] = + tC->confTab.tnsFilterDirection[HIFILT]; + tnsInfo->length[subBlockNumber][HIFILT] = sfbCnt - tC->lpcStartBand[HIFILT]; + + /* disable TNS if predictionGain is less than 3dB or sumSqrCoef is too small + */ + if ((tsbi->predictionGain[HIFILT] > tC->confTab.threshOn[HIFILT]) || + (sumSqrCoef > (tC->confTab.tnsLimitOrder[HIFILT] / 2 + 2))) { + tsbi->tnsActive[HIFILT] = TRUE; + tnsInfo->numOfFilters[subBlockNumber]++; + + /* compute second filter for lower quarter; only allowed for long windows! + */ + if ((blockType != SHORT_WINDOW) && (tC->confTab.filterEnabled[LOFILT]) && + (tC->confTab.seperateFiltersAllowed)) { + /* compute second filter for lower frequencies */ + + /* compute TNS filter in lattice (ParCor) form with LeRoux-Gueguen + * algorithm */ + INT predGain; + { + FIXP_DBL predictionGain_m; + INT predictionGain_e; + + CLpc_AutoToParcor(rxx1, 0, parcor_tmp, + tC->confTab.tnsLimitOrder[LOFILT], + &predictionGain_m, &predictionGain_e); + predGain = + (INT)fMultNorm(predictionGain_m, predictionGain_e, 1000, 31, 31); + } + + /* non-linear quantization of TNS lattice coefficients with given + * resolution */ + FDKaacEnc_Parcor2Index(parcor_tmp, + tnsInfo->coef[subBlockNumber][LOFILT], + tC->confTab.tnsLimitOrder[LOFILT], tC->coefRes); + + /* reduce filter order by truncating trailing zeros, compute + * sum(abs(coefs)) */ + for (i = tC->confTab.tnsLimitOrder[LOFILT] - 1; i >= 0; i--) { + if (tnsInfo->coef[subBlockNumber][LOFILT][i] != 0) { + break; + } + } + tnsInfo->order[subBlockNumber][LOFILT] = i + 1; + + sumSqrCoef = 0; + for (; i >= 0; i--) { + sumSqrCoef += tnsInfo->coef[subBlockNumber][LOFILT][i] * + tnsInfo->coef[subBlockNumber][LOFILT][i]; + } + + tnsInfo->direction[subBlockNumber][LOFILT] = + tC->confTab.tnsFilterDirection[LOFILT]; + tnsInfo->length[subBlockNumber][LOFILT] = + tC->lpcStartBand[HIFILT] - tC->lpcStartBand[LOFILT]; + + /* filter lower quarter if gain is high enough, but not if it's too high + */ + if (((predGain > tC->confTab.threshOn[LOFILT]) && + (predGain < (16000 * tC->confTab.tnsLimitOrder[LOFILT]))) || + ((sumSqrCoef > 9) && + (sumSqrCoef < 22 * tC->confTab.tnsLimitOrder[LOFILT]))) { + /* compare lower to upper filter; if they are very similar, merge them + */ + tsbi->tnsActive[LOFILT] = TRUE; + sumSqrCoef = 0; + for (i = 0; i < tC->confTab.tnsLimitOrder[LOFILT]; i++) { + sumSqrCoef += fAbs(tnsInfo->coef[subBlockNumber][HIFILT][i] - + tnsInfo->coef[subBlockNumber][LOFILT][i]); + } + if ((sumSqrCoef < 2) && + (tnsInfo->direction[subBlockNumber][LOFILT] == + tnsInfo->direction[subBlockNumber][HIFILT])) { + tnsData->filtersMerged = TRUE; + tnsInfo->length[subBlockNumber][HIFILT] = + sfbCnt - tC->lpcStartBand[LOFILT]; + for (; i < tnsInfo->order[subBlockNumber][HIFILT]; i++) { + if (fAbs(tnsInfo->coef[subBlockNumber][HIFILT][i]) > 1) { + break; + } + } + for (i--; i >= 0; i--) { + if (tnsInfo->coef[subBlockNumber][HIFILT][i] != 0) { + break; + } + } + if (i < tnsInfo->order[subBlockNumber][HIFILT]) { + tnsInfo->order[subBlockNumber][HIFILT] = i + 1; + } + } else { + tnsInfo->numOfFilters[subBlockNumber]++; + } + } /* filter lower part */ + tsbi->predictionGain[LOFILT] = predGain; + + } /* second filter allowed */ + } /* if predictionGain > 1437 ... */ + } /* maxOrder > 0 && tnsActive */ + + return 0; +} + +/***************************************************************************/ +/*! + \brief FDKaacLdEnc_TnsSync + + synchronize TNS parameters when TNS gain difference small (relative) + + \param pointer to TNS data structure (destination) + \param pointer to TNS data structure (source) + \param pointer to TNS config structure + \param number of sub-block + \param block type + + \return void +****************************************************************************/ +void FDKaacEnc_TnsSync(TNS_DATA *tnsDataDest, const TNS_DATA *tnsDataSrc, + TNS_INFO *tnsInfoDest, TNS_INFO *tnsInfoSrc, + const INT blockTypeDest, const INT blockTypeSrc, + const TNS_CONFIG *tC) { + int i, w, absDiff, nWindows; + TNS_SUBBLOCK_INFO *sbInfoDest; + const TNS_SUBBLOCK_INFO *sbInfoSrc; + + /* if one channel contains short blocks and the other not, do not synchronize + */ + if ((blockTypeSrc == SHORT_WINDOW && blockTypeDest != SHORT_WINDOW) || + (blockTypeDest == SHORT_WINDOW && blockTypeSrc != SHORT_WINDOW)) { + return; + } + + if (blockTypeDest != SHORT_WINDOW) { + sbInfoDest = &tnsDataDest->dataRaw.Long.subBlockInfo; + sbInfoSrc = &tnsDataSrc->dataRaw.Long.subBlockInfo; + nWindows = 1; + } else { + sbInfoDest = &tnsDataDest->dataRaw.Short.subBlockInfo[0]; + sbInfoSrc = &tnsDataSrc->dataRaw.Short.subBlockInfo[0]; + nWindows = 8; + } + + for (w = 0; w < nWindows; w++) { + const TNS_SUBBLOCK_INFO *pSbInfoSrcW = sbInfoSrc + w; + TNS_SUBBLOCK_INFO *pSbInfoDestW = sbInfoDest + w; + INT doSync = 1, absDiffSum = 0; + + /* if TNS is active in at least one channel, check if ParCor coefficients of + * higher filter are similar */ + if (pSbInfoDestW->tnsActive[HIFILT] || pSbInfoSrcW->tnsActive[HIFILT]) { + for (i = 0; i < tC->maxOrder; i++) { + absDiff = fAbs(tnsInfoDest->coef[w][HIFILT][i] - + tnsInfoSrc->coef[w][HIFILT][i]); + absDiffSum += absDiff; + /* if coefficients diverge too much between channels, do not synchronize + */ + if ((absDiff > 1) || (absDiffSum > 2)) { + doSync = 0; + break; + } + } + + if (doSync) { + /* if no significant difference was detected, synchronize coefficient + * sets */ + if (pSbInfoSrcW->tnsActive[HIFILT]) { + /* no dest filter, or more dest than source filters: use one dest + * filter */ + if ((!pSbInfoDestW->tnsActive[HIFILT]) || + ((pSbInfoDestW->tnsActive[HIFILT]) && + (tnsInfoDest->numOfFilters[w] > tnsInfoSrc->numOfFilters[w]))) { + pSbInfoDestW->tnsActive[HIFILT] = tnsInfoDest->numOfFilters[w] = 1; + } + tnsDataDest->filtersMerged = tnsDataSrc->filtersMerged; + tnsInfoDest->order[w][HIFILT] = tnsInfoSrc->order[w][HIFILT]; + tnsInfoDest->length[w][HIFILT] = tnsInfoSrc->length[w][HIFILT]; + tnsInfoDest->direction[w][HIFILT] = tnsInfoSrc->direction[w][HIFILT]; + tnsInfoDest->coefCompress[w][HIFILT] = + tnsInfoSrc->coefCompress[w][HIFILT]; + + for (i = 0; i < tC->maxOrder; i++) { + tnsInfoDest->coef[w][HIFILT][i] = tnsInfoSrc->coef[w][HIFILT][i]; + } + } else + pSbInfoDestW->tnsActive[HIFILT] = tnsInfoDest->numOfFilters[w] = 0; + } + } + } +} + +/***************************************************************************/ +/*! + \brief FDKaacEnc_TnsEncode + + perform TNS encoding + + \param pointer to TNS info structure + \param pointer to TNS data structure + \param number of sfbs + \param pointer to TNS config structure + \param low-pass line + \param pointer to spectrum + \param number of sub-block + \param block type + + \return ERROR STATUS +****************************************************************************/ +INT FDKaacEnc_TnsEncode(TNS_INFO *tnsInfo, TNS_DATA *tnsData, + const INT numOfSfb, const TNS_CONFIG *tC, + const INT lowPassLine, FIXP_DBL *spectrum, + const INT subBlockNumber, const INT blockType) { + INT i, startLine, stopLine; + + if (((blockType == SHORT_WINDOW) && + (!tnsData->dataRaw.Short.subBlockInfo[subBlockNumber] + .tnsActive[HIFILT])) || + ((blockType != SHORT_WINDOW) && + (!tnsData->dataRaw.Long.subBlockInfo.tnsActive[HIFILT]))) { + return 1; + } + + startLine = (tnsData->filtersMerged) ? tC->lpcStartLine[LOFILT] + : tC->lpcStartLine[HIFILT]; + stopLine = tC->lpcStopLine; + + for (i = 0; i < tnsInfo->numOfFilters[subBlockNumber]; i++) { + INT lpcGainFactor; + FIXP_LPC LpcCoeff[TNS_MAX_ORDER]; + FIXP_DBL workBuffer[TNS_MAX_ORDER]; + FIXP_LPC parcor_tmp[TNS_MAX_ORDER]; + + FDKaacEnc_Index2Parcor(tnsInfo->coef[subBlockNumber][i], parcor_tmp, + tnsInfo->order[subBlockNumber][i], tC->coefRes); + + lpcGainFactor = CLpc_ParcorToLpc( + parcor_tmp, LpcCoeff, tnsInfo->order[subBlockNumber][i], workBuffer); + + FDKmemclear(workBuffer, TNS_MAX_ORDER * sizeof(FIXP_DBL)); + CLpc_Analysis(&spectrum[startLine], stopLine - startLine, LpcCoeff, + lpcGainFactor, tnsInfo->order[subBlockNumber][i], workBuffer, + NULL); + + /* update for second filter */ + startLine = tC->lpcStartLine[LOFILT]; + stopLine = tC->lpcStartLine[HIFILT]; + } + + return (0); +} + +static void FDKaacEnc_CalcGaussWindow(FIXP_DBL *win, const int winSize, + const INT samplingRate, + const INT transformResolution, + const FIXP_DBL timeResolution, + const INT timeResolution_e) { +#define PI_E (2) +#define PI_M FL2FXCONST_DBL(3.1416f / (float)(1 << PI_E)) + +#define EULER_E (2) +#define EULER_M FL2FXCONST_DBL(2.7183 / (float)(1 << EULER_E)) + +#define COEFF_LOOP_SCALE (4) + + INT i, e1, e2, gaussExp_e; + FIXP_DBL gaussExp_m; + + /* calc. window exponent from time resolution: + * + * gaussExp = PI * samplingRate * 0.001f * timeResolution / + * transformResolution; gaussExp = -0.5f * gaussExp * gaussExp; + */ + gaussExp_m = fMultNorm( + timeResolution, + fMult(PI_M, + fDivNorm((FIXP_DBL)(samplingRate), + (FIXP_DBL)(LONG)(transformResolution * 1000.f), &e1)), + &e2); + gaussExp_m = -fPow2Div2(gaussExp_m); + gaussExp_e = 2 * (e1 + e2 + timeResolution_e + PI_E); + + FDK_ASSERT(winSize < (1 << COEFF_LOOP_SCALE)); + + /* calc. window coefficients + * win[i] = (float)exp( gaussExp * (i+0.5) * (i+0.5) ); + */ + for (i = 0; i < winSize; i++) { + win[i] = fPow( + EULER_M, EULER_E, + fMult(gaussExp_m, + fPow2((i * FL2FXCONST_DBL(1.f / (float)(1 << COEFF_LOOP_SCALE)) + + FL2FXCONST_DBL(.5f / (float)(1 << COEFF_LOOP_SCALE))))), + gaussExp_e + 2 * COEFF_LOOP_SCALE, &e1); + + win[i] = scaleValueSaturate(win[i], e1); + } +} + +static INT FDKaacEnc_Search3(FIXP_LPC parcor) { + INT i, index = 0; + + for (i = 0; i < 8; i++) { + if (parcor > FDKaacEnc_tnsCoeff3Borders[i]) index = i; + } + return (index - 4); +} + +static INT FDKaacEnc_Search4(FIXP_LPC parcor) { + INT i, index = 0; + + for (i = 0; i < 16; i++) { + if (parcor > FDKaacEnc_tnsCoeff4Borders[i]) index = i; + } + return (index - 8); +} + +/***************************************************************************** + + functionname: FDKaacEnc_Parcor2Index + +*****************************************************************************/ +static void FDKaacEnc_Parcor2Index(const FIXP_LPC *parcor, INT *RESTRICT index, + const INT order, const INT bitsPerCoeff) { + INT i; + for (i = 0; i < order; i++) { + if (bitsPerCoeff == 3) + index[i] = FDKaacEnc_Search3(parcor[i]); + else + index[i] = FDKaacEnc_Search4(parcor[i]); + } +} + +/***************************************************************************** + + functionname: FDKaacEnc_Index2Parcor + description: inverse quantization for reflection coefficients + returns: - + input: quantized values, ptr. to reflection coefficients, + no. of coefficients, resolution + output: reflection coefficients + +*****************************************************************************/ +static void FDKaacEnc_Index2Parcor(const INT *index, FIXP_LPC *RESTRICT parcor, + const INT order, const INT bitsPerCoeff) { + INT i; + for (i = 0; i < order; i++) + parcor[i] = bitsPerCoeff == 4 ? FDKaacEnc_tnsEncCoeff4[index[i] + 8] + : FDKaacEnc_tnsEncCoeff3[index[i] + 4]; +} diff --git a/fdk-aac/libAACenc/src/aacenc_tns.h b/fdk-aac/libAACenc/src/aacenc_tns.h new file mode 100644 index 0000000..a37f978 --- /dev/null +++ b/fdk-aac/libAACenc/src/aacenc_tns.h @@ -0,0 +1,213 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): Alex Groeschel + + Description: Temporal noise shaping + +*******************************************************************************/ + +#ifndef AACENC_TNS_H +#define AACENC_TNS_H + +#include "common_fix.h" + +#include "psy_const.h" + +#ifndef PI +#define PI 3.1415926535897931f +#endif + +/** + * TNS_ENABLE_MASK + * This bitfield defines which TNS features are enabled + * The TNS mask is composed of 4 bits. + * tnsMask |= 0x1; activate TNS short blocks + * tnsMask |= 0x2; activate TNS for long blocks + * tnsMask |= 0x4; activate TNS PEAK tool for short blocks + * tnsMask |= 0x8; activate TNS PEAK tool for long blocks + */ +#define TNS_ENABLE_MASK 0xf + +/* TNS max filter order for Low Complexity MPEG4 profile */ +#define TNS_MAX_ORDER 12 + +#define MAX_NUM_OF_FILTERS 2 + +#define HIFILT 0 /* index of higher filter */ +#define LOFILT 1 /* index of lower filter */ + +typedef struct { /* stuff that is tabulated dependent on bitrate etc. */ + INT filterEnabled[MAX_NUM_OF_FILTERS]; + INT threshOn[MAX_NUM_OF_FILTERS]; /* min. prediction gain for using tns + TABUL*/ + INT filterStartFreq[MAX_NUM_OF_FILTERS]; /* lowest freq for lpc TABUL*/ + INT tnsLimitOrder[MAX_NUM_OF_FILTERS]; /* Limit for TNS order TABUL*/ + INT tnsFilterDirection[MAX_NUM_OF_FILTERS]; /* Filtering direction, 0=up, + 1=down TABUL */ + INT acfSplit[MAX_NUM_OF_FILTERS]; + FIXP_DBL tnsTimeResolution[MAX_NUM_OF_FILTERS]; /* TNS max. time resolution + TABUL. Should be fract but + MSVC won't compile then */ + INT seperateFiltersAllowed; +} TNS_PARAMETER_TABULATED; + +typedef struct { /*assigned at InitTime*/ + TNS_PARAMETER_TABULATED confTab; + INT isLowDelay; + INT tnsActive; + INT maxOrder; /* max. order of tns filter */ + INT coefRes; + FIXP_DBL acfWindow[MAX_NUM_OF_FILTERS][TNS_MAX_ORDER + 3 + 1]; + /* now some things that only probably can be done at Init time; + could be they have to be split up for each individual (short) window or + even filter. */ + INT lpcStartBand[MAX_NUM_OF_FILTERS]; + INT lpcStartLine[MAX_NUM_OF_FILTERS]; + INT lpcStopBand; + INT lpcStopLine; + +} TNS_CONFIG; + +typedef struct { + INT tnsActive[MAX_NUM_OF_FILTERS]; + INT predictionGain[MAX_NUM_OF_FILTERS]; +} TNS_SUBBLOCK_INFO; + +typedef struct { /*changed at runTime*/ + TNS_SUBBLOCK_INFO subBlockInfo[TRANS_FAC]; + FIXP_DBL ratioMultTable[TRANS_FAC][MAX_SFB_SHORT]; +} TNS_DATA_SHORT; + +typedef struct { /*changed at runTime*/ + TNS_SUBBLOCK_INFO subBlockInfo; + FIXP_DBL ratioMultTable[MAX_SFB_LONG]; +} TNS_DATA_LONG; + +/* can be implemented as union */ +typedef shouldBeUnion { + TNS_DATA_LONG Long; + TNS_DATA_SHORT Short; +} +TNS_DATA_RAW; + +typedef struct { + INT numOfSubblocks; + TNS_DATA_RAW dataRaw; + INT tnsMaxScaleSpec; + INT filtersMerged; +} TNS_DATA; + +typedef struct { + INT numOfFilters[TRANS_FAC]; + INT coefRes[TRANS_FAC]; + INT length[TRANS_FAC][MAX_NUM_OF_FILTERS]; + INT order[TRANS_FAC][MAX_NUM_OF_FILTERS]; + INT direction[TRANS_FAC][MAX_NUM_OF_FILTERS]; + INT coefCompress[TRANS_FAC][MAX_NUM_OF_FILTERS]; + /* for Long: length TNS_MAX_ORDER (12 for LC) is required -> 12 */ + /* for Short: length TRANS_FAC*TNS_MAX_ORDER (only 5 for short LC) is required + * -> 8*5=40 */ + /* Currently TRANS_FAC*TNS_MAX_ORDER = 8*12 = 96 (for LC) is used (per + * channel)! Memory could be saved here! */ + INT coef[TRANS_FAC][MAX_NUM_OF_FILTERS][TNS_MAX_ORDER]; +} TNS_INFO; + +INT FDKaacEnc_FreqToBandWidthRounding(const INT freq, const INT fs, + const INT numOfBands, + const INT *bandStartOffset); + +#endif /* AACENC_TNS_H */ diff --git a/fdk-aac/libAACenc/src/adj_thr.cpp b/fdk-aac/libAACenc/src/adj_thr.cpp new file mode 100644 index 0000000..6e19680 --- /dev/null +++ b/fdk-aac/libAACenc/src/adj_thr.cpp @@ -0,0 +1,2924 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M. Werner + + Description: Threshold compensation + +*******************************************************************************/ + +#include "adj_thr.h" +#include "sf_estim.h" +#include "aacEnc_ram.h" + +#define NUM_NRG_LEVS (8) +#define INV_INT_TAB_SIZE (8) +static const FIXP_DBL invInt[INV_INT_TAB_SIZE] = { + 0x7fffffff, 0x7fffffff, 0x40000000, 0x2aaaaaaa, + 0x20000000, 0x19999999, 0x15555555, 0x12492492}; + +#define INV_SQRT4_TAB_SIZE (8) +static const FIXP_DBL invSqrt4[INV_SQRT4_TAB_SIZE] = { + 0x7fffffff, 0x7fffffff, 0x6ba27e65, 0x61424bb5, + 0x5a827999, 0x55994845, 0x51c8e33c, 0x4eb160d1}; + +/*static const INT invRedExp = 4;*/ +static const FIXP_DBL SnrLdMin1 = + (FIXP_DBL)0xfcad0ddf; /*FL2FXCONST_DBL(FDKlog(0.316)/FDKlog(2.0)/LD_DATA_SCALING);*/ +static const FIXP_DBL SnrLdMin2 = + (FIXP_DBL)0x0351e1a2; /*FL2FXCONST_DBL(FDKlog(3.16) + /FDKlog(2.0)/LD_DATA_SCALING);*/ +static const FIXP_DBL SnrLdFac = + (FIXP_DBL)0xff5b2c3e; /*FL2FXCONST_DBL(FDKlog(0.8) + /FDKlog(2.0)/LD_DATA_SCALING);*/ + +static const FIXP_DBL SnrLdMin3 = + (FIXP_DBL)0xfe000000; /*FL2FXCONST_DBL(FDKlog(0.5) + /FDKlog(2.0)/LD_DATA_SCALING);*/ +static const FIXP_DBL SnrLdMin4 = + (FIXP_DBL)0x02000000; /*FL2FXCONST_DBL(FDKlog(2.0) + /FDKlog(2.0)/LD_DATA_SCALING);*/ +static const FIXP_DBL SnrLdMin5 = + (FIXP_DBL)0xfc000000; /*FL2FXCONST_DBL(FDKlog(0.25) + /FDKlog(2.0)/LD_DATA_SCALING);*/ + +/* +The bits2Pe factors are choosen for the case that some times +the crash recovery strategy will be activated once. +*/ +#define AFTERBURNER_STATI 2 +#define MAX_ALLOWED_EL_CHANNELS 2 + +typedef struct { + INT bitrate; + FIXP_DBL bits2PeFactor[AFTERBURNER_STATI][MAX_ALLOWED_EL_CHANNELS]; +} BIT_PE_SFAC; + +typedef struct { + INT sampleRate; + const BIT_PE_SFAC *pPeTab; + INT nEntries; + +} BITS2PE_CFG_TAB; + +#define FL2B2PE(value) FL2FXCONST_DBL((value) / (1 << 2)) + +static const BIT_PE_SFAC S_Bits2PeTab16000[] = { + /* bitrate| afterburner off | afterburner on | | nCh=1 + | nCh=2 | nCh=1 | nCh=2 */ + {10000, + {{FL2B2PE(1.60f), FL2B2PE(0.00f)}, {FL2B2PE(1.40f), FL2B2PE(0.00f)}}}, + {24000, + {{FL2B2PE(1.80f), FL2B2PE(1.40f)}, {FL2B2PE(1.60f), FL2B2PE(1.20f)}}}, + {32000, + {{FL2B2PE(1.80f), FL2B2PE(1.60f)}, {FL2B2PE(1.60f), FL2B2PE(1.40f)}}}, + {48000, + {{FL2B2PE(1.60f), FL2B2PE(1.80f)}, {FL2B2PE(1.60f), FL2B2PE(1.60f)}}}, + {64000, + {{FL2B2PE(1.20f), FL2B2PE(1.60f)}, {FL2B2PE(1.20f), FL2B2PE(1.60f)}}}, + {96000, + {{FL2B2PE(1.40f), FL2B2PE(1.80f)}, {FL2B2PE(1.40f), FL2B2PE(1.60f)}}}, + {128000, + {{FL2B2PE(1.40f), FL2B2PE(1.80f)}, {FL2B2PE(1.40f), FL2B2PE(1.80f)}}}, + {148000, + {{FL2B2PE(1.40f), FL2B2PE(1.80f)}, {FL2B2PE(1.40f), FL2B2PE(1.40f)}}}}; + +static const BIT_PE_SFAC S_Bits2PeTab22050[] = { + /* bitrate| afterburner off | afterburner on | | nCh=1 + | nCh=2 | nCh=1 | nCh=2 */ + {16000, + {{FL2B2PE(1.60f), FL2B2PE(1.40f)}, {FL2B2PE(1.20f), FL2B2PE(0.80f)}}}, + {24000, + {{FL2B2PE(1.60f), FL2B2PE(1.40f)}, {FL2B2PE(1.40f), FL2B2PE(1.00f)}}}, + {32000, + {{FL2B2PE(1.40f), FL2B2PE(1.40f)}, {FL2B2PE(1.40f), FL2B2PE(1.20f)}}}, + {48000, + {{FL2B2PE(1.20f), FL2B2PE(1.60f)}, {FL2B2PE(1.20f), FL2B2PE(1.40f)}}}, + {64000, + {{FL2B2PE(1.60f), FL2B2PE(1.60f)}, {FL2B2PE(1.60f), FL2B2PE(1.40f)}}}, + {96000, + {{FL2B2PE(1.80f), FL2B2PE(1.60f)}, {FL2B2PE(1.80f), FL2B2PE(1.60f)}}}, + {128000, + {{FL2B2PE(1.80f), FL2B2PE(1.80f)}, {FL2B2PE(1.60f), FL2B2PE(1.60f)}}}, + {148000, + {{FL2B2PE(1.40f), FL2B2PE(1.80f)}, {FL2B2PE(1.40f), FL2B2PE(1.60f)}}}}; + +static const BIT_PE_SFAC S_Bits2PeTab24000[] = { + /* bitrate| afterburner off | afterburner on | | nCh=1 + | nCh=2 | nCh=1 | nCh=2 */ + {16000, + {{FL2B2PE(1.40f), FL2B2PE(1.40f)}, {FL2B2PE(1.20f), FL2B2PE(0.80f)}}}, + {24000, + {{FL2B2PE(1.60f), FL2B2PE(1.20f)}, {FL2B2PE(1.40f), FL2B2PE(1.00f)}}}, + {32000, + {{FL2B2PE(1.40f), FL2B2PE(1.20f)}, {FL2B2PE(1.40f), FL2B2PE(0.80f)}}}, + {48000, + {{FL2B2PE(1.40f), FL2B2PE(1.60f)}, {FL2B2PE(1.40f), FL2B2PE(1.40f)}}}, + {64000, + {{FL2B2PE(1.60f), FL2B2PE(1.60f)}, {FL2B2PE(1.60f), FL2B2PE(1.40f)}}}, + {96000, + {{FL2B2PE(1.80f), FL2B2PE(1.60f)}, {FL2B2PE(1.80f), FL2B2PE(1.60f)}}}, + {128000, + {{FL2B2PE(1.40f), FL2B2PE(1.60f)}, {FL2B2PE(1.80f), FL2B2PE(1.80f)}}}, + {148000, + {{FL2B2PE(1.40f), FL2B2PE(1.60f)}, {FL2B2PE(1.40f), FL2B2PE(1.80f)}}}}; + +static const BIT_PE_SFAC S_Bits2PeTab32000[] = { + /* bitrate| afterburner off | afterburner on | | nCh=1 + | nCh=2 | nCh=1 | nCh=2 */ + {16000, + {{FL2B2PE(1.20f), FL2B2PE(1.40f)}, {FL2B2PE(0.80f), FL2B2PE(0.80f)}}}, + {24000, + {{FL2B2PE(1.40f), FL2B2PE(1.20f)}, {FL2B2PE(1.00f), FL2B2PE(0.60f)}}}, + {32000, + {{FL2B2PE(1.20f), FL2B2PE(1.20f)}, {FL2B2PE(1.00f), FL2B2PE(0.80f)}}}, + {48000, + {{FL2B2PE(1.40f), FL2B2PE(1.40f)}, {FL2B2PE(1.20f), FL2B2PE(1.20f)}}}, + {64000, + {{FL2B2PE(1.60f), FL2B2PE(1.40f)}, {FL2B2PE(1.60f), FL2B2PE(1.20f)}}}, + {96000, + {{FL2B2PE(1.60f), FL2B2PE(1.40f)}, {FL2B2PE(1.60f), FL2B2PE(1.40f)}}}, + {128000, + {{FL2B2PE(1.80f), FL2B2PE(1.60f)}, {FL2B2PE(1.80f), FL2B2PE(1.60f)}}}, + {148000, + {{FL2B2PE(1.80f), FL2B2PE(1.60f)}, {FL2B2PE(1.80f), FL2B2PE(1.60f)}}}, + {160000, + {{FL2B2PE(1.80f), FL2B2PE(1.60f)}, {FL2B2PE(1.80f), FL2B2PE(1.60f)}}}, + {200000, + {{FL2B2PE(1.40f), FL2B2PE(1.60f)}, {FL2B2PE(1.40f), FL2B2PE(1.60f)}}}, + {320000, + {{FL2B2PE(3.20f), FL2B2PE(1.80f)}, {FL2B2PE(3.20f), FL2B2PE(1.80f)}}}}; + +static const BIT_PE_SFAC S_Bits2PeTab44100[] = { + /* bitrate| afterburner off | afterburner on | | nCh=1 + | nCh=2 | nCh=1 | nCh=2 */ + {16000, + {{FL2B2PE(1.20f), FL2B2PE(1.60f)}, {FL2B2PE(0.80f), FL2B2PE(1.00f)}}}, + {24000, + {{FL2B2PE(1.00f), FL2B2PE(1.20f)}, {FL2B2PE(1.00f), FL2B2PE(0.80f)}}}, + {32000, + {{FL2B2PE(1.20f), FL2B2PE(1.20f)}, {FL2B2PE(0.80f), FL2B2PE(0.60f)}}}, + {48000, + {{FL2B2PE(1.20f), FL2B2PE(1.20f)}, {FL2B2PE(1.20f), FL2B2PE(0.80f)}}}, + {64000, + {{FL2B2PE(1.40f), FL2B2PE(1.20f)}, {FL2B2PE(1.20f), FL2B2PE(1.00f)}}}, + {96000, + {{FL2B2PE(1.60f), FL2B2PE(1.20f)}, {FL2B2PE(1.60f), FL2B2PE(1.20f)}}}, + {128000, + {{FL2B2PE(1.60f), FL2B2PE(1.60f)}, {FL2B2PE(1.60f), FL2B2PE(1.40f)}}}, + {148000, + {{FL2B2PE(1.60f), FL2B2PE(1.60f)}, {FL2B2PE(1.60f), FL2B2PE(1.60f)}}}, + {160000, + {{FL2B2PE(1.60f), FL2B2PE(1.60f)}, {FL2B2PE(1.60f), FL2B2PE(1.60f)}}}, + {200000, + {{FL2B2PE(1.80f), FL2B2PE(1.60f)}, {FL2B2PE(1.60f), FL2B2PE(1.60f)}}}, + {320000, + {{FL2B2PE(3.20f), FL2B2PE(1.60f)}, {FL2B2PE(3.20f), FL2B2PE(1.60f)}}}}; + +static const BIT_PE_SFAC S_Bits2PeTab48000[] = { + /* bitrate| afterburner off | afterburner on | | nCh=1 + | nCh=2 | nCh=1 | nCh=2 */ + {16000, + {{FL2B2PE(1.40f), FL2B2PE(0.00f)}, {FL2B2PE(0.80f), FL2B2PE(0.00f)}}}, + {24000, + {{FL2B2PE(1.40f), FL2B2PE(1.20f)}, {FL2B2PE(1.00f), FL2B2PE(0.80f)}}}, + {32000, + {{FL2B2PE(1.00f), FL2B2PE(1.20f)}, {FL2B2PE(0.60f), FL2B2PE(0.80f)}}}, + {48000, + {{FL2B2PE(1.20f), FL2B2PE(1.00f)}, {FL2B2PE(0.80f), FL2B2PE(0.80f)}}}, + {64000, + {{FL2B2PE(1.20f), FL2B2PE(1.20f)}, {FL2B2PE(1.20f), FL2B2PE(1.00f)}}}, + {96000, + {{FL2B2PE(1.60f), FL2B2PE(1.40f)}, {FL2B2PE(1.60f), FL2B2PE(1.20f)}}}, + {128000, + {{FL2B2PE(1.60f), FL2B2PE(1.60f)}, {FL2B2PE(1.60f), FL2B2PE(1.40f)}}}, + {148000, + {{FL2B2PE(1.60f), FL2B2PE(1.60f)}, {FL2B2PE(1.60f), FL2B2PE(1.40f)}}}, + {160000, + {{FL2B2PE(1.60f), FL2B2PE(1.60f)}, {FL2B2PE(1.60f), FL2B2PE(1.40f)}}}, + {200000, + {{FL2B2PE(1.20f), FL2B2PE(1.60f)}, {FL2B2PE(1.60f), FL2B2PE(1.40f)}}}, + {320000, + {{FL2B2PE(3.20f), FL2B2PE(1.60f)}, {FL2B2PE(3.20f), FL2B2PE(1.60f)}}}}; + +static const BITS2PE_CFG_TAB bits2PeConfigTab[] = { + {16000, S_Bits2PeTab16000, sizeof(S_Bits2PeTab16000) / sizeof(BIT_PE_SFAC)}, + {22050, S_Bits2PeTab22050, sizeof(S_Bits2PeTab22050) / sizeof(BIT_PE_SFAC)}, + {24000, S_Bits2PeTab24000, sizeof(S_Bits2PeTab24000) / sizeof(BIT_PE_SFAC)}, + {32000, S_Bits2PeTab32000, sizeof(S_Bits2PeTab32000) / sizeof(BIT_PE_SFAC)}, + {44100, S_Bits2PeTab44100, sizeof(S_Bits2PeTab44100) / sizeof(BIT_PE_SFAC)}, + {48000, S_Bits2PeTab48000, + sizeof(S_Bits2PeTab48000) / sizeof(BIT_PE_SFAC)}}; + +/* values for avoid hole flag */ +enum _avoid_hole_state { NO_AH = 0, AH_INACTIVE = 1, AH_ACTIVE = 2 }; + +/* Q format definitions */ +#define Q_BITFAC \ + (24) /* Q scaling used in FDKaacEnc_bitresCalcBitFac() calculation */ +#define Q_AVGBITS (17) /* scale bit values */ + +/***************************************************************************** + functionname: FDKaacEnc_InitBits2PeFactor + description: retrieve bits2PeFactor from table +*****************************************************************************/ +static void FDKaacEnc_InitBits2PeFactor( + FIXP_DBL *bits2PeFactor_m, INT *bits2PeFactor_e, const INT bitRate, + const INT nChannels, const INT sampleRate, const INT advancedBitsToPe, + const INT dZoneQuantEnable, const INT invQuant) { + /**** 1) Set default bits2pe factor ****/ + FIXP_DBL bit2PE_m = FL2FXCONST_DBL(1.18f / (1 << (1))); + INT bit2PE_e = 1; + + /**** 2) For AAC-(E)LD, make use of advanced bits to pe factor table ****/ + if (advancedBitsToPe && nChannels <= (2)) { + int i; + const BIT_PE_SFAC *peTab = NULL; + INT size = 0; + + /*** 2.1) Get correct table entry ***/ + for (i = 0; i < (INT)(sizeof(bits2PeConfigTab) / sizeof(BITS2PE_CFG_TAB)); + i++) { + if (sampleRate >= bits2PeConfigTab[i].sampleRate) { + peTab = bits2PeConfigTab[i].pPeTab; + size = bits2PeConfigTab[i].nEntries; + } + } + + if ((peTab != NULL) && (size != 0)) { + INT startB = -1; /* bitrate entry in table that is the next-lower to + actual bitrate */ + INT stopB = -1; /* bitrate entry in table that is the next-higher to + actual bitrate */ + FIXP_DBL startPF = + FL2FXCONST_DBL(0.0f); /* bits2PE factor entry in table that is the + next-lower to actual bits2PE factor */ + FIXP_DBL stopPF = FL2FXCONST_DBL(0.0f); /* bits2PE factor entry in table + that is the next-higher to + actual bits2PE factor */ + FIXP_DBL slope = FL2FXCONST_DBL( + 0.0f); /* the slope from the start bits2Pe entry to the next one */ + const int qualityIdx = (invQuant == 0) ? 0 : 1; + + if (bitRate >= peTab[size - 1].bitrate) { + /* Chosen bitrate is higher than the highest bitrate in table. + The slope for extrapolating the bits2PE factor must be zero. + Values are set accordingly. */ + startB = peTab[size - 1].bitrate; + stopB = + bitRate + + 1; /* Can be an arbitrary value greater than startB and bitrate. */ + startPF = peTab[size - 1].bits2PeFactor[qualityIdx][nChannels - 1]; + stopPF = peTab[size - 1].bits2PeFactor[qualityIdx][nChannels - 1]; + } else { + for (i = 0; i < size - 1; i++) { + if ((peTab[i].bitrate <= bitRate) && + (peTab[i + 1].bitrate > bitRate)) { + startB = peTab[i].bitrate; + stopB = peTab[i + 1].bitrate; + startPF = peTab[i].bits2PeFactor[qualityIdx][nChannels - 1]; + stopPF = peTab[i + 1].bits2PeFactor[qualityIdx][nChannels - 1]; + break; + } + } + } + + /*** 2.2) Configuration available? ***/ + if (startB != -1) { + /** 2.2.1) linear interpolate to actual PEfactor **/ + FIXP_DBL bit2PE = 0; + + const FIXP_DBL maxBit2PE = FL2FXCONST_DBL(3.f / 4.f); + + /* bit2PE = ((stopPF-startPF)/(stopB-startB))*(bitRate-startB)+startPF; + */ + slope = fDivNorm(bitRate - startB, stopB - startB); + bit2PE = fMult(slope, stopPF - startPF) + startPF; + + bit2PE = fMin(maxBit2PE, bit2PE); + + /** 2.2.2) sanity check if bits2pe value is high enough **/ + if (bit2PE >= (FL2FXCONST_DBL(0.35f) >> 2)) { + bit2PE_m = bit2PE; + bit2PE_e = 2; /* table is fixed scaled */ + } + } /* br */ + } /* sr */ + } /* advancedBitsToPe */ + + if (dZoneQuantEnable) { + if (bit2PE_m >= (FL2FXCONST_DBL(0.6f)) >> bit2PE_e) { + /* Additional headroom for addition */ + bit2PE_m >>= 1; + bit2PE_e += 1; + } + + /* the quantTendencyCompensator compensates a lower bit consumption due to + * increasing the tendency to quantize low spectral values to the lower + * quantizer border for bitrates below a certain bitrate threshold --> see + * also function calcSfbDistLD in quantize.c */ + if ((bitRate / nChannels > 32000) && (bitRate / nChannels <= 40000)) { + bit2PE_m += (FL2FXCONST_DBL(0.4f)) >> bit2PE_e; + } else if (bitRate / nChannels > 20000) { + bit2PE_m += (FL2FXCONST_DBL(0.3f)) >> bit2PE_e; + } else if (bitRate / nChannels >= 16000) { + bit2PE_m += (FL2FXCONST_DBL(0.3f)) >> bit2PE_e; + } else { + bit2PE_m += (FL2FXCONST_DBL(0.0f)) >> bit2PE_e; + } + } + + /***** 3.) Return bits2pe factor *****/ + *bits2PeFactor_m = bit2PE_m; + *bits2PeFactor_e = bit2PE_e; +} + +/***************************************************************************** +functionname: FDKaacEnc_bits2pe2 +description: convert from bits to pe +*****************************************************************************/ +FDK_INLINE INT FDKaacEnc_bits2pe2(const INT bits, const FIXP_DBL factor_m, + const INT factor_e) { + return (INT)(fMult(factor_m, (FIXP_DBL)(bits << Q_AVGBITS)) >> + (Q_AVGBITS - factor_e)); +} + +/***************************************************************************** +functionname: FDKaacEnc_calcThreshExp +description: loudness calculation (threshold to the power of redExp) +*****************************************************************************/ +static void FDKaacEnc_calcThreshExp( + FIXP_DBL thrExp[(2)][MAX_GROUPED_SFB], + const QC_OUT_CHANNEL *const qcOutChannel[(2)], + const PSY_OUT_CHANNEL *const psyOutChannel[(2)], const INT nChannels) { + INT ch, sfb, sfbGrp; + FIXP_DBL thrExpLdData; + + for (ch = 0; ch < nChannels; ch++) { + for (sfbGrp = 0; sfbGrp < psyOutChannel[ch]->sfbCnt; + sfbGrp += psyOutChannel[ch]->sfbPerGroup) { + for (sfb = 0; sfb < psyOutChannel[ch]->maxSfbPerGroup; sfb++) { + thrExpLdData = psyOutChannel[ch]->sfbThresholdLdData[sfbGrp + sfb] >> 2; + thrExp[ch][sfbGrp + sfb] = CalcInvLdData(thrExpLdData); + } + } + } +} + +/***************************************************************************** + functionname: FDKaacEnc_adaptMinSnr + description: reduce minSnr requirements for bands with relative low +energies +*****************************************************************************/ +static void FDKaacEnc_adaptMinSnr( + QC_OUT_CHANNEL *const qcOutChannel[(2)], + const PSY_OUT_CHANNEL *const psyOutChannel[(2)], + const MINSNR_ADAPT_PARAM *const msaParam, const INT nChannels) { + INT ch, sfb, sfbGrp, nSfb; + FIXP_DBL avgEnLD64, dbRatio, minSnrRed; + FIXP_DBL minSnrLimitLD64 = + FL2FXCONST_DBL(-0.00503012648262f); /* ld64(0.8f) */ + FIXP_DBL nSfbLD64; + FIXP_DBL accu; + + FIXP_DBL msaParam_maxRed = msaParam->maxRed; + FIXP_DBL msaParam_startRatio = msaParam->startRatio; + FIXP_DBL msaParam_redRatioFac = + fMult(msaParam->redRatioFac, FL2FXCONST_DBL(0.3010299956f)); + FIXP_DBL msaParam_redOffs = msaParam->redOffs; + + for (ch = 0; ch < nChannels; ch++) { + /* calc average energy per scalefactor band */ + nSfb = 0; + accu = FL2FXCONST_DBL(0.0f); + + DWORD_ALIGNED(psyOutChannel[ch]->sfbEnergy); + + for (sfbGrp = 0; sfbGrp < psyOutChannel[ch]->sfbCnt; + sfbGrp += psyOutChannel[ch]->sfbPerGroup) { + int maxSfbPerGroup = psyOutChannel[ch]->maxSfbPerGroup; + nSfb += maxSfbPerGroup; + for (sfb = 0; sfb < maxSfbPerGroup; sfb++) { + accu += psyOutChannel[ch]->sfbEnergy[sfbGrp + sfb] >> 6; + } + } + + if ((accu == FL2FXCONST_DBL(0.0f)) || (nSfb == 0)) { + avgEnLD64 = FL2FXCONST_DBL(-1.0f); + } else { + nSfbLD64 = CalcLdInt(nSfb); + avgEnLD64 = CalcLdData(accu); + avgEnLD64 = avgEnLD64 + FL2FXCONST_DBL(0.09375f) - + nSfbLD64; /* 0.09375f: compensate shift with 6 */ + } + + /* reduce minSnr requirement by minSnr^minSnrRed dependent on avgEn/sfbEn */ + int maxSfbPerGroup = psyOutChannel[ch]->maxSfbPerGroup; + int sfbCnt = psyOutChannel[ch]->sfbCnt; + int sfbPerGroup = psyOutChannel[ch]->sfbPerGroup; + + for (sfbGrp = 0; sfbGrp < sfbCnt; sfbGrp += sfbPerGroup) { + FIXP_DBL *RESTRICT psfbEnergyLdData = + &qcOutChannel[ch]->sfbEnergyLdData[sfbGrp]; + FIXP_DBL *RESTRICT psfbMinSnrLdData = + &qcOutChannel[ch]->sfbMinSnrLdData[sfbGrp]; + for (sfb = 0; sfb < maxSfbPerGroup; sfb++) { + FIXP_DBL sfbEnergyLdData = *psfbEnergyLdData++; + FIXP_DBL sfbMinSnrLdData = *psfbMinSnrLdData; + dbRatio = avgEnLD64 - sfbEnergyLdData; + int update = (msaParam_startRatio < dbRatio) ? 1 : 0; + minSnrRed = msaParam_redOffs + fMult(msaParam_redRatioFac, + dbRatio); /* scaled by 1.0f/64.0f*/ + minSnrRed = + fixMax(minSnrRed, msaParam_maxRed); /* scaled by 1.0f/64.0f*/ + minSnrRed = (fMult(sfbMinSnrLdData, minSnrRed)) << 6; + minSnrRed = fixMin(minSnrLimitLD64, minSnrRed); + *psfbMinSnrLdData++ = update ? minSnrRed : sfbMinSnrLdData; + } + } + } +} + +/***************************************************************************** +functionname: FDKaacEnc_initAvoidHoleFlag +description: determine bands where avoid hole is not necessary resp. possible +*****************************************************************************/ +static void FDKaacEnc_initAvoidHoleFlag( + QC_OUT_CHANNEL *const qcOutChannel[(2)], + const PSY_OUT_CHANNEL *const psyOutChannel[(2)], + UCHAR ahFlag[(2)][MAX_GROUPED_SFB], const struct TOOLSINFO *const toolsInfo, + const INT nChannels, const AH_PARAM *const ahParam) { + INT ch, sfb, sfbGrp; + FIXP_DBL sfbEn, sfbEnm1; + FIXP_DBL sfbEnLdData; + FIXP_DBL avgEnLdData; + + /* decrease spread energy by 3dB for long blocks, resp. 2dB for shorts + (avoid more holes in long blocks) */ + for (ch = 0; ch < nChannels; ch++) { + QC_OUT_CHANNEL *const qcOutChan = qcOutChannel[ch]; + + if (psyOutChannel[ch]->lastWindowSequence != SHORT_WINDOW) { + for (sfbGrp = 0; sfbGrp < psyOutChannel[ch]->sfbCnt; + sfbGrp += psyOutChannel[ch]->sfbPerGroup) + for (sfb = 0; sfb < psyOutChannel[ch]->maxSfbPerGroup; sfb++) + qcOutChan->sfbSpreadEnergy[sfbGrp + sfb] >>= 1; + } else { + for (sfbGrp = 0; sfbGrp < psyOutChannel[ch]->sfbCnt; + sfbGrp += psyOutChannel[ch]->sfbPerGroup) + for (sfb = 0; sfb < psyOutChannel[ch]->maxSfbPerGroup; sfb++) + qcOutChan->sfbSpreadEnergy[sfbGrp + sfb] = fMult( + FL2FXCONST_DBL(0.63f), qcOutChan->sfbSpreadEnergy[sfbGrp + sfb]); + } + } + + /* increase minSnr for local peaks, decrease it for valleys */ + if (ahParam->modifyMinSnr) { + for (ch = 0; ch < nChannels; ch++) { + QC_OUT_CHANNEL *const qcOutChan = qcOutChannel[ch]; + for (sfbGrp = 0; sfbGrp < psyOutChannel[ch]->sfbCnt; + sfbGrp += psyOutChannel[ch]->sfbPerGroup) { + for (sfb = 0; sfb < psyOutChannel[ch]->maxSfbPerGroup; sfb++) { + FIXP_DBL sfbEnp1, avgEn; + if (sfb > 0) + sfbEnm1 = qcOutChan->sfbEnergy[sfbGrp + sfb - 1]; + else + sfbEnm1 = qcOutChan->sfbEnergy[sfbGrp + sfb]; + + if (sfb < psyOutChannel[ch]->maxSfbPerGroup - 1) + sfbEnp1 = qcOutChan->sfbEnergy[sfbGrp + sfb + 1]; + else + sfbEnp1 = qcOutChan->sfbEnergy[sfbGrp + sfb]; + + avgEn = (sfbEnm1 >> 1) + (sfbEnp1 >> 1); + avgEnLdData = CalcLdData(avgEn); + sfbEn = qcOutChan->sfbEnergy[sfbGrp + sfb]; + sfbEnLdData = qcOutChan->sfbEnergyLdData[sfbGrp + sfb]; + /* peak ? */ + if (sfbEn > avgEn) { + FIXP_DBL tmpMinSnrLdData; + if (psyOutChannel[ch]->lastWindowSequence == LONG_WINDOW) + tmpMinSnrLdData = + fixMax(SnrLdFac + (FIXP_DBL)(avgEnLdData - sfbEnLdData), + (FIXP_DBL)SnrLdMin1); + else + tmpMinSnrLdData = + fixMax(SnrLdFac + (FIXP_DBL)(avgEnLdData - sfbEnLdData), + (FIXP_DBL)SnrLdMin3); + + qcOutChan->sfbMinSnrLdData[sfbGrp + sfb] = fixMin( + qcOutChan->sfbMinSnrLdData[sfbGrp + sfb], tmpMinSnrLdData); + } + /* valley ? */ + if (((sfbEnLdData + (FIXP_DBL)SnrLdMin4) < (FIXP_DBL)avgEnLdData) && + (sfbEn > FL2FXCONST_DBL(0.0))) { + FIXP_DBL tmpMinSnrLdData = avgEnLdData - sfbEnLdData - + (FIXP_DBL)SnrLdMin4 + + qcOutChan->sfbMinSnrLdData[sfbGrp + sfb]; + tmpMinSnrLdData = fixMin((FIXP_DBL)SnrLdFac, tmpMinSnrLdData); + qcOutChan->sfbMinSnrLdData[sfbGrp + sfb] = + fixMin(tmpMinSnrLdData, + (FIXP_DBL)(qcOutChan->sfbMinSnrLdData[sfbGrp + sfb] + + SnrLdMin2)); + } + } + } + } + } + + /* stereo: adapt the minimum requirements sfbMinSnr of mid and + side channels to avoid spending unnoticable bits */ + if (nChannels == 2) { + QC_OUT_CHANNEL *qcOutChanM = qcOutChannel[0]; + QC_OUT_CHANNEL *qcOutChanS = qcOutChannel[1]; + const PSY_OUT_CHANNEL *const psyOutChanM = psyOutChannel[0]; + for (sfbGrp = 0; sfbGrp < psyOutChanM->sfbCnt; + sfbGrp += psyOutChanM->sfbPerGroup) { + for (sfb = 0; sfb < psyOutChanM->maxSfbPerGroup; sfb++) { + if (toolsInfo->msMask[sfbGrp + sfb]) { + FIXP_DBL maxSfbEnLd = + fixMax(qcOutChanM->sfbEnergyLdData[sfbGrp + sfb], + qcOutChanS->sfbEnergyLdData[sfbGrp + sfb]); + FIXP_DBL maxThrLd, sfbMinSnrTmpLd; + + if (((SnrLdMin5 >> 1) + (maxSfbEnLd >> 1) + + (qcOutChanM->sfbMinSnrLdData[sfbGrp + sfb] >> 1)) <= + FL2FXCONST_DBL(-0.5f)) + maxThrLd = FL2FXCONST_DBL(-1.0f); + else + maxThrLd = SnrLdMin5 + maxSfbEnLd + + qcOutChanM->sfbMinSnrLdData[sfbGrp + sfb]; + + if (qcOutChanM->sfbEnergy[sfbGrp + sfb] > FL2FXCONST_DBL(0.0f)) + sfbMinSnrTmpLd = + maxThrLd - qcOutChanM->sfbEnergyLdData[sfbGrp + sfb]; + else + sfbMinSnrTmpLd = FL2FXCONST_DBL(0.0f); + + qcOutChanM->sfbMinSnrLdData[sfbGrp + sfb] = + fixMax(qcOutChanM->sfbMinSnrLdData[sfbGrp + sfb], sfbMinSnrTmpLd); + + if (qcOutChanM->sfbMinSnrLdData[sfbGrp + sfb] <= FL2FXCONST_DBL(0.0f)) + qcOutChanM->sfbMinSnrLdData[sfbGrp + sfb] = fixMin( + qcOutChanM->sfbMinSnrLdData[sfbGrp + sfb], (FIXP_DBL)SnrLdFac); + + if (qcOutChanS->sfbEnergy[sfbGrp + sfb] > FL2FXCONST_DBL(0.0f)) + sfbMinSnrTmpLd = + maxThrLd - qcOutChanS->sfbEnergyLdData[sfbGrp + sfb]; + else + sfbMinSnrTmpLd = FL2FXCONST_DBL(0.0f); + + qcOutChanS->sfbMinSnrLdData[sfbGrp + sfb] = + fixMax(qcOutChanS->sfbMinSnrLdData[sfbGrp + sfb], sfbMinSnrTmpLd); + + if (qcOutChanS->sfbMinSnrLdData[sfbGrp + sfb] <= FL2FXCONST_DBL(0.0f)) + qcOutChanS->sfbMinSnrLdData[sfbGrp + sfb] = fixMin( + qcOutChanS->sfbMinSnrLdData[sfbGrp + sfb], (FIXP_DBL)SnrLdFac); + + if (qcOutChanM->sfbEnergy[sfbGrp + sfb] > + qcOutChanM->sfbSpreadEnergy[sfbGrp + sfb]) + qcOutChanS->sfbSpreadEnergy[sfbGrp + sfb] = fMult( + qcOutChanS->sfbEnergy[sfbGrp + sfb], FL2FXCONST_DBL(0.9f)); + + if (qcOutChanS->sfbEnergy[sfbGrp + sfb] > + qcOutChanS->sfbSpreadEnergy[sfbGrp + sfb]) + qcOutChanM->sfbSpreadEnergy[sfbGrp + sfb] = fMult( + qcOutChanM->sfbEnergy[sfbGrp + sfb], FL2FXCONST_DBL(0.9f)); + + } /* if (toolsInfo->msMask[sfbGrp+sfb]) */ + } /* sfb */ + } /* sfbGrp */ + } /* nChannels==2 */ + + /* init ahFlag (0: no ah necessary, 1: ah possible, 2: ah active */ + for (ch = 0; ch < nChannels; ch++) { + QC_OUT_CHANNEL *qcOutChan = qcOutChannel[ch]; + const PSY_OUT_CHANNEL *const psyOutChan = psyOutChannel[ch]; + for (sfbGrp = 0; sfbGrp < psyOutChan->sfbCnt; + sfbGrp += psyOutChan->sfbPerGroup) { + for (sfb = 0; sfb < psyOutChan->maxSfbPerGroup; sfb++) { + if ((qcOutChan->sfbSpreadEnergy[sfbGrp + sfb] > + qcOutChan->sfbEnergy[sfbGrp + sfb]) || + (qcOutChan->sfbMinSnrLdData[sfbGrp + sfb] > FL2FXCONST_DBL(0.0f))) { + ahFlag[ch][sfbGrp + sfb] = NO_AH; + } else { + ahFlag[ch][sfbGrp + sfb] = AH_INACTIVE; + } + } + } + } +} + +/** + * \brief Calculate constants that do not change during successive pe + * calculations. + * + * \param peData Pointer to structure containing PE data of + * current element. + * \param psyOutChannel Pointer to PSY_OUT_CHANNEL struct holding + * nChannels elements. + * \param qcOutChannel Pointer to QC_OUT_CHANNEL struct holding + * nChannels elements. + * \param nChannels Number of channels in element. + * \param peOffset Fixed PE offset defined while + * FDKaacEnc_AdjThrInit() depending on bitrate. + * + * \return void + */ +static void FDKaacEnc_preparePe(PE_DATA *const peData, + const PSY_OUT_CHANNEL *const psyOutChannel[(2)], + const QC_OUT_CHANNEL *const qcOutChannel[(2)], + const INT nChannels, const INT peOffset) { + INT ch; + + for (ch = 0; ch < nChannels; ch++) { + const PSY_OUT_CHANNEL *const psyOutChan = psyOutChannel[ch]; + FDKaacEnc_prepareSfbPe( + &peData->peChannelData[ch], psyOutChan->sfbEnergyLdData, + psyOutChan->sfbThresholdLdData, qcOutChannel[ch]->sfbFormFactorLdData, + psyOutChan->sfbOffsets, psyOutChan->sfbCnt, psyOutChan->sfbPerGroup, + psyOutChan->maxSfbPerGroup); + } + peData->offset = peOffset; +} + +/** + * \brief Calculate weighting factor for threshold adjustment. + * + * Calculate weighting factor to be applied at energies and thresholds in ld64 + * format. + * + * \param peData, Pointer to PE data in current element. + * \param psyOutChannel Pointer to PSY_OUT_CHANNEL struct holding + * nChannels elements. + * \param qcOutChannel Pointer to QC_OUT_CHANNEL struct holding + * nChannels elements. + * \param toolsInfo Pointer to tools info struct of current element. + * \param adjThrStateElement Pointer to ATS_ELEMENT holding enFacPatch + * states. + * \param nChannels Number of channels in element. + * \param usePatchTool Apply the weighting tool 0 (no) else (yes). + * + * \return void + */ +static void FDKaacEnc_calcWeighting( + const PE_DATA *const peData, + const PSY_OUT_CHANNEL *const psyOutChannel[(2)], + QC_OUT_CHANNEL *const qcOutChannel[(2)], + const struct TOOLSINFO *const toolsInfo, + ATS_ELEMENT *const adjThrStateElement, const INT nChannels, + const INT usePatchTool) { + int ch, noShortWindowInFrame = TRUE; + INT exePatchM = 0; + + for (ch = 0; ch < nChannels; ch++) { + if (psyOutChannel[ch]->lastWindowSequence == SHORT_WINDOW) { + noShortWindowInFrame = FALSE; + } + FDKmemclear(qcOutChannel[ch]->sfbEnFacLd, + MAX_GROUPED_SFB * sizeof(FIXP_DBL)); + } + + if (usePatchTool == 0) { + return; /* tool is disabled */ + } + + for (ch = 0; ch < nChannels; ch++) { + const PSY_OUT_CHANNEL *const psyOutChan = psyOutChannel[ch]; + + if (noShortWindowInFrame) { /* retain energy ratio between blocks of + different length */ + + FIXP_DBL nrgSum14, nrgSum12, nrgSum34, nrgTotal; + FIXP_DBL nrgFacLd_14, nrgFacLd_12, nrgFacLd_34; + INT usePatch, exePatch; + int sfb, sfbGrp, nLinesSum = 0; + + nrgSum14 = nrgSum12 = nrgSum34 = nrgTotal = FL2FXCONST_DBL(0.f); + + /* calculate flatness of audible spectrum, i.e. spectrum above masking + * threshold. */ + for (sfbGrp = 0; sfbGrp < psyOutChannel[ch]->sfbCnt; + sfbGrp += psyOutChannel[ch]->sfbPerGroup) { + for (sfb = 0; sfb < psyOutChannel[ch]->maxSfbPerGroup; sfb++) { + FIXP_DBL nrgFac12 = CalcInvLdData( + psyOutChan->sfbEnergyLdData[sfbGrp + sfb] >> 1); /* nrg^(1/2) */ + FIXP_DBL nrgFac14 = CalcInvLdData( + psyOutChan->sfbEnergyLdData[sfbGrp + sfb] >> 2); /* nrg^(1/4) */ + + /* maximal number of bands is 64, results scaling factor 6 */ + nLinesSum += peData->peChannelData[ch] + .sfbNLines[sfbGrp + sfb]; /* relevant lines */ + nrgTotal += + (psyOutChan->sfbEnergy[sfbGrp + sfb] >> 6); /* sum up nrg */ + nrgSum12 += (nrgFac12 >> 6); /* sum up nrg^(2/4) */ + nrgSum14 += (nrgFac14 >> 6); /* sum up nrg^(1/4) */ + nrgSum34 += (fMult(nrgFac14, nrgFac12) >> 6); /* sum up nrg^(3/4) */ + } + } + + nrgTotal = CalcLdData(nrgTotal); /* get ld64 of total nrg */ + + nrgFacLd_14 = + CalcLdData(nrgSum14) - nrgTotal; /* ld64(nrgSum14/nrgTotal) */ + nrgFacLd_12 = + CalcLdData(nrgSum12) - nrgTotal; /* ld64(nrgSum12/nrgTotal) */ + nrgFacLd_34 = + CalcLdData(nrgSum34) - nrgTotal; /* ld64(nrgSum34/nrgTotal) */ + + /* Note: nLinesSum cannot be larger than the number of total lines, thats + * taken care of in line_pe.cpp FDKaacEnc_prepareSfbPe() */ + adjThrStateElement->chaosMeasureEnFac[ch] = + fMax(FL2FXCONST_DBL(0.1875f), + fDivNorm(nLinesSum, psyOutChan->sfbOffsets[psyOutChan->sfbCnt])); + + usePatch = (adjThrStateElement->chaosMeasureEnFac[ch] > + FL2FXCONST_DBL(0.78125f)); + exePatch = ((usePatch) && (adjThrStateElement->lastEnFacPatch[ch])); + + for (sfbGrp = 0; sfbGrp < psyOutChannel[ch]->sfbCnt; + sfbGrp += psyOutChannel[ch]->sfbPerGroup) { + for (sfb = 0; sfb < psyOutChannel[ch]->maxSfbPerGroup; sfb++) { + INT sfbExePatch; + /* for MS coupled SFBs, also execute patch in side channel if done in + * mid channel */ + if ((ch == 1) && (toolsInfo->msMask[sfbGrp + sfb])) { + sfbExePatch = exePatchM; + } else { + sfbExePatch = exePatch; + } + + if ((sfbExePatch) && + (psyOutChan->sfbEnergy[sfbGrp + sfb] > FL2FXCONST_DBL(0.f))) { + /* execute patch based on spectral flatness calculated above */ + if (adjThrStateElement->chaosMeasureEnFac[ch] > + FL2FXCONST_DBL(0.8125f)) { + qcOutChannel[ch]->sfbEnFacLd[sfbGrp + sfb] = + ((nrgFacLd_14 + + (psyOutChan->sfbEnergyLdData[sfbGrp + sfb] + + (psyOutChan->sfbEnergyLdData[sfbGrp + sfb] >> 1))) >> + 1); /* sfbEnergy^(3/4) */ + } else if (adjThrStateElement->chaosMeasureEnFac[ch] > + FL2FXCONST_DBL(0.796875f)) { + qcOutChannel[ch]->sfbEnFacLd[sfbGrp + sfb] = + ((nrgFacLd_12 + psyOutChan->sfbEnergyLdData[sfbGrp + sfb]) >> + 1); /* sfbEnergy^(2/4) */ + } else { + qcOutChannel[ch]->sfbEnFacLd[sfbGrp + sfb] = + ((nrgFacLd_34 + + (psyOutChan->sfbEnergyLdData[sfbGrp + sfb] >> 1)) >> + 1); /* sfbEnergy^(1/4) */ + } + qcOutChannel[ch]->sfbEnFacLd[sfbGrp + sfb] = + fixMin(qcOutChannel[ch]->sfbEnFacLd[sfbGrp + sfb], (FIXP_DBL)0); + } + } + } /* sfb loop */ + + adjThrStateElement->lastEnFacPatch[ch] = usePatch; + exePatchM = exePatch; + } else { + /* !noShortWindowInFrame */ + adjThrStateElement->chaosMeasureEnFac[ch] = FL2FXCONST_DBL(0.75f); + adjThrStateElement->lastEnFacPatch[ch] = + TRUE; /* allow use of sfbEnFac patch in upcoming frame */ + } + + } /* ch loop */ +} + +/***************************************************************************** +functionname: FDKaacEnc_calcPe +description: calculate pe for both channels +*****************************************************************************/ +static void FDKaacEnc_calcPe(const PSY_OUT_CHANNEL *const psyOutChannel[(2)], + const QC_OUT_CHANNEL *const qcOutChannel[(2)], + PE_DATA *const peData, const INT nChannels) { + INT ch; + + peData->pe = peData->offset; + peData->constPart = 0; + peData->nActiveLines = 0; + for (ch = 0; ch < nChannels; ch++) { + PE_CHANNEL_DATA *peChanData = &peData->peChannelData[ch]; + + FDKaacEnc_calcSfbPe( + peChanData, qcOutChannel[ch]->sfbWeightedEnergyLdData, + qcOutChannel[ch]->sfbThresholdLdData, psyOutChannel[ch]->sfbCnt, + psyOutChannel[ch]->sfbPerGroup, psyOutChannel[ch]->maxSfbPerGroup, + psyOutChannel[ch]->isBook, psyOutChannel[ch]->isScale); + + peData->pe += peChanData->pe; + peData->constPart += peChanData->constPart; + peData->nActiveLines += peChanData->nActiveLines; + } +} + +void FDKaacEnc_peCalculation(PE_DATA *const peData, + const PSY_OUT_CHANNEL *const psyOutChannel[(2)], + QC_OUT_CHANNEL *const qcOutChannel[(2)], + const struct TOOLSINFO *const toolsInfo, + ATS_ELEMENT *const adjThrStateElement, + const INT nChannels) { + /* constants that will not change during successive pe calculations */ + FDKaacEnc_preparePe(peData, psyOutChannel, qcOutChannel, nChannels, + adjThrStateElement->peOffset); + + /* calculate weighting factor for threshold adjustment */ + FDKaacEnc_calcWeighting(peData, psyOutChannel, qcOutChannel, toolsInfo, + adjThrStateElement, nChannels, 1); + { + /* no weighting of threholds and energies for mlout */ + /* weight energies and thresholds */ + int ch; + for (ch = 0; ch < nChannels; ch++) { + int sfb, sfbGrp; + QC_OUT_CHANNEL *pQcOutCh = qcOutChannel[ch]; + + for (sfbGrp = 0; sfbGrp < psyOutChannel[ch]->sfbCnt; + sfbGrp += psyOutChannel[ch]->sfbPerGroup) { + for (sfb = 0; sfb < psyOutChannel[ch]->maxSfbPerGroup; sfb++) { + pQcOutCh->sfbWeightedEnergyLdData[sfb + sfbGrp] = + pQcOutCh->sfbEnergyLdData[sfb + sfbGrp] - + pQcOutCh->sfbEnFacLd[sfb + sfbGrp]; + pQcOutCh->sfbThresholdLdData[sfb + sfbGrp] -= + pQcOutCh->sfbEnFacLd[sfb + sfbGrp]; + } + } + } + } + + /* pe without reduction */ + FDKaacEnc_calcPe(psyOutChannel, qcOutChannel, peData, nChannels); +} + +/***************************************************************************** +functionname: FDKaacEnc_FDKaacEnc_calcPeNoAH +description: sum the pe data only for bands where avoid hole is inactive +*****************************************************************************/ +#define CONSTPART_HEADROOM 4 +static void FDKaacEnc_FDKaacEnc_calcPeNoAH( + INT *const pe, INT *const constPart, INT *const nActiveLines, + const PE_DATA *const peData, const UCHAR ahFlag[(2)][MAX_GROUPED_SFB], + const PSY_OUT_CHANNEL *const psyOutChannel[(2)], const INT nChannels) { + INT ch, sfb, sfbGrp; + + INT pe_tmp = peData->offset; + INT constPart_tmp = 0; + INT nActiveLines_tmp = 0; + for (ch = 0; ch < nChannels; ch++) { + const PE_CHANNEL_DATA *const peChanData = &peData->peChannelData[ch]; + for (sfbGrp = 0; sfbGrp < psyOutChannel[ch]->sfbCnt; + sfbGrp += psyOutChannel[ch]->sfbPerGroup) { + for (sfb = 0; sfb < psyOutChannel[ch]->maxSfbPerGroup; sfb++) { + if (ahFlag[ch][sfbGrp + sfb] < AH_ACTIVE) { + pe_tmp += peChanData->sfbPe[sfbGrp + sfb]; + constPart_tmp += + peChanData->sfbConstPart[sfbGrp + sfb] >> CONSTPART_HEADROOM; + nActiveLines_tmp += peChanData->sfbNActiveLines[sfbGrp + sfb]; + } + } + } + } + /* correct scaled pe and constPart values */ + *pe = pe_tmp >> PE_CONSTPART_SHIFT; + *constPart = constPart_tmp >> (PE_CONSTPART_SHIFT - CONSTPART_HEADROOM); + + *nActiveLines = nActiveLines_tmp; +} + +/***************************************************************************** +functionname: FDKaacEnc_reduceThresholdsCBR +description: apply reduction formula +*****************************************************************************/ +static const FIXP_DBL limitThrReducedLdData = + (FIXP_DBL)0x00008000; /*FL2FXCONST_DBL(FDKpow(2.0,-LD_DATA_SCALING/4.0));*/ + +static void FDKaacEnc_reduceThresholdsCBR( + QC_OUT_CHANNEL *const qcOutChannel[(2)], + const PSY_OUT_CHANNEL *const psyOutChannel[(2)], + UCHAR ahFlag[(2)][MAX_GROUPED_SFB], + const FIXP_DBL thrExp[(2)][MAX_GROUPED_SFB], const INT nChannels, + const FIXP_DBL redVal_m, const SCHAR redVal_e) { + INT ch, sfb, sfbGrp; + FIXP_DBL sfbEnLdData, sfbThrLdData, sfbThrReducedLdData; + FIXP_DBL sfbThrExp; + + for (ch = 0; ch < nChannels; ch++) { + QC_OUT_CHANNEL *qcOutChan = qcOutChannel[ch]; + for (sfbGrp = 0; sfbGrp < psyOutChannel[ch]->sfbCnt; + sfbGrp += psyOutChannel[ch]->sfbPerGroup) { + for (sfb = 0; sfb < psyOutChannel[ch]->maxSfbPerGroup; sfb++) { + sfbEnLdData = qcOutChan->sfbWeightedEnergyLdData[sfbGrp + sfb]; + sfbThrLdData = qcOutChan->sfbThresholdLdData[sfbGrp + sfb]; + sfbThrExp = thrExp[ch][sfbGrp + sfb]; + if ((sfbEnLdData > sfbThrLdData) && + (ahFlag[ch][sfbGrp + sfb] != AH_ACTIVE)) { + /* threshold reduction formula: + float tmp = thrExp[ch][sfb]+redVal; + tmp *= tmp; + sfbThrReduced = tmp*tmp; + */ + int minScale = fixMin(CountLeadingBits(sfbThrExp), + CountLeadingBits(redVal_m) - redVal_e) - + 1; + + /* 4*log( sfbThrExp + redVal ) */ + sfbThrReducedLdData = + CalcLdData(fAbs(scaleValue(sfbThrExp, minScale) + + scaleValue(redVal_m, redVal_e + minScale))) - + (FIXP_DBL)(minScale << (DFRACT_BITS - 1 - LD_DATA_SHIFT)); + sfbThrReducedLdData <<= 2; + + /* avoid holes */ + if ((sfbThrReducedLdData > + (qcOutChan->sfbMinSnrLdData[sfbGrp + sfb] + sfbEnLdData)) && + (ahFlag[ch][sfbGrp + sfb] != NO_AH)) { + if (qcOutChan->sfbMinSnrLdData[sfbGrp + sfb] > + (FL2FXCONST_DBL(-1.0f) - sfbEnLdData)) { + sfbThrReducedLdData = fixMax( + (qcOutChan->sfbMinSnrLdData[sfbGrp + sfb] + sfbEnLdData), + sfbThrLdData); + } else + sfbThrReducedLdData = sfbThrLdData; + ahFlag[ch][sfbGrp + sfb] = AH_ACTIVE; + } + + /* minimum of 29 dB Ratio for Thresholds */ + if ((sfbEnLdData + (FIXP_DBL)MAXVAL_DBL) > + FL2FXCONST_DBL(9.6336206 / LD_DATA_SCALING)) { + sfbThrReducedLdData = fixMax( + sfbThrReducedLdData, + (sfbEnLdData - FL2FXCONST_DBL(9.6336206 / LD_DATA_SCALING))); + } + + qcOutChan->sfbThresholdLdData[sfbGrp + sfb] = sfbThrReducedLdData; + } + } + } + } +} + +/* similar to prepareSfbPe1() */ +static FIXP_DBL FDKaacEnc_calcChaosMeasure( + const PSY_OUT_CHANNEL *const psyOutChannel, + const FIXP_DBL *const sfbFormFactorLdData) { +#define SCALE_FORM_FAC \ + (4) /* (SCALE_FORM_FAC+FORM_FAC_SHIFT) >= ld(FRAME_LENGTH)*/ +#define SCALE_NRGS (8) +#define SCALE_NLINES (16) +#define SCALE_NRGS_SQRT4 (2) /* 0.25 * SCALE_NRGS */ +#define SCALE_NLINES_P34 (12) /* 0.75 * SCALE_NLINES */ + + INT sfbGrp, sfb; + FIXP_DBL chaosMeasure; + INT frameNLines = 0; + FIXP_DBL frameFormFactor = FL2FXCONST_DBL(0.f); + FIXP_DBL frameEnergy = FL2FXCONST_DBL(0.f); + + for (sfbGrp = 0; sfbGrp < psyOutChannel->sfbCnt; + sfbGrp += psyOutChannel->sfbPerGroup) { + for (sfb = 0; sfb < psyOutChannel->maxSfbPerGroup; sfb++) { + if (psyOutChannel->sfbEnergyLdData[sfbGrp + sfb] > + psyOutChannel->sfbThresholdLdData[sfbGrp + sfb]) { + frameFormFactor += (CalcInvLdData(sfbFormFactorLdData[sfbGrp + sfb]) >> + SCALE_FORM_FAC); + frameNLines += (psyOutChannel->sfbOffsets[sfbGrp + sfb + 1] - + psyOutChannel->sfbOffsets[sfbGrp + sfb]); + frameEnergy += (psyOutChannel->sfbEnergy[sfbGrp + sfb] >> SCALE_NRGS); + } + } + } + + if (frameNLines > 0) { + /* frameNActiveLines = frameFormFactor*2^FORM_FAC_SHIFT * ((frameEnergy + *2^SCALE_NRGS)/frameNLines)^-0.25 chaosMeasure = frameNActiveLines / + frameNLines */ + chaosMeasure = CalcInvLdData( + (((CalcLdData(frameFormFactor) >> 1) - + (CalcLdData(frameEnergy) >> (2 + 1))) - + (fMultDiv2(FL2FXCONST_DBL(0.75f), + CalcLdData((FIXP_DBL)frameNLines + << (DFRACT_BITS - 1 - SCALE_NLINES))) - + (((FIXP_DBL)(-((-SCALE_FORM_FAC + SCALE_NRGS_SQRT4 - FORM_FAC_SHIFT + + SCALE_NLINES_P34) + << (DFRACT_BITS - 1 - LD_DATA_SHIFT)))) >> + 1))) + << 1); + } else { + /* assuming total chaos, if no sfb is above thresholds */ + chaosMeasure = FL2FXCONST_DBL(1.f); + } + + return chaosMeasure; +} + +/* apply reduction formula for VBR-mode */ +static void FDKaacEnc_reduceThresholdsVBR( + QC_OUT_CHANNEL *const qcOutChannel[(2)], + const PSY_OUT_CHANNEL *const psyOutChannel[(2)], + UCHAR ahFlag[(2)][MAX_GROUPED_SFB], + const FIXP_DBL thrExp[(2)][MAX_GROUPED_SFB], const INT nChannels, + const FIXP_DBL vbrQualFactor, FIXP_DBL *const chaosMeasureOld) { + INT ch, sfbGrp, sfb; + FIXP_DBL chGroupEnergy[TRANS_FAC][2]; /*energy for each group and channel*/ + FIXP_DBL chChaosMeasure[2]; + FIXP_DBL frameEnergy = FL2FXCONST_DBL(1e-10f); + FIXP_DBL chaosMeasure = FL2FXCONST_DBL(0.f); + FIXP_DBL sfbEnLdData, sfbThrLdData, sfbThrExp; + FIXP_DBL sfbThrReducedLdData; + FIXP_DBL chaosMeasureAvg; + INT groupCnt; /* loop counter */ + FIXP_DBL redVal[TRANS_FAC]; /* reduction values; in short-block case one + redVal for each group */ + QC_OUT_CHANNEL *qcOutChan = NULL; + const PSY_OUT_CHANNEL *psyOutChan = NULL; + +#define SCALE_GROUP_ENERGY (8) + +#define CONST_CHAOS_MEAS_AVG_FAC_0 (FL2FXCONST_DBL(0.25f)) +#define CONST_CHAOS_MEAS_AVG_FAC_1 (FL2FXCONST_DBL(1.f - 0.25f)) + +#define MIN_LDTHRESH (FL2FXCONST_DBL(-0.515625f)) + + for (ch = 0; ch < nChannels; ch++) { + psyOutChan = psyOutChannel[ch]; + + /* adding up energy for each channel and each group separately */ + FIXP_DBL chEnergy = FL2FXCONST_DBL(0.f); + groupCnt = 0; + + for (sfbGrp = 0; sfbGrp < psyOutChan->sfbCnt; + sfbGrp += psyOutChan->sfbPerGroup, groupCnt++) { + chGroupEnergy[groupCnt][ch] = FL2FXCONST_DBL(0.f); + for (sfb = 0; sfb < psyOutChan->maxSfbPerGroup; sfb++) { + chGroupEnergy[groupCnt][ch] += + (psyOutChan->sfbEnergy[sfbGrp + sfb] >> SCALE_GROUP_ENERGY); + } + chEnergy += chGroupEnergy[groupCnt][ch]; + } + frameEnergy += chEnergy; + + /* chaosMeasure */ + if (psyOutChannel[0]->lastWindowSequence == SHORT_WINDOW) { + chChaosMeasure[ch] = FL2FXCONST_DBL( + 0.5f); /* assume a constant chaos measure of 0.5f for short blocks */ + } else { + chChaosMeasure[ch] = FDKaacEnc_calcChaosMeasure( + psyOutChannel[ch], qcOutChannel[ch]->sfbFormFactorLdData); + } + chaosMeasure += fMult(chChaosMeasure[ch], chEnergy); + } + + if (frameEnergy > chaosMeasure) { + INT scale = CntLeadingZeros(frameEnergy) - 1; + FIXP_DBL num = chaosMeasure << scale; + FIXP_DBL denum = frameEnergy << scale; + chaosMeasure = schur_div(num, denum, 16); + } else { + chaosMeasure = FL2FXCONST_DBL(1.f); + } + + chaosMeasureAvg = fMult(CONST_CHAOS_MEAS_AVG_FAC_0, chaosMeasure) + + fMult(CONST_CHAOS_MEAS_AVG_FAC_1, + *chaosMeasureOld); /* averaging chaos measure */ + *chaosMeasureOld = chaosMeasure = (fixMin( + chaosMeasure, chaosMeasureAvg)); /* use min-value, safe for next frame */ + + /* characteristic curve + chaosMeasure = 0.2f + 0.7f/0.3f * (chaosMeasure - 0.2f); + chaosMeasure = fixMin(1.0f, fixMax(0.1f, chaosMeasure)); + constants scaled by 4.f + */ + chaosMeasure = ((FL2FXCONST_DBL(0.2f) >> 2) + + fMult(FL2FXCONST_DBL(0.7f / (4.f * 0.3f)), + (chaosMeasure - FL2FXCONST_DBL(0.2f)))); + chaosMeasure = + (fixMin((FIXP_DBL)(FL2FXCONST_DBL(1.0f) >> 2), + fixMax((FIXP_DBL)(FL2FXCONST_DBL(0.1f) >> 2), chaosMeasure))) + << 2; + + /* calculation of reduction value */ + if (psyOutChannel[0]->lastWindowSequence == SHORT_WINDOW) { /* short-blocks */ + FDK_ASSERT(TRANS_FAC == 8); +#define WIN_TYPE_SCALE (3) + + groupCnt = 0; + for (sfbGrp = 0; sfbGrp < psyOutChannel[0]->sfbCnt; + sfbGrp += psyOutChannel[0]->sfbPerGroup, groupCnt++) { + FIXP_DBL groupEnergy = FL2FXCONST_DBL(0.f); + + for (ch = 0; ch < nChannels; ch++) { + groupEnergy += + chGroupEnergy[groupCnt] + [ch]; /* adding up the channels groupEnergy */ + } + + FDK_ASSERT(psyOutChannel[0]->groupLen[groupCnt] <= INV_INT_TAB_SIZE); + groupEnergy = fMult( + groupEnergy, + invInt[psyOutChannel[0]->groupLen[groupCnt]]); /* correction of + group energy */ + groupEnergy = fixMin(groupEnergy, + frameEnergy >> WIN_TYPE_SCALE); /* do not allow an + higher redVal as + calculated + framewise */ + + groupEnergy >>= + 2; /* 2*WIN_TYPE_SCALE = 6 => 6+2 = 8 ==> 8/4 = int number */ + + redVal[groupCnt] = + fMult(fMult(vbrQualFactor, chaosMeasure), + CalcInvLdData(CalcLdData(groupEnergy) >> 2)) + << (int)((2 + (2 * WIN_TYPE_SCALE) + SCALE_GROUP_ENERGY) >> 2); + } + } else { /* long-block */ + + redVal[0] = fMult(fMult(vbrQualFactor, chaosMeasure), + CalcInvLdData(CalcLdData(frameEnergy) >> 2)) + << (int)(SCALE_GROUP_ENERGY >> 2); + } + + for (ch = 0; ch < nChannels; ch++) { + qcOutChan = qcOutChannel[ch]; + psyOutChan = psyOutChannel[ch]; + + for (sfbGrp = 0; sfbGrp < psyOutChan->sfbCnt; + sfbGrp += psyOutChan->sfbPerGroup) { + for (sfb = 0; sfb < psyOutChan->maxSfbPerGroup; sfb++) { + sfbEnLdData = (qcOutChan->sfbWeightedEnergyLdData[sfbGrp + sfb]); + sfbThrLdData = (qcOutChan->sfbThresholdLdData[sfbGrp + sfb]); + sfbThrExp = thrExp[ch][sfbGrp + sfb]; + + if ((sfbThrLdData >= MIN_LDTHRESH) && (sfbEnLdData > sfbThrLdData) && + (ahFlag[ch][sfbGrp + sfb] != AH_ACTIVE)) { + /* Short-Window */ + if (psyOutChannel[ch]->lastWindowSequence == SHORT_WINDOW) { + const int groupNumber = (int)sfb / psyOutChan->sfbPerGroup; + + FDK_ASSERT(INV_SQRT4_TAB_SIZE > psyOutChan->groupLen[groupNumber]); + + sfbThrExp = + fMult(sfbThrExp, + fMult(FL2FXCONST_DBL(2.82f / 4.f), + invSqrt4[psyOutChan->groupLen[groupNumber]])) + << 2; + + if (sfbThrExp <= (limitThrReducedLdData - redVal[groupNumber])) { + sfbThrReducedLdData = FL2FXCONST_DBL(-1.0f); + } else { + if ((FIXP_DBL)redVal[groupNumber] >= + FL2FXCONST_DBL(1.0f) - sfbThrExp) + sfbThrReducedLdData = FL2FXCONST_DBL(0.0f); + else { + /* threshold reduction formula */ + sfbThrReducedLdData = + CalcLdData(sfbThrExp + redVal[groupNumber]); + sfbThrReducedLdData <<= 2; + } + } + sfbThrReducedLdData += + (CalcLdInt(psyOutChan->groupLen[groupNumber]) - + ((FIXP_DBL)6 << (DFRACT_BITS - 1 - LD_DATA_SHIFT))); + } + + /* Long-Window */ + else { + if ((FIXP_DBL)redVal[0] >= FL2FXCONST_DBL(1.0f) - sfbThrExp) { + sfbThrReducedLdData = FL2FXCONST_DBL(0.0f); + } else { + /* threshold reduction formula */ + sfbThrReducedLdData = CalcLdData(sfbThrExp + redVal[0]); + sfbThrReducedLdData <<= 2; + } + } + + /* avoid holes */ + if (((sfbThrReducedLdData - sfbEnLdData) > + qcOutChan->sfbMinSnrLdData[sfbGrp + sfb]) && + (ahFlag[ch][sfbGrp + sfb] != NO_AH)) { + if (qcOutChan->sfbMinSnrLdData[sfbGrp + sfb] > + (FL2FXCONST_DBL(-1.0f) - sfbEnLdData)) { + sfbThrReducedLdData = fixMax( + (qcOutChan->sfbMinSnrLdData[sfbGrp + sfb] + sfbEnLdData), + sfbThrLdData); + } else + sfbThrReducedLdData = sfbThrLdData; + ahFlag[ch][sfbGrp + sfb] = AH_ACTIVE; + } + + if (sfbThrReducedLdData < FL2FXCONST_DBL(-0.5f)) + sfbThrReducedLdData = FL2FXCONST_DBL(-1.f); + + /* minimum of 29 dB Ratio for Thresholds */ + if ((sfbEnLdData + FL2FXCONST_DBL(1.0f)) > + FL2FXCONST_DBL(9.6336206 / LD_DATA_SCALING)) { + sfbThrReducedLdData = fixMax( + sfbThrReducedLdData, + sfbEnLdData - FL2FXCONST_DBL(9.6336206 / LD_DATA_SCALING)); + } + + sfbThrReducedLdData = fixMax(MIN_LDTHRESH, sfbThrReducedLdData); + + qcOutChan->sfbThresholdLdData[sfbGrp + sfb] = sfbThrReducedLdData; + } + } + } + } +} + +/***************************************************************************** +functionname: FDKaacEnc_correctThresh +description: if pe difference deltaPe between desired pe and real pe is small +enough, the difference can be distributed among the scale factor bands. New +thresholds can be derived from this pe-difference +*****************************************************************************/ +static void FDKaacEnc_correctThresh( + const CHANNEL_MAPPING *const cm, QC_OUT_ELEMENT *const qcElement[((8))], + const PSY_OUT_ELEMENT *const psyOutElement[((8))], + UCHAR ahFlag[((8))][(2)][MAX_GROUPED_SFB], + const FIXP_DBL thrExp[((8))][(2)][MAX_GROUPED_SFB], const FIXP_DBL redVal_m, + const SCHAR redVal_e, const INT deltaPe, const INT processElements, + const INT elementOffset) { + INT ch, sfb, sfbGrp; + QC_OUT_CHANNEL *qcOutChan; + PSY_OUT_CHANNEL *psyOutChan; + PE_CHANNEL_DATA *peChanData; + FIXP_DBL thrFactorLdData; + FIXP_DBL sfbEnLdData, sfbThrLdData, sfbThrReducedLdData; + FIXP_DBL *sfbPeFactorsLdData[((8))][(2)]; + FIXP_DBL(*sfbNActiveLinesLdData)[(2)][MAX_GROUPED_SFB]; + + INT normFactorInt; + FIXP_DBL normFactorLdData; + + INT nElements = elementOffset + processElements; + INT elementId; + + /* scratch is empty; use temporal memory from quantSpec in QC_OUT_CHANNEL */ + for (elementId = elementOffset; elementId < nElements; elementId++) { + for (ch = 0; ch < cm->elInfo[elementId].nChannelsInEl; ch++) { + /* The reinterpret_cast is used to suppress a compiler warning. We know + * that qcElement[elementId]->qcOutChannel[ch]->quantSpec is sufficiently + * aligned, so the cast is safe */ + sfbPeFactorsLdData[elementId][ch] = + reinterpret_cast<FIXP_DBL *>(reinterpret_cast<void *>( + qcElement[elementId]->qcOutChannel[ch]->quantSpec)); + } + } + /* The reinterpret_cast is used to suppress a compiler warning. We know that + * qcElement[0]->dynMem_SfbNActiveLinesLdData is sufficiently aligned, so the + * cast is safe */ + sfbNActiveLinesLdData = reinterpret_cast<FIXP_DBL(*)[(2)][MAX_GROUPED_SFB]>( + reinterpret_cast<void *>(qcElement[0]->dynMem_SfbNActiveLinesLdData)); + + /* for each sfb calc relative factors for pe changes */ + normFactorInt = 0; + + for (elementId = elementOffset; elementId < nElements; elementId++) { + if (cm->elInfo[elementId].elType != ID_DSE) { + for (ch = 0; ch < cm->elInfo[elementId].nChannelsInEl; ch++) { + psyOutChan = psyOutElement[elementId]->psyOutChannel[ch]; + peChanData = &qcElement[elementId]->peData.peChannelData[ch]; + + for (sfbGrp = 0; sfbGrp < psyOutChan->sfbCnt; + sfbGrp += psyOutChan->sfbPerGroup) { + for (sfb = 0; sfb < psyOutChan->maxSfbPerGroup; sfb++) { + if (peChanData->sfbNActiveLines[sfbGrp + sfb] == 0) { + sfbNActiveLinesLdData[elementId][ch][sfbGrp + sfb] = + FL2FXCONST_DBL(-1.0f); + } else { + /* Both CalcLdInt and CalcLdData can be used! + * No offset has to be subtracted, because sfbNActiveLinesLdData + * is shorted while thrFactor calculation */ + sfbNActiveLinesLdData[elementId][ch][sfbGrp + sfb] = + CalcLdInt(peChanData->sfbNActiveLines[sfbGrp + sfb]); + } + if (((ahFlag[elementId][ch][sfbGrp + sfb] < AH_ACTIVE) || + (deltaPe > 0)) && + peChanData->sfbNActiveLines[sfbGrp + sfb] != 0) { + if (thrExp[elementId][ch][sfbGrp + sfb] > -redVal_m) { + /* sfbPeFactors[ch][sfbGrp+sfb] = + peChanData->sfbNActiveLines[sfbGrp+sfb] / + (thrExp[elementId][ch][sfbGrp+sfb] + + redVal[elementId]); */ + + int minScale = + fixMin( + CountLeadingBits(thrExp[elementId][ch][sfbGrp + sfb]), + CountLeadingBits(redVal_m) - redVal_e) - + 1; + + /* sumld = ld64( sfbThrExp + redVal ) */ + FIXP_DBL sumLd = + CalcLdData(scaleValue(thrExp[elementId][ch][sfbGrp + sfb], + minScale) + + scaleValue(redVal_m, redVal_e + minScale)) - + (FIXP_DBL)(minScale << (DFRACT_BITS - 1 - LD_DATA_SHIFT)); + + if (sumLd < FL2FXCONST_DBL(0.f)) { + sfbPeFactorsLdData[elementId][ch][sfbGrp + sfb] = + sfbNActiveLinesLdData[elementId][ch][sfbGrp + sfb] - + sumLd; + } else { + if (sfbNActiveLinesLdData[elementId][ch][sfbGrp + sfb] > + (FL2FXCONST_DBL(-1.f) + sumLd)) { + sfbPeFactorsLdData[elementId][ch][sfbGrp + sfb] = + sfbNActiveLinesLdData[elementId][ch][sfbGrp + sfb] - + sumLd; + } else { + sfbPeFactorsLdData[elementId][ch][sfbGrp + sfb] = + sfbNActiveLinesLdData[elementId][ch][sfbGrp + sfb]; + } + } + + normFactorInt += (INT)CalcInvLdData( + sfbPeFactorsLdData[elementId][ch][sfbGrp + sfb]); + } else + sfbPeFactorsLdData[elementId][ch][sfbGrp + sfb] = + FL2FXCONST_DBL(1.0f); + } else + sfbPeFactorsLdData[elementId][ch][sfbGrp + sfb] = + FL2FXCONST_DBL(-1.0f); + } + } + } + } + } + + /* normFactorLdData = ld64(deltaPe/normFactorInt) */ + normFactorLdData = + CalcLdData((FIXP_DBL)((deltaPe < 0) ? (-deltaPe) : (deltaPe))) - + CalcLdData((FIXP_DBL)normFactorInt); + + /* distribute the pe difference to the scalefactors + and calculate the according thresholds */ + for (elementId = elementOffset; elementId < nElements; elementId++) { + if (cm->elInfo[elementId].elType != ID_DSE) { + for (ch = 0; ch < cm->elInfo[elementId].nChannelsInEl; ch++) { + qcOutChan = qcElement[elementId]->qcOutChannel[ch]; + psyOutChan = psyOutElement[elementId]->psyOutChannel[ch]; + peChanData = &qcElement[elementId]->peData.peChannelData[ch]; + + for (sfbGrp = 0; sfbGrp < psyOutChan->sfbCnt; + sfbGrp += psyOutChan->sfbPerGroup) { + for (sfb = 0; sfb < psyOutChan->maxSfbPerGroup; sfb++) { + if (peChanData->sfbNActiveLines[sfbGrp + sfb] > 0) { + /* pe difference for this sfb */ + if ((sfbPeFactorsLdData[elementId][ch][sfbGrp + sfb] == + FL2FXCONST_DBL(-1.0f)) || + (deltaPe == 0)) { + thrFactorLdData = FL2FXCONST_DBL(0.f); + } else { + /* new threshold */ + FIXP_DBL tmp = CalcInvLdData( + sfbPeFactorsLdData[elementId][ch][sfbGrp + sfb] + + normFactorLdData - + sfbNActiveLinesLdData[elementId][ch][sfbGrp + sfb] - + FL2FXCONST_DBL((float)LD_DATA_SHIFT / LD_DATA_SCALING)); + + /* limit thrFactor to 60dB */ + tmp = (deltaPe < 0) ? tmp : (-tmp); + thrFactorLdData = + fMin(tmp, FL2FXCONST_DBL(20.f / LD_DATA_SCALING)); + } + + /* new threshold */ + sfbThrLdData = qcOutChan->sfbThresholdLdData[sfbGrp + sfb]; + sfbEnLdData = qcOutChan->sfbWeightedEnergyLdData[sfbGrp + sfb]; + + if (thrFactorLdData < FL2FXCONST_DBL(0.f)) { + if (sfbThrLdData > (FL2FXCONST_DBL(-1.f) - thrFactorLdData)) { + sfbThrReducedLdData = sfbThrLdData + thrFactorLdData; + } else { + sfbThrReducedLdData = FL2FXCONST_DBL(-1.f); + } + } else { + sfbThrReducedLdData = sfbThrLdData + thrFactorLdData; + } + + /* avoid hole */ + if ((sfbThrReducedLdData - sfbEnLdData > + qcOutChan->sfbMinSnrLdData[sfbGrp + sfb]) && + (ahFlag[elementId][ch][sfbGrp + sfb] == AH_INACTIVE)) { + /* sfbThrReduced = max(psyOutChan[ch]->sfbMinSnr[i] * sfbEn, + * sfbThr); */ + if (sfbEnLdData > + (sfbThrLdData - qcOutChan->sfbMinSnrLdData[sfbGrp + sfb])) { + sfbThrReducedLdData = + qcOutChan->sfbMinSnrLdData[sfbGrp + sfb] + sfbEnLdData; + } else { + sfbThrReducedLdData = sfbThrLdData; + } + ahFlag[elementId][ch][sfbGrp + sfb] = AH_ACTIVE; + } + + qcOutChan->sfbThresholdLdData[sfbGrp + sfb] = sfbThrReducedLdData; + } + } + } + } + } + } +} + +/***************************************************************************** + functionname: FDKaacEnc_reduceMinSnr + description: if the desired pe can not be reached, reduce pe by + reducing minSnr +*****************************************************************************/ +static void FDKaacEnc_reduceMinSnr( + const CHANNEL_MAPPING *const cm, QC_OUT_ELEMENT *const qcElement[((8))], + const PSY_OUT_ELEMENT *const psyOutElement[((8))], + const UCHAR ahFlag[((8))][(2)][MAX_GROUPED_SFB], const INT desiredPe, + INT *const redPeGlobal, const INT processElements, const INT elementOffset) + +{ + INT ch, elementId, globalMaxSfb = 0; + const INT nElements = elementOffset + processElements; + INT newGlobalPe = *redPeGlobal; + + if (newGlobalPe <= desiredPe) { + goto bail; + } + + /* global maximum of maxSfbPerGroup */ + for (elementId = elementOffset; elementId < nElements; elementId++) { + if (cm->elInfo[elementId].elType != ID_DSE) { + for (ch = 0; ch < cm->elInfo[elementId].nChannelsInEl; ch++) { + globalMaxSfb = + fMax(globalMaxSfb, + psyOutElement[elementId]->psyOutChannel[ch]->maxSfbPerGroup); + } + } + } + + /* as long as globalPE is above desirePE reduce SNR to 1.0 dB, starting at + * highest SFB */ + while ((newGlobalPe > desiredPe) && (--globalMaxSfb >= 0)) { + for (elementId = elementOffset; elementId < nElements; elementId++) { + if (cm->elInfo[elementId].elType != ID_DSE) { + PE_DATA *peData = &qcElement[elementId]->peData; + + for (ch = 0; ch < cm->elInfo[elementId].nChannelsInEl; ch++) { + QC_OUT_CHANNEL *qcOutChan = qcElement[elementId]->qcOutChannel[ch]; + PSY_OUT_CHANNEL *psyOutChan = + psyOutElement[elementId]->psyOutChannel[ch]; + + /* try to reduce SNR of channel's uppermost SFB(s) */ + if (globalMaxSfb < psyOutChan->maxSfbPerGroup) { + INT sfb, deltaPe = 0; + + for (sfb = globalMaxSfb; sfb < psyOutChan->sfbCnt; + sfb += psyOutChan->sfbPerGroup) { + if (ahFlag[elementId][ch][sfb] != NO_AH && + qcOutChan->sfbMinSnrLdData[sfb] < SnrLdFac && + (qcOutChan->sfbWeightedEnergyLdData[sfb] > + qcOutChan->sfbThresholdLdData[sfb] - SnrLdFac)) { + /* increase threshold to new minSnr of 1dB */ + qcOutChan->sfbMinSnrLdData[sfb] = SnrLdFac; + qcOutChan->sfbThresholdLdData[sfb] = + qcOutChan->sfbWeightedEnergyLdData[sfb] + SnrLdFac; + + /* calc new pe */ + /* C2 + C3*ld(1/0.8) = 1.5 */ + deltaPe -= peData->peChannelData[ch].sfbPe[sfb]; + + /* sfbPe = 1.5 * sfbNLines */ + peData->peChannelData[ch].sfbPe[sfb] = + (3 * peData->peChannelData[ch].sfbNLines[sfb]) + << (PE_CONSTPART_SHIFT - 1); + deltaPe += peData->peChannelData[ch].sfbPe[sfb]; + } + + } /* sfb loop */ + + deltaPe >>= PE_CONSTPART_SHIFT; + peData->pe += deltaPe; + peData->peChannelData[ch].pe += deltaPe; + newGlobalPe += deltaPe; + + } /* if globalMaxSfb < maxSfbPerGroup */ + + /* stop if enough has been saved */ + if (newGlobalPe <= desiredPe) { + goto bail; + } + + } /* ch loop */ + } /* != ID_DSE */ + } /* elementId loop */ + } /* while ( newGlobalPe > desiredPe) && (--globalMaxSfb >= 0) ) */ + +bail: + /* update global PE */ + *redPeGlobal = newGlobalPe; +} + +/***************************************************************************** + functionname: FDKaacEnc_allowMoreHoles + description: if the desired pe can not be reached, some more scalefactor + bands have to be quantized to zero +*****************************************************************************/ +static void FDKaacEnc_allowMoreHoles( + const CHANNEL_MAPPING *const cm, QC_OUT_ELEMENT *const qcElement[((8))], + const PSY_OUT_ELEMENT *const psyOutElement[((8))], + const ATS_ELEMENT *const AdjThrStateElement[((8))], + UCHAR ahFlag[((8))][(2)][MAX_GROUPED_SFB], const INT desiredPe, + const INT currentPe, const int processElements, const int elementOffset) { + INT elementId; + INT nElements = elementOffset + processElements; + INT actPe = currentPe; + + if (actPe <= desiredPe) { + return; /* nothing to do */ + } + + for (elementId = elementOffset; elementId < nElements; elementId++) { + if (cm->elInfo[elementId].elType != ID_DSE) { + INT ch, sfb, sfbGrp; + + PE_DATA *peData = &qcElement[elementId]->peData; + const INT nChannels = cm->elInfo[elementId].nChannelsInEl; + + QC_OUT_CHANNEL *qcOutChannel[(2)] = {NULL}; + PSY_OUT_CHANNEL *psyOutChannel[(2)] = {NULL}; + + for (ch = 0; ch < nChannels; ch++) { + /* init pointers */ + qcOutChannel[ch] = qcElement[elementId]->qcOutChannel[ch]; + psyOutChannel[ch] = psyOutElement[elementId]->psyOutChannel[ch]; + + for (sfbGrp = 0; sfbGrp < psyOutChannel[ch]->sfbCnt; + sfbGrp += psyOutChannel[ch]->sfbPerGroup) { + for (sfb = psyOutChannel[ch]->maxSfbPerGroup; + sfb < psyOutChannel[ch]->sfbPerGroup; sfb++) { + peData->peChannelData[ch].sfbPe[sfbGrp + sfb] = 0; + } + } + } + + /* for MS allow hole in the channel with less energy */ + if (nChannels == 2 && psyOutChannel[0]->lastWindowSequence == + psyOutChannel[1]->lastWindowSequence) { + for (sfb = psyOutChannel[0]->maxSfbPerGroup - 1; sfb >= 0; sfb--) { + for (sfbGrp = 0; sfbGrp < psyOutChannel[0]->sfbCnt; + sfbGrp += psyOutChannel[0]->sfbPerGroup) { + if (psyOutElement[elementId]->toolsInfo.msMask[sfbGrp + sfb]) { + FIXP_DBL EnergyLd_L = + qcOutChannel[0]->sfbWeightedEnergyLdData[sfbGrp + sfb]; + FIXP_DBL EnergyLd_R = + qcOutChannel[1]->sfbWeightedEnergyLdData[sfbGrp + sfb]; + + /* allow hole in side channel ? */ + if ((ahFlag[elementId][1][sfbGrp + sfb] != NO_AH) && + (((FL2FXCONST_DBL(-0.02065512648f) >> 1) + + (qcOutChannel[0]->sfbMinSnrLdData[sfbGrp + sfb] >> 1)) > + ((EnergyLd_R >> 1) - (EnergyLd_L >> 1)))) { + ahFlag[elementId][1][sfbGrp + sfb] = NO_AH; + qcOutChannel[1]->sfbThresholdLdData[sfbGrp + sfb] = + FL2FXCONST_DBL(0.015625f) + EnergyLd_R; + actPe -= peData->peChannelData[1].sfbPe[sfbGrp + sfb] >> + PE_CONSTPART_SHIFT; + } + /* allow hole in mid channel ? */ + else if ((ahFlag[elementId][0][sfbGrp + sfb] != NO_AH) && + (((FL2FXCONST_DBL(-0.02065512648f) >> 1) + + (qcOutChannel[1]->sfbMinSnrLdData[sfbGrp + sfb] >> + 1)) > ((EnergyLd_L >> 1) - (EnergyLd_R >> 1)))) { + ahFlag[elementId][0][sfbGrp + sfb] = NO_AH; + qcOutChannel[0]->sfbThresholdLdData[sfbGrp + sfb] = + FL2FXCONST_DBL(0.015625f) + EnergyLd_L; + actPe -= peData->peChannelData[0].sfbPe[sfbGrp + sfb] >> + PE_CONSTPART_SHIFT; + } /* if (ahFlag) */ + } /* if MS */ + } /* sfbGrp */ + if (actPe <= desiredPe) { + return; /* stop if enough has been saved */ + } + } /* sfb */ + } /* MS possible ? */ + + } /* EOF DSE-suppression */ + } /* EOF for all elements... */ + + if (actPe > desiredPe) { + /* more holes necessary? subsequently erase bands starting with low energies + */ + INT ch, sfb, sfbGrp; + INT minSfb, maxSfb; + INT enIdx, ahCnt, done; + INT startSfb[(8)]; + INT sfbCnt[(8)]; + INT sfbPerGroup[(8)]; + INT maxSfbPerGroup[(8)]; + FIXP_DBL avgEn; + FIXP_DBL minEnLD64; + FIXP_DBL avgEnLD64; + FIXP_DBL enLD64[NUM_NRG_LEVS]; + INT avgEn_e; + + /* get the scaling factor over all audio elements and channels */ + maxSfb = 0; + for (elementId = elementOffset; elementId < nElements; elementId++) { + if (cm->elInfo[elementId].elType != ID_DSE) { + for (ch = 0; ch < cm->elInfo[elementId].nChannelsInEl; ch++) { + for (sfbGrp = 0; + sfbGrp < psyOutElement[elementId]->psyOutChannel[ch]->sfbCnt; + sfbGrp += + psyOutElement[elementId]->psyOutChannel[ch]->sfbPerGroup) { + maxSfb += + psyOutElement[elementId]->psyOutChannel[ch]->maxSfbPerGroup; + } + } + } + } + avgEn_e = + (DFRACT_BITS - fixnormz_D((LONG)fMax(0, maxSfb - 1))); /* ilog2() */ + + ahCnt = 0; + maxSfb = 0; + minSfb = MAX_SFB; + avgEn = FL2FXCONST_DBL(0.0f); + minEnLD64 = FL2FXCONST_DBL(0.0f); + + for (elementId = elementOffset; elementId < nElements; elementId++) { + if (cm->elInfo[elementId].elType != ID_DSE) { + for (ch = 0; ch < cm->elInfo[elementId].nChannelsInEl; ch++) { + const INT chIdx = cm->elInfo[elementId].ChannelIndex[ch]; + QC_OUT_CHANNEL *qcOutChannel = qcElement[elementId]->qcOutChannel[ch]; + PSY_OUT_CHANNEL *psyOutChannel = + psyOutElement[elementId]->psyOutChannel[ch]; + + maxSfbPerGroup[chIdx] = psyOutChannel->maxSfbPerGroup; + sfbCnt[chIdx] = psyOutChannel->sfbCnt; + sfbPerGroup[chIdx] = psyOutChannel->sfbPerGroup; + + maxSfb = fMax(maxSfb, psyOutChannel->maxSfbPerGroup); + + if (psyOutChannel->lastWindowSequence != SHORT_WINDOW) { + startSfb[chIdx] = AdjThrStateElement[elementId]->ahParam.startSfbL; + } else { + startSfb[chIdx] = AdjThrStateElement[elementId]->ahParam.startSfbS; + } + + minSfb = fMin(minSfb, startSfb[chIdx]); + + sfbGrp = 0; + sfb = startSfb[chIdx]; + + do { + for (; sfb < psyOutChannel->maxSfbPerGroup; sfb++) { + if ((ahFlag[elementId][ch][sfbGrp + sfb] != NO_AH) && + (qcOutChannel->sfbWeightedEnergyLdData[sfbGrp + sfb] > + qcOutChannel->sfbThresholdLdData[sfbGrp + sfb])) { + minEnLD64 = fixMin(minEnLD64, + qcOutChannel->sfbEnergyLdData[sfbGrp + sfb]); + avgEn += qcOutChannel->sfbEnergy[sfbGrp + sfb] >> avgEn_e; + ahCnt++; + } + } + + sfbGrp += psyOutChannel->sfbPerGroup; + sfb = startSfb[chIdx]; + + } while (sfbGrp < psyOutChannel->sfbCnt); + } + } /* (cm->elInfo[elementId].elType != ID_DSE) */ + } /* (elementId = elementOffset;elementId<nElements;elementId++) */ + + if ((avgEn == FL2FXCONST_DBL(0.0f)) || (ahCnt == 0)) { + avgEnLD64 = FL2FXCONST_DBL(0.0f); + } else { + avgEnLD64 = CalcLdData(avgEn) + + (FIXP_DBL)(avgEn_e << (DFRACT_BITS - 1 - LD_DATA_SHIFT)) - + CalcLdInt(ahCnt); + } + + /* calc some energy borders between minEn and avgEn */ + + /* for (enIdx = 0; enIdx < NUM_NRG_LEVS; enIdx++) { + en[enIdx] = (2.0f*enIdx+1.0f)/(2.0f*NUM_NRG_LEVS-1.0f); + } */ + enLD64[0] = + minEnLD64 + fMult((avgEnLD64 - minEnLD64), FL2FXCONST_DBL(0.06666667f)); + enLD64[1] = + minEnLD64 + fMult((avgEnLD64 - minEnLD64), FL2FXCONST_DBL(0.20000000f)); + enLD64[2] = + minEnLD64 + fMult((avgEnLD64 - minEnLD64), FL2FXCONST_DBL(0.33333334f)); + enLD64[3] = + minEnLD64 + fMult((avgEnLD64 - minEnLD64), FL2FXCONST_DBL(0.46666667f)); + enLD64[4] = + minEnLD64 + fMult((avgEnLD64 - minEnLD64), FL2FXCONST_DBL(0.60000002f)); + enLD64[5] = + minEnLD64 + fMult((avgEnLD64 - minEnLD64), FL2FXCONST_DBL(0.73333335f)); + enLD64[6] = + minEnLD64 + fMult((avgEnLD64 - minEnLD64), FL2FXCONST_DBL(0.86666667f)); + enLD64[7] = minEnLD64 + (avgEnLD64 - minEnLD64); + + done = 0; + enIdx = 0; + sfb = maxSfb - 1; + + while (!done) { + for (elementId = elementOffset; elementId < nElements; elementId++) { + if (cm->elInfo[elementId].elType != ID_DSE) { + PE_DATA *peData = &qcElement[elementId]->peData; + for (ch = 0; ch < cm->elInfo[elementId].nChannelsInEl; ch++) { + const INT chIdx = cm->elInfo[elementId].ChannelIndex[ch]; + QC_OUT_CHANNEL *qcOutChannel = + qcElement[elementId]->qcOutChannel[ch]; + if (sfb >= startSfb[chIdx] && sfb < maxSfbPerGroup[chIdx]) { + for (sfbGrp = 0; sfbGrp < sfbCnt[chIdx]; + sfbGrp += sfbPerGroup[chIdx]) { + /* sfb energy below border ? */ + if (ahFlag[elementId][ch][sfbGrp + sfb] != NO_AH && + qcOutChannel->sfbEnergyLdData[sfbGrp + sfb] < + enLD64[enIdx]) { + /* allow hole */ + ahFlag[elementId][ch][sfbGrp + sfb] = NO_AH; + qcOutChannel->sfbThresholdLdData[sfbGrp + sfb] = + FL2FXCONST_DBL(0.015625f) + + qcOutChannel->sfbWeightedEnergyLdData[sfbGrp + sfb]; + actPe -= peData->peChannelData[ch].sfbPe[sfbGrp + sfb] >> + PE_CONSTPART_SHIFT; + } + if (actPe <= desiredPe) { + return; /* stop if enough has been saved */ + } + } /* sfbGrp */ + } /* sfb */ + } /* nChannelsInEl */ + } /* ID_DSE */ + } /* elementID */ + + sfb--; + if (sfb < minSfb) { + /* restart with next energy border */ + sfb = maxSfb; + enIdx++; + if (enIdx >= NUM_NRG_LEVS) { + done = 1; + } + } + } /* done */ + } /* (actPe <= desiredPe) */ +} + +/* reset avoid hole flags from AH_ACTIVE to AH_INACTIVE */ +static void FDKaacEnc_resetAHFlags( + UCHAR ahFlag[(2)][MAX_GROUPED_SFB], const INT nChannels, + const PSY_OUT_CHANNEL *const psyOutChannel[(2)]) { + int ch, sfb, sfbGrp; + + for (ch = 0; ch < nChannels; ch++) { + for (sfbGrp = 0; sfbGrp < psyOutChannel[ch]->sfbCnt; + sfbGrp += psyOutChannel[ch]->sfbPerGroup) { + for (sfb = 0; sfb < psyOutChannel[ch]->maxSfbPerGroup; sfb++) { + if (ahFlag[ch][sfbGrp + sfb] == AH_ACTIVE) { + ahFlag[ch][sfbGrp + sfb] = AH_INACTIVE; + } + } + } + } +} + +static FIXP_DBL CalcRedValPower(FIXP_DBL num, FIXP_DBL denum, INT *scaling) { + FIXP_DBL value = FL2FXCONST_DBL(0.f); + + if (num >= FL2FXCONST_DBL(0.f)) { + value = fDivNorm(num, denum, scaling); + } else { + value = -fDivNorm(-num, denum, scaling); + } + value = f2Pow(value, *scaling, scaling); + + return value; +} + +/***************************************************************************** +functionname: FDKaacEnc_adaptThresholdsToPe +description: two guesses for the reduction value and one final correction of +the thresholds +*****************************************************************************/ +static void FDKaacEnc_adaptThresholdsToPe( + const CHANNEL_MAPPING *const cm, + ATS_ELEMENT *const AdjThrStateElement[((8))], + QC_OUT_ELEMENT *const qcElement[((8))], + const PSY_OUT_ELEMENT *const psyOutElement[((8))], const INT desiredPe, + const INT maxIter2ndGuess, const INT processElements, + const INT elementOffset) { + FIXP_DBL reductionValue_m; + SCHAR reductionValue_e; + UCHAR(*pAhFlag)[(2)][MAX_GROUPED_SFB]; + FIXP_DBL(*pThrExp)[(2)][MAX_GROUPED_SFB]; + int iter; + + INT constPartGlobal, noRedPeGlobal, nActiveLinesGlobal, redPeGlobal; + constPartGlobal = noRedPeGlobal = nActiveLinesGlobal = redPeGlobal = 0; + + int elementId; + + int nElements = elementOffset + processElements; + if (nElements > cm->nElements) { + nElements = cm->nElements; + } + + /* The reinterpret_cast is used to suppress a compiler warning. We know that + * qcElement[0]->dynMem_Ah_Flag is sufficiently aligned, so the cast is safe + */ + pAhFlag = reinterpret_cast<UCHAR(*)[(2)][MAX_GROUPED_SFB]>( + reinterpret_cast<void *>(qcElement[0]->dynMem_Ah_Flag)); + /* The reinterpret_cast is used to suppress a compiler warning. We know that + * qcElement[0]->dynMem_Thr_Exp is sufficiently aligned, so the cast is safe + */ + pThrExp = reinterpret_cast<FIXP_DBL(*)[(2)][MAX_GROUPED_SFB]>( + reinterpret_cast<void *>(qcElement[0]->dynMem_Thr_Exp)); + + /* ------------------------------------------------------- */ + /* Part I: Initialize data structures and variables... */ + /* ------------------------------------------------------- */ + for (elementId = elementOffset; elementId < nElements; elementId++) { + if (cm->elInfo[elementId].elType != ID_DSE) { + INT nChannels = cm->elInfo[elementId].nChannelsInEl; + PE_DATA *peData = &qcElement[elementId]->peData; + + /* thresholds to the power of redExp */ + FDKaacEnc_calcThreshExp( + pThrExp[elementId], qcElement[elementId]->qcOutChannel, + psyOutElement[elementId]->psyOutChannel, nChannels); + + /* lower the minSnr requirements for low energies compared to the average + energy in this frame */ + FDKaacEnc_adaptMinSnr(qcElement[elementId]->qcOutChannel, + psyOutElement[elementId]->psyOutChannel, + &AdjThrStateElement[elementId]->minSnrAdaptParam, + nChannels); + + /* init ahFlag (0: no ah necessary, 1: ah possible, 2: ah active */ + FDKaacEnc_initAvoidHoleFlag( + qcElement[elementId]->qcOutChannel, + psyOutElement[elementId]->psyOutChannel, pAhFlag[elementId], + &psyOutElement[elementId]->toolsInfo, nChannels, + &AdjThrStateElement[elementId]->ahParam); + + /* sum up */ + constPartGlobal += peData->constPart; + noRedPeGlobal += peData->pe; + nActiveLinesGlobal += fixMax((INT)peData->nActiveLines, 1); + + } /* EOF DSE-suppression */ + } /* EOF for all elements... */ + + /* + First guess of reduction value: + avgThrExp = (float)pow(2.0f, (constPartGlobal - noRedPeGlobal)/(4.0f * + nActiveLinesGlobal)); redVal = (float)pow(2.0f, (constPartGlobal - + desiredPe)/(4.0f * nActiveLinesGlobal)) - avgThrExp; redVal = max(0.f, + redVal); + */ + int redVal_e, avgThrExp_e, result_e; + FIXP_DBL redVal_m, avgThrExp_m; + + redVal_m = CalcRedValPower(constPartGlobal - desiredPe, + 4 * nActiveLinesGlobal, &redVal_e); + avgThrExp_m = CalcRedValPower(constPartGlobal - noRedPeGlobal, + 4 * nActiveLinesGlobal, &avgThrExp_e); + result_e = fMax(redVal_e, avgThrExp_e) + 1; + + reductionValue_m = fMax(FL2FXCONST_DBL(0.f), + scaleValue(redVal_m, redVal_e - result_e) - + scaleValue(avgThrExp_m, avgThrExp_e - result_e)); + reductionValue_e = result_e; + + /* ----------------------------------------------------------------------- */ + /* Part II: Calculate bit consumption of initial bit constraints setup */ + /* ----------------------------------------------------------------------- */ + for (elementId = elementOffset; elementId < nElements; elementId++) { + if (cm->elInfo[elementId].elType != ID_DSE) { + INT nChannels = cm->elInfo[elementId].nChannelsInEl; + PE_DATA *peData = &qcElement[elementId]->peData; + + /* reduce thresholds */ + FDKaacEnc_reduceThresholdsCBR( + qcElement[elementId]->qcOutChannel, + psyOutElement[elementId]->psyOutChannel, pAhFlag[elementId], + pThrExp[elementId], nChannels, reductionValue_m, reductionValue_e); + + /* pe after first guess */ + FDKaacEnc_calcPe(psyOutElement[elementId]->psyOutChannel, + qcElement[elementId]->qcOutChannel, peData, nChannels); + + redPeGlobal += peData->pe; + } /* EOF DSE-suppression */ + } /* EOF for all elements... */ + + /* -------------------------------------------------- */ + /* Part III: Iterate until bit constraints are met */ + /* -------------------------------------------------- */ + iter = 0; + while ((fixp_abs(redPeGlobal - desiredPe) > + fMultI(FL2FXCONST_DBL(0.05f), desiredPe)) && + (iter < maxIter2ndGuess)) { + INT desiredPeNoAHGlobal; + INT redPeNoAHGlobal = 0; + INT constPartNoAHGlobal = 0; + INT nActiveLinesNoAHGlobal = 0; + + for (elementId = elementOffset; elementId < nElements; elementId++) { + if (cm->elInfo[elementId].elType != ID_DSE) { + INT redPeNoAH, constPartNoAH, nActiveLinesNoAH; + INT nChannels = cm->elInfo[elementId].nChannelsInEl; + PE_DATA *peData = &qcElement[elementId]->peData; + + /* pe for bands where avoid hole is inactive */ + FDKaacEnc_FDKaacEnc_calcPeNoAH( + &redPeNoAH, &constPartNoAH, &nActiveLinesNoAH, peData, + pAhFlag[elementId], psyOutElement[elementId]->psyOutChannel, + nChannels); + + redPeNoAHGlobal += redPeNoAH; + constPartNoAHGlobal += constPartNoAH; + nActiveLinesNoAHGlobal += nActiveLinesNoAH; + } /* EOF DSE-suppression */ + } /* EOF for all elements... */ + + /* Calculate new redVal ... */ + if (desiredPe < redPeGlobal) { + /* new desired pe without bands where avoid hole is active */ + desiredPeNoAHGlobal = desiredPe - (redPeGlobal - redPeNoAHGlobal); + + /* limit desiredPeNoAH to positive values, as the PE can not become + * negative */ + desiredPeNoAHGlobal = fMax(0, desiredPeNoAHGlobal); + + /* second guess (only if there are bands left where avoid hole is + * inactive)*/ + if (nActiveLinesNoAHGlobal > 0) { + /* + avgThrExp = (float)pow(2.0f, (constPartNoAHGlobal - redPeNoAHGlobal) / + (4.0f * nActiveLinesNoAHGlobal)); redVal += (float)pow(2.0f, + (constPartNoAHGlobal - desiredPeNoAHGlobal) / (4.0f * + nActiveLinesNoAHGlobal)) - avgThrExp; redVal = max(0.0f, redVal); + */ + + redVal_m = CalcRedValPower(constPartNoAHGlobal - desiredPeNoAHGlobal, + 4 * nActiveLinesNoAHGlobal, &redVal_e); + avgThrExp_m = CalcRedValPower(constPartNoAHGlobal - redPeNoAHGlobal, + 4 * nActiveLinesNoAHGlobal, &avgThrExp_e); + result_e = fMax(reductionValue_e, fMax(redVal_e, avgThrExp_e) + 1) + 1; + + reductionValue_m = + fMax(FL2FXCONST_DBL(0.f), + scaleValue(reductionValue_m, reductionValue_e - result_e) + + scaleValue(redVal_m, redVal_e - result_e) - + scaleValue(avgThrExp_m, avgThrExp_e - result_e)); + reductionValue_e = result_e; + + } /* nActiveLinesNoAHGlobal > 0 */ + } else { + /* redVal *= redPeGlobal/desiredPe; */ + int sc0, sc1; + reductionValue_m = fMultNorm( + reductionValue_m, + fDivNorm((FIXP_DBL)redPeGlobal, (FIXP_DBL)desiredPe, &sc0), &sc1); + reductionValue_e += sc0 + sc1; + + for (elementId = elementOffset; elementId < nElements; elementId++) { + if (cm->elInfo[elementId].elType != ID_DSE) { + FDKaacEnc_resetAHFlags(pAhFlag[elementId], + cm->elInfo[elementId].nChannelsInEl, + psyOutElement[elementId]->psyOutChannel); + } /* EOF DSE-suppression */ + } /* EOF for all elements... */ + } + + redPeGlobal = 0; + /* Calculate new redVal's PE... */ + for (elementId = elementOffset; elementId < nElements; elementId++) { + if (cm->elInfo[elementId].elType != ID_DSE) { + INT nChannels = cm->elInfo[elementId].nChannelsInEl; + PE_DATA *peData = &qcElement[elementId]->peData; + + /* reduce thresholds */ + FDKaacEnc_reduceThresholdsCBR( + qcElement[elementId]->qcOutChannel, + psyOutElement[elementId]->psyOutChannel, pAhFlag[elementId], + pThrExp[elementId], nChannels, reductionValue_m, reductionValue_e); + + /* pe after second guess */ + FDKaacEnc_calcPe(psyOutElement[elementId]->psyOutChannel, + qcElement[elementId]->qcOutChannel, peData, nChannels); + redPeGlobal += peData->pe; + + } /* EOF DSE-suppression */ + } /* EOF for all elements... */ + + iter++; + } /* EOF while */ + + /* ------------------------------------------------------- */ + /* Part IV: if still required, further reduce constraints */ + /* ------------------------------------------------------- */ + /* 1.0* 1.15* 1.20* + * desiredPe desiredPe desiredPe + * | | | + * ...XXXXXXXXXXXXXXXXXXXXXXXXXXX| | + * | | |XXXXXXXXXXX... + * | |XXXXXXXXXXX| + * --- A --- | --- B --- | --- C --- + * + * (X): redPeGlobal + * (A): FDKaacEnc_correctThresh() + * (B): FDKaacEnc_allowMoreHoles() + * (C): FDKaacEnc_reduceMinSnr() + */ + + /* correct thresholds to get closer to the desired pe */ + if (redPeGlobal > desiredPe) { + FDKaacEnc_correctThresh(cm, qcElement, psyOutElement, pAhFlag, pThrExp, + reductionValue_m, reductionValue_e, + desiredPe - redPeGlobal, processElements, + elementOffset); + + /* update PE */ + redPeGlobal = 0; + for (elementId = elementOffset; elementId < nElements; elementId++) { + if (cm->elInfo[elementId].elType != ID_DSE) { + INT nChannels = cm->elInfo[elementId].nChannelsInEl; + PE_DATA *peData = &qcElement[elementId]->peData; + + /* pe after correctThresh */ + FDKaacEnc_calcPe(psyOutElement[elementId]->psyOutChannel, + qcElement[elementId]->qcOutChannel, peData, nChannels); + redPeGlobal += peData->pe; + + } /* EOF DSE-suppression */ + } /* EOF for all elements... */ + } + + if (redPeGlobal > desiredPe) { + /* reduce pe by reducing minSnr requirements */ + FDKaacEnc_reduceMinSnr( + cm, qcElement, psyOutElement, pAhFlag, + (fMultI(FL2FXCONST_DBL(0.15f), desiredPe) + desiredPe), &redPeGlobal, + processElements, elementOffset); + + /* reduce pe by allowing additional spectral holes */ + FDKaacEnc_allowMoreHoles(cm, qcElement, psyOutElement, AdjThrStateElement, + pAhFlag, desiredPe, redPeGlobal, processElements, + elementOffset); + } +} + +/* similar to FDKaacEnc_adaptThresholdsToPe(), for VBR-mode */ +static void FDKaacEnc_AdaptThresholdsVBR( + QC_OUT_CHANNEL *const qcOutChannel[(2)], + const PSY_OUT_CHANNEL *const psyOutChannel[(2)], + ATS_ELEMENT *const AdjThrStateElement, + const struct TOOLSINFO *const toolsInfo, const INT nChannels) { + UCHAR(*pAhFlag)[MAX_GROUPED_SFB]; + FIXP_DBL(*pThrExp)[MAX_GROUPED_SFB]; + + /* allocate scratch memory */ + C_ALLOC_SCRATCH_START(_pAhFlag, UCHAR, (2) * MAX_GROUPED_SFB) + C_ALLOC_SCRATCH_START(_pThrExp, FIXP_DBL, (2) * MAX_GROUPED_SFB) + pAhFlag = (UCHAR(*)[MAX_GROUPED_SFB])_pAhFlag; + pThrExp = (FIXP_DBL(*)[MAX_GROUPED_SFB])_pThrExp; + + /* thresholds to the power of redExp */ + FDKaacEnc_calcThreshExp(pThrExp, qcOutChannel, psyOutChannel, nChannels); + + /* lower the minSnr requirements for low energies compared to the average + energy in this frame */ + FDKaacEnc_adaptMinSnr(qcOutChannel, psyOutChannel, + &AdjThrStateElement->minSnrAdaptParam, nChannels); + + /* init ahFlag (0: no ah necessary, 1: ah possible, 2: ah active */ + FDKaacEnc_initAvoidHoleFlag(qcOutChannel, psyOutChannel, pAhFlag, toolsInfo, + nChannels, &AdjThrStateElement->ahParam); + + /* reduce thresholds */ + FDKaacEnc_reduceThresholdsVBR(qcOutChannel, psyOutChannel, pAhFlag, pThrExp, + nChannels, AdjThrStateElement->vbrQualFactor, + &AdjThrStateElement->chaosMeasureOld); + + /* free scratch memory */ + C_ALLOC_SCRATCH_END(_pThrExp, FIXP_DBL, (2) * MAX_GROUPED_SFB) + C_ALLOC_SCRATCH_END(_pAhFlag, UCHAR, (2) * MAX_GROUPED_SFB) +} + +/***************************************************************************** + + functionname: FDKaacEnc_calcBitSave + description: Calculates percentage of bit save, see figure below + returns: + input: parameters and bitres-fullness + output: percentage of bit save + +*****************************************************************************/ +/* + bitsave + maxBitSave(%)| clipLow + |---\ + | \ + | \ + | \ + | \ + |--------\--------------> bitres + | \ + minBitSave(%)| \------------ + clipHigh maxBitres +*/ +static FIXP_DBL FDKaacEnc_calcBitSave(FIXP_DBL fillLevel, + const FIXP_DBL clipLow, + const FIXP_DBL clipHigh, + const FIXP_DBL minBitSave, + const FIXP_DBL maxBitSave, + const FIXP_DBL bitsave_slope) { + FIXP_DBL bitsave; + + fillLevel = fixMax(fillLevel, clipLow); + fillLevel = fixMin(fillLevel, clipHigh); + + bitsave = maxBitSave - fMult((fillLevel - clipLow), bitsave_slope); + + return (bitsave); +} + +/***************************************************************************** + + functionname: FDKaacEnc_calcBitSpend + description: Calculates percentage of bit spend, see figure below + returns: + input: parameters and bitres-fullness + output: percentage of bit spend + +*****************************************************************************/ +/* + bitspend clipHigh + maxBitSpend(%)| /-----------maxBitres + | / + | / + | / + | / + | / + |----/-----------------> bitres + | / + minBitSpend(%)|--/ + clipLow +*/ +static FIXP_DBL FDKaacEnc_calcBitSpend(FIXP_DBL fillLevel, + const FIXP_DBL clipLow, + const FIXP_DBL clipHigh, + const FIXP_DBL minBitSpend, + const FIXP_DBL maxBitSpend, + const FIXP_DBL bitspend_slope) { + FIXP_DBL bitspend; + + fillLevel = fixMax(fillLevel, clipLow); + fillLevel = fixMin(fillLevel, clipHigh); + + bitspend = minBitSpend + fMult(fillLevel - clipLow, bitspend_slope); + + return (bitspend); +} + +/***************************************************************************** + + functionname: FDKaacEnc_adjustPeMinMax() + description: adjusts peMin and peMax parameters over time + returns: + input: current pe, peMin, peMax, bitres size + output: adjusted peMin/peMax + +*****************************************************************************/ +static void FDKaacEnc_adjustPeMinMax(const INT currPe, INT *peMin, INT *peMax) { + FIXP_DBL minFacHi = FL2FXCONST_DBL(0.3f), maxFacHi = (FIXP_DBL)MAXVAL_DBL, + minFacLo = FL2FXCONST_DBL(0.14f), maxFacLo = FL2FXCONST_DBL(0.07f); + INT diff; + + INT minDiff_fix = fMultI(FL2FXCONST_DBL(0.1666666667f), currPe); + + if (currPe > *peMax) { + diff = (currPe - *peMax); + *peMin += fMultI(minFacHi, diff); + *peMax += fMultI(maxFacHi, diff); + } else if (currPe < *peMin) { + diff = (*peMin - currPe); + *peMin -= fMultI(minFacLo, diff); + *peMax -= fMultI(maxFacLo, diff); + } else { + *peMin += fMultI(minFacHi, (currPe - *peMin)); + *peMax -= fMultI(maxFacLo, (*peMax - currPe)); + } + + if ((*peMax - *peMin) < minDiff_fix) { + INT peMax_fix = *peMax, peMin_fix = *peMin; + FIXP_DBL partLo_fix, partHi_fix; + + partLo_fix = (FIXP_DBL)fixMax(0, currPe - peMin_fix); + partHi_fix = (FIXP_DBL)fixMax(0, peMax_fix - currPe); + + peMax_fix = + (INT)(currPe + fMultI(fDivNorm(partHi_fix, (partLo_fix + partHi_fix)), + minDiff_fix)); + peMin_fix = + (INT)(currPe - fMultI(fDivNorm(partLo_fix, (partLo_fix + partHi_fix)), + minDiff_fix)); + peMin_fix = fixMax(0, peMin_fix); + + *peMax = peMax_fix; + *peMin = peMin_fix; + } +} + +/***************************************************************************** + + functionname: BitresCalcBitFac + description: calculates factor of spending bits for one frame + 1.0 : take all frame dynpart bits + >1.0 : take all frame dynpart bits + bitres + <1.0 : put bits in bitreservoir + returns: BitFac + input: bitres-fullness, pe, blockType, parameter-settings + output: + +*****************************************************************************/ +/* + bitfac(%) pemax + bitspend(%) | /-----------maxBitres + | / + | / + | / + | / + | / + |----/-----------------> pe + | / + bitsave(%) |--/ + pemin +*/ + +void FDKaacEnc_bitresCalcBitFac(const INT bitresBits, const INT maxBitresBits, + const INT pe, const INT lastWindowSequence, + const INT avgBits, const FIXP_DBL maxBitFac, + const ADJ_THR_STATE *const AdjThr, + ATS_ELEMENT *const adjThrChan, + FIXP_DBL *const pBitresFac, + INT *const pBitresFac_e) { + const BRES_PARAM *bresParam; + INT pex; + FIXP_DBL fillLevel; + INT fillLevel_e = 0; + + FIXP_DBL bitresFac; + INT bitresFac_e; + + FIXP_DBL bitSave, bitSpend; + FIXP_DBL bitsave_slope, bitspend_slope; + FIXP_DBL fillLevel_fix = MAXVAL_DBL; + + FIXP_DBL slope = MAXVAL_DBL; + + if (lastWindowSequence != SHORT_WINDOW) { + bresParam = &(AdjThr->bresParamLong); + bitsave_slope = FL2FXCONST_DBL(0.466666666); + bitspend_slope = FL2FXCONST_DBL(0.666666666); + } else { + bresParam = &(AdjThr->bresParamShort); + bitsave_slope = (FIXP_DBL)0x2E8BA2E9; + bitspend_slope = (FIXP_DBL)0x7fffffff; + } + + // fillLevel = (float)(bitresBits+avgBits) / (float)(maxBitresBits + avgBits); + if (bitresBits < maxBitresBits) { + fillLevel_fix = fDivNorm(bitresBits, maxBitresBits); + } + + pex = fMax(pe, adjThrChan->peMin); + pex = fMin(pex, adjThrChan->peMax); + + bitSave = FDKaacEnc_calcBitSave( + fillLevel_fix, bresParam->clipSaveLow, bresParam->clipSaveHigh, + bresParam->minBitSave, bresParam->maxBitSave, bitsave_slope); + + bitSpend = FDKaacEnc_calcBitSpend( + fillLevel_fix, bresParam->clipSpendLow, bresParam->clipSpendHigh, + bresParam->minBitSpend, bresParam->maxBitSpend, bitspend_slope); + + slope = schur_div((pex - adjThrChan->peMin), + (adjThrChan->peMax - adjThrChan->peMin), 31); + + /* scale down by 1 bit because the result of the following addition can be + * bigger than 1 (though smaller than 2) */ + bitresFac = ((FIXP_DBL)(MAXVAL_DBL >> 1) - (bitSave >> 1)); + bitresFac_e = 1; /* exp=1 */ + bitresFac = fMultAddDiv2(bitresFac, slope, bitSpend + bitSave); /* exp=1 */ + + /*** limit bitresFac for small bitreservoir ***/ + fillLevel = fDivNorm(bitresBits, avgBits, &fillLevel_e); + if (fillLevel_e < 0) { + fillLevel = scaleValue(fillLevel, fillLevel_e); + fillLevel_e = 0; + } + /* shift down value by 1 because of summation, ... */ + fillLevel >>= 1; + fillLevel_e += 1; + /* ..., this summation: */ + fillLevel += scaleValue(FL2FXCONST_DBL(0.7f), -fillLevel_e); + /* set bitresfactor to same exponent as fillLevel */ + if (scaleValue(bitresFac, -fillLevel_e + 1) > fillLevel) { + bitresFac = fillLevel; + bitresFac_e = fillLevel_e; + } + + /* limit bitresFac for high bitrates */ + if (scaleValue(bitresFac, bitresFac_e - (DFRACT_BITS - 1 - 24)) > maxBitFac) { + bitresFac = maxBitFac; + bitresFac_e = (DFRACT_BITS - 1 - 24); + } + + FDKaacEnc_adjustPeMinMax(pe, &adjThrChan->peMin, &adjThrChan->peMax); + + /* output values */ + *pBitresFac = bitresFac; + *pBitresFac_e = bitresFac_e; +} + +/***************************************************************************** +functionname: FDKaacEnc_AdjThrNew +description: allocate ADJ_THR_STATE +*****************************************************************************/ +INT FDKaacEnc_AdjThrNew(ADJ_THR_STATE **phAdjThr, INT nElements) { + INT err = 0; + INT i; + ADJ_THR_STATE *hAdjThr = GetRam_aacEnc_AdjustThreshold(); + if (hAdjThr == NULL) { + err = 1; + goto bail; + } + + for (i = 0; i < nElements; i++) { + hAdjThr->adjThrStateElem[i] = GetRam_aacEnc_AdjThrStateElement(i); + if (hAdjThr->adjThrStateElem[i] == NULL) { + err = 1; + goto bail; + } + } + +bail: + *phAdjThr = hAdjThr; + return err; +} + +/***************************************************************************** +functionname: FDKaacEnc_AdjThrInit +description: initialize ADJ_THR_STATE +*****************************************************************************/ +void FDKaacEnc_AdjThrInit( + ADJ_THR_STATE *const hAdjThr, const INT meanPe, const INT invQuant, + const CHANNEL_MAPPING *const channelMapping, const INT sampleRate, + const INT totalBitrate, const INT isLowDelay, + const AACENC_BITRES_MODE bitResMode, const INT dZoneQuantEnable, + const INT bitDistributionMode, const FIXP_DBL vbrQualFactor) { + INT i; + + FIXP_DBL POINT8 = FL2FXCONST_DBL(0.8f); + FIXP_DBL POINT6 = FL2FXCONST_DBL(0.6f); + + if (bitDistributionMode == 1) { + hAdjThr->bitDistributionMode = AACENC_BD_MODE_INTRA_ELEMENT; + } else { + hAdjThr->bitDistributionMode = AACENC_BD_MODE_INTER_ELEMENT; + } + + /* Max number of iterations in second guess is 3 for lowdelay aot and for + configurations with multiple audio elements in general, otherwise iteration + value is always 1. */ + hAdjThr->maxIter2ndGuess = + (isLowDelay != 0 || channelMapping->nElements > 1) ? 3 : 1; + + /* common for all elements: */ + /* parameters for bitres control */ + hAdjThr->bresParamLong.clipSaveLow = + (FIXP_DBL)0x1999999a; /* FL2FXCONST_DBL(0.2f); */ + hAdjThr->bresParamLong.clipSaveHigh = + (FIXP_DBL)0x7999999a; /* FL2FXCONST_DBL(0.95f); */ + hAdjThr->bresParamLong.minBitSave = + (FIXP_DBL)0xf999999a; /* FL2FXCONST_DBL(-0.05f); */ + hAdjThr->bresParamLong.maxBitSave = + (FIXP_DBL)0x26666666; /* FL2FXCONST_DBL(0.3f); */ + hAdjThr->bresParamLong.clipSpendLow = + (FIXP_DBL)0x1999999a; /* FL2FXCONST_DBL(0.2f); */ + hAdjThr->bresParamLong.clipSpendHigh = + (FIXP_DBL)0x7999999a; /* FL2FXCONST_DBL(0.95f); */ + hAdjThr->bresParamLong.minBitSpend = + (FIXP_DBL)0xf3333333; /* FL2FXCONST_DBL(-0.10f); */ + hAdjThr->bresParamLong.maxBitSpend = + (FIXP_DBL)0x33333333; /* FL2FXCONST_DBL(0.4f); */ + + hAdjThr->bresParamShort.clipSaveLow = + (FIXP_DBL)0x199999a0; /* FL2FXCONST_DBL(0.2f); */ + hAdjThr->bresParamShort.clipSaveHigh = + (FIXP_DBL)0x5fffffff; /* FL2FXCONST_DBL(0.75f); */ + hAdjThr->bresParamShort.minBitSave = + (FIXP_DBL)0x00000000; /* FL2FXCONST_DBL(0.0f); */ + hAdjThr->bresParamShort.maxBitSave = + (FIXP_DBL)0x199999a0; /* FL2FXCONST_DBL(0.2f); */ + hAdjThr->bresParamShort.clipSpendLow = + (FIXP_DBL)0x199999a0; /* FL2FXCONST_DBL(0.2f); */ + hAdjThr->bresParamShort.clipSpendHigh = + (FIXP_DBL)0x5fffffff; /* FL2FXCONST_DBL(0.75f); */ + hAdjThr->bresParamShort.minBitSpend = + (FIXP_DBL)0xf9999998; /* FL2FXCONST_DBL(-0.05f); */ + hAdjThr->bresParamShort.maxBitSpend = + (FIXP_DBL)0x40000000; /* FL2FXCONST_DBL(0.5f); */ + + /* specific for each element: */ + for (i = 0; i < channelMapping->nElements; i++) { + const FIXP_DBL relativeBits = channelMapping->elInfo[i].relativeBits; + const INT nChannelsInElement = channelMapping->elInfo[i].nChannelsInEl; + const INT bitrateInElement = + (relativeBits != (FIXP_DBL)MAXVAL_DBL) + ? (INT)fMultNorm(relativeBits, (FIXP_DBL)totalBitrate) + : totalBitrate; + const INT chBitrate = bitrateInElement >> (nChannelsInElement == 1 ? 0 : 1); + + ATS_ELEMENT *atsElem = hAdjThr->adjThrStateElem[i]; + MINSNR_ADAPT_PARAM *msaParam = &atsElem->minSnrAdaptParam; + + /* parameters for bitres control */ + if (isLowDelay) { + atsElem->peMin = fMultI(POINT8, meanPe); + atsElem->peMax = fMultI(POINT6, meanPe) << 1; + } else { + atsElem->peMin = fMultI(POINT8, meanPe) >> 1; + atsElem->peMax = fMultI(POINT6, meanPe); + } + + /* for use in FDKaacEnc_reduceThresholdsVBR */ + atsElem->chaosMeasureOld = FL2FXCONST_DBL(0.3f); + + /* additional pe offset to correct pe2bits for low bitrates */ + /* ---- no longer necessary, set by table ----- */ + atsElem->peOffset = 0; + + /* vbr initialisation */ + atsElem->vbrQualFactor = vbrQualFactor; + if (chBitrate < 32000) { + atsElem->peOffset = + fixMax(50, 100 - fMultI((FIXP_DBL)0x666667, chBitrate)); + } + + /* avoid hole parameters */ + if (chBitrate >= 20000) { + atsElem->ahParam.modifyMinSnr = TRUE; + atsElem->ahParam.startSfbL = 15; + atsElem->ahParam.startSfbS = 3; + } else { + atsElem->ahParam.modifyMinSnr = FALSE; + atsElem->ahParam.startSfbL = 0; + atsElem->ahParam.startSfbS = 0; + } + + /* minSnr adaptation */ + msaParam->maxRed = FL2FXCONST_DBL(0.00390625f); /* 0.25f/64.0f */ + /* start adaptation of minSnr for avgEn/sfbEn > startRatio */ + msaParam->startRatio = FL2FXCONST_DBL(0.05190512648f); /* ld64(10.0f) */ + /* maximum minSnr reduction to minSnr^maxRed is reached for + avgEn/sfbEn >= maxRatio */ + /* msaParam->maxRatio = 1000.0f; */ + /*msaParam->redRatioFac = ((float)1.0f - msaParam->maxRed) / + * ((float)10.0f*log10(msaParam->startRatio/msaParam->maxRatio)/log10(2.0f)*(float)0.3010299956f);*/ + msaParam->redRatioFac = FL2FXCONST_DBL(-0.375f); /* -0.0375f * 10.0f */ + /*msaParam->redOffs = (float)1.0f - msaParam->redRatioFac * (float)10.0f * + * log10(msaParam->startRatio)/log10(2.0f) * (float)0.3010299956f;*/ + msaParam->redOffs = FL2FXCONST_DBL(0.021484375); /* 1.375f/64.0f */ + + /* init pe correction */ + atsElem->peCorrectionFactor_m = FL2FXCONST_DBL(0.5f); /* 1.0 */ + atsElem->peCorrectionFactor_e = 1; + + atsElem->dynBitsLast = -1; + atsElem->peLast = 0; + + /* init bits to pe factor */ + + /* init bits2PeFactor */ + FDKaacEnc_InitBits2PeFactor( + &atsElem->bits2PeFactor_m, &atsElem->bits2PeFactor_e, bitrateInElement, + nChannelsInElement, sampleRate, isLowDelay, dZoneQuantEnable, invQuant); + + } /* for nElements */ +} + +/***************************************************************************** + functionname: FDKaacEnc_FDKaacEnc_calcPeCorrection + description: calc desired pe +*****************************************************************************/ +static void FDKaacEnc_FDKaacEnc_calcPeCorrection( + FIXP_DBL *const correctionFac_m, INT *const correctionFac_e, + const INT peAct, const INT peLast, const INT bitsLast, + const FIXP_DBL bits2PeFactor_m, const INT bits2PeFactor_e) { + if ((bitsLast > 0) && (peAct < 1.5f * peLast) && (peAct > 0.7f * peLast) && + (FDKaacEnc_bits2pe2(bitsLast, + fMult(FL2FXCONST_DBL(1.2f / 2.f), bits2PeFactor_m), + bits2PeFactor_e + 1) > peLast) && + (FDKaacEnc_bits2pe2(bitsLast, + fMult(FL2FXCONST_DBL(0.65f), bits2PeFactor_m), + bits2PeFactor_e) < peLast)) { + FIXP_DBL corrFac = *correctionFac_m; + + int scaling = 0; + FIXP_DBL denum = (FIXP_DBL)FDKaacEnc_bits2pe2(bitsLast, bits2PeFactor_m, + bits2PeFactor_e); + FIXP_DBL newFac = fDivNorm((FIXP_DBL)peLast, denum, &scaling); + + /* dead zone, newFac and corrFac are scaled by 0.5 */ + if ((FIXP_DBL)peLast <= denum) { /* ratio <= 1.f */ + newFac = fixMax( + scaleValue(fixMin(fMult(FL2FXCONST_DBL(1.1f / 2.f), newFac), + scaleValue(FL2FXCONST_DBL(1.f / 2.f), -scaling)), + scaling), + FL2FXCONST_DBL(0.85f / 2.f)); + } else { /* ratio < 1.f */ + newFac = fixMax( + fixMin(scaleValue(fMult(FL2FXCONST_DBL(0.9f / 2.f), newFac), scaling), + FL2FXCONST_DBL(1.15f / 2.f)), + FL2FXCONST_DBL(1.f / 2.f)); + } + + if (((newFac > FL2FXCONST_DBL(1.f / 2.f)) && + (corrFac < FL2FXCONST_DBL(1.f / 2.f))) || + ((newFac < FL2FXCONST_DBL(1.f / 2.f)) && + (corrFac > FL2FXCONST_DBL(1.f / 2.f)))) { + corrFac = FL2FXCONST_DBL(1.f / 2.f); + } + + /* faster adaptation towards 1.0, slower in the other direction */ + if ((corrFac < FL2FXCONST_DBL(1.f / 2.f) && newFac < corrFac) || + (corrFac > FL2FXCONST_DBL(1.f / 2.f) && newFac > corrFac)) { + corrFac = fMult(FL2FXCONST_DBL(0.85f), corrFac) + + fMult(FL2FXCONST_DBL(0.15f), newFac); + } else { + corrFac = fMult(FL2FXCONST_DBL(0.7f), corrFac) + + fMult(FL2FXCONST_DBL(0.3f), newFac); + } + + corrFac = fixMax(fixMin(corrFac, FL2FXCONST_DBL(1.15f / 2.f)), + FL2FXCONST_DBL(0.85 / 2.f)); + + *correctionFac_m = corrFac; + *correctionFac_e = 1; + } else { + *correctionFac_m = FL2FXCONST_DBL(1.f / 2.f); + *correctionFac_e = 1; + } +} + +static void FDKaacEnc_calcPeCorrectionLowBitRes( + FIXP_DBL *const correctionFac_m, INT *const correctionFac_e, + const INT peLast, const INT bitsLast, const INT bitresLevel, + const INT nChannels, const FIXP_DBL bits2PeFactor_m, + const INT bits2PeFactor_e) { + /* tuning params */ + const FIXP_DBL amp = FL2FXCONST_DBL(0.005); + const FIXP_DBL maxDiff = FL2FXCONST_DBL(0.25f); + + if (bitsLast > 0) { + /* Estimate deviation of granted and used dynamic bits in previous frame, in + * PE units */ + const int bitsBalLast = + peLast - FDKaacEnc_bits2pe2(bitsLast, bits2PeFactor_m, bits2PeFactor_e); + + /* reserve n bits per channel */ + int headroom = (bitresLevel >= 50 * nChannels) ? 0 : (100 * nChannels); + + /* in PE units */ + headroom = FDKaacEnc_bits2pe2(headroom, bits2PeFactor_m, bits2PeFactor_e); + + /* + * diff = amp * ((bitsBalLast - headroom) / (bitresLevel + headroom) + * diff = max ( min ( diff, maxDiff, -maxDiff)) / 2 + */ + FIXP_DBL denominator = (FIXP_DBL)FDKaacEnc_bits2pe2( + bitresLevel, bits2PeFactor_m, bits2PeFactor_e) + + (FIXP_DBL)headroom; + + int scaling = 0; + FIXP_DBL diff = + (bitsBalLast >= headroom) + ? fMult(amp, fDivNorm((FIXP_DBL)(bitsBalLast - headroom), + denominator, &scaling)) + : -fMult(amp, fDivNorm(-(FIXP_DBL)(bitsBalLast - headroom), + denominator, &scaling)); + + scaling -= 1; /* divide by 2 */ + + diff = (scaling <= 0) + ? fMax(fMin(diff >> (-scaling), maxDiff >> 1), -maxDiff >> 1) + : fMax(fMin(diff, maxDiff >> (1 + scaling)), + -maxDiff >> (1 + scaling)) + << scaling; + + /* + * corrFac += diff + * corrFac = max ( min ( corrFac/2.f, 1.f/2.f, 0.75f/2.f ) ) + */ + *correctionFac_m = + fMax(fMin((*correctionFac_m) + diff, FL2FXCONST_DBL(1.0f / 2.f)), + FL2FXCONST_DBL(0.75f / 2.f)); + *correctionFac_e = 1; + } else { + *correctionFac_m = FL2FXCONST_DBL(0.75 / 2.f); + *correctionFac_e = 1; + } +} + +void FDKaacEnc_DistributeBits( + ADJ_THR_STATE *adjThrState, ATS_ELEMENT *AdjThrStateElement, + PSY_OUT_CHANNEL *psyOutChannel[(2)], PE_DATA *peData, INT *grantedPe, + INT *grantedPeCorr, const INT nChannels, const INT commonWindow, + const INT grantedDynBits, const INT bitresBits, const INT maxBitresBits, + const FIXP_DBL maxBitFac, const AACENC_BITRES_MODE bitResMode) { + FIXP_DBL bitFactor; + INT bitFactor_e; + INT noRedPe = peData->pe; + + /* prefer short windows for calculation of bitFactor */ + INT curWindowSequence = LONG_WINDOW; + if (nChannels == 2) { + if ((psyOutChannel[0]->lastWindowSequence == SHORT_WINDOW) || + (psyOutChannel[1]->lastWindowSequence == SHORT_WINDOW)) { + curWindowSequence = SHORT_WINDOW; + } + } else { + curWindowSequence = psyOutChannel[0]->lastWindowSequence; + } + + if (grantedDynBits >= 1) { + if (bitResMode != AACENC_BR_MODE_FULL) { + /* small or disabled bitreservoir */ + *grantedPe = FDKaacEnc_bits2pe2(grantedDynBits, + AdjThrStateElement->bits2PeFactor_m, + AdjThrStateElement->bits2PeFactor_e); + } else { + /* factor dependend on current fill level and pe */ + FDKaacEnc_bitresCalcBitFac( + bitresBits, maxBitresBits, noRedPe, curWindowSequence, grantedDynBits, + maxBitFac, adjThrState, AdjThrStateElement, &bitFactor, &bitFactor_e); + + /* desired pe for actual frame */ + /* Worst case max of grantedDynBits is = 1024 * 5.27 * 2 */ + *grantedPe = FDKaacEnc_bits2pe2( + grantedDynBits, fMult(bitFactor, AdjThrStateElement->bits2PeFactor_m), + AdjThrStateElement->bits2PeFactor_e + bitFactor_e); + } + } else { + *grantedPe = 0; /* prevent divsion by 0 */ + } + + /* correction of pe value */ + switch (bitResMode) { + case AACENC_BR_MODE_DISABLED: + case AACENC_BR_MODE_REDUCED: + /* correction of pe value for low bitres */ + FDKaacEnc_calcPeCorrectionLowBitRes( + &AdjThrStateElement->peCorrectionFactor_m, + &AdjThrStateElement->peCorrectionFactor_e, AdjThrStateElement->peLast, + AdjThrStateElement->dynBitsLast, bitresBits, nChannels, + AdjThrStateElement->bits2PeFactor_m, + AdjThrStateElement->bits2PeFactor_e); + break; + case AACENC_BR_MODE_FULL: + default: + /* correction of pe value for high bitres */ + FDKaacEnc_FDKaacEnc_calcPeCorrection( + &AdjThrStateElement->peCorrectionFactor_m, + &AdjThrStateElement->peCorrectionFactor_e, + fixMin(*grantedPe, noRedPe), AdjThrStateElement->peLast, + AdjThrStateElement->dynBitsLast, AdjThrStateElement->bits2PeFactor_m, + AdjThrStateElement->bits2PeFactor_e); + break; + } + + *grantedPeCorr = + (INT)(fMult((FIXP_DBL)(*grantedPe << Q_AVGBITS), + AdjThrStateElement->peCorrectionFactor_m) >> + (Q_AVGBITS - AdjThrStateElement->peCorrectionFactor_e)); + + /* update last pe */ + AdjThrStateElement->peLast = *grantedPe; + AdjThrStateElement->dynBitsLast = -1; +} + +/***************************************************************************** +functionname: FDKaacEnc_AdjustThresholds +description: adjust thresholds +*****************************************************************************/ +void FDKaacEnc_AdjustThresholds( + ADJ_THR_STATE *const hAdjThr, QC_OUT_ELEMENT *const qcElement[((8))], + QC_OUT *const qcOut, const PSY_OUT_ELEMENT *const psyOutElement[((8))], + const INT CBRbitrateMode, const CHANNEL_MAPPING *const cm) { + int i; + + if (CBRbitrateMode) { + /* In case, no bits must be shifted between different elements, */ + /* an element-wise execution of the pe-dependent threshold- */ + /* adaption becomes necessary... */ + if (hAdjThr->bitDistributionMode == AACENC_BD_MODE_INTRA_ELEMENT) { + for (i = 0; i < cm->nElements; i++) { + ELEMENT_INFO elInfo = cm->elInfo[i]; + + if ((elInfo.elType == ID_SCE) || (elInfo.elType == ID_CPE) || + (elInfo.elType == ID_LFE)) { + /* qcElement[i]->grantedPe = 2000; */ /* Use this only for debugging + */ + // if (totalGrantedPeCorr < totalNoRedPe) { + if (qcElement[i]->grantedPeCorr < qcElement[i]->peData.pe) { + /* calc threshold necessary for desired pe */ + FDKaacEnc_adaptThresholdsToPe( + cm, hAdjThr->adjThrStateElem, qcElement, psyOutElement, + qcElement[i]->grantedPeCorr, hAdjThr->maxIter2ndGuess, + 1, /* Process only 1 element */ + i /* Process exactly THIS element */ + ); + } + } /* -end- if(ID_SCE || ID_CPE || ID_LFE) */ + } /* -end- element loop */ + } /* AACENC_BD_MODE_INTRA_ELEMENT */ + else if (hAdjThr->bitDistributionMode == AACENC_BD_MODE_INTER_ELEMENT) { + /* Use global Pe to obtain the thresholds? */ + if (qcOut->totalGrantedPeCorr < qcOut->totalNoRedPe) { + /* add equal loadness quantization noise to match the */ + /* desired pe calc threshold necessary for desired pe */ + /* Now carried out globally to cover all(!) channels. */ + FDKaacEnc_adaptThresholdsToPe(cm, hAdjThr->adjThrStateElem, qcElement, + psyOutElement, qcOut->totalGrantedPeCorr, + hAdjThr->maxIter2ndGuess, + cm->nElements, /* Process all elements */ + 0); /* Process exactly THIS element */ + } else { + /* In case global pe doesn't need to be reduced check each element to + hold estimated bitrate below maximum element bitrate. */ + for (i = 0; i < cm->nElements; i++) { + if ((cm->elInfo[i].elType == ID_SCE) || + (cm->elInfo[i].elType == ID_CPE) || + (cm->elInfo[i].elType == ID_LFE)) { + /* Element pe applies to dynamic bits of maximum element bitrate. */ + const int maxElementPe = FDKaacEnc_bits2pe2( + (cm->elInfo[i].nChannelsInEl * MIN_BUFSIZE_PER_EFF_CHAN) - + qcElement[i]->staticBitsUsed - qcElement[i]->extBitsUsed, + hAdjThr->adjThrStateElem[i]->bits2PeFactor_m, + hAdjThr->adjThrStateElem[i]->bits2PeFactor_e); + + if (maxElementPe < qcElement[i]->peData.pe) { + FDKaacEnc_adaptThresholdsToPe( + cm, hAdjThr->adjThrStateElem, qcElement, psyOutElement, + maxElementPe, hAdjThr->maxIter2ndGuess, 1, i); + } + } /* -end- if(ID_SCE || ID_CPE || ID_LFE) */ + } /* -end- element loop */ + } /* (qcOut->totalGrantedPeCorr < qcOut->totalNoRedPe) */ + } /* AACENC_BD_MODE_INTER_ELEMENT */ + } else { + for (i = 0; i < cm->nElements; i++) { + ELEMENT_INFO elInfo = cm->elInfo[i]; + + if ((elInfo.elType == ID_SCE) || (elInfo.elType == ID_CPE) || + (elInfo.elType == ID_LFE)) { + /* for VBR-mode */ + FDKaacEnc_AdaptThresholdsVBR( + qcElement[i]->qcOutChannel, psyOutElement[i]->psyOutChannel, + hAdjThr->adjThrStateElem[i], &psyOutElement[i]->toolsInfo, + cm->elInfo[i].nChannelsInEl); + } /* -end- if(ID_SCE || ID_CPE || ID_LFE) */ + + } /* -end- element loop */ + } + for (i = 0; i < cm->nElements; i++) { + int ch, sfb, sfbGrp; + /* no weighting of threholds and energies for mlout */ + /* weight energies and thresholds */ + for (ch = 0; ch < cm->elInfo[i].nChannelsInEl; ch++) { + QC_OUT_CHANNEL *pQcOutCh = qcElement[i]->qcOutChannel[ch]; + for (sfbGrp = 0; sfbGrp < psyOutElement[i]->psyOutChannel[ch]->sfbCnt; + sfbGrp += psyOutElement[i]->psyOutChannel[ch]->sfbPerGroup) { + for (sfb = 0; sfb < psyOutElement[i]->psyOutChannel[ch]->maxSfbPerGroup; + sfb++) { + pQcOutCh->sfbThresholdLdData[sfb + sfbGrp] += + pQcOutCh->sfbEnFacLd[sfb + sfbGrp]; + } + } + } + } +} + +void FDKaacEnc_AdjThrClose(ADJ_THR_STATE **phAdjThr) { + INT i; + ADJ_THR_STATE *hAdjThr = *phAdjThr; + + if (hAdjThr != NULL) { + for (i = 0; i < ((8)); i++) { + if (hAdjThr->adjThrStateElem[i] != NULL) { + FreeRam_aacEnc_AdjThrStateElement(&hAdjThr->adjThrStateElem[i]); + } + } + FreeRam_aacEnc_AdjustThreshold(phAdjThr); + } +} diff --git a/fdk-aac/libAACenc/src/adj_thr.h b/fdk-aac/libAACenc/src/adj_thr.h new file mode 100644 index 0000000..1f5f998 --- /dev/null +++ b/fdk-aac/libAACenc/src/adj_thr.h @@ -0,0 +1,166 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M. Werner + + Description: Threshold compensation + +*******************************************************************************/ + +#ifndef ADJ_THR_H +#define ADJ_THR_H + +#include "common_fix.h" +#include "adj_thr_data.h" +#include "qc_data.h" +#include "line_pe.h" +#include "interface.h" + +/***************************************************************************** + functionname: FDKaacEnc_peCalculation + description: +*****************************************************************************/ +void FDKaacEnc_peCalculation(PE_DATA *const peData, + const PSY_OUT_CHANNEL *const psyOutChannel[(2)], + QC_OUT_CHANNEL *const qcOutChannel[(2)], + const struct TOOLSINFO *const toolsInfo, + ATS_ELEMENT *const adjThrStateElement, + const INT nChannels); + +/***************************************************************************** +functionname: FDKaacEnc_AdjThrNew +description: allocate ADJ_THR_STATE +*****************************************************************************/ +INT FDKaacEnc_AdjThrNew(ADJ_THR_STATE **phAdjThr, INT nElements); + +/***************************************************************************** +functionname: FDKaacEnc_AdjThrInit +description: initialize ADJ_THR_STATE +*****************************************************************************/ +void FDKaacEnc_AdjThrInit( + ADJ_THR_STATE *const hAdjThr, const INT meanPe, const INT invQuant, + const CHANNEL_MAPPING *const channelMapping, const INT sampleRate, + const INT totalBitrate, const INT isLowDelay, + const AACENC_BITRES_MODE bitResMode, const INT dZoneQuantEnable, + const INT bitDistributionMode, const FIXP_DBL vbrQualFactor); + +/***************************************************************************** +functionname: FDKaacEnc_DistributeBits +description: +*****************************************************************************/ +void FDKaacEnc_DistributeBits( + ADJ_THR_STATE *adjThrState, ATS_ELEMENT *AdjThrStateElement, + PSY_OUT_CHANNEL *psyOutChannel[(2)], PE_DATA *peData, INT *grantedPe, + INT *grantedPeCorr, const INT nChannels, const INT commonWindow, + const INT avgBits, const INT bitresBits, const INT maxBitresBits, + const FIXP_DBL maxBitFac, const AACENC_BITRES_MODE bitResMode); + +/***************************************************************************** +functionname: FDKaacEnc_AdjustThresholds +description: adjust thresholds +*****************************************************************************/ +void FDKaacEnc_AdjustThresholds( + ADJ_THR_STATE *const hAdjThr, QC_OUT_ELEMENT *const qcElement[((8))], + QC_OUT *const qcOut, const PSY_OUT_ELEMENT *const psyOutElement[((8))], + const INT CBRbitrateMode, const CHANNEL_MAPPING *const cm); + +/***************************************************************************** +functionname: FDKaacEnc_AdjThrClose +description: +*****************************************************************************/ +void FDKaacEnc_AdjThrClose(ADJ_THR_STATE **hAdjThr); + +#endif diff --git a/fdk-aac/libAACenc/src/adj_thr_data.h b/fdk-aac/libAACenc/src/adj_thr_data.h new file mode 100644 index 0000000..4cd1299 --- /dev/null +++ b/fdk-aac/libAACenc/src/adj_thr_data.h @@ -0,0 +1,175 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M. Schug / A. Groeschel + + Description: threshold calculations + +*******************************************************************************/ + +#ifndef ADJ_THR_DATA_H +#define ADJ_THR_DATA_H + +#include "psy_const.h" + +typedef enum { + AACENC_BD_MODE_INTER_ELEMENT = 0, + AACENC_BD_MODE_INTRA_ELEMENT = 1 +} AACENC_BIT_DISTRIBUTION_MODE; + +typedef enum { + AACENC_BR_MODE_FULL = 0, + AACENC_BR_MODE_REDUCED = 1, + AACENC_BR_MODE_DISABLED = 2 +} AACENC_BITRES_MODE; + +typedef struct { + FIXP_DBL clipSaveLow, clipSaveHigh; + FIXP_DBL minBitSave, maxBitSave; + FIXP_DBL clipSpendLow, clipSpendHigh; + FIXP_DBL minBitSpend, maxBitSpend; +} BRES_PARAM; + +typedef struct { + INT modifyMinSnr; + INT startSfbL, startSfbS; +} AH_PARAM; + +typedef struct { + FIXP_DBL maxRed; + FIXP_DBL startRatio; + FIXP_DBL maxRatio; + FIXP_DBL redRatioFac; + FIXP_DBL redOffs; +} MINSNR_ADAPT_PARAM; + +typedef struct { + /* parameters for bitreservoir control */ + INT peMin, peMax; + /* constant offset to pe */ + INT peOffset; + /* constant PeFactor */ + FIXP_DBL bits2PeFactor_m; + INT bits2PeFactor_e; + /* avoid hole parameters */ + AH_PARAM ahParam; + /* parameters for adaptation of minSnr */ + MINSNR_ADAPT_PARAM minSnrAdaptParam; + + /* values for correction of pe */ + INT peLast; + INT dynBitsLast; + FIXP_DBL peCorrectionFactor_m; + INT peCorrectionFactor_e; + + /* vbr encoding */ + FIXP_DBL vbrQualFactor; + FIXP_DBL chaosMeasureOld; + + /* threshold weighting */ + FIXP_DBL chaosMeasureEnFac[(2)]; + INT lastEnFacPatch[(2)]; + +} ATS_ELEMENT; + +typedef struct { + BRES_PARAM bresParamLong, bresParamShort; + ATS_ELEMENT* adjThrStateElem[((8))]; + AACENC_BIT_DISTRIBUTION_MODE bitDistributionMode; + INT maxIter2ndGuess; +} ADJ_THR_STATE; + +#endif diff --git a/fdk-aac/libAACenc/src/band_nrg.cpp b/fdk-aac/libAACenc/src/band_nrg.cpp new file mode 100644 index 0000000..fb22dbb --- /dev/null +++ b/fdk-aac/libAACenc/src/band_nrg.cpp @@ -0,0 +1,361 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M. Werner + + Description: Band/Line energy calculations + +*******************************************************************************/ + +#include "band_nrg.h" + +/***************************************************************************** + functionname: FDKaacEnc_CalcSfbMaxScaleSpec + description: + input: + output: +*****************************************************************************/ +void FDKaacEnc_CalcSfbMaxScaleSpec(const FIXP_DBL *RESTRICT mdctSpectrum, + const INT *RESTRICT bandOffset, + INT *RESTRICT sfbMaxScaleSpec, + const INT numBands) { + INT i, j; + FIXP_DBL maxSpc, tmp; + + for (i = 0; i < numBands; i++) { + maxSpc = (FIXP_DBL)0; + + DWORD_ALIGNED(mdctSpectrum); + + for (j = bandOffset[i]; j < bandOffset[i + 1]; j++) { + tmp = fixp_abs(mdctSpectrum[j]); + maxSpc = fixMax(maxSpc, tmp); + } + j = CntLeadingZeros(maxSpc) - 1; + sfbMaxScaleSpec[i] = fixMin((DFRACT_BITS - 2), j); + /* CountLeadingBits() is not necessary here since test value is always > 0 + */ + } +} + +/***************************************************************************** + functionname: FDKaacEnc_CheckBandEnergyOptim + description: + input: + output: +*****************************************************************************/ +FIXP_DBL +FDKaacEnc_CheckBandEnergyOptim(const FIXP_DBL *const RESTRICT mdctSpectrum, + const INT *const RESTRICT sfbMaxScaleSpec, + const INT *const RESTRICT bandOffset, + const INT numBands, + FIXP_DBL *RESTRICT bandEnergy, + FIXP_DBL *RESTRICT bandEnergyLdData, + const INT minSpecShift) { + INT i, j, scale, nr = 0; + FIXP_DBL maxNrgLd = FL2FXCONST_DBL(-1.0f); + FIXP_DBL maxNrg = 0; + FIXP_DBL spec; + + for (i = 0; i < numBands; i++) { + scale = fixMax(0, sfbMaxScaleSpec[i] - 4); + FIXP_DBL tmp = 0; + + DWORD_ALIGNED(mdctSpectrum); + + for (j = bandOffset[i]; j < bandOffset[i + 1]; j++) { + spec = mdctSpectrum[j] << scale; + tmp = fPow2AddDiv2(tmp, spec); + } + bandEnergy[i] = tmp << 1; + + /* calculate ld of bandNrg, subtract scaling */ + bandEnergyLdData[i] = CalcLdData(bandEnergy[i]); + if (bandEnergyLdData[i] != FL2FXCONST_DBL(-1.0f)) { + bandEnergyLdData[i] -= scale * FL2FXCONST_DBL(2.0 / 64); + } + /* find index of maxNrg */ + if (bandEnergyLdData[i] > maxNrgLd) { + maxNrgLd = bandEnergyLdData[i]; + nr = i; + } + } + + /* return unscaled maxNrg*/ + scale = fixMax(0, sfbMaxScaleSpec[nr] - 4); + scale = fixMax(2 * (minSpecShift - scale), -(DFRACT_BITS - 1)); + + maxNrg = scaleValue(bandEnergy[nr], scale); + + return maxNrg; +} + +/***************************************************************************** + functionname: FDKaacEnc_CalcBandEnergyOptimLong + description: + input: + output: +*****************************************************************************/ +INT FDKaacEnc_CalcBandEnergyOptimLong(const FIXP_DBL *RESTRICT mdctSpectrum, + INT *RESTRICT sfbMaxScaleSpec, + const INT *RESTRICT bandOffset, + const INT numBands, + FIXP_DBL *RESTRICT bandEnergy, + FIXP_DBL *RESTRICT bandEnergyLdData) { + INT i, j, shiftBits = 0; + FIXP_DBL maxNrgLd = FL2FXCONST_DBL(0.0f); + + FIXP_DBL spec; + + for (i = 0; i < numBands; i++) { + INT leadingBits = sfbMaxScaleSpec[i] - + 4; /* max sfbWidth = 96 ; 2^7=128 => 7/2 = 4 (spc*spc) */ + FIXP_DBL tmp = FL2FXCONST_DBL(0.0); + /* don't use scaleValue() here, it increases workload quite sufficiently... + */ + if (leadingBits >= 0) { + for (j = bandOffset[i]; j < bandOffset[i + 1]; j++) { + spec = mdctSpectrum[j] << leadingBits; + tmp = fPow2AddDiv2(tmp, spec); + } + } else { + INT shift = -leadingBits; + for (j = bandOffset[i]; j < bandOffset[i + 1]; j++) { + spec = mdctSpectrum[j] >> shift; + tmp = fPow2AddDiv2(tmp, spec); + } + } + bandEnergy[i] = tmp << 1; + } + + /* calculate ld of bandNrg, subtract scaling */ + LdDataVector(bandEnergy, bandEnergyLdData, numBands); + for (i = numBands; i-- != 0;) { + FIXP_DBL scaleDiff = (sfbMaxScaleSpec[i] - 4) * FL2FXCONST_DBL(2.0 / 64); + + bandEnergyLdData[i] = (bandEnergyLdData[i] >= + ((FL2FXCONST_DBL(-1.f) >> 1) + (scaleDiff >> 1))) + ? bandEnergyLdData[i] - scaleDiff + : FL2FXCONST_DBL(-1.f); + /* find maxNrgLd */ + maxNrgLd = fixMax(maxNrgLd, bandEnergyLdData[i]); + } + + if (maxNrgLd <= (FIXP_DBL)0) { + for (i = numBands; i-- != 0;) { + INT scale = fixMin((sfbMaxScaleSpec[i] - 4) << 1, (DFRACT_BITS - 1)); + bandEnergy[i] = scaleValue(bandEnergy[i], -scale); + } + return 0; + } else { /* scale down NRGs */ + while (maxNrgLd > FL2FXCONST_DBL(0.0f)) { + maxNrgLd -= FL2FXCONST_DBL(2.0 / 64); + shiftBits++; + } + for (i = numBands; i-- != 0;) { + INT scale = fixMin(((sfbMaxScaleSpec[i] - 4) + shiftBits) << 1, + (DFRACT_BITS - 1)); + bandEnergyLdData[i] -= shiftBits * FL2FXCONST_DBL(2.0 / 64); + bandEnergy[i] = scaleValue(bandEnergy[i], -scale); + } + return shiftBits; + } +} + +/***************************************************************************** + functionname: FDKaacEnc_CalcBandEnergyOptimShort + description: + input: + output: +*****************************************************************************/ +void FDKaacEnc_CalcBandEnergyOptimShort(const FIXP_DBL *RESTRICT mdctSpectrum, + INT *RESTRICT sfbMaxScaleSpec, + const INT *RESTRICT bandOffset, + const INT numBands, + FIXP_DBL *RESTRICT bandEnergy) { + INT i, j; + + for (i = 0; i < numBands; i++) { + int leadingBits = sfbMaxScaleSpec[i] - + 3; /* max sfbWidth = 36 ; 2^6=64 => 6/2 = 3 (spc*spc) */ + FIXP_DBL tmp = FL2FXCONST_DBL(0.0); + for (j = bandOffset[i]; j < bandOffset[i + 1]; j++) { + FIXP_DBL spec = scaleValue(mdctSpectrum[j], leadingBits); + tmp = fPow2AddDiv2(tmp, spec); + } + bandEnergy[i] = tmp; + } + + for (i = 0; i < numBands; i++) { + INT scale = (2 * (sfbMaxScaleSpec[i] - 3)) - + 1; /* max sfbWidth = 36 ; 2^6=64 => 6/2 = 3 (spc*spc) */ + scale = fixMax(fixMin(scale, (DFRACT_BITS - 1)), -(DFRACT_BITS - 1)); + bandEnergy[i] = scaleValueSaturate(bandEnergy[i], -scale); + } +} + +/***************************************************************************** + functionname: FDKaacEnc_CalcBandNrgMSOpt + description: + input: + output: +*****************************************************************************/ +void FDKaacEnc_CalcBandNrgMSOpt( + const FIXP_DBL *RESTRICT mdctSpectrumLeft, + const FIXP_DBL *RESTRICT mdctSpectrumRight, + INT *RESTRICT sfbMaxScaleSpecLeft, INT *RESTRICT sfbMaxScaleSpecRight, + const INT *RESTRICT bandOffset, const INT numBands, + FIXP_DBL *RESTRICT bandEnergyMid, FIXP_DBL *RESTRICT bandEnergySide, + INT calcLdData, FIXP_DBL *RESTRICT bandEnergyMidLdData, + FIXP_DBL *RESTRICT bandEnergySideLdData) { + INT i, j, minScale; + FIXP_DBL NrgMid, NrgSide, specm, specs; + + for (i = 0; i < numBands; i++) { + NrgMid = NrgSide = FL2FXCONST_DBL(0.0); + minScale = fixMin(sfbMaxScaleSpecLeft[i], sfbMaxScaleSpecRight[i]) - 4; + minScale = fixMax(0, minScale); + + if (minScale > 0) { + for (j = bandOffset[i]; j < bandOffset[i + 1]; j++) { + FIXP_DBL specL = mdctSpectrumLeft[j] << (minScale - 1); + FIXP_DBL specR = mdctSpectrumRight[j] << (minScale - 1); + specm = specL + specR; + specs = specL - specR; + NrgMid = fPow2AddDiv2(NrgMid, specm); + NrgSide = fPow2AddDiv2(NrgSide, specs); + } + } else { + for (j = bandOffset[i]; j < bandOffset[i + 1]; j++) { + FIXP_DBL specL = mdctSpectrumLeft[j] >> 1; + FIXP_DBL specR = mdctSpectrumRight[j] >> 1; + specm = specL + specR; + specs = specL - specR; + NrgMid = fPow2AddDiv2(NrgMid, specm); + NrgSide = fPow2AddDiv2(NrgSide, specs); + } + } + bandEnergyMid[i] = fMin(NrgMid, (FIXP_DBL)MAXVAL_DBL >> 1) << 1; + bandEnergySide[i] = fMin(NrgSide, (FIXP_DBL)MAXVAL_DBL >> 1) << 1; + } + + if (calcLdData) { + LdDataVector(bandEnergyMid, bandEnergyMidLdData, numBands); + LdDataVector(bandEnergySide, bandEnergySideLdData, numBands); + } + + for (i = 0; i < numBands; i++) { + minScale = fixMin(sfbMaxScaleSpecLeft[i], sfbMaxScaleSpecRight[i]); + INT scale = fixMax(0, 2 * (minScale - 4)); + + if (calcLdData) { + /* using the minimal scaling of left and right channel can cause very + small energies; check ldNrg before subtract scaling multiplication: + fract*INT we don't need fMult */ + + int minus = scale * FL2FXCONST_DBL(1.0 / 64); + + if (bandEnergyMidLdData[i] != FL2FXCONST_DBL(-1.0f)) + bandEnergyMidLdData[i] -= minus; + + if (bandEnergySideLdData[i] != FL2FXCONST_DBL(-1.0f)) + bandEnergySideLdData[i] -= minus; + } + scale = fixMin(scale, (DFRACT_BITS - 1)); + bandEnergyMid[i] >>= scale; + bandEnergySide[i] >>= scale; + } +} diff --git a/fdk-aac/libAACenc/src/band_nrg.h b/fdk-aac/libAACenc/src/band_nrg.h new file mode 100644 index 0000000..4137565 --- /dev/null +++ b/fdk-aac/libAACenc/src/band_nrg.h @@ -0,0 +1,142 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M. Werner + + Description: Band/Line energy calculation + +*******************************************************************************/ + +#ifndef BAND_NRG_H +#define BAND_NRG_H + +#include "common_fix.h" + +void FDKaacEnc_CalcSfbMaxScaleSpec(const FIXP_DBL *mdctSpectrum, + const INT *bandOffset, INT *sfbMaxScaleSpec, + const INT numBands); + +FIXP_DBL +FDKaacEnc_CheckBandEnergyOptim(const FIXP_DBL *const RESTRICT mdctSpectrum, + const INT *const RESTRICT sfbMaxScaleSpec, + const INT *const RESTRICT bandOffset, + const INT numBands, + FIXP_DBL *RESTRICT bandEnergy, + FIXP_DBL *RESTRICT bandEnergyLdData, + const INT minSpecShift); + +INT FDKaacEnc_CalcBandEnergyOptimLong(const FIXP_DBL *mdctSpectrum, + INT *sfbMaxScaleSpec, + const INT *bandOffset, const INT numBands, + FIXP_DBL *bandEnergy, + FIXP_DBL *bandEnergyLdData); + +void FDKaacEnc_CalcBandEnergyOptimShort(const FIXP_DBL *mdctSpectrum, + INT *sfbMaxScaleSpec, + const INT *bandOffset, + const INT numBands, + FIXP_DBL *bandEnergy); + +void FDKaacEnc_CalcBandNrgMSOpt( + const FIXP_DBL *RESTRICT mdctSpectrumLeft, + const FIXP_DBL *RESTRICT mdctSpectrumRight, + INT *RESTRICT sfbMaxScaleSpecLeft, INT *RESTRICT sfbMaxScaleSpecRight, + const INT *RESTRICT bandOffset, const INT numBands, + FIXP_DBL *RESTRICT bandEnergyMid, FIXP_DBL *RESTRICT bandEnergySide, + INT calcLdData, FIXP_DBL *RESTRICT bandEnergyMidLdData, + FIXP_DBL *RESTRICT bandEnergySideLdData); + +#endif diff --git a/fdk-aac/libAACenc/src/bandwidth.cpp b/fdk-aac/libAACenc/src/bandwidth.cpp new file mode 100644 index 0000000..36cd64d --- /dev/null +++ b/fdk-aac/libAACenc/src/bandwidth.cpp @@ -0,0 +1,360 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M. Schug / A. Groeschel + + Description: bandwidth expert + +*******************************************************************************/ + +#include "channel_map.h" +#include "bandwidth.h" +#include "aacEnc_ram.h" + +typedef struct { + INT chanBitRate; + INT bandWidthMono; + INT bandWidth2AndMoreChan; + +} BANDWIDTH_TAB; + +static const BANDWIDTH_TAB bandWidthTable[] = { + {0, 3700, 5000}, {12000, 5000, 6400}, {20000, 6900, 9640}, + {28000, 9600, 13050}, {40000, 12060, 14260}, {56000, 13950, 15500}, + {72000, 14200, 16120}, {96000, 17000, 17000}, {576001, 17000, 17000}}; + +static const BANDWIDTH_TAB bandWidthTable_LD_22050[] = { + {8000, 2000, 2400}, {12000, 2500, 2700}, {16000, 3300, 3100}, + {24000, 6250, 7200}, {32000, 9200, 10500}, {40000, 16000, 16000}, + {48000, 16000, 16000}, {282241, 16000, 16000}}; + +static const BANDWIDTH_TAB bandWidthTable_LD_24000[] = { + {8000, 2000, 2000}, {12000, 2000, 2300}, {16000, 2200, 2500}, + {24000, 5650, 7200}, {32000, 11600, 12000}, {40000, 12000, 16000}, + {48000, 16000, 16000}, {64000, 16000, 16000}, {307201, 16000, 16000}}; + +static const BANDWIDTH_TAB bandWidthTable_LD_32000[] = { + {8000, 2000, 2000}, {12000, 2000, 2000}, {24000, 4250, 7200}, + {32000, 8400, 9000}, {40000, 9400, 11300}, {48000, 11900, 14700}, + {64000, 14800, 16000}, {76000, 16000, 16000}, {409601, 16000, 16000}}; + +static const BANDWIDTH_TAB bandWidthTable_LD_44100[] = { + {8000, 2000, 2000}, {24000, 2000, 2000}, {32000, 4400, 5700}, + {40000, 7400, 8800}, {48000, 9000, 10700}, {56000, 11000, 12900}, + {64000, 14400, 15500}, {80000, 16000, 16200}, {96000, 16500, 16000}, + {128000, 16000, 16000}, {564481, 16000, 16000}}; + +static const BANDWIDTH_TAB bandWidthTable_LD_48000[] = { + {8000, 2000, 2000}, {24000, 2000, 2000}, {32000, 4400, 5700}, + {40000, 7400, 8800}, {48000, 9000, 10700}, {56000, 11000, 12800}, + {64000, 14300, 15400}, {80000, 16000, 16200}, {96000, 16500, 16000}, + {128000, 16000, 16000}, {614401, 16000, 16000}}; + +typedef struct { + AACENC_BITRATE_MODE bitrateMode; + int bandWidthMono; + int bandWidth2AndMoreChan; +} BANDWIDTH_TAB_VBR; + +static const BANDWIDTH_TAB_VBR bandWidthTableVBR[] = { + {AACENC_BR_MODE_CBR, 0, 0}, + {AACENC_BR_MODE_VBR_1, 13050, 13050}, + {AACENC_BR_MODE_VBR_2, 13050, 13050}, + {AACENC_BR_MODE_VBR_3, 14260, 14260}, + {AACENC_BR_MODE_VBR_4, 15500, 15500}, + {AACENC_BR_MODE_VBR_5, 48000, 48000}, + {AACENC_BR_MODE_SFR, 0, 0}, + {AACENC_BR_MODE_FF, 0, 0} + +}; + +static INT GetBandwidthEntry(const INT frameLength, const INT sampleRate, + const INT chanBitRate, const INT entryNo) { + INT bandwidth = -1; + const BANDWIDTH_TAB *pBwTab = NULL; + INT bwTabSize = 0; + + switch (frameLength) { + case 960: + case 1024: + pBwTab = bandWidthTable; + bwTabSize = sizeof(bandWidthTable) / sizeof(BANDWIDTH_TAB); + break; + case 120: + case 128: + case 240: + case 256: + case 480: + case 512: + switch (sampleRate) { + case 8000: + case 11025: + case 12000: + case 16000: + case 22050: + pBwTab = bandWidthTable_LD_22050; + bwTabSize = sizeof(bandWidthTable_LD_22050) / sizeof(BANDWIDTH_TAB); + break; + case 24000: + pBwTab = bandWidthTable_LD_24000; + bwTabSize = sizeof(bandWidthTable_LD_24000) / sizeof(BANDWIDTH_TAB); + break; + case 32000: + pBwTab = bandWidthTable_LD_32000; + bwTabSize = sizeof(bandWidthTable_LD_32000) / sizeof(BANDWIDTH_TAB); + break; + case 44100: + pBwTab = bandWidthTable_LD_44100; + bwTabSize = sizeof(bandWidthTable_LD_44100) / sizeof(BANDWIDTH_TAB); + break; + case 48000: + case 64000: + case 88200: + case 96000: + pBwTab = bandWidthTable_LD_48000; + bwTabSize = sizeof(bandWidthTable_LD_48000) / sizeof(BANDWIDTH_TAB); + break; + } + break; + default: + pBwTab = NULL; + bwTabSize = 0; + } + + if (pBwTab != NULL) { + int i; + for (i = 0; i < bwTabSize - 1; i++) { + if (chanBitRate >= pBwTab[i].chanBitRate && + chanBitRate < pBwTab[i + 1].chanBitRate) { + switch (frameLength) { + case 960: + case 1024: + bandwidth = (entryNo == 0) ? pBwTab[i].bandWidthMono + : pBwTab[i].bandWidth2AndMoreChan; + break; + case 120: + case 128: + case 240: + case 256: + case 480: + case 512: { + INT q_res = 0; + INT startBw = (entryNo == 0) ? pBwTab[i].bandWidthMono + : pBwTab[i].bandWidth2AndMoreChan; + INT endBw = (entryNo == 0) ? pBwTab[i + 1].bandWidthMono + : pBwTab[i + 1].bandWidth2AndMoreChan; + INT startBr = pBwTab[i].chanBitRate; + INT endBr = pBwTab[i + 1].chanBitRate; + + FIXP_DBL bwFac_fix = + fDivNorm(chanBitRate - startBr, endBr - startBr, &q_res); + bandwidth = + (INT)scaleValue(fMult(bwFac_fix, (FIXP_DBL)(endBw - startBw)), + q_res) + + startBw; + } break; + default: + bandwidth = -1; + } + break; + } /* within bitrate range */ + } + } /* pBwTab!=NULL */ + + return bandwidth; +} + +AAC_ENCODER_ERROR FDKaacEnc_DetermineBandWidth( + const INT proposedBandWidth, const INT bitrate, + const AACENC_BITRATE_MODE bitrateMode, const INT sampleRate, + const INT frameLength, const CHANNEL_MAPPING *const cm, + const CHANNEL_MODE encoderMode, INT *const bandWidth) { + AAC_ENCODER_ERROR ErrorStatus = AAC_ENC_OK; + INT chanBitRate = bitrate / cm->nChannelsEff; + + switch (bitrateMode) { + case AACENC_BR_MODE_VBR_1: + case AACENC_BR_MODE_VBR_2: + case AACENC_BR_MODE_VBR_3: + case AACENC_BR_MODE_VBR_4: + case AACENC_BR_MODE_VBR_5: + if (proposedBandWidth != 0) { + /* use given bw */ + *bandWidth = proposedBandWidth; + } else { + /* take bw from table */ + switch (encoderMode) { + case MODE_1: + *bandWidth = bandWidthTableVBR[bitrateMode].bandWidthMono; + break; + case MODE_2: + case MODE_1_2: + case MODE_1_2_1: + case MODE_1_2_2: + case MODE_1_2_2_1: + case MODE_6_1: + case MODE_1_2_2_2_1: + case MODE_7_1_REAR_SURROUND: + case MODE_7_1_FRONT_CENTER: + case MODE_7_1_BACK: + case MODE_7_1_TOP_FRONT: + *bandWidth = bandWidthTableVBR[bitrateMode].bandWidth2AndMoreChan; + break; + default: + return AAC_ENC_UNSUPPORTED_CHANNELCONFIG; + } + } + break; + case AACENC_BR_MODE_CBR: + case AACENC_BR_MODE_SFR: + case AACENC_BR_MODE_FF: + + /* bandwidth limiting */ + if (proposedBandWidth != 0) { + *bandWidth = fMin(proposedBandWidth, fMin(20000, sampleRate >> 1)); + } else { /* search reasonable bandwidth */ + + int entryNo = 0; + + switch (encoderMode) { + case MODE_1: /* mono */ + entryNo = 0; /* use mono bandwidth settings */ + break; + + case MODE_2: /* stereo */ + case MODE_1_2: /* sce + cpe */ + case MODE_1_2_1: /* sce + cpe + sce */ + case MODE_1_2_2: /* sce + cpe + cpe */ + case MODE_1_2_2_1: /* (5.1) sce + cpe + cpe + lfe */ + case MODE_6_1: + case MODE_1_2_2_2_1: + case MODE_7_1_REAR_SURROUND: + case MODE_7_1_FRONT_CENTER: + case MODE_7_1_BACK: + case MODE_7_1_TOP_FRONT: + entryNo = 1; /* use stereo bandwidth settings */ + break; + + default: + return AAC_ENC_UNSUPPORTED_CHANNELCONFIG; + } + + *bandWidth = + GetBandwidthEntry(frameLength, sampleRate, chanBitRate, entryNo); + + if (*bandWidth == -1) { + switch (frameLength) { + case 120: + case 128: + case 240: + case 256: + *bandWidth = 16000; + break; + default: + ErrorStatus = AAC_ENC_INVALID_CHANNEL_BITRATE; + } + } + } + break; + default: + *bandWidth = 0; + return AAC_ENC_UNSUPPORTED_BITRATE_MODE; + } + + *bandWidth = fMin(*bandWidth, sampleRate / 2); + + return ErrorStatus; +} diff --git a/fdk-aac/libAACenc/src/bandwidth.h b/fdk-aac/libAACenc/src/bandwidth.h new file mode 100644 index 0000000..088e829 --- /dev/null +++ b/fdk-aac/libAACenc/src/bandwidth.h @@ -0,0 +1,114 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M. Schug / A. Groeschel + + Description: bandwidth expert + +*******************************************************************************/ + +#ifndef BANDWIDTH_H +#define BANDWIDTH_H + +#include "qc_data.h" + +AAC_ENCODER_ERROR FDKaacEnc_DetermineBandWidth( + const INT proposedBandWidth, const INT bitrate, + const AACENC_BITRATE_MODE bitrateMode, const INT sampleRate, + const INT frameLength, const CHANNEL_MAPPING *const cm, + const CHANNEL_MODE encoderMode, INT *const bandWidth); + +#endif /* BANDWIDTH_H */ diff --git a/fdk-aac/libAACenc/src/bit_cnt.cpp b/fdk-aac/libAACenc/src/bit_cnt.cpp new file mode 100644 index 0000000..579df8c --- /dev/null +++ b/fdk-aac/libAACenc/src/bit_cnt.cpp @@ -0,0 +1,950 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M.Werner + + Description: Huffman Bitcounter & coder + +*******************************************************************************/ + +#include "bit_cnt.h" + +#include "aacEnc_ram.h" + +#define HI_LTAB(a) (a >> 16) +#define LO_LTAB(a) (a & 0xffff) + +/***************************************************************************** + + + functionname: FDKaacEnc_count1_2_3_4_5_6_7_8_9_10_11 + description: counts tables 1-11 + returns: + input: quantized spectrum + output: bitCount for tables 1-11 + +*****************************************************************************/ + +static void FDKaacEnc_count1_2_3_4_5_6_7_8_9_10_11(const SHORT *const values, + const INT width, + INT *RESTRICT bitCount) { + INT i; + INT bc1_2, bc3_4, bc5_6, bc7_8, bc9_10, bc11, sc; + INT t0, t1, t2, t3; + bc1_2 = 0; + bc3_4 = 0; + bc5_6 = 0; + bc7_8 = 0; + bc9_10 = 0; + bc11 = 0; + sc = 0; + + DWORD_ALIGNED(values); + + for (i = 0; i < width; i += 4) { + t0 = values[i + 0]; + t1 = values[i + 1]; + t2 = values[i + 2]; + t3 = values[i + 3]; + + bc1_2 += (INT)FDKaacEnc_huff_ltab1_2[t0 + 1][t1 + 1][t2 + 1][t3 + 1]; + bc5_6 += (INT)FDKaacEnc_huff_ltab5_6[t0 + 4][t1 + 4] + + (INT)FDKaacEnc_huff_ltab5_6[t2 + 4][t3 + 4]; + + t0 = fixp_abs(t0); + sc += (t0 > 0); + t1 = fixp_abs(t1); + sc += (t1 > 0); + t2 = fixp_abs(t2); + sc += (t2 > 0); + t3 = fixp_abs(t3); + sc += (t3 > 0); + + bc3_4 += (INT)FDKaacEnc_huff_ltab3_4[t0][t1][t2][t3]; + bc7_8 += (INT)FDKaacEnc_huff_ltab7_8[t0][t1] + + (INT)FDKaacEnc_huff_ltab7_8[t2][t3]; + bc9_10 += (INT)FDKaacEnc_huff_ltab9_10[t0][t1] + + (INT)FDKaacEnc_huff_ltab9_10[t2][t3]; + bc11 += + (INT)FDKaacEnc_huff_ltab11[t0][t1] + (INT)FDKaacEnc_huff_ltab11[t2][t3]; + } + bitCount[1] = HI_LTAB(bc1_2); + bitCount[2] = LO_LTAB(bc1_2); + bitCount[3] = HI_LTAB(bc3_4) + sc; + bitCount[4] = LO_LTAB(bc3_4) + sc; + bitCount[5] = HI_LTAB(bc5_6); + bitCount[6] = LO_LTAB(bc5_6); + bitCount[7] = HI_LTAB(bc7_8) + sc; + bitCount[8] = LO_LTAB(bc7_8) + sc; + bitCount[9] = HI_LTAB(bc9_10) + sc; + bitCount[10] = LO_LTAB(bc9_10) + sc; + bitCount[11] = bc11 + sc; +} + +/***************************************************************************** + + functionname: FDKaacEnc_count3_4_5_6_7_8_9_10_11 + description: counts tables 3-11 + returns: + input: quantized spectrum + output: bitCount for tables 3-11 + +*****************************************************************************/ + +static void FDKaacEnc_count3_4_5_6_7_8_9_10_11(const SHORT *const values, + const INT width, + INT *RESTRICT bitCount) { + INT i; + INT bc3_4, bc5_6, bc7_8, bc9_10, bc11, sc; + INT t0, t1, t2, t3; + + bc3_4 = 0; + bc5_6 = 0; + bc7_8 = 0; + bc9_10 = 0; + bc11 = 0; + sc = 0; + + DWORD_ALIGNED(values); + + for (i = 0; i < width; i += 4) { + t0 = values[i + 0]; + t1 = values[i + 1]; + t2 = values[i + 2]; + t3 = values[i + 3]; + + bc5_6 += (INT)FDKaacEnc_huff_ltab5_6[t0 + 4][t1 + 4] + + (INT)FDKaacEnc_huff_ltab5_6[t2 + 4][t3 + 4]; + + t0 = fixp_abs(t0); + sc += (t0 > 0); + t1 = fixp_abs(t1); + sc += (t1 > 0); + t2 = fixp_abs(t2); + sc += (t2 > 0); + t3 = fixp_abs(t3); + sc += (t3 > 0); + + bc3_4 += (INT)FDKaacEnc_huff_ltab3_4[t0][t1][t2][t3]; + bc7_8 += (INT)FDKaacEnc_huff_ltab7_8[t0][t1] + + (INT)FDKaacEnc_huff_ltab7_8[t2][t3]; + bc9_10 += (INT)FDKaacEnc_huff_ltab9_10[t0][t1] + + (INT)FDKaacEnc_huff_ltab9_10[t2][t3]; + bc11 += + (INT)FDKaacEnc_huff_ltab11[t0][t1] + (INT)FDKaacEnc_huff_ltab11[t2][t3]; + } + + bitCount[1] = INVALID_BITCOUNT; + bitCount[2] = INVALID_BITCOUNT; + bitCount[3] = HI_LTAB(bc3_4) + sc; + bitCount[4] = LO_LTAB(bc3_4) + sc; + bitCount[5] = HI_LTAB(bc5_6); + bitCount[6] = LO_LTAB(bc5_6); + bitCount[7] = HI_LTAB(bc7_8) + sc; + bitCount[8] = LO_LTAB(bc7_8) + sc; + bitCount[9] = HI_LTAB(bc9_10) + sc; + bitCount[10] = LO_LTAB(bc9_10) + sc; + bitCount[11] = bc11 + sc; +} + +/***************************************************************************** + + functionname: FDKaacEnc_count5_6_7_8_9_10_11 + description: counts tables 5-11 + returns: + input: quantized spectrum + output: bitCount for tables 5-11 + +*****************************************************************************/ + +static void FDKaacEnc_count5_6_7_8_9_10_11(const SHORT *const values, + const INT width, + INT *RESTRICT bitCount) { + INT i; + INT bc5_6, bc7_8, bc9_10, bc11, sc; + INT t0, t1, t2, t3; + bc5_6 = 0; + bc7_8 = 0; + bc9_10 = 0; + bc11 = 0; + sc = 0; + + DWORD_ALIGNED(values); + + for (i = 0; i < width; i += 4) { + t0 = values[i + 0]; + t1 = values[i + 1]; + t2 = values[i + 2]; + t3 = values[i + 3]; + + bc5_6 += (INT)FDKaacEnc_huff_ltab5_6[t0 + 4][t1 + 4] + + (INT)FDKaacEnc_huff_ltab5_6[t2 + 4][t3 + 4]; + + t0 = fixp_abs(t0); + sc += (t0 > 0); + t1 = fixp_abs(t1); + sc += (t1 > 0); + t2 = fixp_abs(t2); + sc += (t2 > 0); + t3 = fixp_abs(t3); + sc += (t3 > 0); + + bc7_8 += (INT)FDKaacEnc_huff_ltab7_8[t0][t1] + + (INT)FDKaacEnc_huff_ltab7_8[t2][t3]; + bc9_10 += (INT)FDKaacEnc_huff_ltab9_10[t0][t1] + + (INT)FDKaacEnc_huff_ltab9_10[t2][t3]; + bc11 += + (INT)FDKaacEnc_huff_ltab11[t0][t1] + (INT)FDKaacEnc_huff_ltab11[t2][t3]; + } + bitCount[1] = INVALID_BITCOUNT; + bitCount[2] = INVALID_BITCOUNT; + bitCount[3] = INVALID_BITCOUNT; + bitCount[4] = INVALID_BITCOUNT; + bitCount[5] = HI_LTAB(bc5_6); + bitCount[6] = LO_LTAB(bc5_6); + bitCount[7] = HI_LTAB(bc7_8) + sc; + bitCount[8] = LO_LTAB(bc7_8) + sc; + bitCount[9] = HI_LTAB(bc9_10) + sc; + bitCount[10] = LO_LTAB(bc9_10) + sc; + bitCount[11] = bc11 + sc; +} + +/***************************************************************************** + + functionname: FDKaacEnc_count7_8_9_10_11 + description: counts tables 7-11 + returns: + input: quantized spectrum + output: bitCount for tables 7-11 + +*****************************************************************************/ + +static void FDKaacEnc_count7_8_9_10_11(const SHORT *const values, + const INT width, + INT *RESTRICT bitCount) { + INT i; + INT bc7_8, bc9_10, bc11, sc; + INT t0, t1, t2, t3; + + bc7_8 = 0; + bc9_10 = 0; + bc11 = 0; + sc = 0; + + DWORD_ALIGNED(values); + + for (i = 0; i < width; i += 4) { + t0 = values[i + 0]; + t1 = values[i + 1]; + t2 = values[i + 2]; + t3 = values[i + 3]; + + t0 = fixp_abs(t0); + sc += (t0 > 0); + t1 = fixp_abs(t1); + sc += (t1 > 0); + t2 = fixp_abs(t2); + sc += (t2 > 0); + t3 = fixp_abs(t3); + sc += (t3 > 0); + + bc7_8 += (INT)FDKaacEnc_huff_ltab7_8[t0][t1] + + (INT)FDKaacEnc_huff_ltab7_8[t2][t3]; + bc9_10 += (INT)FDKaacEnc_huff_ltab9_10[t0][t1] + + (INT)FDKaacEnc_huff_ltab9_10[t2][t3]; + bc11 += + (INT)FDKaacEnc_huff_ltab11[t0][t1] + (INT)FDKaacEnc_huff_ltab11[t2][t3]; + } + + bitCount[1] = INVALID_BITCOUNT; + bitCount[2] = INVALID_BITCOUNT; + bitCount[3] = INVALID_BITCOUNT; + bitCount[4] = INVALID_BITCOUNT; + bitCount[5] = INVALID_BITCOUNT; + bitCount[6] = INVALID_BITCOUNT; + bitCount[7] = HI_LTAB(bc7_8) + sc; + bitCount[8] = LO_LTAB(bc7_8) + sc; + bitCount[9] = HI_LTAB(bc9_10) + sc; + bitCount[10] = LO_LTAB(bc9_10) + sc; + bitCount[11] = bc11 + sc; +} + +/***************************************************************************** + + functionname: FDKaacEnc_count9_10_11 + description: counts tables 9-11 + returns: + input: quantized spectrum + output: bitCount for tables 9-11 + +*****************************************************************************/ + +static void FDKaacEnc_count9_10_11(const SHORT *const values, const INT width, + INT *RESTRICT bitCount) { + INT i; + INT bc9_10, bc11, sc; + INT t0, t1, t2, t3; + + bc9_10 = 0; + bc11 = 0; + sc = 0; + + DWORD_ALIGNED(values); + + for (i = 0; i < width; i += 4) { + t0 = values[i + 0]; + t1 = values[i + 1]; + t2 = values[i + 2]; + t3 = values[i + 3]; + + t0 = fixp_abs(t0); + sc += (t0 > 0); + t1 = fixp_abs(t1); + sc += (t1 > 0); + t2 = fixp_abs(t2); + sc += (t2 > 0); + t3 = fixp_abs(t3); + sc += (t3 > 0); + + bc9_10 += (INT)FDKaacEnc_huff_ltab9_10[t0][t1] + + (INT)FDKaacEnc_huff_ltab9_10[t2][t3]; + bc11 += + (INT)FDKaacEnc_huff_ltab11[t0][t1] + (INT)FDKaacEnc_huff_ltab11[t2][t3]; + } + + bitCount[1] = INVALID_BITCOUNT; + bitCount[2] = INVALID_BITCOUNT; + bitCount[3] = INVALID_BITCOUNT; + bitCount[4] = INVALID_BITCOUNT; + bitCount[5] = INVALID_BITCOUNT; + bitCount[6] = INVALID_BITCOUNT; + bitCount[7] = INVALID_BITCOUNT; + bitCount[8] = INVALID_BITCOUNT; + bitCount[9] = HI_LTAB(bc9_10) + sc; + bitCount[10] = LO_LTAB(bc9_10) + sc; + bitCount[11] = bc11 + sc; +} + +/***************************************************************************** + + functionname: FDKaacEnc_count11 + description: counts table 11 + returns: + input: quantized spectrum + output: bitCount for table 11 + +*****************************************************************************/ + +static void FDKaacEnc_count11(const SHORT *const values, const INT width, + INT *RESTRICT bitCount) { + INT i; + INT bc11, sc; + INT t0, t1, t2, t3; + + bc11 = 0; + sc = 0; + + DWORD_ALIGNED(values); + + for (i = 0; i < width; i += 4) { + t0 = values[i + 0]; + t1 = values[i + 1]; + t2 = values[i + 2]; + t3 = values[i + 3]; + + t0 = fixp_abs(t0); + sc += (t0 > 0); + t1 = fixp_abs(t1); + sc += (t1 > 0); + t2 = fixp_abs(t2); + sc += (t2 > 0); + t3 = fixp_abs(t3); + sc += (t3 > 0); + + bc11 += + (INT)FDKaacEnc_huff_ltab11[t0][t1] + (INT)FDKaacEnc_huff_ltab11[t2][t3]; + } + + bitCount[1] = INVALID_BITCOUNT; + bitCount[2] = INVALID_BITCOUNT; + bitCount[3] = INVALID_BITCOUNT; + bitCount[4] = INVALID_BITCOUNT; + bitCount[5] = INVALID_BITCOUNT; + bitCount[6] = INVALID_BITCOUNT; + bitCount[7] = INVALID_BITCOUNT; + bitCount[8] = INVALID_BITCOUNT; + bitCount[9] = INVALID_BITCOUNT; + bitCount[10] = INVALID_BITCOUNT; + bitCount[11] = bc11 + sc; +} + +/***************************************************************************** + + functionname: FDKaacEnc_countEsc + description: counts table 11 (with Esc) + returns: + input: quantized spectrum + output: bitCount for tables 11 (with Esc) + +*****************************************************************************/ + +static void FDKaacEnc_countEsc(const SHORT *const values, const INT width, + INT *RESTRICT bitCount) { + INT i; + INT bc11, ec, sc; + INT t0, t1, t00, t01; + + bc11 = 0; + sc = 0; + ec = 0; + for (i = 0; i < width; i += 2) { + t0 = fixp_abs(values[i + 0]); + t1 = fixp_abs(values[i + 1]); + + sc += (t0 > 0) + (t1 > 0); + + t00 = fixMin(t0, 16); + t01 = fixMin(t1, 16); + bc11 += (INT)FDKaacEnc_huff_ltab11[t00][t01]; + + if (t0 >= 16) { + ec += 5; + while ((t0 >>= 1) >= 16) ec += 2; + } + + if (t1 >= 16) { + ec += 5; + while ((t1 >>= 1) >= 16) ec += 2; + } + } + + for (i = 0; i < 11; i++) bitCount[i] = INVALID_BITCOUNT; + + bitCount[11] = bc11 + sc + ec; +} + +typedef void (*COUNT_FUNCTION)(const SHORT *const values, const INT width, + INT *RESTRICT bitCount); + +static const COUNT_FUNCTION countFuncTable[CODE_BOOK_ESC_LAV + 1] = { + + FDKaacEnc_count1_2_3_4_5_6_7_8_9_10_11, /* 0 */ + FDKaacEnc_count1_2_3_4_5_6_7_8_9_10_11, /* 1 */ + FDKaacEnc_count3_4_5_6_7_8_9_10_11, /* 2 */ + FDKaacEnc_count5_6_7_8_9_10_11, /* 3 */ + FDKaacEnc_count5_6_7_8_9_10_11, /* 4 */ + FDKaacEnc_count7_8_9_10_11, /* 5 */ + FDKaacEnc_count7_8_9_10_11, /* 6 */ + FDKaacEnc_count7_8_9_10_11, /* 7 */ + FDKaacEnc_count9_10_11, /* 8 */ + FDKaacEnc_count9_10_11, /* 9 */ + FDKaacEnc_count9_10_11, /* 10 */ + FDKaacEnc_count9_10_11, /* 11 */ + FDKaacEnc_count9_10_11, /* 12 */ + FDKaacEnc_count11, /* 13 */ + FDKaacEnc_count11, /* 14 */ + FDKaacEnc_count11, /* 15 */ + FDKaacEnc_countEsc /* 16 */ +}; + +INT FDKaacEnc_bitCount(const SHORT *const values, const INT width, + const INT maxVal, INT *const RESTRICT bitCount) { + /* + check if we can use codebook 0 + */ + + bitCount[0] = (maxVal == 0) ? 0 : INVALID_BITCOUNT; + + countFuncTable[fixMin(maxVal, (INT)CODE_BOOK_ESC_LAV)](values, width, + bitCount); + + return (0); +} + +/* + count difference between actual and zeroed lines +*/ +INT FDKaacEnc_countValues(SHORT *RESTRICT values, INT width, INT codeBook) { + INT i, t0, t1, t2, t3; + INT bitCnt = 0; + + switch (codeBook) { + case CODE_BOOK_ZERO_NO: + break; + + case CODE_BOOK_1_NO: + for (i = 0; i < width; i += 4) { + t0 = values[i + 0]; + t1 = values[i + 1]; + t2 = values[i + 2]; + t3 = values[i + 3]; + bitCnt += + HI_LTAB(FDKaacEnc_huff_ltab1_2[t0 + 1][t1 + 1][t2 + 1][t3 + 1]); + } + break; + + case CODE_BOOK_2_NO: + for (i = 0; i < width; i += 4) { + t0 = values[i + 0]; + t1 = values[i + 1]; + t2 = values[i + 2]; + t3 = values[i + 3]; + bitCnt += + LO_LTAB(FDKaacEnc_huff_ltab1_2[t0 + 1][t1 + 1][t2 + 1][t3 + 1]); + } + break; + + case CODE_BOOK_3_NO: + for (i = 0; i < width; i += 4) { + t0 = fixp_abs(values[i + 0]); + bitCnt += (t0 > 0); + t1 = fixp_abs(values[i + 1]); + bitCnt += (t1 > 0); + t2 = fixp_abs(values[i + 2]); + bitCnt += (t2 > 0); + t3 = fixp_abs(values[i + 3]); + bitCnt += (t3 > 0); + bitCnt += HI_LTAB(FDKaacEnc_huff_ltab3_4[t0][t1][t2][t3]); + } + break; + + case CODE_BOOK_4_NO: + for (i = 0; i < width; i += 4) { + t0 = fixp_abs(values[i + 0]); + bitCnt += (t0 > 0); + t1 = fixp_abs(values[i + 1]); + bitCnt += (t1 > 0); + t2 = fixp_abs(values[i + 2]); + bitCnt += (t2 > 0); + t3 = fixp_abs(values[i + 3]); + bitCnt += (t3 > 0); + bitCnt += LO_LTAB(FDKaacEnc_huff_ltab3_4[t0][t1][t2][t3]); + } + break; + + case CODE_BOOK_5_NO: + for (i = 0; i < width; i += 4) { + t0 = values[i + 0]; + t1 = values[i + 1]; + t2 = values[i + 2]; + t3 = values[i + 3]; + bitCnt += HI_LTAB(FDKaacEnc_huff_ltab5_6[t0 + 4][t1 + 4]) + + HI_LTAB(FDKaacEnc_huff_ltab5_6[t2 + 4][t3 + 4]); + } + break; + + case CODE_BOOK_6_NO: + for (i = 0; i < width; i += 4) { + t0 = values[i + 0]; + t1 = values[i + 1]; + t2 = values[i + 2]; + t3 = values[i + 3]; + bitCnt += LO_LTAB(FDKaacEnc_huff_ltab5_6[t0 + 4][t1 + 4]) + + LO_LTAB(FDKaacEnc_huff_ltab5_6[t2 + 4][t3 + 4]); + } + break; + + case CODE_BOOK_7_NO: + for (i = 0; i < width; i += 4) { + t0 = fixp_abs(values[i + 0]); + bitCnt += (t0 > 0); + t1 = fixp_abs(values[i + 1]); + bitCnt += (t1 > 0); + t2 = fixp_abs(values[i + 2]); + bitCnt += (t2 > 0); + t3 = fixp_abs(values[i + 3]); + bitCnt += (t3 > 0); + bitCnt += HI_LTAB(FDKaacEnc_huff_ltab7_8[t0][t1]) + + HI_LTAB(FDKaacEnc_huff_ltab7_8[t2][t3]); + } + break; + + case CODE_BOOK_8_NO: + for (i = 0; i < width; i += 4) { + t0 = fixp_abs(values[i + 0]); + bitCnt += (t0 > 0); + t1 = fixp_abs(values[i + 1]); + bitCnt += (t1 > 0); + t2 = fixp_abs(values[i + 2]); + bitCnt += (t2 > 0); + t3 = fixp_abs(values[i + 3]); + bitCnt += (t3 > 0); + bitCnt += LO_LTAB(FDKaacEnc_huff_ltab7_8[t0][t1]) + + LO_LTAB(FDKaacEnc_huff_ltab7_8[t2][t3]); + } + break; + + case CODE_BOOK_9_NO: + for (i = 0; i < width; i += 4) { + t0 = fixp_abs(values[i + 0]); + bitCnt += (t0 > 0); + t1 = fixp_abs(values[i + 1]); + bitCnt += (t1 > 0); + t2 = fixp_abs(values[i + 2]); + bitCnt += (t2 > 0); + t3 = fixp_abs(values[i + 3]); + bitCnt += (t3 > 0); + bitCnt += HI_LTAB(FDKaacEnc_huff_ltab9_10[t0][t1]) + + HI_LTAB(FDKaacEnc_huff_ltab9_10[t2][t3]); + } + break; + + case CODE_BOOK_10_NO: + for (i = 0; i < width; i += 4) { + t0 = fixp_abs(values[i + 0]); + bitCnt += (t0 > 0); + t1 = fixp_abs(values[i + 1]); + bitCnt += (t1 > 0); + t2 = fixp_abs(values[i + 2]); + bitCnt += (t2 > 0); + t3 = fixp_abs(values[i + 3]); + bitCnt += (t3 > 0); + bitCnt += LO_LTAB(FDKaacEnc_huff_ltab9_10[t0][t1]) + + LO_LTAB(FDKaacEnc_huff_ltab9_10[t2][t3]); + } + break; + + case CODE_BOOK_ESC_NO: + for (i = 0; i < width; i += 2) { + t0 = fixp_abs(values[i + 0]); + bitCnt += (t0 > 0); + t1 = fixp_abs(values[i + 1]); + bitCnt += (t1 > 0); + bitCnt += (INT)FDKaacEnc_huff_ltab11[fixMin(t0, 16)][fixMin(t1, 16)]; + if (t0 >= 16) { + bitCnt += 5; + while ((t0 >>= 1) >= 16) bitCnt += 2; + } + if (t1 >= 16) { + bitCnt += 5; + while ((t1 >>= 1) >= 16) bitCnt += 2; + } + } + break; + + default: + break; + } + + return (bitCnt); +} + +INT FDKaacEnc_codeValues(SHORT *RESTRICT values, INT width, INT codeBook, + HANDLE_FDK_BITSTREAM hBitstream) { + INT i, t0, t1, t2, t3, t00, t01; + INT codeWord, codeLength; + INT sign, signLength; + + DWORD_ALIGNED(values); + + switch (codeBook) { + case CODE_BOOK_ZERO_NO: + break; + + case CODE_BOOK_1_NO: + for (i = 0; i < width; i += 4) { + t0 = values[i + 0] + 1; + t1 = values[i + 1] + 1; + t2 = values[i + 2] + 1; + t3 = values[i + 3] + 1; + codeWord = FDKaacEnc_huff_ctab1[t0][t1][t2][t3]; + codeLength = HI_LTAB(FDKaacEnc_huff_ltab1_2[t0][t1][t2][t3]); + FDKwriteBits(hBitstream, codeWord, codeLength); + } + break; + + case CODE_BOOK_2_NO: + for (i = 0; i < width; i += 4) { + t0 = values[i + 0] + 1; + t1 = values[i + 1] + 1; + t2 = values[i + 2] + 1; + t3 = values[i + 3] + 1; + codeWord = FDKaacEnc_huff_ctab2[t0][t1][t2][t3]; + codeLength = LO_LTAB(FDKaacEnc_huff_ltab1_2[t0][t1][t2][t3]); + FDKwriteBits(hBitstream, codeWord, codeLength); + } + break; + + case CODE_BOOK_3_NO: + for (i = 0; i < (width >> 2); i++) { + sign = 0; + signLength = 0; + int index[4]; + for (int j = 0; j < 4; j++) { + int ti = *values++; + int zero = (ti == 0) ? 0 : 1; + signLength += zero; + sign = (sign << zero) + ((UINT)ti >> 31); + index[j] = fixp_abs(ti); + } + codeWord = FDKaacEnc_huff_ctab3[index[0]][index[1]][index[2]][index[3]]; + codeLength = HI_LTAB( + FDKaacEnc_huff_ltab3_4[index[0]][index[1]][index[2]][index[3]]); + FDKwriteBits(hBitstream, (codeWord << signLength) | sign, + codeLength + signLength); + } + break; + + case CODE_BOOK_4_NO: + for (i = 0; i < width; i += 4) { + sign = 0; + signLength = 0; + int index[4]; + for (int j = 0; j < 4; j++) { + int ti = *values++; + int zero = (ti == 0) ? 0 : 1; + signLength += zero; + sign = (sign << zero) + ((UINT)ti >> 31); + index[j] = fixp_abs(ti); + } + codeWord = FDKaacEnc_huff_ctab4[index[0]][index[1]][index[2]][index[3]]; + codeLength = LO_LTAB( + FDKaacEnc_huff_ltab3_4[index[0]][index[1]][index[2]][index[3]]); + FDKwriteBits(hBitstream, (codeWord << signLength) | sign, + codeLength + signLength); + } + break; + + case CODE_BOOK_5_NO: + for (i = 0; i < (width >> 2); i++) { + t0 = *values++ + 4; + t1 = *values++ + 4; + t2 = *values++ + 4; + t3 = *values++ + 4; + codeWord = FDKaacEnc_huff_ctab5[t0][t1]; + codeLength = + HI_LTAB(FDKaacEnc_huff_ltab5_6[t2][t3]); /* length of 2nd cw */ + codeWord = (codeWord << codeLength) + FDKaacEnc_huff_ctab5[t2][t3]; + codeLength += HI_LTAB(FDKaacEnc_huff_ltab5_6[t0][t1]); + FDKwriteBits(hBitstream, codeWord, codeLength); + } + break; + + case CODE_BOOK_6_NO: + for (i = 0; i < (width >> 2); i++) { + t0 = *values++ + 4; + t1 = *values++ + 4; + t2 = *values++ + 4; + t3 = *values++ + 4; + codeWord = FDKaacEnc_huff_ctab6[t0][t1]; + codeLength = + LO_LTAB(FDKaacEnc_huff_ltab5_6[t2][t3]); /* length of 2nd cw */ + codeWord = (codeWord << codeLength) + FDKaacEnc_huff_ctab6[t2][t3]; + codeLength += LO_LTAB(FDKaacEnc_huff_ltab5_6[t0][t1]); + FDKwriteBits(hBitstream, codeWord, codeLength); + } + break; + + case CODE_BOOK_7_NO: + for (i = 0; i < (width >> 1); i++) { + t0 = *values++; + sign = ((UINT)t0 >> 31); + t0 = fixp_abs(t0); + signLength = (t0 == 0) ? 0 : 1; + t1 = *values++; + INT zero = (t1 == 0) ? 0 : 1; + signLength += zero; + sign = (sign << zero) + ((UINT)t1 >> 31); + t1 = fixp_abs(t1); + codeWord = FDKaacEnc_huff_ctab7[t0][t1]; + codeLength = HI_LTAB(FDKaacEnc_huff_ltab7_8[t0][t1]); + FDKwriteBits(hBitstream, (codeWord << signLength) | sign, + codeLength + signLength); + } + break; + + case CODE_BOOK_8_NO: + for (i = 0; i < (width >> 1); i++) { + t0 = *values++; + sign = ((UINT)t0 >> 31); + t0 = fixp_abs(t0); + signLength = (t0 == 0) ? 0 : 1; + t1 = *values++; + INT zero = (t1 == 0) ? 0 : 1; + signLength += zero; + sign = (sign << zero) + ((UINT)t1 >> 31); + t1 = fixp_abs(t1); + codeWord = FDKaacEnc_huff_ctab8[t0][t1]; + codeLength = LO_LTAB(FDKaacEnc_huff_ltab7_8[t0][t1]); + FDKwriteBits(hBitstream, (codeWord << signLength) | sign, + codeLength + signLength); + } + break; + + case CODE_BOOK_9_NO: + for (i = 0; i < (width >> 1); i++) { + t0 = *values++; + sign = ((UINT)t0 >> 31); + t0 = fixp_abs(t0); + signLength = (t0 == 0) ? 0 : 1; + t1 = *values++; + INT zero = (t1 == 0) ? 0 : 1; + signLength += zero; + sign = (sign << zero) + ((UINT)t1 >> 31); + t1 = fixp_abs(t1); + codeWord = FDKaacEnc_huff_ctab9[t0][t1]; + codeLength = HI_LTAB(FDKaacEnc_huff_ltab9_10[t0][t1]); + FDKwriteBits(hBitstream, (codeWord << signLength) | sign, + codeLength + signLength); + } + break; + + case CODE_BOOK_10_NO: + for (i = 0; i < (width >> 1); i++) { + t0 = *values++; + sign = ((UINT)t0 >> 31); + t0 = fixp_abs(t0); + signLength = (t0 == 0) ? 0 : 1; + t1 = *values++; + INT zero = (t1 == 0) ? 0 : 1; + signLength += zero; + sign = (sign << zero) + ((UINT)t1 >> 31); + t1 = fixp_abs(t1); + codeWord = FDKaacEnc_huff_ctab10[t0][t1]; + codeLength = LO_LTAB(FDKaacEnc_huff_ltab9_10[t0][t1]); + FDKwriteBits(hBitstream, (codeWord << signLength) | sign, + codeLength + signLength); + } + break; + + case CODE_BOOK_ESC_NO: + for (i = 0; i < (width >> 1); i++) { + t0 = *values++; + sign = ((UINT)t0 >> 31); + t0 = fixp_abs(t0); + signLength = (t0 == 0) ? 0 : 1; + t1 = *values++; + INT zero = (t1 == 0) ? 0 : 1; + signLength += zero; + sign = (sign << zero) + ((UINT)t1 >> 31); + t1 = fixp_abs(t1); + + t00 = fixMin(t0, 16); + t01 = fixMin(t1, 16); + + codeWord = FDKaacEnc_huff_ctab11[t00][t01]; + codeLength = (INT)FDKaacEnc_huff_ltab11[t00][t01]; + FDKwriteBits(hBitstream, (codeWord << signLength) | sign, + codeLength + signLength); + for (int j = 0; j < 2; j++) { + if (t0 >= 16) { + INT n = 4, p = t0; + for (; (p >>= 1) >= 16;) n++; + FDKwriteBits(hBitstream, + (((1 << (n - 3)) - 2) << n) | (t0 - (1 << n)), + n + n - 3); + } + t0 = t1; + } + } + break; + + default: + break; + } + return (0); +} + +INT FDKaacEnc_codeScalefactorDelta(INT delta, HANDLE_FDK_BITSTREAM hBitstream) { + INT codeWord, codeLength; + + if (fixp_abs(delta) > CODE_BOOK_SCF_LAV) return (1); + + codeWord = FDKaacEnc_huff_ctabscf[delta + CODE_BOOK_SCF_LAV]; + codeLength = (INT)FDKaacEnc_huff_ltabscf[delta + CODE_BOOK_SCF_LAV]; + FDKwriteBits(hBitstream, codeWord, codeLength); + return (0); +} diff --git a/fdk-aac/libAACenc/src/bit_cnt.h b/fdk-aac/libAACenc/src/bit_cnt.h new file mode 100644 index 0000000..7f4c450 --- /dev/null +++ b/fdk-aac/libAACenc/src/bit_cnt.h @@ -0,0 +1,200 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M.Werner + + Description: Huffman Bitcounter & coder + +*******************************************************************************/ + +#ifndef BIT_CNT_H +#define BIT_CNT_H + +#include "common_fix.h" +#include "FDK_bitstream.h" +#include "aacEnc_rom.h" + +#define INVALID_BITCOUNT (FDK_INT_MAX / 4) + +/* + code book number table +*/ + +enum codeBookNo { + CODE_BOOK_ZERO_NO = 0, + CODE_BOOK_1_NO = 1, + CODE_BOOK_2_NO = 2, + CODE_BOOK_3_NO = 3, + CODE_BOOK_4_NO = 4, + CODE_BOOK_5_NO = 5, + CODE_BOOK_6_NO = 6, + CODE_BOOK_7_NO = 7, + CODE_BOOK_8_NO = 8, + CODE_BOOK_9_NO = 9, + CODE_BOOK_10_NO = 10, + CODE_BOOK_ESC_NO = 11, + CODE_BOOK_RES_NO = 12, + CODE_BOOK_PNS_NO = 13, + CODE_BOOK_IS_OUT_OF_PHASE_NO = 14, + CODE_BOOK_IS_IN_PHASE_NO = 15 + +}; + +/* + code book index table +*/ + +enum codeBookNdx { + CODE_BOOK_ZERO_NDX, + CODE_BOOK_1_NDX, + CODE_BOOK_2_NDX, + CODE_BOOK_3_NDX, + CODE_BOOK_4_NDX, + CODE_BOOK_5_NDX, + CODE_BOOK_6_NDX, + CODE_BOOK_7_NDX, + CODE_BOOK_8_NDX, + CODE_BOOK_9_NDX, + CODE_BOOK_10_NDX, + CODE_BOOK_ESC_NDX, + CODE_BOOK_RES_NDX, + CODE_BOOK_PNS_NDX, + CODE_BOOK_IS_OUT_OF_PHASE_NDX, + CODE_BOOK_IS_IN_PHASE_NDX, + NUMBER_OF_CODE_BOOKS +}; + +/* + code book lav table +*/ + +enum codeBookLav { + CODE_BOOK_ZERO_LAV = 0, + CODE_BOOK_1_LAV = 1, + CODE_BOOK_2_LAV = 1, + CODE_BOOK_3_LAV = 2, + CODE_BOOK_4_LAV = 2, + CODE_BOOK_5_LAV = 4, + CODE_BOOK_6_LAV = 4, + CODE_BOOK_7_LAV = 7, + CODE_BOOK_8_LAV = 7, + CODE_BOOK_9_LAV = 12, + CODE_BOOK_10_LAV = 12, + CODE_BOOK_ESC_LAV = 16, + CODE_BOOK_SCF_LAV = 60, + CODE_BOOK_PNS_LAV = 60 +}; + +INT FDKaacEnc_bitCount(const SHORT *aQuantSpectrum, const INT noOfSpecLines, + INT maxVal, INT *bitCountLut); + +INT FDKaacEnc_countValues(SHORT *values, INT width, INT codeBook); + +INT FDKaacEnc_codeValues(SHORT *values, INT width, INT codeBook, + HANDLE_FDK_BITSTREAM hBitstream); + +INT FDKaacEnc_codeScalefactorDelta(INT scalefactor, + HANDLE_FDK_BITSTREAM hBitstream); + +inline INT FDKaacEnc_bitCountScalefactorDelta(const INT delta) { + FDK_ASSERT((0 <= (delta + CODE_BOOK_SCF_LAV)) && + ((delta + CODE_BOOK_SCF_LAV) < + (int)(sizeof(FDKaacEnc_huff_ltabscf) / + sizeof((FDKaacEnc_huff_ltabscf[0]))))); + return ((INT)FDKaacEnc_huff_ltabscf[delta + CODE_BOOK_SCF_LAV]); +} + +#endif diff --git a/fdk-aac/libAACenc/src/bitenc.cpp b/fdk-aac/libAACenc/src/bitenc.cpp new file mode 100644 index 0000000..512d596 --- /dev/null +++ b/fdk-aac/libAACenc/src/bitenc.cpp @@ -0,0 +1,1362 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M. Werner + + Description: Bitstream encoder + +*******************************************************************************/ + +#include <stdio.h> +#include "bitenc.h" +#include "bit_cnt.h" +#include "dyn_bits.h" +#include "qc_data.h" +#include "interface.h" +#include "aacEnc_ram.h" + +#include "tpenc_lib.h" + +#include "FDK_tools_rom.h" /* needed for the bitstream syntax tables */ + +static const int globalGainOffset = 100; +static const int icsReservedBit = 0; +static const int noiseOffset = 90; + +/***************************************************************************** + + functionname: FDKaacEnc_encodeSpectralData + description: encode spectral data + returns: the number of written bits + input: + output: + +*****************************************************************************/ +static INT FDKaacEnc_encodeSpectralData(INT *sfbOffset, + SECTION_DATA *sectionData, + SHORT *quantSpectrum, + HANDLE_FDK_BITSTREAM hBitStream) { + INT i, sfb; + INT dbgVal = FDKgetValidBits(hBitStream); + + for (i = 0; i < sectionData->noOfSections; i++) { + if (sectionData->huffsection[i].codeBook != CODE_BOOK_PNS_NO) { + /* huffencode spectral data for this huffsection */ + INT tmp = sectionData->huffsection[i].sfbStart + + sectionData->huffsection[i].sfbCnt; + for (sfb = sectionData->huffsection[i].sfbStart; sfb < tmp; sfb++) { + FDKaacEnc_codeValues(quantSpectrum + sfbOffset[sfb], + sfbOffset[sfb + 1] - sfbOffset[sfb], + sectionData->huffsection[i].codeBook, hBitStream); + } + } + } + return (FDKgetValidBits(hBitStream) - dbgVal); +} + +/***************************************************************************** + + functionname:FDKaacEnc_encodeGlobalGain + description: encodes Global Gain (common scale factor) + returns: the number of static bits + input: + output: + +*****************************************************************************/ +static INT FDKaacEnc_encodeGlobalGain(INT globalGain, INT scalefac, + HANDLE_FDK_BITSTREAM hBitStream, + INT mdctScale) { + if (hBitStream != NULL) { + FDKwriteBits(hBitStream, + globalGain - scalefac + globalGainOffset - + 4 * (LOG_NORM_PCM - mdctScale), + 8); + } + return (8); +} + +/***************************************************************************** + + functionname:FDKaacEnc_encodeIcsInfo + description: encodes Ics Info + returns: the number of static bits + input: + output: + +*****************************************************************************/ + +static INT FDKaacEnc_encodeIcsInfo(INT blockType, INT windowShape, + INT groupingMask, INT maxSfbPerGroup, + HANDLE_FDK_BITSTREAM hBitStream, + UINT syntaxFlags) { + INT statBits; + + if (blockType == SHORT_WINDOW) { + statBits = 8 + TRANS_FAC - 1; + } else { + if (syntaxFlags & AC_ELD) { + statBits = 6; + } else { + statBits = (!(syntaxFlags & AC_SCALABLE)) ? 11 : 10; + } + } + + if (hBitStream != NULL) { + if (!(syntaxFlags & AC_ELD)) { + FDKwriteBits(hBitStream, icsReservedBit, 1); + FDKwriteBits(hBitStream, blockType, 2); + FDKwriteBits(hBitStream, + (windowShape == LOL_WINDOW) ? KBD_WINDOW : windowShape, 1); + } + + switch (blockType) { + case LONG_WINDOW: + case START_WINDOW: + case STOP_WINDOW: + FDKwriteBits(hBitStream, maxSfbPerGroup, 6); + + if (!(syntaxFlags & + (AC_SCALABLE | AC_ELD))) { /* If not scalable syntax then ... */ + /* No predictor data present */ + FDKwriteBits(hBitStream, 0, 1); + } + break; + + case SHORT_WINDOW: + FDKwriteBits(hBitStream, maxSfbPerGroup, 4); + + /* Write grouping bits */ + FDKwriteBits(hBitStream, groupingMask, TRANS_FAC - 1); + break; + } + } + + return (statBits); +} + +/***************************************************************************** + + functionname: FDKaacEnc_encodeSectionData + description: encode section data (common Huffman codebooks for adjacent + SFB's) + returns: none + input: + output: + +*****************************************************************************/ +static INT FDKaacEnc_encodeSectionData(SECTION_DATA *sectionData, + HANDLE_FDK_BITSTREAM hBitStream, + UINT useVCB11) { + if (hBitStream != NULL) { + INT sectEscapeVal = 0, sectLenBits = 0; + INT sectLen; + INT i; + INT dbgVal = FDKgetValidBits(hBitStream); + INT sectCbBits = 4; + + switch (sectionData->blockType) { + case LONG_WINDOW: + case START_WINDOW: + case STOP_WINDOW: + sectEscapeVal = SECT_ESC_VAL_LONG; + sectLenBits = SECT_BITS_LONG; + break; + + case SHORT_WINDOW: + sectEscapeVal = SECT_ESC_VAL_SHORT; + sectLenBits = SECT_BITS_SHORT; + break; + } + + for (i = 0; i < sectionData->noOfSections; i++) { + INT codeBook = sectionData->huffsection[i].codeBook; + + FDKwriteBits(hBitStream, codeBook, sectCbBits); + + { + sectLen = sectionData->huffsection[i].sfbCnt; + + while (sectLen >= sectEscapeVal) { + FDKwriteBits(hBitStream, sectEscapeVal, sectLenBits); + sectLen -= sectEscapeVal; + } + FDKwriteBits(hBitStream, sectLen, sectLenBits); + } + } + return (FDKgetValidBits(hBitStream) - dbgVal); + } + return (0); +} + +/***************************************************************************** + + functionname: FDKaacEnc_encodeScaleFactorData + description: encode DPCM coded scale factors + returns: none + input: + output: + +*****************************************************************************/ +static INT FDKaacEnc_encodeScaleFactorData(UINT *maxValueInSfb, + SECTION_DATA *sectionData, + INT *scalefac, + HANDLE_FDK_BITSTREAM hBitStream, + INT *RESTRICT noiseNrg, + const INT *isScale, INT globalGain) { + if (hBitStream != NULL) { + INT i, j, lastValScf, deltaScf; + INT deltaPns; + INT lastValPns = 0; + INT noisePCMFlag = TRUE; + INT lastValIs; + + INT dbgVal = FDKgetValidBits(hBitStream); + + lastValScf = scalefac[sectionData->firstScf]; + lastValPns = globalGain - scalefac[sectionData->firstScf] + + globalGainOffset - 4 * LOG_NORM_PCM - noiseOffset; + lastValIs = 0; + + for (i = 0; i < sectionData->noOfSections; i++) { + if (sectionData->huffsection[i].codeBook != CODE_BOOK_ZERO_NO) { + if ((sectionData->huffsection[i].codeBook == + CODE_BOOK_IS_OUT_OF_PHASE_NO) || + (sectionData->huffsection[i].codeBook == + CODE_BOOK_IS_IN_PHASE_NO)) { + INT sfbStart = sectionData->huffsection[i].sfbStart; + INT tmp = sfbStart + sectionData->huffsection[i].sfbCnt; + for (j = sfbStart; j < tmp; j++) { + INT deltaIs = isScale[j] - lastValIs; + lastValIs = isScale[j]; + if (FDKaacEnc_codeScalefactorDelta(deltaIs, hBitStream)) { + return (1); + } + } /* sfb */ + } else if (sectionData->huffsection[i].codeBook == CODE_BOOK_PNS_NO) { + INT sfbStart = sectionData->huffsection[i].sfbStart; + INT tmp = sfbStart + sectionData->huffsection[i].sfbCnt; + for (j = sfbStart; j < tmp; j++) { + deltaPns = noiseNrg[j] - lastValPns; + lastValPns = noiseNrg[j]; + + if (noisePCMFlag) { + FDKwriteBits(hBitStream, deltaPns + (1 << (PNS_PCM_BITS - 1)), + PNS_PCM_BITS); + noisePCMFlag = FALSE; + } else { + if (FDKaacEnc_codeScalefactorDelta(deltaPns, hBitStream)) { + return (1); + } + } + } /* sfb */ + } else { + INT tmp = sectionData->huffsection[i].sfbStart + + sectionData->huffsection[i].sfbCnt; + for (j = sectionData->huffsection[i].sfbStart; j < tmp; j++) { + /* + check if we can repeat the last value to save bits + */ + if (maxValueInSfb[j] == 0) + deltaScf = 0; + else { + deltaScf = -(scalefac[j] - lastValScf); + lastValScf = scalefac[j]; + } + if (FDKaacEnc_codeScalefactorDelta(deltaScf, hBitStream)) { + return (1); + } + } /* sfb */ + } /* code scalefactor */ + } /* sectionData->huffsection[i].codeBook != CODE_BOOK_ZERO_NO */ + } /* section loop */ + + return (FDKgetValidBits(hBitStream) - dbgVal); + } /* if (hBitStream != NULL) */ + + return (0); +} + +/***************************************************************************** + + functionname:encodeMsInfo + description: encodes MS-Stereo Info + returns: the number of static bits + input: + output: + +*****************************************************************************/ +static INT FDKaacEnc_encodeMSInfo(INT sfbCnt, INT grpSfb, INT maxSfb, + INT msDigest, INT *jsFlags, + HANDLE_FDK_BITSTREAM hBitStream) { + INT sfb, sfbOff, msBits = 0; + + if (hBitStream != NULL) { + switch (msDigest) { + case MS_NONE: + FDKwriteBits(hBitStream, SI_MS_MASK_NONE, 2); + msBits += 2; + break; + + case MS_ALL: + FDKwriteBits(hBitStream, SI_MS_MASK_ALL, 2); + msBits += 2; + break; + + case MS_SOME: + FDKwriteBits(hBitStream, SI_MS_MASK_SOME, 2); + msBits += 2; + for (sfbOff = 0; sfbOff < sfbCnt; sfbOff += grpSfb) { + for (sfb = 0; sfb < maxSfb; sfb++) { + if (jsFlags[sfbOff + sfb] & MS_ON) { + FDKwriteBits(hBitStream, 1, 1); + } else { + FDKwriteBits(hBitStream, 0, 1); + } + msBits += 1; + } + } + break; + } + } else { + msBits += 2; + if (msDigest == MS_SOME) { + for (sfbOff = 0; sfbOff < sfbCnt; sfbOff += grpSfb) { + for (sfb = 0; sfb < maxSfb; sfb++) { + msBits += 1; + } + } + } + } + return (msBits); +} + +/***************************************************************************** + + functionname: FDKaacEnc_encodeTnsDataPresent + description: encode TNS data (filter order, coeffs, ..) + returns: the number of static bits + input: + output: + +*****************************************************************************/ +static INT FDKaacEnc_encodeTnsDataPresent(TNS_INFO *tnsInfo, INT blockType, + HANDLE_FDK_BITSTREAM hBitStream) { + if ((hBitStream != NULL) && (tnsInfo != NULL)) { + INT i, tnsPresent = 0; + INT numOfWindows = (blockType == SHORT_WINDOW ? TRANS_FAC : 1); + + for (i = 0; i < numOfWindows; i++) { + if (tnsInfo->numOfFilters[i] != 0) { + tnsPresent = 1; + break; + } + } + + if (tnsPresent == 0) { + FDKwriteBits(hBitStream, 0, 1); + } else { + FDKwriteBits(hBitStream, 1, 1); + } + } + return (1); +} + +/***************************************************************************** + + functionname: FDKaacEnc_encodeTnsData + description: encode TNS data (filter order, coeffs, ..) + returns: the number of static bits + input: + output: + +*****************************************************************************/ +static INT FDKaacEnc_encodeTnsData(TNS_INFO *tnsInfo, INT blockType, + HANDLE_FDK_BITSTREAM hBitStream) { + INT tnsBits = 0; + + if (tnsInfo != NULL) { + INT i, j, k; + INT tnsPresent = 0; + INT coefBits; + INT numOfWindows = (blockType == SHORT_WINDOW ? TRANS_FAC : 1); + + for (i = 0; i < numOfWindows; i++) { + if (tnsInfo->numOfFilters[i] != 0) { + tnsPresent = 1; + } + } + + if (hBitStream != NULL) { + if (tnsPresent == 1) { /* there is data to be written*/ + for (i = 0; i < numOfWindows; i++) { + FDKwriteBits(hBitStream, tnsInfo->numOfFilters[i], + (blockType == SHORT_WINDOW ? 1 : 2)); + tnsBits += (blockType == SHORT_WINDOW ? 1 : 2); + if (tnsInfo->numOfFilters[i]) { + FDKwriteBits(hBitStream, (tnsInfo->coefRes[i] == 4 ? 1 : 0), 1); + tnsBits += 1; + } + for (j = 0; j < tnsInfo->numOfFilters[i]; j++) { + FDKwriteBits(hBitStream, tnsInfo->length[i][j], + (blockType == SHORT_WINDOW ? 4 : 6)); + tnsBits += (blockType == SHORT_WINDOW ? 4 : 6); + FDK_ASSERT(tnsInfo->order[i][j] <= 12); + FDKwriteBits(hBitStream, tnsInfo->order[i][j], + (blockType == SHORT_WINDOW ? 3 : 5)); + tnsBits += (blockType == SHORT_WINDOW ? 3 : 5); + if (tnsInfo->order[i][j]) { + FDKwriteBits(hBitStream, tnsInfo->direction[i][j], 1); + tnsBits += 1; /*direction*/ + if (tnsInfo->coefRes[i] == 4) { + coefBits = 3; + for (k = 0; k < tnsInfo->order[i][j]; k++) { + if (tnsInfo->coef[i][j][k] > 3 || + tnsInfo->coef[i][j][k] < -4) { + coefBits = 4; + break; + } + } + } else { + coefBits = 2; + for (k = 0; k < tnsInfo->order[i][j]; k++) { + if (tnsInfo->coef[i][j][k] > 1 || + tnsInfo->coef[i][j][k] < -2) { + coefBits = 3; + break; + } + } + } + FDKwriteBits(hBitStream, -(coefBits - tnsInfo->coefRes[i]), + 1); /*coef_compres*/ + tnsBits += 1; /*coef_compression */ + for (k = 0; k < tnsInfo->order[i][j]; k++) { + static const INT rmask[] = {0, 1, 3, 7, 15}; + FDKwriteBits(hBitStream, + tnsInfo->coef[i][j][k] & rmask[coefBits], + coefBits); + tnsBits += coefBits; + } + } + } + } + } + } else { + if (tnsPresent != 0) { + for (i = 0; i < numOfWindows; i++) { + tnsBits += (blockType == SHORT_WINDOW ? 1 : 2); + if (tnsInfo->numOfFilters[i]) { + tnsBits += 1; + for (j = 0; j < tnsInfo->numOfFilters[i]; j++) { + tnsBits += (blockType == SHORT_WINDOW ? 4 : 6); + tnsBits += (blockType == SHORT_WINDOW ? 3 : 5); + if (tnsInfo->order[i][j]) { + tnsBits += 1; /*direction*/ + tnsBits += 1; /*coef_compression */ + if (tnsInfo->coefRes[i] == 4) { + coefBits = 3; + for (k = 0; k < tnsInfo->order[i][j]; k++) { + if (tnsInfo->coef[i][j][k] > 3 || + tnsInfo->coef[i][j][k] < -4) { + coefBits = 4; + break; + } + } + } else { + coefBits = 2; + for (k = 0; k < tnsInfo->order[i][j]; k++) { + if (tnsInfo->coef[i][j][k] > 1 || + tnsInfo->coef[i][j][k] < -2) { + coefBits = 3; + break; + } + } + } + for (k = 0; k < tnsInfo->order[i][j]; k++) { + tnsBits += coefBits; + } + } + } + } + } + } + } + } /* (tnsInfo!=NULL) */ + + return (tnsBits); +} + +/***************************************************************************** + + functionname: FDKaacEnc_encodeGainControlData + description: unsupported + returns: none + input: + output: + +*****************************************************************************/ +static INT FDKaacEnc_encodeGainControlData(HANDLE_FDK_BITSTREAM hBitStream) { + if (hBitStream != NULL) { + FDKwriteBits(hBitStream, 0, 1); + } + return (1); +} + +/***************************************************************************** + + functionname: FDKaacEnc_encodePulseData + description: not supported yet (dummy) + returns: none + input: + output: + +*****************************************************************************/ +static INT FDKaacEnc_encodePulseData(HANDLE_FDK_BITSTREAM hBitStream) { + if (hBitStream != NULL) { + FDKwriteBits(hBitStream, 0, 1); + } + return (1); +} + +/***************************************************************************** + + functionname: FDKaacEnc_writeExtensionPayload + description: write extension payload to bitstream + returns: number of written bits + input: + output: + +*****************************************************************************/ +static INT FDKaacEnc_writeExtensionPayload(HANDLE_FDK_BITSTREAM hBitStream, + EXT_PAYLOAD_TYPE extPayloadType, + const UCHAR *extPayloadData, + INT extPayloadBits) { +#define EXT_TYPE_BITS (4) +#define DATA_EL_VERSION_BITS (4) +#define FILL_NIBBLE_BITS (4) + + INT extBitsUsed = 0; + + if (extPayloadBits >= EXT_TYPE_BITS) { + UCHAR fillByte = 0x00; /* for EXT_FIL and EXT_FILL_DATA */ + + if (hBitStream != NULL) { + FDKwriteBits(hBitStream, extPayloadType, EXT_TYPE_BITS); + } + extBitsUsed += EXT_TYPE_BITS; + + switch (extPayloadType) { + /* case EXT_SAC_DATA: */ + case EXT_LDSAC_DATA: + if (hBitStream != NULL) { + FDKwriteBits(hBitStream, *extPayloadData++, 4); /* nibble */ + } + extBitsUsed += 4; + FDK_FALLTHROUGH; + case EXT_DYNAMIC_RANGE: + case EXT_SBR_DATA: + case EXT_SBR_DATA_CRC: + if (hBitStream != NULL) { + int i, writeBits = extPayloadBits; + for (i = 0; writeBits >= 8; i++) { + FDKwriteBits(hBitStream, *extPayloadData++, 8); + writeBits -= 8; + } + if (writeBits > 0) { + FDKwriteBits(hBitStream, (*extPayloadData) >> (8 - writeBits), + writeBits); + } + } + extBitsUsed += extPayloadBits; + break; + + case EXT_DATA_ELEMENT: { + INT dataElementLength = (extPayloadBits + 7) >> 3; + INT cnt = dataElementLength; + int loopCounter = 1; + + while (dataElementLength >= 255) { + loopCounter++; + dataElementLength -= 255; + } + + if (hBitStream != NULL) { + int i; + FDKwriteBits( + hBitStream, 0x00, + DATA_EL_VERSION_BITS); /* data_element_version = ANC_DATA */ + + for (i = 1; i < loopCounter; i++) { + FDKwriteBits(hBitStream, 255, 8); + } + FDKwriteBits(hBitStream, dataElementLength, 8); + + for (i = 0; i < cnt; i++) { + FDKwriteBits(hBitStream, extPayloadData[i], 8); + } + } + extBitsUsed += DATA_EL_VERSION_BITS + (loopCounter * 8) + (cnt * 8); + } break; + + case EXT_FILL_DATA: + fillByte = 0xA5; + FDK_FALLTHROUGH; + case EXT_FIL: + default: + if (hBitStream != NULL) { + int writeBits = extPayloadBits; + FDKwriteBits(hBitStream, 0x00, FILL_NIBBLE_BITS); + writeBits -= + 8; /* acount for the extension type and the fill nibble */ + while (writeBits >= 8) { + FDKwriteBits(hBitStream, fillByte, 8); + writeBits -= 8; + } + } + extBitsUsed += FILL_NIBBLE_BITS + (extPayloadBits & ~0x7) - 8; + break; + } + } + + return (extBitsUsed); +} + +/***************************************************************************** + + functionname: FDKaacEnc_writeDataStreamElement + description: write data stream elements like ancillary data ... + returns: the amount of used bits + input: + output: + +******************************************************************************/ +static INT FDKaacEnc_writeDataStreamElement(HANDLE_TRANSPORTENC hTpEnc, + INT elementInstanceTag, + INT dataPayloadBytes, + UCHAR *dataBuffer, + UINT alignAnchor) { +#define DATA_BYTE_ALIGN_FLAG (0) + +#define EL_INSTANCE_TAG_BITS (4) +#define DATA_BYTE_ALIGN_FLAG_BITS (1) +#define DATA_LEN_COUNT_BITS (8) +#define DATA_LEN_ESC_COUNT_BITS (8) + +#define MAX_DATA_ALIGN_BITS (7) +#define MAX_DSE_DATA_BYTES (510) + + INT dseBitsUsed = 0; + + while (dataPayloadBytes > 0) { + int esc_count = -1; + int cnt = 0; + INT crcReg = -1; + + dseBitsUsed += EL_ID_BITS + EL_INSTANCE_TAG_BITS + + DATA_BYTE_ALIGN_FLAG_BITS + DATA_LEN_COUNT_BITS; + + if (DATA_BYTE_ALIGN_FLAG) { + dseBitsUsed += MAX_DATA_ALIGN_BITS; + } + + cnt = fixMin(MAX_DSE_DATA_BYTES, dataPayloadBytes); + if (cnt >= 255) { + esc_count = cnt - 255; + dseBitsUsed += DATA_LEN_ESC_COUNT_BITS; + } + + dataPayloadBytes -= cnt; + dseBitsUsed += cnt * 8; + + if (hTpEnc != NULL) { + HANDLE_FDK_BITSTREAM hBitStream = transportEnc_GetBitstream(hTpEnc); + int i; + + FDKwriteBits(hBitStream, ID_DSE, EL_ID_BITS); + + crcReg = transportEnc_CrcStartReg(hTpEnc, 0); + + FDKwriteBits(hBitStream, elementInstanceTag, EL_INSTANCE_TAG_BITS); + FDKwriteBits(hBitStream, DATA_BYTE_ALIGN_FLAG, DATA_BYTE_ALIGN_FLAG_BITS); + + /* write length field(s) */ + if (esc_count >= 0) { + FDKwriteBits(hBitStream, 255, DATA_LEN_COUNT_BITS); + FDKwriteBits(hBitStream, esc_count, DATA_LEN_ESC_COUNT_BITS); + } else { + FDKwriteBits(hBitStream, cnt, DATA_LEN_COUNT_BITS); + } + + if (DATA_BYTE_ALIGN_FLAG) { + INT tmp = (INT)FDKgetValidBits(hBitStream); + FDKbyteAlign(hBitStream, alignAnchor); + /* count actual bits */ + dseBitsUsed += + (INT)FDKgetValidBits(hBitStream) - tmp - MAX_DATA_ALIGN_BITS; + } + + /* write payload */ + for (i = 0; i < cnt; i++) { + FDKwriteBits(hBitStream, dataBuffer[i], 8); + } + transportEnc_CrcEndReg(hTpEnc, crcReg); + } + } + + return (dseBitsUsed); +} + +/***************************************************************************** + + functionname: FDKaacEnc_writeExtensionData + description: write extension payload to bitstream + returns: number of written bits + input: + output: + +*****************************************************************************/ +INT FDKaacEnc_writeExtensionData(HANDLE_TRANSPORTENC hTpEnc, + QC_OUT_EXTENSION *pExtension, + INT elInstanceTag, /* for DSE only */ + UINT alignAnchor, /* for DSE only */ + UINT syntaxFlags, AUDIO_OBJECT_TYPE aot, + SCHAR epConfig) { +#define FILL_EL_COUNT_BITS (4) +#define FILL_EL_ESC_COUNT_BITS (8) +#define MAX_FILL_DATA_BYTES (269) + + HANDLE_FDK_BITSTREAM hBitStream = NULL; + INT payloadBits = pExtension->nPayloadBits; + INT extBitsUsed = 0; + + if (hTpEnc != NULL) { + hBitStream = transportEnc_GetBitstream(hTpEnc); + } + + if (syntaxFlags & (AC_SCALABLE | AC_ER)) { + { + if ((syntaxFlags & AC_ELD) && ((pExtension->type == EXT_SBR_DATA) || + (pExtension->type == EXT_SBR_DATA_CRC))) { + if (hBitStream != NULL) { + int i, writeBits = payloadBits; + UCHAR *extPayloadData = pExtension->pPayload; + + for (i = 0; writeBits >= 8; i++) { + FDKwriteBits(hBitStream, extPayloadData[i], 8); + writeBits -= 8; + } + if (writeBits > 0) { + FDKwriteBits(hBitStream, extPayloadData[i] >> (8 - writeBits), + writeBits); + } + } + extBitsUsed += payloadBits; + } else { + /* ER or scalable syntax -> write extension en bloc */ + extBitsUsed += FDKaacEnc_writeExtensionPayload( + hBitStream, pExtension->type, pExtension->pPayload, payloadBits); + } + } + } else { + /* We have normal GA bitstream payload (AOT 2,5,29) so pack + the data into a fill elements or DSEs */ + + if (pExtension->type == EXT_DATA_ELEMENT) { + extBitsUsed += FDKaacEnc_writeDataStreamElement( + hTpEnc, elInstanceTag, pExtension->nPayloadBits >> 3, + pExtension->pPayload, alignAnchor); + } else { + while (payloadBits >= (EL_ID_BITS + FILL_EL_COUNT_BITS)) { + INT cnt, esc_count = -1, alignBits = 7; + + if ((pExtension->type == EXT_FILL_DATA) || + (pExtension->type == EXT_FIL)) { + payloadBits -= EL_ID_BITS + FILL_EL_COUNT_BITS; + if (payloadBits >= 15 * 8) { + payloadBits -= FILL_EL_ESC_COUNT_BITS; + esc_count = 0; /* write esc_count even if cnt becomes smaller 15 */ + } + alignBits = 0; + } + + cnt = fixMin(MAX_FILL_DATA_BYTES, (payloadBits + alignBits) >> 3); + + if (cnt >= 15) { + esc_count = cnt - 15 + 1; + } + + if (hBitStream != NULL) { + /* write bitstream */ + FDKwriteBits(hBitStream, ID_FIL, EL_ID_BITS); + if (esc_count >= 0) { + FDKwriteBits(hBitStream, 15, FILL_EL_COUNT_BITS); + FDKwriteBits(hBitStream, esc_count, FILL_EL_ESC_COUNT_BITS); + } else { + FDKwriteBits(hBitStream, cnt, FILL_EL_COUNT_BITS); + } + } + + extBitsUsed += EL_ID_BITS + FILL_EL_COUNT_BITS + + ((esc_count >= 0) ? FILL_EL_ESC_COUNT_BITS : 0); + + cnt = fixMin(cnt * 8, payloadBits); /* convert back to bits */ + extBitsUsed += FDKaacEnc_writeExtensionPayload( + hBitStream, pExtension->type, pExtension->pPayload, cnt); + payloadBits -= cnt; + } + } + } + + return (extBitsUsed); +} + +/***************************************************************************** + + functionname: FDKaacEnc_ByteAlignment + description: + returns: + input: + output: + +*****************************************************************************/ +static void FDKaacEnc_ByteAlignment(HANDLE_FDK_BITSTREAM hBitStream, + int alignBits) { + FDKwriteBits(hBitStream, 0, alignBits); +} + +AAC_ENCODER_ERROR FDKaacEnc_ChannelElementWrite( + HANDLE_TRANSPORTENC hTpEnc, ELEMENT_INFO *pElInfo, + QC_OUT_CHANNEL *qcOutChannel[(2)], PSY_OUT_ELEMENT *psyOutElement, + PSY_OUT_CHANNEL *psyOutChannel[(2)], UINT syntaxFlags, + AUDIO_OBJECT_TYPE aot, SCHAR epConfig, INT *pBitDemand, UCHAR minCnt) { + AAC_ENCODER_ERROR error = AAC_ENC_OK; + HANDLE_FDK_BITSTREAM hBitStream = NULL; + INT bitDemand = 0; + const element_list_t *list; + int i, ch, decision_bit; + INT crcReg1 = -1, crcReg2 = -1; + UCHAR numberOfChannels; + + if (hTpEnc != NULL) { + /* Get bitstream handle */ + hBitStream = transportEnc_GetBitstream(hTpEnc); + } + + if ((pElInfo->elType == ID_SCE) || (pElInfo->elType == ID_LFE)) { + numberOfChannels = 1; + } else { + numberOfChannels = 2; + } + + /* Get channel element sequence table */ + list = getBitstreamElementList(aot, epConfig, numberOfChannels, 0, 0); + if (list == NULL) { + error = AAC_ENC_UNSUPPORTED_AOT; + goto bail; + } + + if (!(syntaxFlags & (AC_SCALABLE | AC_ER))) { + if (hBitStream != NULL) { + FDKwriteBits(hBitStream, pElInfo->elType, EL_ID_BITS); + } + bitDemand += EL_ID_BITS; + } + + /* Iterate through sequence table */ + i = 0; + ch = 0; + decision_bit = 0; + do { + /* some tmp values */ + SECTION_DATA *pChSectionData = NULL; + INT *pChScf = NULL; + UINT *pChMaxValueInSfb = NULL; + TNS_INFO *pTnsInfo = NULL; + INT chGlobalGain = 0; + INT chBlockType = 0; + INT chMaxSfbPerGrp = 0; + INT chSfbPerGrp = 0; + INT chSfbCnt = 0; + INT chFirstScf = 0; + + if (minCnt == 0) { + if (qcOutChannel != NULL) { + pChSectionData = &(qcOutChannel[ch]->sectionData); + pChScf = qcOutChannel[ch]->scf; + chGlobalGain = qcOutChannel[ch]->globalGain; + pChMaxValueInSfb = qcOutChannel[ch]->maxValueInSfb; + chBlockType = pChSectionData->blockType; + chMaxSfbPerGrp = pChSectionData->maxSfbPerGroup; + chSfbPerGrp = pChSectionData->sfbPerGroup; + chSfbCnt = pChSectionData->sfbCnt; + chFirstScf = pChScf[pChSectionData->firstScf]; + } else { + /* get values from PSY */ + chSfbCnt = psyOutChannel[ch]->sfbCnt; + chSfbPerGrp = psyOutChannel[ch]->sfbPerGroup; + chMaxSfbPerGrp = psyOutChannel[ch]->maxSfbPerGroup; + } + pTnsInfo = &psyOutChannel[ch]->tnsInfo; + } /* minCnt==0 */ + + if (qcOutChannel == NULL) { + chBlockType = psyOutChannel[ch]->lastWindowSequence; + } + + switch (list->id[i]) { + case element_instance_tag: + /* Write element instance tag */ + if (hBitStream != NULL) { + FDKwriteBits(hBitStream, pElInfo->instanceTag, 4); + } + bitDemand += 4; + break; + + case common_window: + /* Write common window flag */ + decision_bit = psyOutElement->commonWindow; + if (hBitStream != NULL) { + FDKwriteBits(hBitStream, psyOutElement->commonWindow, 1); + } + bitDemand += 1; + break; + + case ics_info: + /* Write individual channel info */ + bitDemand += + FDKaacEnc_encodeIcsInfo(chBlockType, psyOutChannel[ch]->windowShape, + psyOutChannel[ch]->groupingMask, + chMaxSfbPerGrp, hBitStream, syntaxFlags); + break; + + case ltp_data_present: + /* Write LTP data present flag */ + if (hBitStream != NULL) { + FDKwriteBits(hBitStream, 0, 1); + } + bitDemand += 1; + break; + + case ltp_data: + /* Predictor data not supported. + Nothing to do here. */ + break; + + case ms: + /* Write MS info */ + bitDemand += FDKaacEnc_encodeMSInfo( + chSfbCnt, chSfbPerGrp, chMaxSfbPerGrp, + (minCnt == 0) ? psyOutElement->toolsInfo.msDigest : MS_NONE, + psyOutElement->toolsInfo.msMask, hBitStream); + break; + + case global_gain: + bitDemand += FDKaacEnc_encodeGlobalGain( + chGlobalGain, chFirstScf, hBitStream, psyOutChannel[ch]->mdctScale); + break; + + case section_data: { + INT siBits = FDKaacEnc_encodeSectionData( + pChSectionData, hBitStream, (syntaxFlags & AC_ER_VCB11) ? 1 : 0); + if (hBitStream != NULL) { + if (siBits != qcOutChannel[ch]->sectionData.sideInfoBits) { + error = AAC_ENC_WRITE_SEC_ERROR; + } + } + bitDemand += siBits; + } break; + + case scale_factor_data: { + INT sfDataBits = FDKaacEnc_encodeScaleFactorData( + pChMaxValueInSfb, pChSectionData, pChScf, hBitStream, + psyOutChannel[ch]->noiseNrg, psyOutChannel[ch]->isScale, + chGlobalGain); + if ((hBitStream != NULL) && + (sfDataBits != (qcOutChannel[ch]->sectionData.scalefacBits + + qcOutChannel[ch]->sectionData.noiseNrgBits))) { + error = AAC_ENC_WRITE_SCAL_ERROR; + } + bitDemand += sfDataBits; + } break; + + case esc2_rvlc: + if (syntaxFlags & AC_ER_RVLC) { + /* write RVLC data into bitstream (error sens. cat. 2) */ + error = AAC_ENC_UNSUPPORTED_AOT; + } + break; + + case pulse: + /* Write pulse data */ + bitDemand += FDKaacEnc_encodePulseData(hBitStream); + break; + + case tns_data_present: + /* Write TNS data present flag */ + bitDemand += + FDKaacEnc_encodeTnsDataPresent(pTnsInfo, chBlockType, hBitStream); + break; + case tns_data: + /* Write TNS data */ + bitDemand += FDKaacEnc_encodeTnsData(pTnsInfo, chBlockType, hBitStream); + break; + + case gain_control_data: + /* Nothing to do here */ + break; + + case gain_control_data_present: + bitDemand += FDKaacEnc_encodeGainControlData(hBitStream); + break; + + case esc1_hcr: + if (syntaxFlags & AC_ER_HCR) { + error = AAC_ENC_UNKNOWN; + } + break; + + case spectral_data: + if (hBitStream != NULL) { + INT spectralBits = 0; + + spectralBits = FDKaacEnc_encodeSpectralData( + psyOutChannel[ch]->sfbOffsets, pChSectionData, + qcOutChannel[ch]->quantSpec, hBitStream); + + if (spectralBits != qcOutChannel[ch]->sectionData.huffmanBits) { + return AAC_ENC_WRITE_SPEC_ERROR; + } + bitDemand += spectralBits; + } + break; + + /* Non data cases */ + case adtscrc_start_reg1: + if (hTpEnc != NULL) { + crcReg1 = transportEnc_CrcStartReg(hTpEnc, 192); + } + break; + case adtscrc_start_reg2: + if (hTpEnc != NULL) { + crcReg2 = transportEnc_CrcStartReg(hTpEnc, 128); + } + break; + case adtscrc_end_reg1: + case drmcrc_end_reg: + if (hTpEnc != NULL) { + transportEnc_CrcEndReg(hTpEnc, crcReg1); + } + break; + case adtscrc_end_reg2: + if (hTpEnc != NULL) { + transportEnc_CrcEndReg(hTpEnc, crcReg2); + } + break; + case drmcrc_start_reg: + if (hTpEnc != NULL) { + crcReg1 = transportEnc_CrcStartReg(hTpEnc, 0); + } + break; + case next_channel: + ch = (ch + 1) % numberOfChannels; + break; + case link_sequence: + list = list->next[decision_bit]; + i = -1; + break; + + default: + error = AAC_ENC_UNKNOWN; + break; + } + + if (error != AAC_ENC_OK) { + return error; + } + + i++; + + } while (list->id[i] != end_of_sequence); + +bail: + if (pBitDemand != NULL) { + *pBitDemand = bitDemand; + } + + return error; +} + +//----------------------------------------------------------------------------------------------- + +AAC_ENCODER_ERROR FDKaacEnc_WriteBitstream(HANDLE_TRANSPORTENC hTpEnc, + CHANNEL_MAPPING *channelMapping, + QC_OUT *qcOut, PSY_OUT *psyOut, + QC_STATE *qcKernel, + AUDIO_OBJECT_TYPE aot, + UINT syntaxFlags, SCHAR epConfig) { + HANDLE_FDK_BITSTREAM hBs = transportEnc_GetBitstream(hTpEnc); + AAC_ENCODER_ERROR ErrorStatus = AAC_ENC_OK; + int i, n, doByteAlign = 1; + INT bitMarkUp; + INT frameBits; + /* Get first bit of raw data block. + In case of ADTS+PCE, AU would start at PCE. + This is okay because PCE assures alignment. */ + UINT alignAnchor = FDKgetValidBits(hBs); + + frameBits = bitMarkUp = alignAnchor; + + + /* Write DSEs first in case of DAB */ + for (n = 0; (n < qcOut->nExtensions) && (n < (2+2)); n++) { + if ( (syntaxFlags & AC_DAB) && + (qcOut->extension[n].type == EXT_DATA_ELEMENT) ) { + FDKaacEnc_writeExtensionData( hTpEnc, + &qcOut->extension[n], + 0, + alignAnchor, + syntaxFlags, + aot, + epConfig ); + } + + /* For EXT_FIL or EXT_FILL_DATA we could do an additional sanity check here */ + } + + /* Channel element loop */ + for (i = 0; i < channelMapping->nElements; i++) { + ELEMENT_INFO elInfo = channelMapping->elInfo[i]; + INT elementUsedBits = 0; + + switch (elInfo.elType) { + case ID_SCE: /* single channel */ + case ID_CPE: /* channel pair */ + case ID_LFE: /* low freq effects channel */ + { + if (AAC_ENC_OK != + (ErrorStatus = FDKaacEnc_ChannelElementWrite( + hTpEnc, &elInfo, qcOut->qcElement[i]->qcOutChannel, + psyOut->psyOutElement[i], + psyOut->psyOutElement[i]->psyOutChannel, + syntaxFlags, /* syntaxFlags (ER tools ...) */ + aot, /* aot: AOT_AAC_LC, AOT_SBR, AOT_PS */ + epConfig, /* epConfig -1, 0, 1 */ + NULL, 0))) { + return ErrorStatus; + } + + if (!(syntaxFlags & AC_ER)) { + /* Write associated extension payload */ + for (n = 0; n < qcOut->qcElement[i]->nExtensions; n++) { + FDKaacEnc_writeExtensionData( + hTpEnc, &qcOut->qcElement[i]->extension[n], 0, alignAnchor, + syntaxFlags, aot, epConfig); + } + } + } break; + + /* In FDK, DSE signalling explicit done in elDSE. See channel_map.cpp */ + default: + return AAC_ENC_INVALID_ELEMENTINFO_TYPE; + + } /* switch */ + + if (elInfo.elType != ID_DSE) { + elementUsedBits -= bitMarkUp; + bitMarkUp = FDKgetValidBits(hBs); + elementUsedBits += bitMarkUp; + frameBits += elementUsedBits; + } + + } /* for (i=0; i<channelMapping.nElements; i++) */ + + if ((syntaxFlags & AC_ER) && !(syntaxFlags & AC_DRM)) { + UCHAR channelElementExtensionWritten[((8))][( + 1)]; /* 0: extension not touched, 1: extension already written */ + + FDKmemclear(channelElementExtensionWritten, + sizeof(channelElementExtensionWritten)); + + if (syntaxFlags & AC_ELD) { + for (i = 0; i < channelMapping->nElements; i++) { + for (n = 0; n < qcOut->qcElement[i]->nExtensions; n++) { + if ((qcOut->qcElement[i]->extension[n].type == EXT_SBR_DATA) || + (qcOut->qcElement[i]->extension[n].type == EXT_SBR_DATA_CRC)) { + /* Write sbr extension payload */ + FDKaacEnc_writeExtensionData( + hTpEnc, &qcOut->qcElement[i]->extension[n], 0, alignAnchor, + syntaxFlags, aot, epConfig); + + channelElementExtensionWritten[i][n] = 1; + } /* SBR */ + } /* n */ + } /* i */ + } /* AC_ELD */ + + for (i = 0; i < channelMapping->nElements; i++) { + for (n = 0; n < qcOut->qcElement[i]->nExtensions; n++) { + if (channelElementExtensionWritten[i][n] == 0) { + /* Write all ramaining extension payloads in element */ + FDKaacEnc_writeExtensionData(hTpEnc, + &qcOut->qcElement[i]->extension[n], 0, + alignAnchor, syntaxFlags, aot, epConfig); + } + } /* n */ + } /* i */ + } /* if AC_ER */ + + /* Extend global extension payload table with fill bits */ + n = qcOut->nExtensions; + + /* Add fill data / stuffing bits */ + n = qcOut->nExtensions; + +// if (!(syntaxFlags & AC_DAB)) { + qcOut->extension[n].type = EXT_FILL_DATA; + qcOut->extension[n].nPayloadBits = qcOut->totFillBits; + qcOut->nExtensions++; +// } else { +// doByteAlign = 0; +// } + if (syntaxFlags & AC_DAB) + doByteAlign = 0; + + /* Write global extension payload and fill data */ + for (n = 0; (n < qcOut->nExtensions) && (n < (2+2)); n++) + { + if ( !(syntaxFlags & AC_DAB) || + ( (syntaxFlags & AC_DAB) && + (qcOut->extension[n].type != EXT_DATA_ELEMENT) + ) + ) { + FDKaacEnc_writeExtensionData( hTpEnc, + &qcOut->extension[n], + 0, + alignAnchor, + syntaxFlags, + aot, + epConfig ); + } + + /* For EXT_FIL or EXT_FILL_DATA we could do an additional sanity check here + */ + } + + if (!(syntaxFlags & (AC_SCALABLE | AC_ER | AC_DAB))) { + FDKwriteBits(hBs, ID_END, EL_ID_BITS); + } + + if (doByteAlign) { + /* Assure byte alignment*/ + if (((FDKgetValidBits(hBs) - alignAnchor + qcOut->alignBits) & 0x7) != 0) { + return AAC_ENC_WRITTEN_BITS_ERROR; + } + + FDKaacEnc_ByteAlignment(hBs, qcOut->alignBits); + } + + frameBits -= bitMarkUp; + frameBits += FDKgetValidBits(hBs); + + transportEnc_EndAccessUnit(hTpEnc, &frameBits); + + if (frameBits != qcOut->totalBits + qcKernel->globHdrBits){ + fprintf(stderr, "frameBits != qcOut->totalBits + qcKernel->globHdrBits: %d != %d + %d", frameBits, qcOut->totalBits, qcKernel->globHdrBits); + return AAC_ENC_WRITTEN_BITS_ERROR; + } + + //fprintf(stderr, "ErrorStatus=%d", ErrorStatus); + return ErrorStatus; +} diff --git a/fdk-aac/libAACenc/src/bitenc.h b/fdk-aac/libAACenc/src/bitenc.h new file mode 100644 index 0000000..75dc068 --- /dev/null +++ b/fdk-aac/libAACenc/src/bitenc.h @@ -0,0 +1,184 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M. Werner + + Description: Bitstream encoder + +*******************************************************************************/ + +#ifndef BITENC_H +#define BITENC_H + +#include "qc_data.h" +#include "aacenc_tns.h" +#include "channel_map.h" +#include "interface.h" /* obsolete, when PSY_OUT is thrown out of the WritBS-call! */ +#include "FDK_audio.h" +#include "aacenc.h" + +#include "tpenc_lib.h" + +typedef enum { + MAX_ENCODER_CHANNELS = 9, + MAX_BLOCK_TYPES = 4, + MAX_AAC_LAYERS = 9, + MAX_LAYERS = MAX_AAC_LAYERS, /* only one core layer if present */ + FIRST_LAY = 1 /* default layer number for AAC nonscalable */ +} _MAX_CONST; + +#define BUFFER_MX_HUFFCB_SIZE \ + (32 * sizeof(INT)) /* our FDK_bitbuffer needs size of power 2 */ + +#define EL_ID_BITS (3) + +/** + * \brief Arbitrary order bitstream writer. This function can either assemble a + * bit stream and write into the bit buffer of hTpEnc or calculate the number of + * static bits (signal independent) TpEnc handle must be NULL in this + * case. Or also Calculate the minimum possible number of static bits + * which by disabling all tools e.g. MS, TNS and sbfCnt=0. The minCnt + * parameter has to be 1 in this latter case. + * \param hTpEnc Transport encoder handle. If NULL, the number of static bits + * will be returned into *pBitDemand. + * \param pElInfo + * \param qcOutChannel + * \param hReorderInfo + * \param psyOutElement + * \param psyOutChannel + * \param syntaxFlags Bit stream syntax flags as defined in FDK_audio.h (Audio + * Codec flags). + * \param aot + * \param epConfig + * \param pBitDemand Pointer to an int where the amount of bits is returned + * into. The returned value depends on if hTpEnc is NULL and minCnt. + * \param minCnt If non-zero the value returned into *pBitDemand is the absolute + * minimum required amount of static bits in order to write a valid bit stream. + * \return AAC_ENCODER_ERROR error code + */ +AAC_ENCODER_ERROR FDKaacEnc_ChannelElementWrite( + HANDLE_TRANSPORTENC hTpEnc, ELEMENT_INFO *pElInfo, + QC_OUT_CHANNEL *qcOutChannel[(2)], PSY_OUT_ELEMENT *psyOutElement, + PSY_OUT_CHANNEL *psyOutChannel[(2)], UINT syntaxFlags, + AUDIO_OBJECT_TYPE aot, SCHAR epConfig, INT *pBitDemand, UCHAR minCnt); +/** + * \brief Write bit stream or account static bits + * \param hTpEnc transport encoder handle. If NULL, the function will + * not write any bit stream data but only count the amount + * of static (signal independent) bits + * \param channelMapping Channel mapping info + * \param qcOut + * \param psyOut + * \param qcKernel + * \param hBSE + * \param aot Audio Object Type being encoded + * \param syntaxFlags Flags indicating format specific detail + * \param epConfig Error protection config + */ +AAC_ENCODER_ERROR FDKaacEnc_WriteBitstream(HANDLE_TRANSPORTENC hTpEnc, + CHANNEL_MAPPING *channelMapping, + QC_OUT *qcOut, PSY_OUT *psyOut, + QC_STATE *qcKernel, + AUDIO_OBJECT_TYPE aot, + UINT syntaxFlags, SCHAR epConfig); + +INT FDKaacEnc_writeExtensionData(HANDLE_TRANSPORTENC hTpEnc, + QC_OUT_EXTENSION *pExtension, + INT elInstanceTag, UINT alignAnchor, + UINT syntaxFlags, AUDIO_OBJECT_TYPE aot, + SCHAR epConfig); + +#endif /* BITENC_H */ diff --git a/fdk-aac/libAACenc/src/block_switch.cpp b/fdk-aac/libAACenc/src/block_switch.cpp new file mode 100644 index 0000000..c132253 --- /dev/null +++ b/fdk-aac/libAACenc/src/block_switch.cpp @@ -0,0 +1,582 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M. Werner, Tobias Chalupka + + Description: Block switching + +*******************************************************************************/ + +/****************** Includes *****************************/ + +#include "block_switch.h" +#include "genericStds.h" + +#define LOWOV_WINDOW _LOWOV_WINDOW + +/**************** internal function prototypes ***********/ + +static FIXP_DBL FDKaacEnc_GetWindowEnergy(const FIXP_DBL in[], + const INT blSwWndIdx); + +static void FDKaacEnc_CalcWindowEnergy( + BLOCK_SWITCHING_CONTROL *RESTRICT blockSwitchingControl, INT windowLen, + const INT_PCM *pTimeSignal); + +/****************** Constants *****************************/ +/* LONG START + * SHORT STOP LOWOV */ +static const INT blockType2windowShape[2][5] = { + {SINE_WINDOW, KBD_WINDOW, WRONG_WINDOW, SINE_WINDOW, KBD_WINDOW}, /* LD */ + {KBD_WINDOW, SINE_WINDOW, SINE_WINDOW, KBD_WINDOW, WRONG_WINDOW}}; /* LC */ + +/* IIR high pass coeffs */ + +#ifndef SINETABLE_16BIT + +static const FIXP_DBL hiPassCoeff[BLOCK_SWITCHING_IIR_LEN] = { + FL2FXCONST_DBL(-0.5095), FL2FXCONST_DBL(0.7548)}; + +static const FIXP_DBL accWindowNrgFac = + FL2FXCONST_DBL(0.3f); /* factor for accumulating filtered window energies */ +static const FIXP_DBL oneMinusAccWindowNrgFac = FL2FXCONST_DBL(0.7f); +/* static const float attackRatio = 10.0; */ /* lower ratio limit for attacks */ +static const FIXP_DBL invAttackRatio = + FL2FXCONST_DBL(0.1f); /* inverted lower ratio limit for attacks */ + +/* The next constants are scaled, because they are used for comparison with + * scaled values*/ +/* minimum energy for attacks */ +static const FIXP_DBL minAttackNrg = + (FL2FXCONST_DBL(1e+6f * NORM_PCM_ENERGY) >> + BLOCK_SWITCH_ENERGY_SHIFT); /* minimum energy for attacks */ + +#else + +static const FIXP_SGL hiPassCoeff[BLOCK_SWITCHING_IIR_LEN] = { + FL2FXCONST_SGL(-0.5095), FL2FXCONST_SGL(0.7548)}; + +static const FIXP_DBL accWindowNrgFac = + FL2FXCONST_DBL(0.3f); /* factor for accumulating filtered window energies */ +static const FIXP_SGL oneMinusAccWindowNrgFac = FL2FXCONST_SGL(0.7f); +/* static const float attackRatio = 10.0; */ /* lower ratio limit for attacks */ +static const FIXP_SGL invAttackRatio = + FL2FXCONST_SGL(0.1f); /* inverted lower ratio limit for attacks */ +/* minimum energy for attacks */ +static const FIXP_DBL minAttackNrg = + (FL2FXCONST_DBL(1e+6f * NORM_PCM_ENERGY) >> + BLOCK_SWITCH_ENERGY_SHIFT); /* minimum energy for attacks */ + +#endif + +/**************** internal function prototypes ***********/ + +/****************** Routines ****************************/ +void FDKaacEnc_InitBlockSwitching( + BLOCK_SWITCHING_CONTROL *blockSwitchingControl, INT isLowDelay) { + FDKmemclear(blockSwitchingControl, sizeof(BLOCK_SWITCHING_CONTROL)); + + if (isLowDelay) { + blockSwitchingControl->nBlockSwitchWindows = 4; + blockSwitchingControl->allowShortFrames = 0; + blockSwitchingControl->allowLookAhead = 0; + } else { + blockSwitchingControl->nBlockSwitchWindows = 8; + blockSwitchingControl->allowShortFrames = 1; + blockSwitchingControl->allowLookAhead = 1; + } + + blockSwitchingControl->noOfGroups = MAX_NO_OF_GROUPS; + + /* Initialize startvalue for blocktype */ + blockSwitchingControl->lastWindowSequence = LONG_WINDOW; + blockSwitchingControl->windowShape = + blockType2windowShape[blockSwitchingControl->allowShortFrames] + [blockSwitchingControl->lastWindowSequence]; +} + +static const INT suggestedGroupingTable[TRANS_FAC][MAX_NO_OF_GROUPS] = { + /* Attack in Window 0 */ {1, 3, 3, 1}, + /* Attack in Window 1 */ {1, 1, 3, 3}, + /* Attack in Window 2 */ {2, 1, 3, 2}, + /* Attack in Window 3 */ {3, 1, 3, 1}, + /* Attack in Window 4 */ {3, 1, 1, 3}, + /* Attack in Window 5 */ {3, 2, 1, 2}, + /* Attack in Window 6 */ {3, 3, 1, 1}, + /* Attack in Window 7 */ {3, 3, 1, 1}}; + +/* change block type depending on current blocktype and whether there's an + * attack */ +/* assume no look-ahead */ +static const INT chgWndSq[2][N_BLOCKTYPES] = { + /* LONG WINDOW START_WINDOW SHORT_WINDOW STOP_WINDOW, + LOWOV_WINDOW, WRONG_WINDOW */ + /*no attack*/ {LONG_WINDOW, STOP_WINDOW, WRONG_WINDOW, LONG_WINDOW, + STOP_WINDOW, WRONG_WINDOW}, + /*attack */ {START_WINDOW, LOWOV_WINDOW, WRONG_WINDOW, START_WINDOW, + LOWOV_WINDOW, WRONG_WINDOW}}; + +/* change block type depending on current blocktype and whether there's an + * attack */ +/* assume look-ahead */ +static const INT chgWndSqLkAhd[2][2][N_BLOCKTYPES] = { + /*attack LONG WINDOW START_WINDOW SHORT_WINDOW STOP_WINDOW LOWOV_WINDOW, WRONG_WINDOW */ /* last attack */ + /*no attack*/ { + {LONG_WINDOW, SHORT_WINDOW, STOP_WINDOW, LONG_WINDOW, WRONG_WINDOW, + WRONG_WINDOW}, /* no attack */ + /*attack */ {START_WINDOW, SHORT_WINDOW, SHORT_WINDOW, START_WINDOW, + WRONG_WINDOW, WRONG_WINDOW}}, /* no attack */ + /*no attack*/ {{LONG_WINDOW, SHORT_WINDOW, SHORT_WINDOW, LONG_WINDOW, + WRONG_WINDOW, WRONG_WINDOW}, /* attack */ + /*attack */ {START_WINDOW, SHORT_WINDOW, SHORT_WINDOW, + START_WINDOW, WRONG_WINDOW, + WRONG_WINDOW}} /* attack */ +}; + +int FDKaacEnc_BlockSwitching(BLOCK_SWITCHING_CONTROL *blockSwitchingControl, + const INT granuleLength, const int isLFE, + const INT_PCM *pTimeSignal) { + UINT i; + FIXP_DBL enM1, enMax; + + UINT nBlockSwitchWindows = blockSwitchingControl->nBlockSwitchWindows; + + /* for LFE : only LONG window allowed */ + if (isLFE) { + /* case LFE: */ + /* only long blocks, always use sine windows (MPEG2 AAC, MPEG4 AAC) */ + blockSwitchingControl->lastWindowSequence = LONG_WINDOW; + blockSwitchingControl->windowShape = SINE_WINDOW; + blockSwitchingControl->noOfGroups = 1; + blockSwitchingControl->groupLen[0] = 1; + + return (0); + }; + + /* Save current attack index as last attack index */ + blockSwitchingControl->lastattack = blockSwitchingControl->attack; + blockSwitchingControl->lastAttackIndex = blockSwitchingControl->attackIndex; + + /* Save current window energy as last window energy */ + FDKmemcpy(blockSwitchingControl->windowNrg[0], + blockSwitchingControl->windowNrg[1], + sizeof(blockSwitchingControl->windowNrg[0])); + FDKmemcpy(blockSwitchingControl->windowNrgF[0], + blockSwitchingControl->windowNrgF[1], + sizeof(blockSwitchingControl->windowNrgF[0])); + + if (blockSwitchingControl->allowShortFrames) { + /* Calculate suggested grouping info for the last frame */ + + /* Reset grouping info */ + FDKmemclear(blockSwitchingControl->groupLen, + sizeof(blockSwitchingControl->groupLen)); + + /* Set grouping info */ + blockSwitchingControl->noOfGroups = MAX_NO_OF_GROUPS; + + FDKmemcpy(blockSwitchingControl->groupLen, + suggestedGroupingTable[blockSwitchingControl->lastAttackIndex], + sizeof(blockSwitchingControl->groupLen)); + + if (blockSwitchingControl->attack == TRUE) + blockSwitchingControl->maxWindowNrg = + FDKaacEnc_GetWindowEnergy(blockSwitchingControl->windowNrg[0], + blockSwitchingControl->lastAttackIndex); + else + blockSwitchingControl->maxWindowNrg = FL2FXCONST_DBL(0.0); + } + + /* Calculate unfiltered and filtered energies in subwindows and combine to + * segments */ + FDKaacEnc_CalcWindowEnergy( + blockSwitchingControl, + granuleLength >> (nBlockSwitchWindows == 4 ? 2 : 3), pTimeSignal); + + /* now calculate if there is an attack */ + + /* reset attack */ + blockSwitchingControl->attack = FALSE; + + /* look for attack */ + enMax = FL2FXCONST_DBL(0.0f); + enM1 = blockSwitchingControl->windowNrgF[0][nBlockSwitchWindows - 1]; + + for (i = 0; i < nBlockSwitchWindows; i++) { + FIXP_DBL tmp = + fMultDiv2(oneMinusAccWindowNrgFac, blockSwitchingControl->accWindowNrg); + blockSwitchingControl->accWindowNrg = fMultAdd(tmp, accWindowNrgFac, enM1); + + if (fMult(blockSwitchingControl->windowNrgF[1][i], invAttackRatio) > + blockSwitchingControl->accWindowNrg) { + blockSwitchingControl->attack = TRUE; + blockSwitchingControl->attackIndex = i; + } + enM1 = blockSwitchingControl->windowNrgF[1][i]; + enMax = fixMax(enMax, enM1); + } + + if (enMax < minAttackNrg) blockSwitchingControl->attack = FALSE; + + /* Check if attack spreads over frame border */ + if ((blockSwitchingControl->attack == FALSE) && + (blockSwitchingControl->lastattack == TRUE)) { + /* if attack is in last window repeat SHORT_WINDOW */ + if (((blockSwitchingControl->windowNrgF[0][nBlockSwitchWindows - 1] >> 4) > + fMult((FIXP_DBL)(10 << (DFRACT_BITS - 1 - 4)), + blockSwitchingControl->windowNrgF[1][1])) && + (blockSwitchingControl->lastAttackIndex == + (INT)nBlockSwitchWindows - 1)) { + blockSwitchingControl->attack = TRUE; + blockSwitchingControl->attackIndex = 0; + } + } + + if (blockSwitchingControl->allowLookAhead) { + blockSwitchingControl->lastWindowSequence = + chgWndSqLkAhd[blockSwitchingControl->lastattack] + [blockSwitchingControl->attack] + [blockSwitchingControl->lastWindowSequence]; + } else { + /* Low Delay */ + blockSwitchingControl->lastWindowSequence = + chgWndSq[blockSwitchingControl->attack] + [blockSwitchingControl->lastWindowSequence]; + } + + /* update window shape */ + blockSwitchingControl->windowShape = + blockType2windowShape[blockSwitchingControl->allowShortFrames] + [blockSwitchingControl->lastWindowSequence]; + + return (0); +} + +static FIXP_DBL FDKaacEnc_GetWindowEnergy(const FIXP_DBL in[], + const INT blSwWndIdx) { + /* For coherency, change FDKaacEnc_GetWindowEnergy() to calcluate the energy + for a block switching analysis windows, not for a short block. The same is + done FDKaacEnc_CalcWindowEnergy(). The result of + FDKaacEnc_GetWindowEnergy() is used for a comparision of the max energy of + left/right channel. */ + + return in[blSwWndIdx]; +} + +static void FDKaacEnc_CalcWindowEnergy( + BLOCK_SWITCHING_CONTROL *RESTRICT blockSwitchingControl, INT windowLen, + const INT_PCM *pTimeSignal) { + INT i; + UINT w; + +#ifndef SINETABLE_16BIT + const FIXP_DBL hiPassCoeff0 = hiPassCoeff[0]; + const FIXP_DBL hiPassCoeff1 = hiPassCoeff[1]; +#else + const FIXP_SGL hiPassCoeff0 = hiPassCoeff[0]; + const FIXP_SGL hiPassCoeff1 = hiPassCoeff[1]; +#endif + + FIXP_DBL temp_iirState0 = blockSwitchingControl->iirStates[0]; + FIXP_DBL temp_iirState1 = blockSwitchingControl->iirStates[1]; + + /* sum up scalarproduct of timesignal as windowed Energies */ + for (w = 0; w < blockSwitchingControl->nBlockSwitchWindows; w++) { + ULONG temp_windowNrg = 0x0; + ULONG temp_windowNrgF = 0x0; + + /* windowNrg = sum(timesample^2) */ + for (i = 0; i < windowLen; i++) { + FIXP_DBL tempUnfiltered, t1, t2; + /* tempUnfiltered is scaled with 1 to prevent overflows during calculation + * of tempFiltred */ +#if SAMPLE_BITS == DFRACT_BITS + tempUnfiltered = (FIXP_DBL)*pTimeSignal++ >> 1; +#else + tempUnfiltered = (FIXP_DBL)*pTimeSignal++ + << (DFRACT_BITS - SAMPLE_BITS - 1); +#endif + t1 = fMultDiv2(hiPassCoeff1, tempUnfiltered - temp_iirState0); + t2 = fMultDiv2(hiPassCoeff0, temp_iirState1); + temp_iirState0 = tempUnfiltered; + temp_iirState1 = (t1 - t2) << 1; + + temp_windowNrg += (LONG)fPow2Div2(temp_iirState0) >> + (BLOCK_SWITCH_ENERGY_SHIFT - 1 - 2); + temp_windowNrgF += (LONG)fPow2Div2(temp_iirState1) >> + (BLOCK_SWITCH_ENERGY_SHIFT - 1 - 2); + } + blockSwitchingControl->windowNrg[1][w] = + (LONG)fMin(temp_windowNrg, (UINT)MAXVAL_DBL); + blockSwitchingControl->windowNrgF[1][w] = + (LONG)fMin(temp_windowNrgF, (UINT)MAXVAL_DBL); + } + blockSwitchingControl->iirStates[0] = temp_iirState0; + blockSwitchingControl->iirStates[1] = temp_iirState1; +} + +static const UCHAR synchronizedBlockTypeTable[5][5] = { + /* LONG_WINDOW START_WINDOW SHORT_WINDOW STOP_WINDOW + LOWOV_WINDOW*/ + /* LONG_WINDOW */ {LONG_WINDOW, START_WINDOW, SHORT_WINDOW, STOP_WINDOW, + LOWOV_WINDOW}, + /* START_WINDOW */ + {START_WINDOW, START_WINDOW, SHORT_WINDOW, SHORT_WINDOW, LOWOV_WINDOW}, + /* SHORT_WINDOW */ + {SHORT_WINDOW, SHORT_WINDOW, SHORT_WINDOW, SHORT_WINDOW, WRONG_WINDOW}, + /* STOP_WINDOW */ + {STOP_WINDOW, SHORT_WINDOW, SHORT_WINDOW, STOP_WINDOW, LOWOV_WINDOW}, + /* LOWOV_WINDOW */ + {LOWOV_WINDOW, LOWOV_WINDOW, WRONG_WINDOW, LOWOV_WINDOW, LOWOV_WINDOW}, +}; + +int FDKaacEnc_SyncBlockSwitching( + BLOCK_SWITCHING_CONTROL *blockSwitchingControlLeft, + BLOCK_SWITCHING_CONTROL *blockSwitchingControlRight, const INT nChannels, + const INT commonWindow) { + UCHAR patchType = LONG_WINDOW; + + if (nChannels == 2 && commonWindow == TRUE) { + /* could be better with a channel loop (need a handle to psy_data) */ + /* get suggested Block Types and synchronize */ + patchType = synchronizedBlockTypeTable[patchType][blockSwitchingControlLeft + ->lastWindowSequence]; + patchType = synchronizedBlockTypeTable[patchType][blockSwitchingControlRight + ->lastWindowSequence]; + + /* sanity check (no change from low overlap window to short winow and vice + * versa) */ + if (patchType == WRONG_WINDOW) return -1; /* mixed up AAC-LC and AAC-LD */ + + /* Set synchronized Blocktype */ + blockSwitchingControlLeft->lastWindowSequence = patchType; + blockSwitchingControlRight->lastWindowSequence = patchType; + + /* update window shape */ + blockSwitchingControlLeft->windowShape = + blockType2windowShape[blockSwitchingControlLeft->allowShortFrames] + [blockSwitchingControlLeft->lastWindowSequence]; + blockSwitchingControlRight->windowShape = + blockType2windowShape[blockSwitchingControlLeft->allowShortFrames] + [blockSwitchingControlRight->lastWindowSequence]; + } + + if (blockSwitchingControlLeft->allowShortFrames) { + int i; + + if (nChannels == 2) { + if (commonWindow == TRUE) { + /* Synchronize grouping info */ + int windowSequenceLeftOld = + blockSwitchingControlLeft->lastWindowSequence; + int windowSequenceRightOld = + blockSwitchingControlRight->lastWindowSequence; + + /* Long Blocks */ + if (patchType != SHORT_WINDOW) { + /* Set grouping info */ + blockSwitchingControlLeft->noOfGroups = 1; + blockSwitchingControlRight->noOfGroups = 1; + blockSwitchingControlLeft->groupLen[0] = 1; + blockSwitchingControlRight->groupLen[0] = 1; + + for (i = 1; i < MAX_NO_OF_GROUPS; i++) { + blockSwitchingControlLeft->groupLen[i] = 0; + blockSwitchingControlRight->groupLen[i] = 0; + } + } + + /* Short Blocks */ + else { + /* in case all two channels were detected as short-blocks before + * syncing, use the grouping of channel with higher maxWindowNrg */ + if ((windowSequenceLeftOld == SHORT_WINDOW) && + (windowSequenceRightOld == SHORT_WINDOW)) { + if (blockSwitchingControlLeft->maxWindowNrg > + blockSwitchingControlRight->maxWindowNrg) { + /* Left Channel wins */ + blockSwitchingControlRight->noOfGroups = + blockSwitchingControlLeft->noOfGroups; + for (i = 0; i < MAX_NO_OF_GROUPS; i++) { + blockSwitchingControlRight->groupLen[i] = + blockSwitchingControlLeft->groupLen[i]; + } + } else { + /* Right Channel wins */ + blockSwitchingControlLeft->noOfGroups = + blockSwitchingControlRight->noOfGroups; + for (i = 0; i < MAX_NO_OF_GROUPS; i++) { + blockSwitchingControlLeft->groupLen[i] = + blockSwitchingControlRight->groupLen[i]; + } + } + } else if ((windowSequenceLeftOld == SHORT_WINDOW) && + (windowSequenceRightOld != SHORT_WINDOW)) { + /* else use grouping of short-block channel */ + blockSwitchingControlRight->noOfGroups = + blockSwitchingControlLeft->noOfGroups; + for (i = 0; i < MAX_NO_OF_GROUPS; i++) { + blockSwitchingControlRight->groupLen[i] = + blockSwitchingControlLeft->groupLen[i]; + } + } else if ((windowSequenceRightOld == SHORT_WINDOW) && + (windowSequenceLeftOld != SHORT_WINDOW)) { + blockSwitchingControlLeft->noOfGroups = + blockSwitchingControlRight->noOfGroups; + for (i = 0; i < MAX_NO_OF_GROUPS; i++) { + blockSwitchingControlLeft->groupLen[i] = + blockSwitchingControlRight->groupLen[i]; + } + } else { + /* syncing a start and stop window ... */ + blockSwitchingControlLeft->noOfGroups = + blockSwitchingControlRight->noOfGroups = 2; + blockSwitchingControlLeft->groupLen[0] = + blockSwitchingControlRight->groupLen[0] = 4; + blockSwitchingControlLeft->groupLen[1] = + blockSwitchingControlRight->groupLen[1] = 4; + } + } /* Short Blocks */ + } else { + /* stereo, no common window */ + if (blockSwitchingControlLeft->lastWindowSequence != SHORT_WINDOW) { + blockSwitchingControlLeft->noOfGroups = 1; + blockSwitchingControlLeft->groupLen[0] = 1; + for (i = 1; i < MAX_NO_OF_GROUPS; i++) { + blockSwitchingControlLeft->groupLen[i] = 0; + } + } + if (blockSwitchingControlRight->lastWindowSequence != SHORT_WINDOW) { + blockSwitchingControlRight->noOfGroups = 1; + blockSwitchingControlRight->groupLen[0] = 1; + for (i = 1; i < MAX_NO_OF_GROUPS; i++) { + blockSwitchingControlRight->groupLen[i] = 0; + } + } + } /* common window */ + } else { + /* Mono */ + if (blockSwitchingControlLeft->lastWindowSequence != SHORT_WINDOW) { + blockSwitchingControlLeft->noOfGroups = 1; + blockSwitchingControlLeft->groupLen[0] = 1; + + for (i = 1; i < MAX_NO_OF_GROUPS; i++) { + blockSwitchingControlLeft->groupLen[i] = 0; + } + } + } + } /* allowShortFrames */ + + /* Translate LOWOV_WINDOW block type to a meaningful window shape. */ + if (!blockSwitchingControlLeft->allowShortFrames) { + if (blockSwitchingControlLeft->lastWindowSequence != LONG_WINDOW && + blockSwitchingControlLeft->lastWindowSequence != STOP_WINDOW) { + blockSwitchingControlLeft->lastWindowSequence = LONG_WINDOW; + blockSwitchingControlLeft->windowShape = LOL_WINDOW; + } + } + if (nChannels == 2) { + if (!blockSwitchingControlRight->allowShortFrames) { + if (blockSwitchingControlRight->lastWindowSequence != LONG_WINDOW && + blockSwitchingControlRight->lastWindowSequence != STOP_WINDOW) { + blockSwitchingControlRight->lastWindowSequence = LONG_WINDOW; + blockSwitchingControlRight->windowShape = LOL_WINDOW; + } + } + } + + return 0; +} diff --git a/fdk-aac/libAACenc/src/block_switch.h b/fdk-aac/libAACenc/src/block_switch.h new file mode 100644 index 0000000..ff20f84 --- /dev/null +++ b/fdk-aac/libAACenc/src/block_switch.h @@ -0,0 +1,162 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M. Werner + + Description: Block switching + +*******************************************************************************/ + +#ifndef BLOCK_SWITCH_H +#define BLOCK_SWITCH_H + +#include "common_fix.h" + +#include "psy_const.h" + +/****************** Defines ******************************/ +#define BLOCK_SWITCH_WINDOWS 8 /* number of windows for energy calculation */ + +#define BLOCK_SWITCHING_IIR_LEN \ + 2 /* Length of HighPass-IIR-Filter for Attack-Detection */ +#define BLOCK_SWITCH_ENERGY_SHIFT \ + 7 /* should be logDualis(BLOCK_SWITCH_WINDOW_LEN) to avoid overflow in \ + windowNrgs. */ + +#define LAST_WINDOW 0 +#define THIS_WINDOW 1 + +/****************** Structures ***************************/ +typedef struct { + INT lastWindowSequence; + INT windowShape; + INT lastWindowShape; + UINT nBlockSwitchWindows; /* number of windows for energy calculation */ + INT attack; + INT lastattack; + INT attackIndex; + INT lastAttackIndex; + INT allowShortFrames; /* for Low Delay, don't allow short frames */ + INT allowLookAhead; /* for Low Delay, don't do look-ahead */ + INT noOfGroups; + INT groupLen[MAX_NO_OF_GROUPS]; + FIXP_DBL maxWindowNrg; /* max energy in subwindows */ + + FIXP_DBL + windowNrg[2][BLOCK_SWITCH_WINDOWS]; /* time signal energy in Subwindows + (last and current) */ + FIXP_DBL windowNrgF[2][BLOCK_SWITCH_WINDOWS]; /* filtered time signal energy + in segments (last and + current) */ + FIXP_DBL accWindowNrg; /* recursively accumulated windowNrgF */ + + FIXP_DBL iirStates[BLOCK_SWITCHING_IIR_LEN]; /* filter delay-line */ + +} BLOCK_SWITCHING_CONTROL; + +void FDKaacEnc_InitBlockSwitching( + BLOCK_SWITCHING_CONTROL *blockSwitchingControl, INT isLowDelay); + +int FDKaacEnc_BlockSwitching(BLOCK_SWITCHING_CONTROL *blockSwitchingControl, + const INT granuleLength, const int isLFE, + const INT_PCM *pTimeSignal); + +int FDKaacEnc_SyncBlockSwitching( + BLOCK_SWITCHING_CONTROL *blockSwitchingControlLeft, + BLOCK_SWITCHING_CONTROL *blockSwitchingControlRight, const INT noOfChannels, + const INT commonWindow); + +#endif /* #ifndef BLOCK_SWITCH_H */ diff --git a/fdk-aac/libAACenc/src/channel_map.cpp b/fdk-aac/libAACenc/src/channel_map.cpp new file mode 100644 index 0000000..6ee91d5 --- /dev/null +++ b/fdk-aac/libAACenc/src/channel_map.cpp @@ -0,0 +1,664 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): A. Groeschel + + Description: channel mapping functionality + +*******************************************************************************/ + +#include "channel_map.h" +#include "bitenc.h" +#include "psy_const.h" +#include "qc_data.h" +#include "aacEnc_ram.h" +#include "FDK_tools_rom.h" + +/* channel_assignment treats the relationship of Input file channels + to the encoder channels. + This is necessary because the usual order in RIFF files (.wav) + is different from the elements order in the coder given + by Table 8.1 (implicit speaker mapping) of the AAC standard. + + In mono and stereo case, this is trivial. + In mc case, it looks like this: + + Channel Input file coder chan +5ch: + front center 2 0 (SCE channel) + left center 0 1 (1st of 1st CPE) + right center 1 2 (2nd of 1st CPE) + left surround 3 3 (1st of 2nd CPE) + right surround 4 4 (2nd of 2nd CPE) + +5.1ch: + front center 2 0 (SCE channel) + left center 0 1 (1st of 1st CPE) + right center 1 2 (2nd of 1st CPE) + left surround 4 3 (1st of 2nd CPE) + right surround 5 4 (2nd of 2nd CPE) + LFE 3 5 (LFE) +*/ + +/* Channel mode configuration tab provides, + corresponding number of channels and elements +*/ +static const CHANNEL_MODE_CONFIG_TAB channelModeConfig[] = { + {MODE_1, 1, 1, 1}, /* chCfg 1, SCE */ + {MODE_2, 2, 2, 1}, /* chCfg 2, CPE */ + {MODE_1_2, 3, 3, 2}, /* chCfg 3, SCE,CPE */ + {MODE_1_2_1, 4, 4, 3}, /* chCfg 4, SCE,CPE,SCE */ + {MODE_1_2_2, 5, 5, 3}, /* chCfg 5, SCE,CPE,CPE */ + {MODE_1_2_2_1, 6, 5, 4}, /* chCfg 6, SCE,CPE,CPE,LFE */ + {MODE_1_2_2_2_1, 8, 7, 5}, /* chCfg 7, SCE,CPE,CPE,CPE,LFE */ + {MODE_6_1, 7, 6, 5}, /* chCfg 11, SCE,CPE,CPE,SCE,LFE */ + {MODE_7_1_BACK, 8, 7, 5}, /* chCfg 12, SCE,CPE,CPE,CPE,LFE */ + {MODE_7_1_TOP_FRONT, 8, 7, 5}, /* chCfg 14, SCE,CPE,CPE,LFE,CPE */ + {MODE_7_1_REAR_SURROUND, 8, 7, + 5}, /* same as MODE_7_1_BACK, SCE,CPE,CPE,CPE,LFE */ + {MODE_7_1_FRONT_CENTER, 8, 7, + 5}, /* same as MODE_1_2_2_2_1, SCE,CPE,CPE,CPE,LFE */ + +}; + +AAC_ENCODER_ERROR FDKaacEnc_DetermineEncoderMode(CHANNEL_MODE* mode, + INT nChannels) { + INT i; + CHANNEL_MODE encMode = MODE_INVALID; + + if (*mode == MODE_UNKNOWN) { + for (i = 0; i < (INT)sizeof(channelModeConfig) / + (INT)sizeof(CHANNEL_MODE_CONFIG_TAB); + i++) { + if (channelModeConfig[i].nChannels == nChannels) { + encMode = channelModeConfig[i].encMode; + break; + } + } + *mode = encMode; + } else { + /* check if valid channel configuration */ + if (FDKaacEnc_GetChannelModeConfiguration(*mode)->nChannels == nChannels) { + encMode = *mode; + } + } + + if (encMode == MODE_INVALID) { + return AAC_ENC_UNSUPPORTED_CHANNELCONFIG; + } + + return AAC_ENC_OK; +} + +static INT FDKaacEnc_initElement(ELEMENT_INFO* elInfo, MP4_ELEMENT_ID elType, + INT* cnt, FDK_channelMapDescr* mapDescr, + UINT mapIdx, INT* it_cnt, + const FIXP_DBL relBits) { + INT error = 0; + INT counter = *cnt; + + elInfo->elType = elType; + elInfo->relativeBits = relBits; + + switch (elInfo->elType) { + case ID_SCE: + case ID_LFE: + case ID_CCE: + elInfo->nChannelsInEl = 1; + elInfo->ChannelIndex[0] = + FDK_chMapDescr_getMapValue(mapDescr, counter++, mapIdx); + elInfo->instanceTag = it_cnt[elType]++; + break; + case ID_CPE: + elInfo->nChannelsInEl = 2; + elInfo->ChannelIndex[0] = + FDK_chMapDescr_getMapValue(mapDescr, counter++, mapIdx); + elInfo->ChannelIndex[1] = + FDK_chMapDescr_getMapValue(mapDescr, counter++, mapIdx); + elInfo->instanceTag = it_cnt[elType]++; + break; + case ID_DSE: + elInfo->nChannelsInEl = 0; + elInfo->ChannelIndex[0] = 0; + elInfo->ChannelIndex[1] = 0; + elInfo->instanceTag = it_cnt[elType]++; + break; + default: + error = 1; + }; + *cnt = counter; + return error; +} + +AAC_ENCODER_ERROR FDKaacEnc_InitChannelMapping(CHANNEL_MODE mode, + CHANNEL_ORDER co, + CHANNEL_MAPPING* cm) { + INT count = 0; /* count through coder channels */ + INT it_cnt[ID_END + 1]; + INT i; + UINT mapIdx; + FDK_channelMapDescr mapDescr; + + for (i = 0; i < ID_END; i++) it_cnt[i] = 0; + + FDKmemclear(cm, sizeof(CHANNEL_MAPPING)); + + /* init channel mapping*/ + for (i = 0; i < (INT)sizeof(channelModeConfig) / + (INT)sizeof(CHANNEL_MODE_CONFIG_TAB); + i++) { + if (channelModeConfig[i].encMode == mode) { + cm->encMode = channelModeConfig[i].encMode; + cm->nChannels = channelModeConfig[i].nChannels; + cm->nChannelsEff = channelModeConfig[i].nChannelsEff; + cm->nElements = channelModeConfig[i].nElements; + + break; + } + } + + /* init map descriptor */ + FDK_chMapDescr_init(&mapDescr, NULL, 0, (co == CH_ORDER_MPEG) ? 1 : 0); + switch (mode) { + case MODE_7_1_REAR_SURROUND: /* MODE_7_1_REAR_SURROUND is equivalent to + MODE_7_1_BACK */ + mapIdx = (INT)MODE_7_1_BACK; + break; + case MODE_7_1_FRONT_CENTER: /* MODE_7_1_FRONT_CENTER is equivalent to + MODE_1_2_2_2_1 */ + mapIdx = (INT)MODE_1_2_2_2_1; + break; + default: + mapIdx = + (INT)mode > 14 + ? 0 + : (INT) + mode; /* if channel config > 14 MPEG mapping will be used */ + } + + /* init element info struct */ + switch (mode) { + case MODE_1: + /* (mono) sce */ + FDKaacEnc_initElement(&cm->elInfo[0], ID_SCE, &count, &mapDescr, mapIdx, + it_cnt, (FIXP_DBL)MAXVAL_DBL); + break; + case MODE_2: + /* (stereo) cpe */ + FDKaacEnc_initElement(&cm->elInfo[0], ID_CPE, &count, &mapDescr, mapIdx, + it_cnt, (FIXP_DBL)MAXVAL_DBL); + break; + + case MODE_1_2: + /* sce + cpe */ + FDKaacEnc_initElement(&cm->elInfo[0], ID_SCE, &count, &mapDescr, mapIdx, + it_cnt, FL2FXCONST_DBL(0.4f)); + FDKaacEnc_initElement(&cm->elInfo[1], ID_CPE, &count, &mapDescr, mapIdx, + it_cnt, FL2FXCONST_DBL(0.6f)); + break; + + case MODE_1_2_1: + /* sce + cpe + sce */ + FDKaacEnc_initElement(&cm->elInfo[0], ID_SCE, &count, &mapDescr, mapIdx, + it_cnt, FL2FXCONST_DBL(0.3f)); + FDKaacEnc_initElement(&cm->elInfo[1], ID_CPE, &count, &mapDescr, mapIdx, + it_cnt, FL2FXCONST_DBL(0.4f)); + FDKaacEnc_initElement(&cm->elInfo[2], ID_SCE, &count, &mapDescr, mapIdx, + it_cnt, FL2FXCONST_DBL(0.3f)); + break; + + case MODE_1_2_2: + /* sce + cpe + cpe */ + FDKaacEnc_initElement(&cm->elInfo[0], ID_SCE, &count, &mapDescr, mapIdx, + it_cnt, FL2FXCONST_DBL(0.26f)); + FDKaacEnc_initElement(&cm->elInfo[1], ID_CPE, &count, &mapDescr, mapIdx, + it_cnt, FL2FXCONST_DBL(0.37f)); + FDKaacEnc_initElement(&cm->elInfo[2], ID_CPE, &count, &mapDescr, mapIdx, + it_cnt, FL2FXCONST_DBL(0.37f)); + break; + + case MODE_1_2_2_1: + /* (5.1) sce + cpe + cpe + lfe */ + FDKaacEnc_initElement(&cm->elInfo[0], ID_SCE, &count, &mapDescr, mapIdx, + it_cnt, FL2FXCONST_DBL(0.24f)); + FDKaacEnc_initElement(&cm->elInfo[1], ID_CPE, &count, &mapDescr, mapIdx, + it_cnt, FL2FXCONST_DBL(0.35f)); + FDKaacEnc_initElement(&cm->elInfo[2], ID_CPE, &count, &mapDescr, mapIdx, + it_cnt, FL2FXCONST_DBL(0.35f)); + FDKaacEnc_initElement(&cm->elInfo[3], ID_LFE, &count, &mapDescr, mapIdx, + it_cnt, FL2FXCONST_DBL(0.06f)); + break; + + case MODE_6_1: + /* (6.1) sce + cpe + cpe + sce + lfe */ + FDKaacEnc_initElement(&cm->elInfo[0], ID_SCE, &count, &mapDescr, mapIdx, + it_cnt, FL2FXCONST_DBL(0.2f)); + FDKaacEnc_initElement(&cm->elInfo[1], ID_CPE, &count, &mapDescr, mapIdx, + it_cnt, FL2FXCONST_DBL(0.275f)); + FDKaacEnc_initElement(&cm->elInfo[2], ID_CPE, &count, &mapDescr, mapIdx, + it_cnt, FL2FXCONST_DBL(0.275f)); + FDKaacEnc_initElement(&cm->elInfo[3], ID_SCE, &count, &mapDescr, mapIdx, + it_cnt, FL2FXCONST_DBL(0.2f)); + FDKaacEnc_initElement(&cm->elInfo[4], ID_LFE, &count, &mapDescr, mapIdx, + it_cnt, FL2FXCONST_DBL(0.05f)); + break; + + case MODE_1_2_2_2_1: + case MODE_7_1_BACK: + case MODE_7_1_TOP_FRONT: + case MODE_7_1_REAR_SURROUND: + case MODE_7_1_FRONT_CENTER: { + /* (7.1) sce + cpe + cpe + cpe + lfe */ + /* (7.1 top) sce + cpe + cpe + lfe + cpe */ + + FDKaacEnc_initElement(&cm->elInfo[0], ID_SCE, &count, &mapDescr, mapIdx, + it_cnt, FL2FXCONST_DBL(0.18f)); + FDKaacEnc_initElement(&cm->elInfo[1], ID_CPE, &count, &mapDescr, mapIdx, + it_cnt, FL2FXCONST_DBL(0.26f)); + FDKaacEnc_initElement(&cm->elInfo[2], ID_CPE, &count, &mapDescr, mapIdx, + it_cnt, FL2FXCONST_DBL(0.26f)); + if (mode != MODE_7_1_TOP_FRONT) { + FDKaacEnc_initElement(&cm->elInfo[3], ID_CPE, &count, &mapDescr, mapIdx, + it_cnt, FL2FXCONST_DBL(0.26f)); + FDKaacEnc_initElement(&cm->elInfo[4], ID_LFE, &count, &mapDescr, mapIdx, + it_cnt, FL2FXCONST_DBL(0.04f)); + } else { + FDKaacEnc_initElement(&cm->elInfo[3], ID_LFE, &count, &mapDescr, mapIdx, + it_cnt, FL2FXCONST_DBL(0.04f)); + FDKaacEnc_initElement(&cm->elInfo[4], ID_CPE, &count, &mapDescr, mapIdx, + it_cnt, FL2FXCONST_DBL(0.26f)); + } + break; + } + + default: + //*chMap=0; + return AAC_ENC_UNSUPPORTED_CHANNELCONFIG; + }; + + FDK_ASSERT(cm->nElements <= ((8))); + + return AAC_ENC_OK; +} + +AAC_ENCODER_ERROR FDKaacEnc_InitElementBits(QC_STATE* hQC, CHANNEL_MAPPING* cm, + INT bitrateTot, INT averageBitsTot, + INT maxChannelBits) { + int sc_brTot = CountLeadingBits(bitrateTot); + + switch (cm->encMode) { + case MODE_1: + hQC->elementBits[0]->chBitrateEl = bitrateTot; + + hQC->elementBits[0]->maxBitsEl = maxChannelBits; + + hQC->elementBits[0]->relativeBitsEl = cm->elInfo[0].relativeBits; + break; + + case MODE_2: + hQC->elementBits[0]->chBitrateEl = bitrateTot >> 1; + + hQC->elementBits[0]->maxBitsEl = 2 * maxChannelBits; + + hQC->elementBits[0]->relativeBitsEl = cm->elInfo[0].relativeBits; + break; + case MODE_1_2: { + hQC->elementBits[0]->relativeBitsEl = cm->elInfo[0].relativeBits; + hQC->elementBits[1]->relativeBitsEl = cm->elInfo[1].relativeBits; + FIXP_DBL sceRate = cm->elInfo[0].relativeBits; + FIXP_DBL cpeRate = cm->elInfo[1].relativeBits; + + hQC->elementBits[0]->chBitrateEl = + fMult(sceRate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> sc_brTot; + hQC->elementBits[1]->chBitrateEl = + fMult(cpeRate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> (sc_brTot + 1); + + hQC->elementBits[0]->maxBitsEl = maxChannelBits; + hQC->elementBits[1]->maxBitsEl = 2 * maxChannelBits; + break; + } + case MODE_1_2_1: { + /* sce + cpe + sce */ + hQC->elementBits[0]->relativeBitsEl = cm->elInfo[0].relativeBits; + hQC->elementBits[1]->relativeBitsEl = cm->elInfo[1].relativeBits; + hQC->elementBits[2]->relativeBitsEl = cm->elInfo[2].relativeBits; + FIXP_DBL sce1Rate = cm->elInfo[0].relativeBits; + FIXP_DBL cpeRate = cm->elInfo[1].relativeBits; + FIXP_DBL sce2Rate = cm->elInfo[2].relativeBits; + + hQC->elementBits[0]->chBitrateEl = + fMult(sce1Rate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> sc_brTot; + hQC->elementBits[1]->chBitrateEl = + fMult(cpeRate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> (sc_brTot + 1); + hQC->elementBits[2]->chBitrateEl = + fMult(sce2Rate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> sc_brTot; + + hQC->elementBits[0]->maxBitsEl = maxChannelBits; + hQC->elementBits[1]->maxBitsEl = 2 * maxChannelBits; + hQC->elementBits[2]->maxBitsEl = maxChannelBits; + break; + } + case MODE_1_2_2: { + /* sce + cpe + cpe */ + hQC->elementBits[0]->relativeBitsEl = cm->elInfo[0].relativeBits; + hQC->elementBits[1]->relativeBitsEl = cm->elInfo[1].relativeBits; + hQC->elementBits[2]->relativeBitsEl = cm->elInfo[2].relativeBits; + FIXP_DBL sceRate = cm->elInfo[0].relativeBits; + FIXP_DBL cpe1Rate = cm->elInfo[1].relativeBits; + FIXP_DBL cpe2Rate = cm->elInfo[2].relativeBits; + + hQC->elementBits[0]->chBitrateEl = + fMult(sceRate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> sc_brTot; + hQC->elementBits[1]->chBitrateEl = + fMult(cpe1Rate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> (sc_brTot + 1); + hQC->elementBits[2]->chBitrateEl = + fMult(cpe2Rate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> (sc_brTot + 1); + + hQC->elementBits[0]->maxBitsEl = maxChannelBits; + hQC->elementBits[1]->maxBitsEl = 2 * maxChannelBits; + hQC->elementBits[2]->maxBitsEl = 2 * maxChannelBits; + break; + } + case MODE_1_2_2_1: { + /* (5.1) sce + cpe + cpe + lfe */ + hQC->elementBits[0]->relativeBitsEl = cm->elInfo[0].relativeBits; + hQC->elementBits[1]->relativeBitsEl = cm->elInfo[1].relativeBits; + hQC->elementBits[2]->relativeBitsEl = cm->elInfo[2].relativeBits; + hQC->elementBits[3]->relativeBitsEl = cm->elInfo[3].relativeBits; + FIXP_DBL sceRate = cm->elInfo[0].relativeBits; + FIXP_DBL cpe1Rate = cm->elInfo[1].relativeBits; + FIXP_DBL cpe2Rate = cm->elInfo[2].relativeBits; + FIXP_DBL lfeRate = cm->elInfo[3].relativeBits; + + int maxBitsTot = + maxChannelBits * 5; /* LFE does not add to bit reservoir */ + int sc = CountLeadingBits(fixMax(maxChannelBits, averageBitsTot)); + int maxLfeBits = (int)fMax( + (INT)((fMult(lfeRate, (FIXP_DBL)(maxChannelBits << sc)) >> sc) << 1), + (INT)((fMult(FL2FXCONST_DBL(1.1f / 2.f), + fMult(lfeRate, (FIXP_DBL)(averageBitsTot << sc))) + << 1) >> + sc)); + + maxChannelBits = (maxBitsTot - maxLfeBits); + sc = CountLeadingBits(maxChannelBits); + + maxChannelBits = + fMult((FIXP_DBL)maxChannelBits << sc, GetInvInt(5)) >> sc; + + hQC->elementBits[0]->chBitrateEl = + fMult(sceRate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> sc_brTot; + hQC->elementBits[1]->chBitrateEl = + fMult(cpe1Rate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> (sc_brTot + 1); + hQC->elementBits[2]->chBitrateEl = + fMult(cpe2Rate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> (sc_brTot + 1); + hQC->elementBits[3]->chBitrateEl = + fMult(lfeRate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> sc_brTot; + + hQC->elementBits[0]->maxBitsEl = maxChannelBits; + hQC->elementBits[1]->maxBitsEl = 2 * maxChannelBits; + hQC->elementBits[2]->maxBitsEl = 2 * maxChannelBits; + hQC->elementBits[3]->maxBitsEl = maxLfeBits; + + break; + } + case MODE_6_1: { + /* (6.1) sce + cpe + cpe + sce + lfe */ + FIXP_DBL sceRate = hQC->elementBits[0]->relativeBitsEl = + cm->elInfo[0].relativeBits; + FIXP_DBL cpe1Rate = hQC->elementBits[1]->relativeBitsEl = + cm->elInfo[1].relativeBits; + FIXP_DBL cpe2Rate = hQC->elementBits[2]->relativeBitsEl = + cm->elInfo[2].relativeBits; + FIXP_DBL sce2Rate = hQC->elementBits[3]->relativeBitsEl = + cm->elInfo[3].relativeBits; + FIXP_DBL lfeRate = hQC->elementBits[4]->relativeBitsEl = + cm->elInfo[4].relativeBits; + + int maxBitsTot = + maxChannelBits * 6; /* LFE does not add to bit reservoir */ + int sc = CountLeadingBits(fixMax(maxChannelBits, averageBitsTot)); + int maxLfeBits = (int)fMax( + (INT)((fMult(lfeRate, (FIXP_DBL)(maxChannelBits << sc)) >> sc) << 1), + (INT)((fMult(FL2FXCONST_DBL(1.1f / 2.f), + fMult(lfeRate, (FIXP_DBL)(averageBitsTot << sc))) + << 1) >> + sc)); + + maxChannelBits = (maxBitsTot - maxLfeBits) / 6; + + hQC->elementBits[0]->chBitrateEl = + fMult(sceRate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> sc_brTot; + hQC->elementBits[1]->chBitrateEl = + fMult(cpe1Rate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> (sc_brTot + 1); + hQC->elementBits[2]->chBitrateEl = + fMult(cpe2Rate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> (sc_brTot + 1); + hQC->elementBits[3]->chBitrateEl = + fMult(sce2Rate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> (sc_brTot + 1); + hQC->elementBits[4]->chBitrateEl = + fMult(lfeRate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> sc_brTot; + + hQC->elementBits[0]->maxBitsEl = maxChannelBits; + hQC->elementBits[1]->maxBitsEl = 2 * maxChannelBits; + hQC->elementBits[2]->maxBitsEl = 2 * maxChannelBits; + hQC->elementBits[3]->maxBitsEl = maxChannelBits; + hQC->elementBits[4]->maxBitsEl = maxLfeBits; + break; + } + case MODE_7_1_TOP_FRONT: + case MODE_7_1_BACK: + case MODE_7_1_REAR_SURROUND: + case MODE_7_1_FRONT_CENTER: + case MODE_1_2_2_2_1: { + int cpe3Idx = (cm->encMode != MODE_7_1_TOP_FRONT) ? 3 : 4; + int lfeIdx = (cm->encMode != MODE_7_1_TOP_FRONT) ? 4 : 3; + + /* (7.1) sce + cpe + cpe + cpe + lfe */ + FIXP_DBL sceRate = hQC->elementBits[0]->relativeBitsEl = + cm->elInfo[0].relativeBits; + FIXP_DBL cpe1Rate = hQC->elementBits[1]->relativeBitsEl = + cm->elInfo[1].relativeBits; + FIXP_DBL cpe2Rate = hQC->elementBits[2]->relativeBitsEl = + cm->elInfo[2].relativeBits; + FIXP_DBL cpe3Rate = hQC->elementBits[cpe3Idx]->relativeBitsEl = + cm->elInfo[cpe3Idx].relativeBits; + FIXP_DBL lfeRate = hQC->elementBits[lfeIdx]->relativeBitsEl = + cm->elInfo[lfeIdx].relativeBits; + + int maxBitsTot = + maxChannelBits * 7; /* LFE does not add to bit reservoir */ + int sc = CountLeadingBits(fixMax(maxChannelBits, averageBitsTot)); + int maxLfeBits = (int)fMax( + (INT)((fMult(lfeRate, (FIXP_DBL)(maxChannelBits << sc)) >> sc) << 1), + (INT)((fMult(FL2FXCONST_DBL(1.1f / 2.f), + fMult(lfeRate, (FIXP_DBL)(averageBitsTot << sc))) + << 1) >> + sc)); + + maxChannelBits = (maxBitsTot - maxLfeBits) / 7; + + hQC->elementBits[0]->chBitrateEl = + fMult(sceRate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> sc_brTot; + hQC->elementBits[1]->chBitrateEl = + fMult(cpe1Rate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> (sc_brTot + 1); + hQC->elementBits[2]->chBitrateEl = + fMult(cpe2Rate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> (sc_brTot + 1); + hQC->elementBits[cpe3Idx]->chBitrateEl = + fMult(cpe3Rate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> (sc_brTot + 1); + hQC->elementBits[lfeIdx]->chBitrateEl = + fMult(lfeRate, (FIXP_DBL)(bitrateTot << sc_brTot)) >> sc_brTot; + + hQC->elementBits[0]->maxBitsEl = maxChannelBits; + hQC->elementBits[1]->maxBitsEl = 2 * maxChannelBits; + hQC->elementBits[2]->maxBitsEl = 2 * maxChannelBits; + hQC->elementBits[cpe3Idx]->maxBitsEl = 2 * maxChannelBits; + hQC->elementBits[lfeIdx]->maxBitsEl = maxLfeBits; + break; + } + + default: + return AAC_ENC_UNSUPPORTED_CHANNELCONFIG; + } + + return AAC_ENC_OK; +} + +/********************************************************************************/ +/* */ +/* function: GetMonoStereoMODE(const CHANNEL_MODE mode) */ +/* */ +/* description: Determines encoder setting from channel mode. */ +/* Multichannel modes are mapped to mono or stereo modes */ +/* returns MODE_MONO in case of mono, */ +/* MODE_STEREO in case of stereo */ +/* MODE_INVALID in case of error */ +/* */ +/* input: CHANNEL_MODE mode: Encoder mode (see qc_data.h). */ +/* output: return: CM_STEREO_MODE monoStereoSetting */ +/* (MODE_INVALID: error, */ +/* MODE_MONO: mono */ +/* MODE_STEREO: stereo). */ +/* */ +/* misc: No memory is allocated. */ +/* */ +/********************************************************************************/ + +ELEMENT_MODE FDKaacEnc_GetMonoStereoMode(const CHANNEL_MODE mode) { + ELEMENT_MODE monoStereoSetting = EL_MODE_INVALID; + + switch (mode) { + case MODE_1: /* mono setups */ + monoStereoSetting = EL_MODE_MONO; + break; + + case MODE_2: /* stereo setups */ + case MODE_1_2: + case MODE_1_2_1: + case MODE_1_2_2: + case MODE_1_2_2_1: + case MODE_6_1: + case MODE_1_2_2_2_1: + case MODE_7_1_REAR_SURROUND: + case MODE_7_1_FRONT_CENTER: + case MODE_7_1_BACK: + case MODE_7_1_TOP_FRONT: + monoStereoSetting = EL_MODE_STEREO; + break; + + default: /* error */ + monoStereoSetting = EL_MODE_INVALID; + break; + } + + return monoStereoSetting; +} + +const CHANNEL_MODE_CONFIG_TAB* FDKaacEnc_GetChannelModeConfiguration( + const CHANNEL_MODE mode) { + INT i; + const CHANNEL_MODE_CONFIG_TAB* cm_config = NULL; + + /* get channel mode config */ + for (i = 0; i < (INT)sizeof(channelModeConfig) / + (INT)sizeof(CHANNEL_MODE_CONFIG_TAB); + i++) { + if (channelModeConfig[i].encMode == mode) { + cm_config = &channelModeConfig[i]; + break; + } + } + return cm_config; +} diff --git a/fdk-aac/libAACenc/src/channel_map.h b/fdk-aac/libAACenc/src/channel_map.h new file mode 100644 index 0000000..f9154cd --- /dev/null +++ b/fdk-aac/libAACenc/src/channel_map.h @@ -0,0 +1,136 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): A. Groeschel + + Description: channel mapping functionality + +*******************************************************************************/ + +#ifndef CHANNEL_MAP_H +#define CHANNEL_MAP_H + +#include "aacenc.h" +#include "psy_const.h" +#include "qc_data.h" + +typedef struct { + CHANNEL_MODE encMode; + INT nChannels; + INT nChannelsEff; + INT nElements; +} CHANNEL_MODE_CONFIG_TAB; + +/* Element mode */ +typedef enum { EL_MODE_INVALID = 0, EL_MODE_MONO, EL_MODE_STEREO } ELEMENT_MODE; + +AAC_ENCODER_ERROR FDKaacEnc_DetermineEncoderMode(CHANNEL_MODE* mode, + INT nChannels); + +AAC_ENCODER_ERROR FDKaacEnc_InitChannelMapping(CHANNEL_MODE mode, + CHANNEL_ORDER co, + CHANNEL_MAPPING* chMap); + +AAC_ENCODER_ERROR FDKaacEnc_InitElementBits(QC_STATE* hQC, CHANNEL_MAPPING* cm, + INT bitrateTot, INT averageBitsTot, + INT maxChannelBits); + +ELEMENT_MODE FDKaacEnc_GetMonoStereoMode(const CHANNEL_MODE mode); + +const CHANNEL_MODE_CONFIG_TAB* FDKaacEnc_GetChannelModeConfiguration( + const CHANNEL_MODE mode); + +#endif /* CHANNEL_MAP_H */ diff --git a/fdk-aac/libAACenc/src/chaosmeasure.cpp b/fdk-aac/libAACenc/src/chaosmeasure.cpp new file mode 100644 index 0000000..664284b --- /dev/null +++ b/fdk-aac/libAACenc/src/chaosmeasure.cpp @@ -0,0 +1,191 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M. Werner + + Description: Chaos measure calculation + +*******************************************************************************/ + +#include "chaosmeasure.h" + +/***************************************************************************** + functionname: FDKaacEnc_FDKaacEnc_CalculateChaosMeasurePeakFast + description: Eberlein method of chaos measure calculation by high-pass + filtering amplitude spectrum + A special case of FDKaacEnc_CalculateChaosMeasureTonalGeneric +-- highly optimized +*****************************************************************************/ +static void FDKaacEnc_FDKaacEnc_CalculateChaosMeasurePeakFast( + FIXP_DBL *RESTRICT paMDCTDataNM0, INT numberOfLines, + FIXP_DBL *RESTRICT chaosMeasure) { + INT i, j; + + /* calculate chaos measure by "peak filter" */ + /* make even and odd pass through data */ + FIXP_DBL left_0_div2, + center_0; /* left, center tap of filter, even numbered */ + FIXP_DBL left_1_div2, center_1; /* left, center tap of filter, odd numbered */ + + left_0_div2 = (FIXP_DBL)(((LONG)paMDCTDataNM0[0] ^ + ((LONG)paMDCTDataNM0[0] >> (DFRACT_BITS - 1))) >> + 1); + left_1_div2 = (FIXP_DBL)(((LONG)paMDCTDataNM0[1] ^ + ((LONG)paMDCTDataNM0[1] >> (DFRACT_BITS - 1))) >> + 1); + center_0 = (FIXP_DBL)((LONG)paMDCTDataNM0[2] ^ + ((LONG)paMDCTDataNM0[2] >> (DFRACT_BITS - 1))); + center_1 = (FIXP_DBL)((LONG)paMDCTDataNM0[3] ^ + ((LONG)paMDCTDataNM0[3] >> (DFRACT_BITS - 1))); + + for (j = 2; j < numberOfLines - 2; j += 2) { + FIXP_DBL right_0 = + (FIXP_DBL)((LONG)paMDCTDataNM0[j + 2] ^ + ((LONG)paMDCTDataNM0[j + 2] >> (DFRACT_BITS - 1))); + FIXP_DBL tmp_0 = left_0_div2 + (right_0 >> 1); + FIXP_DBL right_1 = + (FIXP_DBL)((LONG)paMDCTDataNM0[j + 3] ^ + ((LONG)paMDCTDataNM0[j + 3] >> (DFRACT_BITS - 1))); + FIXP_DBL tmp_1 = left_1_div2 + (right_1 >> 1); + + if (tmp_0 < center_0) { + INT leadingBits = CntLeadingZeros(center_0) - 1; + tmp_0 = schur_div(tmp_0 << leadingBits, center_0 << leadingBits, 8); + tmp_0 = fMult(tmp_0, tmp_0); + } else { + tmp_0 = (FIXP_DBL)MAXVAL_DBL; + } + chaosMeasure[j + 0] = tmp_0; + left_0_div2 = center_0 >> 1; + center_0 = right_0; + + if (tmp_1 < center_1) { + INT leadingBits = CntLeadingZeros(center_1) - 1; + tmp_1 = schur_div(tmp_1 << leadingBits, center_1 << leadingBits, 8); + tmp_1 = fMult(tmp_1, tmp_1); + } else { + tmp_1 = (FIXP_DBL)MAXVAL_DBL; + } + + left_1_div2 = center_1 >> 1; + center_1 = right_1; + chaosMeasure[j + 1] = tmp_1; + } + + /* provide chaos measure for first few lines */ + chaosMeasure[0] = chaosMeasure[2]; + chaosMeasure[1] = chaosMeasure[2]; + + /* provide chaos measure for last few lines */ + for (i = (numberOfLines - 3); i < numberOfLines; i++) + chaosMeasure[i] = FL2FXCONST_DBL(0.5); +} + +/***************************************************************************** + functionname: FDKaacEnc_CalculateChaosMeasure + description: calculates a chaosmeasure for every line, different methods + are available. 0 means tonal, 1 means noiselike + returns: + input: MDCT data, number of lines + output: chaosMeasure +*****************************************************************************/ +void FDKaacEnc_CalculateChaosMeasure(FIXP_DBL *paMDCTDataNM0, INT numberOfLines, + FIXP_DBL *chaosMeasure) + +{ + FDKaacEnc_FDKaacEnc_CalculateChaosMeasurePeakFast( + paMDCTDataNM0, numberOfLines, chaosMeasure); +} diff --git a/fdk-aac/libAACenc/src/chaosmeasure.h b/fdk-aac/libAACenc/src/chaosmeasure.h new file mode 100644 index 0000000..60d4137 --- /dev/null +++ b/fdk-aac/libAACenc/src/chaosmeasure.h @@ -0,0 +1,112 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M. Werner + + Description: Chaos measure calculation + +*******************************************************************************/ + +#ifndef CHAOSMEASURE_H +#define CHAOSMEASURE_H + +#include "common_fix.h" +#include "psy_const.h" + +void FDKaacEnc_CalculateChaosMeasure(FIXP_DBL *paMDCTDataNM0, INT numberOfLines, + FIXP_DBL *chaosMeasure); + +#endif /* CHAOSMEASURE_H */ diff --git a/fdk-aac/libAACenc/src/dyn_bits.cpp b/fdk-aac/libAACenc/src/dyn_bits.cpp new file mode 100644 index 0000000..b52dc2e --- /dev/null +++ b/fdk-aac/libAACenc/src/dyn_bits.cpp @@ -0,0 +1,665 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M.Werner + + Description: Noiseless coder module + +*******************************************************************************/ + +#include "dyn_bits.h" +#include "bit_cnt.h" +#include "psy_const.h" +#include "aacenc_pns.h" +#include "aacEnc_ram.h" +#include "aacEnc_rom.h" + +typedef INT (*lookUpTable)[CODE_BOOK_ESC_NDX + 1]; + +static INT FDKaacEnc_getSideInfoBits(const SECTION_INFO* const huffsection, + const SHORT* const sideInfoTab, + const INT useHCR) { + INT sideInfoBits; + + if (useHCR && + ((huffsection->codeBook == 11) || (huffsection->codeBook >= 16))) { + sideInfoBits = 5; + } else { + sideInfoBits = sideInfoTab[huffsection->sfbCnt]; + } + + return (sideInfoBits); +} + +/* count bits using all possible tables */ +static void FDKaacEnc_buildBitLookUp( + const SHORT* const quantSpectrum, const INT maxSfb, + const INT* const sfbOffset, const UINT* const sfbMax, + INT bitLookUp[MAX_SFB_LONG][CODE_BOOK_ESC_NDX + 1], + SECTION_INFO* const huffsection) { + INT i, sfbWidth; + + for (i = 0; i < maxSfb; i++) { + huffsection[i].sfbCnt = 1; + huffsection[i].sfbStart = i; + huffsection[i].sectionBits = INVALID_BITCOUNT; + huffsection[i].codeBook = -1; + sfbWidth = sfbOffset[i + 1] - sfbOffset[i]; + FDKaacEnc_bitCount(quantSpectrum + sfbOffset[i], sfbWidth, sfbMax[i], + bitLookUp[i]); + } +} + +/* essential helper functions */ +static inline INT FDKaacEnc_findBestBook(const INT* const bc, INT* const book, + const INT useVCB11) { + INT minBits = INVALID_BITCOUNT, j; + + int end = CODE_BOOK_ESC_NDX; + + for (j = 0; j <= end; j++) { + if (bc[j] < minBits) { + minBits = bc[j]; + *book = j; + } + } + return (minBits); +} + +static inline INT FDKaacEnc_findMinMergeBits(const INT* const bc1, + const INT* const bc2, + const INT useVCB11) { + INT minBits = INVALID_BITCOUNT, j; + + DWORD_ALIGNED(bc1); + DWORD_ALIGNED(bc2); + + for (j = 0; j <= CODE_BOOK_ESC_NDX; j++) { + minBits = fixMin(minBits, bc1[j] + bc2[j]); + } + return (minBits); +} + +static inline void FDKaacEnc_mergeBitLookUp(INT* const RESTRICT bc1, + const INT* const RESTRICT bc2) { + int j; + + for (j = 0; j <= CODE_BOOK_ESC_NDX; j++) { + FDK_ASSERT(INVALID_BITCOUNT == 0x1FFFFFFF); + bc1[j] = fixMin(bc1[j] + bc2[j], INVALID_BITCOUNT); + } +} + +static inline INT FDKaacEnc_findMaxMerge(const INT* const mergeGainLookUp, + const SECTION_INFO* const huffsection, + const INT maxSfb, INT* const maxNdx) { + INT i, maxMergeGain = 0; + int lastMaxNdx = 0; + + for (i = 0; i + huffsection[i].sfbCnt < maxSfb; i += huffsection[i].sfbCnt) { + if (mergeGainLookUp[i] > maxMergeGain) { + maxMergeGain = mergeGainLookUp[i]; + lastMaxNdx = i; + } + } + *maxNdx = lastMaxNdx; + return (maxMergeGain); +} + +static inline INT FDKaacEnc_CalcMergeGain( + const SECTION_INFO* const huffsection, + const INT bitLookUp[MAX_SFB_LONG][CODE_BOOK_ESC_NDX + 1], + const SHORT* const sideInfoTab, const INT ndx1, const INT ndx2, + const INT useVCB11) { + INT MergeGain, MergeBits, SplitBits; + + MergeBits = + sideInfoTab[huffsection[ndx1].sfbCnt + huffsection[ndx2].sfbCnt] + + FDKaacEnc_findMinMergeBits(bitLookUp[ndx1], bitLookUp[ndx2], useVCB11); + SplitBits = + huffsection[ndx1].sectionBits + + huffsection[ndx2].sectionBits; /* Bit amount for splitted huffsections */ + MergeGain = SplitBits - MergeBits; + + if ((huffsection[ndx1].codeBook == CODE_BOOK_PNS_NO) || + (huffsection[ndx2].codeBook == CODE_BOOK_PNS_NO) || + (huffsection[ndx1].codeBook == CODE_BOOK_IS_OUT_OF_PHASE_NO) || + (huffsection[ndx2].codeBook == CODE_BOOK_IS_OUT_OF_PHASE_NO) || + (huffsection[ndx1].codeBook == CODE_BOOK_IS_IN_PHASE_NO) || + (huffsection[ndx2].codeBook == CODE_BOOK_IS_IN_PHASE_NO)) { + MergeGain = -1; + } + + return (MergeGain); +} + +/* sectioning Stage 0:find minimum codbooks */ +static void FDKaacEnc_gmStage0( + SECTION_INFO* const RESTRICT huffsection, + const INT bitLookUp[MAX_SFB_LONG][CODE_BOOK_ESC_NDX + 1], const INT maxSfb, + const INT* const noiseNrg, const INT* const isBook) { + INT i; + + for (i = 0; i < maxSfb; i++) { + /* Side-Info bits will be calculated in Stage 1! */ + if (huffsection[i].sectionBits == INVALID_BITCOUNT) { + /* intensity and pns codebooks are already allocated in bitcount.c */ + if (noiseNrg[i] != NO_NOISE_PNS) { + huffsection[i].codeBook = CODE_BOOK_PNS_NO; + huffsection[i].sectionBits = 0; + } else if (isBook[i]) { + huffsection[i].codeBook = isBook[i]; + huffsection[i].sectionBits = 0; + } else { + huffsection[i].sectionBits = + FDKaacEnc_findBestBook(bitLookUp[i], &(huffsection[i].codeBook), + 0); /* useVCB11 must be 0!!! */ + } + } + } +} + +/* + sectioning Stage 1:merge all connected regions with the same code book and + calculate side info + */ +static void FDKaacEnc_gmStage1( + SECTION_INFO* const RESTRICT huffsection, + INT bitLookUp[MAX_SFB_LONG][CODE_BOOK_ESC_NDX + 1], const INT maxSfb, + const SHORT* const sideInfoTab, const INT useVCB11) { + INT mergeStart = 0, mergeEnd; + + do { + for (mergeEnd = mergeStart + 1; mergeEnd < maxSfb; mergeEnd++) { + if (huffsection[mergeStart].codeBook != huffsection[mergeEnd].codeBook) + break; + + /* we can merge. update tables, side info bits will be updated outside of + * this loop */ + huffsection[mergeStart].sfbCnt++; + huffsection[mergeStart].sectionBits += huffsection[mergeEnd].sectionBits; + + /* update bit look up for all code books */ + FDKaacEnc_mergeBitLookUp(bitLookUp[mergeStart], bitLookUp[mergeEnd]); + } + + /* add side info info bits */ + huffsection[mergeStart].sectionBits += FDKaacEnc_getSideInfoBits( + &huffsection[mergeStart], sideInfoTab, useVCB11); + huffsection[mergeEnd - 1].sfbStart = + huffsection[mergeStart].sfbStart; /* speed up prev search */ + + mergeStart = mergeEnd; + + } while (mergeStart < maxSfb); +} + +/* + sectioning Stage 2:greedy merge algorithm, merge connected sections with + maximum bit gain until no more gain is possible + */ +static inline void FDKaacEnc_gmStage2( + SECTION_INFO* const RESTRICT huffsection, + INT* const RESTRICT mergeGainLookUp, + INT bitLookUp[MAX_SFB_LONG][CODE_BOOK_ESC_NDX + 1], const INT maxSfb, + const SHORT* const sideInfoTab, const INT useVCB11) { + INT i; + + for (i = 0; i + huffsection[i].sfbCnt < maxSfb; i += huffsection[i].sfbCnt) { + mergeGainLookUp[i] = + FDKaacEnc_CalcMergeGain(huffsection, bitLookUp, sideInfoTab, i, + i + huffsection[i].sfbCnt, useVCB11); + } + + while (TRUE) { + INT maxMergeGain, maxNdx, maxNdxNext, maxNdxLast; + + maxMergeGain = + FDKaacEnc_findMaxMerge(mergeGainLookUp, huffsection, maxSfb, &maxNdx); + + /* exit while loop if no more gain is possible */ + if (maxMergeGain <= 0) break; + + maxNdxNext = maxNdx + huffsection[maxNdx].sfbCnt; + + /* merge sections with maximum bit gain */ + huffsection[maxNdx].sfbCnt += huffsection[maxNdxNext].sfbCnt; + huffsection[maxNdx].sectionBits += + huffsection[maxNdxNext].sectionBits - maxMergeGain; + + /* update bit look up table for merged huffsection */ + FDKaacEnc_mergeBitLookUp(bitLookUp[maxNdx], bitLookUp[maxNdxNext]); + + /* update mergeLookUpTable */ + if (maxNdx != 0) { + maxNdxLast = huffsection[maxNdx - 1].sfbStart; + mergeGainLookUp[maxNdxLast] = FDKaacEnc_CalcMergeGain( + huffsection, bitLookUp, sideInfoTab, maxNdxLast, maxNdx, useVCB11); + } + maxNdxNext = maxNdx + huffsection[maxNdx].sfbCnt; + + huffsection[maxNdxNext - 1].sfbStart = huffsection[maxNdx].sfbStart; + + if (maxNdxNext < maxSfb) + mergeGainLookUp[maxNdx] = FDKaacEnc_CalcMergeGain( + huffsection, bitLookUp, sideInfoTab, maxNdx, maxNdxNext, useVCB11); + } +} + +/* count bits used by the noiseless coder */ +static void FDKaacEnc_noiselessCounter( + SECTION_DATA* const RESTRICT sectionData, INT mergeGainLookUp[MAX_SFB_LONG], + INT bitLookUp[MAX_SFB_LONG][CODE_BOOK_ESC_NDX + 1], + const SHORT* const quantSpectrum, const UINT* const maxValueInSfb, + const INT* const sfbOffset, const INT blockType, const INT* const noiseNrg, + const INT* const isBook, const INT useVCB11) { + INT grpNdx, i; + const SHORT* sideInfoTab = NULL; + SECTION_INFO* huffsection; + + /* use appropriate side info table */ + switch (blockType) { + case LONG_WINDOW: + case START_WINDOW: + case STOP_WINDOW: + default: + sideInfoTab = FDKaacEnc_sideInfoTabLong; + break; + case SHORT_WINDOW: + sideInfoTab = FDKaacEnc_sideInfoTabShort; + break; + } + + FDK_ASSERT(sideInfoTab != NULL); + + sectionData->noOfSections = 0; + sectionData->huffmanBits = 0; + sectionData->sideInfoBits = 0; + + if (sectionData->maxSfbPerGroup == 0) return; + + /* loop trough groups */ + for (grpNdx = 0; grpNdx < sectionData->sfbCnt; + grpNdx += sectionData->sfbPerGroup) { + huffsection = sectionData->huffsection + sectionData->noOfSections; + + /* count bits in this group */ + FDKaacEnc_buildBitLookUp(quantSpectrum, sectionData->maxSfbPerGroup, + sfbOffset + grpNdx, maxValueInSfb + grpNdx, + bitLookUp, huffsection); + + /* 0.Stage :Find minimum Codebooks */ + FDKaacEnc_gmStage0(huffsection, bitLookUp, sectionData->maxSfbPerGroup, + noiseNrg + grpNdx, isBook + grpNdx); + + /* 1.Stage :Merge all connected regions with the same code book */ + FDKaacEnc_gmStage1(huffsection, bitLookUp, sectionData->maxSfbPerGroup, + sideInfoTab, useVCB11); + + /* + 2.Stage + greedy merge algorithm, merge connected huffsections with maximum bit + gain until no more gain is possible + */ + + FDKaacEnc_gmStage2(huffsection, mergeGainLookUp, bitLookUp, + sectionData->maxSfbPerGroup, sideInfoTab, useVCB11); + + /* + compress output, calculate total huff and side bits + since we did not update the actual codebook in stage 2 + to save time, we must set it here for later use in bitenc + */ + + for (i = 0; i < sectionData->maxSfbPerGroup; i += huffsection[i].sfbCnt) { + if ((huffsection[i].codeBook == CODE_BOOK_PNS_NO) || + (huffsection[i].codeBook == CODE_BOOK_IS_OUT_OF_PHASE_NO) || + (huffsection[i].codeBook == CODE_BOOK_IS_IN_PHASE_NO)) { + huffsection[i].sectionBits = 0; + } else { + /* the sections in the sectionData are now marked with the optimal code + * book */ + + FDKaacEnc_findBestBook(bitLookUp[i], &(huffsection[i].codeBook), + useVCB11); + + sectionData->huffmanBits += + huffsection[i].sectionBits - + FDKaacEnc_getSideInfoBits(&huffsection[i], sideInfoTab, useVCB11); + } + + huffsection[i].sfbStart += grpNdx; + + /* sum up side info bits (section data bits) */ + sectionData->sideInfoBits += + FDKaacEnc_getSideInfoBits(&huffsection[i], sideInfoTab, useVCB11); + sectionData->huffsection[sectionData->noOfSections++] = huffsection[i]; + } + } +} + +/******************************************************************************* + + functionname: FDKaacEnc_scfCount + returns : --- + description : count bits used by scalefactors. + + not in all cases if maxValueInSfb[] == 0 we set deltaScf + to zero. only if the difference of the last and future + scalefacGain is not greater then CODE_BOOK_SCF_LAV (60). + + example: + ^ + scalefacGain | + | + | last 75 + | | + | | + | | + | | current 50 + | | | + | | | + | | | + | | | + | | | future 5 + | | | | + --- ... ---------------------------- ... ---------> + sfb + + + if maxValueInSfb[] of current is zero because of a + notfallstrategie, we do not save bits and transmit a + deltaScf of 25. otherwise the deltaScf between the last + scalfacGain (75) and the future scalefacGain (5) is 70. + +********************************************************************************/ +static void FDKaacEnc_scfCount(const INT* const scalefacGain, + const UINT* const maxValueInSfb, + SECTION_DATA* const RESTRICT sectionData, + const INT* const isScale) { + INT i, j, k, m, n; + + INT lastValScf = 0; + INT deltaScf = 0; + INT found = 0; + INT scfSkipCounter = 0; + INT lastValIs = 0; + + sectionData->scalefacBits = 0; + + if (scalefacGain == NULL) return; + + sectionData->firstScf = 0; + + for (i = 0; i < sectionData->noOfSections; i++) { + if (sectionData->huffsection[i].codeBook != CODE_BOOK_ZERO_NO) { + sectionData->firstScf = sectionData->huffsection[i].sfbStart; + lastValScf = scalefacGain[sectionData->firstScf]; + break; + } + } + + for (i = 0; i < sectionData->noOfSections; i++) { + if ((sectionData->huffsection[i].codeBook == + CODE_BOOK_IS_OUT_OF_PHASE_NO) || + (sectionData->huffsection[i].codeBook == CODE_BOOK_IS_IN_PHASE_NO)) { + for (j = sectionData->huffsection[i].sfbStart; + j < sectionData->huffsection[i].sfbStart + + sectionData->huffsection[i].sfbCnt; + j++) { + INT deltaIs = isScale[j] - lastValIs; + lastValIs = isScale[j]; + sectionData->scalefacBits += + FDKaacEnc_bitCountScalefactorDelta(deltaIs); + } + } /* Intensity */ + else if ((sectionData->huffsection[i].codeBook != CODE_BOOK_ZERO_NO) && + (sectionData->huffsection[i].codeBook != CODE_BOOK_PNS_NO)) { + INT tmp = sectionData->huffsection[i].sfbStart + + sectionData->huffsection[i].sfbCnt; + for (j = sectionData->huffsection[i].sfbStart; j < tmp; j++) { + /* check if we can repeat the last value to save bits */ + if (maxValueInSfb[j] == 0) { + found = 0; + /* are scalefactors skipped? */ + if (scfSkipCounter == 0) { + /* end of section */ + if (j == (tmp - 1)) + found = 0; /* search in other sections for maxValueInSfb != 0 */ + else { + /* search in this section for the next maxValueInSfb[] != 0 */ + for (k = (j + 1); k < tmp; k++) { + if (maxValueInSfb[k] != 0) { + found = 1; + if ((fixp_abs(scalefacGain[k] - lastValScf)) <= + CODE_BOOK_SCF_LAV) + deltaScf = 0; /* save bits */ + else { + /* do not save bits */ + deltaScf = lastValScf - scalefacGain[j]; + lastValScf = scalefacGain[j]; + scfSkipCounter = 0; + } + break; + } + /* count scalefactor skip */ + scfSkipCounter++; + } + } + + /* search for the next maxValueInSfb[] != 0 in all other sections */ + for (m = (i + 1); (m < sectionData->noOfSections) && (found == 0); + m++) { + if ((sectionData->huffsection[m].codeBook != CODE_BOOK_ZERO_NO) && + (sectionData->huffsection[m].codeBook != CODE_BOOK_PNS_NO)) { + INT end = sectionData->huffsection[m].sfbStart + + sectionData->huffsection[m].sfbCnt; + for (n = sectionData->huffsection[m].sfbStart; n < end; n++) { + if (maxValueInSfb[n] != 0) { + found = 1; + if (fixp_abs(scalefacGain[n] - lastValScf) <= + CODE_BOOK_SCF_LAV) + deltaScf = 0; /* save bits */ + else { + /* do not save bits */ + deltaScf = lastValScf - scalefacGain[j]; + lastValScf = scalefacGain[j]; + scfSkipCounter = 0; + } + break; + } + /* count scalefactor skip */ + scfSkipCounter++; + } + } + } + /* no maxValueInSfb[] != 0 found */ + if (found == 0) { + deltaScf = 0; + scfSkipCounter = 0; + } + } else { + /* consider skipped scalefactors */ + deltaScf = 0; + scfSkipCounter--; + } + } else { + deltaScf = lastValScf - scalefacGain[j]; + lastValScf = scalefacGain[j]; + } + sectionData->scalefacBits += + FDKaacEnc_bitCountScalefactorDelta(deltaScf); + } + } + } /* for (i=0; i<sectionData->noOfSections; i++) */ +} + +/* count bits used by pns */ +static void FDKaacEnc_noiseCount(SECTION_DATA* const RESTRICT sectionData, + const INT* const noiseNrg) { + INT noisePCMFlag = TRUE; + INT lastValPns = 0, deltaPns; + int i, j; + + sectionData->noiseNrgBits = 0; + + for (i = 0; i < sectionData->noOfSections; i++) { + if (sectionData->huffsection[i].codeBook == CODE_BOOK_PNS_NO) { + int sfbStart = sectionData->huffsection[i].sfbStart; + int sfbEnd = sfbStart + sectionData->huffsection[i].sfbCnt; + for (j = sfbStart; j < sfbEnd; j++) { + if (noisePCMFlag) { + sectionData->noiseNrgBits += PNS_PCM_BITS; + lastValPns = noiseNrg[j]; + noisePCMFlag = FALSE; + } else { + deltaPns = noiseNrg[j] - lastValPns; + lastValPns = noiseNrg[j]; + sectionData->noiseNrgBits += + FDKaacEnc_bitCountScalefactorDelta(deltaPns); + } + } + } + } +} + +INT FDKaacEnc_dynBitCount(BITCNTR_STATE* const hBC, + const SHORT* const quantSpectrum, + const UINT* const maxValueInSfb, + const INT* const scalefac, const INT blockType, + const INT sfbCnt, const INT maxSfbPerGroup, + const INT sfbPerGroup, const INT* const sfbOffset, + SECTION_DATA* const RESTRICT sectionData, + const INT* const noiseNrg, const INT* const isBook, + const INT* const isScale, const UINT syntaxFlags) { + sectionData->blockType = blockType; + sectionData->sfbCnt = sfbCnt; + sectionData->sfbPerGroup = sfbPerGroup; + sectionData->noOfGroups = sfbCnt / sfbPerGroup; + sectionData->maxSfbPerGroup = maxSfbPerGroup; + + FDKaacEnc_noiselessCounter(sectionData, hBC->mergeGainLookUp, + (lookUpTable)hBC->bitLookUp, quantSpectrum, + maxValueInSfb, sfbOffset, blockType, noiseNrg, + isBook, (syntaxFlags & AC_ER_VCB11) ? 1 : 0); + + FDKaacEnc_scfCount(scalefac, maxValueInSfb, sectionData, isScale); + + FDKaacEnc_noiseCount(sectionData, noiseNrg); + + return (sectionData->huffmanBits + sectionData->sideInfoBits + + sectionData->scalefacBits + sectionData->noiseNrgBits); +} + +INT FDKaacEnc_BCNew(BITCNTR_STATE** phBC, UCHAR* dynamic_RAM) { + BITCNTR_STATE* hBC = GetRam_aacEnc_BitCntrState(); + + if (hBC) { + *phBC = hBC; + hBC->bitLookUp = GetRam_aacEnc_BitLookUp(0, dynamic_RAM); + hBC->mergeGainLookUp = GetRam_aacEnc_MergeGainLookUp(0, dynamic_RAM); + if (hBC->bitLookUp == 0 || hBC->mergeGainLookUp == 0) { + return 1; + } + } + return (hBC == 0) ? 1 : 0; +} + +void FDKaacEnc_BCClose(BITCNTR_STATE** phBC) { + if (*phBC != NULL) { + FreeRam_aacEnc_BitCntrState(phBC); + } +} diff --git a/fdk-aac/libAACenc/src/dyn_bits.h b/fdk-aac/libAACenc/src/dyn_bits.h new file mode 100644 index 0000000..a727a30 --- /dev/null +++ b/fdk-aac/libAACenc/src/dyn_bits.h @@ -0,0 +1,160 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M.Werner + + Description: Noiseless coder module + +*******************************************************************************/ + +#ifndef DYN_BITS_H +#define DYN_BITS_H + +#include "common_fix.h" + +#include "psy_const.h" +#include "aacenc_tns.h" + +#define MAX_SECTIONS MAX_GROUPED_SFB +#define SECT_ESC_VAL_LONG 31 +#define SECT_ESC_VAL_SHORT 7 +#define CODE_BOOK_BITS 4 +#define SECT_BITS_LONG 5 +#define SECT_BITS_SHORT 3 +#define PNS_PCM_BITS 9 + +typedef struct { + INT codeBook; + INT sfbStart; + INT sfbCnt; + INT sectionBits; /* huff + si ! */ +} SECTION_INFO; + +typedef struct { + INT blockType; + INT noOfGroups; + INT sfbCnt; + INT maxSfbPerGroup; + INT sfbPerGroup; + INT noOfSections; + SECTION_INFO huffsection[MAX_SECTIONS]; + INT sideInfoBits; /* sectioning bits */ + INT huffmanBits; /* huffman coded bits */ + INT scalefacBits; /* scalefac coded bits */ + INT noiseNrgBits; /* noiseEnergy coded bits */ + INT firstScf; /* first scf to be coded */ +} SECTION_DATA; + +struct BITCNTR_STATE { + INT* bitLookUp; + INT* mergeGainLookUp; +}; + +INT FDKaacEnc_BCNew(BITCNTR_STATE** phBC, UCHAR* dynamic_RAM); + +void FDKaacEnc_BCClose(BITCNTR_STATE** phBC); + +INT FDKaacEnc_dynBitCount(BITCNTR_STATE* const hBC, + const SHORT* const quantSpectrum, + const UINT* const maxValueInSfb, + const INT* const scalefac, const INT blockType, + const INT sfbCnt, const INT maxSfbPerGroup, + const INT sfbPerGroup, const INT* const sfbOffset, + SECTION_DATA* const RESTRICT sectionData, + const INT* const noiseNrg, const INT* const isBook, + const INT* const isScale, const UINT syntaxFlags); + +#endif diff --git a/fdk-aac/libAACenc/src/grp_data.cpp b/fdk-aac/libAACenc/src/grp_data.cpp new file mode 100644 index 0000000..bc9d85f --- /dev/null +++ b/fdk-aac/libAACenc/src/grp_data.cpp @@ -0,0 +1,264 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M.Werner + + Description: Short block grouping + +*******************************************************************************/ + +#include "psy_const.h" +#include "interface.h" + +/* + * this routine does not work in-place + */ + +/* + * Don't use fAddSaturate2() because it looses one bit accuracy which is + * usefull for quality. + */ +static inline FIXP_DBL nrgAddSaturate(const FIXP_DBL a, const FIXP_DBL b) { + return ((a >= (FIXP_DBL)MAXVAL_DBL - b) ? (FIXP_DBL)MAXVAL_DBL : (a + b)); +} + +void FDKaacEnc_groupShortData(FIXP_DBL *mdctSpectrum, /* in-out */ + SFB_THRESHOLD *sfbThreshold, /* in-out */ + SFB_ENERGY *sfbEnergy, /* in-out */ + SFB_ENERGY *sfbEnergyMS, /* in-out */ + SFB_ENERGY *sfbSpreadEnergy, const INT sfbCnt, + const INT sfbActive, const INT *sfbOffset, + const FIXP_DBL *sfbMinSnrLdData, + INT *groupedSfbOffset, /* out */ + INT *maxSfbPerGroup, /* out */ + FIXP_DBL *groupedSfbMinSnrLdData, + const INT noOfGroups, const INT *groupLen, + const INT granuleLength) { + INT i, j; + INT line; /* counts through lines */ + INT sfb; /* counts through scalefactor bands */ + INT grp; /* counts through groups */ + INT wnd; /* counts through windows in a group */ + INT offset; /* needed in sfbOffset grouping */ + INT highestSfb; + INT granuleLength_short = granuleLength / TRANS_FAC; + + C_ALLOC_SCRATCH_START(tmpSpectrum, FIXP_DBL, (1024)) + + /* for short blocks: regroup spectrum and */ + /* group energies and thresholds according to grouping */ + + /* calculate maxSfbPerGroup */ + highestSfb = 0; + for (wnd = 0; wnd < TRANS_FAC; wnd++) { + for (sfb = sfbActive - 1; sfb >= highestSfb; sfb--) { + for (line = sfbOffset[sfb + 1] - 1; line >= sfbOffset[sfb]; line--) { + if (mdctSpectrum[wnd * granuleLength_short + line] != + FL2FXCONST_SPC(0.0)) + break; /* this band is not completely zero */ + } + if (line >= sfbOffset[sfb]) break; /* this band was not completely zero */ + } + highestSfb = fixMax(highestSfb, sfb); + } + highestSfb = highestSfb > 0 ? highestSfb : 0; + *maxSfbPerGroup = highestSfb + 1; + + /* calculate groupedSfbOffset */ + i = 0; + offset = 0; + for (grp = 0; grp < noOfGroups; grp++) { + for (sfb = 0; sfb < sfbActive + 1; sfb++) { + groupedSfbOffset[i++] = offset + sfbOffset[sfb] * groupLen[grp]; + } + i += sfbCnt - sfb; + offset += groupLen[grp] * granuleLength_short; + } + groupedSfbOffset[i++] = granuleLength; + + /* calculate groupedSfbMinSnr */ + i = 0; + for (grp = 0; grp < noOfGroups; grp++) { + for (sfb = 0; sfb < sfbActive; sfb++) { + groupedSfbMinSnrLdData[i++] = sfbMinSnrLdData[sfb]; + } + i += sfbCnt - sfb; + } + + /* sum up sfbThresholds */ + wnd = 0; + i = 0; + for (grp = 0; grp < noOfGroups; grp++) { + for (sfb = 0; sfb < sfbActive; sfb++) { + FIXP_DBL thresh = sfbThreshold->Short[wnd][sfb]; + for (j = 1; j < groupLen[grp]; j++) { + thresh = nrgAddSaturate(thresh, sfbThreshold->Short[wnd + j][sfb]); + } + sfbThreshold->Long[i++] = thresh; + } + i += sfbCnt - sfb; + wnd += groupLen[grp]; + } + + /* sum up sfbEnergies left/right */ + wnd = 0; + i = 0; + for (grp = 0; grp < noOfGroups; grp++) { + for (sfb = 0; sfb < sfbActive; sfb++) { + FIXP_DBL energy = sfbEnergy->Short[wnd][sfb]; + for (j = 1; j < groupLen[grp]; j++) { + energy = nrgAddSaturate(energy, sfbEnergy->Short[wnd + j][sfb]); + } + sfbEnergy->Long[i++] = energy; + } + i += sfbCnt - sfb; + wnd += groupLen[grp]; + } + + /* sum up sfbEnergies mid/side */ + wnd = 0; + i = 0; + for (grp = 0; grp < noOfGroups; grp++) { + for (sfb = 0; sfb < sfbActive; sfb++) { + FIXP_DBL energy = sfbEnergyMS->Short[wnd][sfb]; + for (j = 1; j < groupLen[grp]; j++) { + energy = nrgAddSaturate(energy, sfbEnergyMS->Short[wnd + j][sfb]); + } + sfbEnergyMS->Long[i++] = energy; + } + i += sfbCnt - sfb; + wnd += groupLen[grp]; + } + + /* sum up sfbSpreadEnergies */ + wnd = 0; + i = 0; + for (grp = 0; grp < noOfGroups; grp++) { + for (sfb = 0; sfb < sfbActive; sfb++) { + FIXP_DBL energy = sfbSpreadEnergy->Short[wnd][sfb]; + for (j = 1; j < groupLen[grp]; j++) { + energy = nrgAddSaturate(energy, sfbSpreadEnergy->Short[wnd + j][sfb]); + } + sfbSpreadEnergy->Long[i++] = energy; + } + i += sfbCnt - sfb; + wnd += groupLen[grp]; + } + + /* re-group spectrum */ + wnd = 0; + i = 0; + for (grp = 0; grp < noOfGroups; grp++) { + for (sfb = 0; sfb < sfbActive; sfb++) { + int width = sfbOffset[sfb + 1] - sfbOffset[sfb]; + FIXP_DBL *pMdctSpectrum = + &mdctSpectrum[sfbOffset[sfb]] + wnd * granuleLength_short; + for (j = 0; j < groupLen[grp]; j++) { + FIXP_DBL *pTmp = pMdctSpectrum; + for (line = width; line > 0; line--) { + tmpSpectrum[i++] = *pTmp++; + } + pMdctSpectrum += granuleLength_short; + } + } + i += (groupLen[grp] * (sfbOffset[sfbCnt] - sfbOffset[sfb])); + wnd += groupLen[grp]; + } + + FDKmemcpy(mdctSpectrum, tmpSpectrum, granuleLength * sizeof(FIXP_DBL)); + + C_ALLOC_SCRATCH_END(tmpSpectrum, FIXP_DBL, (1024)) +} diff --git a/fdk-aac/libAACenc/src/grp_data.h b/fdk-aac/libAACenc/src/grp_data.h new file mode 100644 index 0000000..3e1a708 --- /dev/null +++ b/fdk-aac/libAACenc/src/grp_data.h @@ -0,0 +1,123 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M.Werner + + Description: Short block grouping + +*******************************************************************************/ + +#ifndef GRP_DATA_H +#define GRP_DATA_H + +#include "common_fix.h" + +#include "psy_data.h" + +void FDKaacEnc_groupShortData(FIXP_DBL *mdctSpectrum, /* in-out */ + SFB_THRESHOLD *sfbThreshold, /* in-out */ + SFB_ENERGY *sfbEnergy, /* in-out */ + SFB_ENERGY *sfbEnergyMS, /* in-out */ + SFB_ENERGY *sfbSpreadEnergy, const INT sfbCnt, + const INT sfbActive, const INT *sfbOffset, + const FIXP_DBL *sfbMinSnrLdData, + INT *groupedSfbOffset, /* out */ + INT *maxSfbPerGroup, + FIXP_DBL *groupedSfbMinSnrLdData, + const INT noOfGroups, const INT *groupLen, + const INT granuleLength); + +#endif /* _INTERFACE_H */ diff --git a/fdk-aac/libAACenc/src/intensity.cpp b/fdk-aac/libAACenc/src/intensity.cpp new file mode 100644 index 0000000..8cb1b45 --- /dev/null +++ b/fdk-aac/libAACenc/src/intensity.cpp @@ -0,0 +1,810 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): A. Horndasch (code originally from lwr) / Josef Hoepfl (FDK) + + Description: intensity stereo processing + +*******************************************************************************/ + +#include "intensity.h" + +#include "interface.h" +#include "psy_configuration.h" +#include "psy_const.h" +#include "qc_main.h" +#include "bit_cnt.h" + +/* only set an IS seed it left/right channel correlation is above IS_CORR_THRESH + */ +#define IS_CORR_THRESH FL2FXCONST_DBL(0.95f) + +/* when expanding the IS region to more SFBs only accept an error that is + * not more than IS_TOTAL_ERROR_THRESH overall and + * not more than IS_LOCAL_ERROR_THRESH for the current SFB */ +#define IS_TOTAL_ERROR_THRESH FL2FXCONST_DBL(0.04f) +#define IS_LOCAL_ERROR_THRESH FL2FXCONST_DBL(0.01f) + +/* the maximum allowed change of the intensity direction (unit: IS scale) - + * scaled with factor 0.25 - */ +#define IS_DIRECTION_DEVIATION_THRESH_SF 2 +#define IS_DIRECTION_DEVIATION_THRESH \ + FL2FXCONST_DBL(2.0f / (1 << IS_DIRECTION_DEVIATION_THRESH_SF)) + +/* IS regions need to have a minimal percentage of the overall loudness, e.g. + * 0.06 == 6% */ +#define IS_REGION_MIN_LOUDNESS FL2FXCONST_DBL(0.1f) + +/* only perform IS if IS_MIN_SFBS neighboring SFBs can be processed */ +#define IS_MIN_SFBS 6 + +/* only do IS if + * if IS_LEFT_RIGHT_RATIO_THRESH < sfbEnergyLeft[sfb]/sfbEnergyRight[sfb] < 1 / + * IS_LEFT_RIGHT_RATIO_THRESH + * -> no IS if the panning angle is not far from the middle, MS will do */ +/* this is equivalent to a scale of +/-1.02914634566 */ +#define IS_LEFT_RIGHT_RATIO_THRESH FL2FXCONST_DBL(0.7f) + +/* scalefactor of realScale */ +#define REAL_SCALE_SF 1 + +/* scalefactor overallLoudness */ +#define OVERALL_LOUDNESS_SF 6 + +/* scalefactor for sum over max samples per goup */ +#define MAX_SFB_PER_GROUP_SF 6 + +/* scalefactor for sum of mdct spectrum */ +#define MDCT_SPEC_SF 6 + +typedef struct { + FIXP_DBL corr_thresh; /*!< Only set an IS seed it left/right channel + correlation is above corr_thresh */ + + FIXP_DBL total_error_thresh; /*!< When expanding the IS region to more SFBs + only accept an error that is not more than + 'total_error_thresh' overall. */ + + FIXP_DBL local_error_thresh; /*!< When expanding the IS region to more SFBs + only accept an error that is not more than + 'local_error_thresh' for the current SFB. */ + + FIXP_DBL direction_deviation_thresh; /*!< The maximum allowed change of the + intensity direction (unit: IS scale) + */ + + FIXP_DBL is_region_min_loudness; /*!< IS regions need to have a minimal + percentage of the overall loudness, e.g. + 0.06 == 6% */ + + INT min_is_sfbs; /*!< Only perform IS if 'min_is_sfbs' neighboring SFBs can be + processed */ + + FIXP_DBL left_right_ratio_threshold; /*!< No IS if the panning angle is not + far from the middle, MS will do */ + +} INTENSITY_PARAMETERS; + +/***************************************************************************** + + functionname: calcSfbMaxScale + + description: Calc max value in scalefactor band + + input: *mdctSpectrum + l1 + l2 + + output: none + + returns: scalefactor + +*****************************************************************************/ +static INT calcSfbMaxScale(const FIXP_DBL *mdctSpectrum, const INT l1, + const INT l2) { + INT i; + INT sfbMaxScale; + FIXP_DBL maxSpc; + + maxSpc = FL2FXCONST_DBL(0.0); + for (i = l1; i < l2; i++) { + FIXP_DBL tmp = fixp_abs((FIXP_DBL)mdctSpectrum[i]); + maxSpc = fixMax(maxSpc, tmp); + } + sfbMaxScale = (maxSpc == FL2FXCONST_DBL(0.0)) ? (DFRACT_BITS - 2) + : CntLeadingZeros(maxSpc) - 1; + + return sfbMaxScale; +} + +/***************************************************************************** + + functionname: FDKaacEnc_initIsParams + + description: Initialization of intensity parameters + + input: isParams + + output: isParams + + returns: none + +*****************************************************************************/ +static void FDKaacEnc_initIsParams(INTENSITY_PARAMETERS *isParams) { + isParams->corr_thresh = IS_CORR_THRESH; + isParams->total_error_thresh = IS_TOTAL_ERROR_THRESH; + isParams->local_error_thresh = IS_LOCAL_ERROR_THRESH; + isParams->direction_deviation_thresh = IS_DIRECTION_DEVIATION_THRESH; + isParams->is_region_min_loudness = IS_REGION_MIN_LOUDNESS; + isParams->min_is_sfbs = IS_MIN_SFBS; + isParams->left_right_ratio_threshold = IS_LEFT_RIGHT_RATIO_THRESH; +} + +/***************************************************************************** + + functionname: FDKaacEnc_prepareIntensityDecision + + description: Prepares intensity decision + + input: sfbEnergyLeft + sfbEnergyRight + sfbEnergyLdDataLeft + sfbEnergyLdDataRight + mdctSpectrumLeft + sfbEnergyLdDataRight + isParams + + output: hrrErr scale: none + isMask scale: none + realScale scale: LD_DATA_SHIFT + REAL_SCALE_SF + normSfbLoudness scale: none + + returns: none + +*****************************************************************************/ +static void FDKaacEnc_prepareIntensityDecision( + const FIXP_DBL *sfbEnergyLeft, const FIXP_DBL *sfbEnergyRight, + const FIXP_DBL *sfbEnergyLdDataLeft, const FIXP_DBL *sfbEnergyLdDataRight, + const FIXP_DBL *mdctSpectrumLeft, const FIXP_DBL *mdctSpectrumRight, + const INTENSITY_PARAMETERS *isParams, FIXP_DBL *hrrErr, INT *isMask, + FIXP_DBL *realScale, FIXP_DBL *normSfbLoudness, const INT sfbCnt, + const INT sfbPerGroup, const INT maxSfbPerGroup, const INT *sfbOffset) { + INT j, sfb, sfboffs; + INT grpCounter; + + /* temporary variables to compute loudness */ + FIXP_DBL overallLoudness[MAX_NO_OF_GROUPS]; + + /* temporary variables to compute correlation */ + FIXP_DBL channelCorr[MAX_GROUPED_SFB]; + FIXP_DBL ml, mr; + FIXP_DBL prod_lr; + FIXP_DBL square_l, square_r; + FIXP_DBL tmp_l, tmp_r; + FIXP_DBL inv_n; + + FDKmemclear(channelCorr, MAX_GROUPED_SFB * sizeof(FIXP_DBL)); + FDKmemclear(normSfbLoudness, MAX_GROUPED_SFB * sizeof(FIXP_DBL)); + FDKmemclear(overallLoudness, MAX_NO_OF_GROUPS * sizeof(FIXP_DBL)); + FDKmemclear(realScale, MAX_GROUPED_SFB * sizeof(FIXP_DBL)); + + for (grpCounter = 0, sfboffs = 0; sfboffs < sfbCnt; + sfboffs += sfbPerGroup, grpCounter++) { + overallLoudness[grpCounter] = FL2FXCONST_DBL(0.0f); + for (sfb = 0; sfb < maxSfbPerGroup; sfb++) { + INT sL, sR, s; + FIXP_DBL isValue = sfbEnergyLdDataLeft[sfb + sfboffs] - + sfbEnergyLdDataRight[sfb + sfboffs]; + + /* delimitate intensity scale value to representable range */ + realScale[sfb + sfboffs] = fixMin( + FL2FXCONST_DBL(60.f / (1 << (REAL_SCALE_SF + LD_DATA_SHIFT))), + fixMax(FL2FXCONST_DBL(-60.f / (1 << (REAL_SCALE_SF + LD_DATA_SHIFT))), + isValue)); + + sL = fixMax(0, (CntLeadingZeros(sfbEnergyLeft[sfb + sfboffs]) - 1)); + sR = fixMax(0, (CntLeadingZeros(sfbEnergyRight[sfb + sfboffs]) - 1)); + s = (fixMin(sL, sR) >> 2) << 2; + normSfbLoudness[sfb + sfboffs] = + sqrtFixp(sqrtFixp(((sfbEnergyLeft[sfb + sfboffs] << s) >> 1) + + ((sfbEnergyRight[sfb + sfboffs] << s) >> 1))) >> + (s >> 2); + + overallLoudness[grpCounter] += + normSfbLoudness[sfb + sfboffs] >> OVERALL_LOUDNESS_SF; + /* don't do intensity if + * - panning angle is too close to the middle or + * - one channel is non-existent or + * - if it is dual mono */ + if ((sfbEnergyLeft[sfb + sfboffs] >= + fMult(isParams->left_right_ratio_threshold, + sfbEnergyRight[sfb + sfboffs])) && + (fMult(isParams->left_right_ratio_threshold, + sfbEnergyLeft[sfb + sfboffs]) <= + sfbEnergyRight[sfb + sfboffs])) { + /* this will prevent post processing from considering this SFB for + * merging */ + hrrErr[sfb + sfboffs] = FL2FXCONST_DBL(1.0 / 8.0); + } + } + } + + for (grpCounter = 0, sfboffs = 0; sfboffs < sfbCnt; + sfboffs += sfbPerGroup, grpCounter++) { + INT invOverallLoudnessSF; + FIXP_DBL invOverallLoudness; + + if (overallLoudness[grpCounter] == FL2FXCONST_DBL(0.0)) { + invOverallLoudness = FL2FXCONST_DBL(0.0); + invOverallLoudnessSF = 0; + } else { + invOverallLoudness = + fDivNorm((FIXP_DBL)MAXVAL_DBL, overallLoudness[grpCounter], + &invOverallLoudnessSF); + invOverallLoudnessSF = + invOverallLoudnessSF - OVERALL_LOUDNESS_SF + + 1; /* +1: compensate fMultDiv2() in subsequent loop */ + } + invOverallLoudnessSF = fixMin( + fixMax(invOverallLoudnessSF, -(DFRACT_BITS - 1)), DFRACT_BITS - 1); + + for (sfb = 0; sfb < maxSfbPerGroup; sfb++) { + FIXP_DBL tmp; + + tmp = fMultDiv2((normSfbLoudness[sfb + sfboffs] >> OVERALL_LOUDNESS_SF) + << OVERALL_LOUDNESS_SF, + invOverallLoudness); + + normSfbLoudness[sfb + sfboffs] = scaleValue(tmp, invOverallLoudnessSF); + + channelCorr[sfb + sfboffs] = FL2FXCONST_DBL(0.0f); + + /* max width of scalefactorband is 96; width's are always even */ + /* inv_n is scaled with factor 2 to compensate fMultDiv2() in subsequent + * loops */ + inv_n = GetInvInt( + (sfbOffset[sfb + sfboffs + 1] - sfbOffset[sfb + sfboffs]) >> 1); + + if (inv_n > FL2FXCONST_DBL(0.0f)) { + INT s, sL, sR; + + /* correlation := Pearson's product-moment coefficient */ + /* compute correlation between channels and check if it is over + * threshold */ + ml = FL2FXCONST_DBL(0.0f); + mr = FL2FXCONST_DBL(0.0f); + prod_lr = FL2FXCONST_DBL(0.0f); + square_l = FL2FXCONST_DBL(0.0f); + square_r = FL2FXCONST_DBL(0.0f); + + sL = calcSfbMaxScale(mdctSpectrumLeft, sfbOffset[sfb + sfboffs], + sfbOffset[sfb + sfboffs + 1]); + sR = calcSfbMaxScale(mdctSpectrumRight, sfbOffset[sfb + sfboffs], + sfbOffset[sfb + sfboffs + 1]); + s = fixMin(sL, sR); + + for (j = sfbOffset[sfb + sfboffs]; j < sfbOffset[sfb + sfboffs + 1]; + j++) { + ml += fMultDiv2((mdctSpectrumLeft[j] << s), + inv_n); // scaled with mdctScale - s + inv_n + mr += fMultDiv2((mdctSpectrumRight[j] << s), + inv_n); // scaled with mdctScale - s + inv_n + } + ml = fMultDiv2(ml, inv_n); // scaled with mdctScale - s + inv_n + mr = fMultDiv2(mr, inv_n); // scaled with mdctScale - s + inv_n + + for (j = sfbOffset[sfb + sfboffs]; j < sfbOffset[sfb + sfboffs + 1]; + j++) { + tmp_l = fMultDiv2((mdctSpectrumLeft[j] << s), inv_n) - + ml; // scaled with mdctScale - s + inv_n + tmp_r = fMultDiv2((mdctSpectrumRight[j] << s), inv_n) - + mr; // scaled with mdctScale - s + inv_n + + prod_lr += fMultDiv2( + tmp_l, tmp_r); // scaled with 2*(mdctScale - s + inv_n) + 1 + square_l += + fPow2Div2(tmp_l); // scaled with 2*(mdctScale - s + inv_n) + 1 + square_r += + fPow2Div2(tmp_r); // scaled with 2*(mdctScale - s + inv_n) + 1 + } + prod_lr = prod_lr << 1; // scaled with 2*(mdctScale - s + inv_n) + square_l = square_l << 1; // scaled with 2*(mdctScale - s + inv_n) + square_r = square_r << 1; // scaled with 2*(mdctScale - s + inv_n) + + if (square_l > FL2FXCONST_DBL(0.0f) && + square_r > FL2FXCONST_DBL(0.0f)) { + INT channelCorrSF = 0; + + /* local scaling of square_l and square_r is compensated after sqrt + * calculation */ + sL = fixMax(0, (CntLeadingZeros(square_l) - 1)); + sR = fixMax(0, (CntLeadingZeros(square_r) - 1)); + s = ((sL + sR) >> 1) << 1; + sL = fixMin(sL, s); + sR = s - sL; + tmp = fMult(square_l << sL, square_r << sR); + tmp = sqrtFixp(tmp); + + FDK_ASSERT(tmp > FL2FXCONST_DBL(0.0f)); + + /* numerator and denominator have the same scaling */ + if (prod_lr < FL2FXCONST_DBL(0.0f)) { + channelCorr[sfb + sfboffs] = + -(fDivNorm(-prod_lr, tmp, &channelCorrSF)); + + } else { + channelCorr[sfb + sfboffs] = + (fDivNorm(prod_lr, tmp, &channelCorrSF)); + } + channelCorrSF = fixMin( + fixMax((channelCorrSF + ((sL + sR) >> 1)), -(DFRACT_BITS - 1)), + DFRACT_BITS - 1); + + if (channelCorrSF < 0) { + channelCorr[sfb + sfboffs] = + channelCorr[sfb + sfboffs] >> (-channelCorrSF); + } else { + /* avoid overflows due to limited computational accuracy */ + if (fAbs(channelCorr[sfb + sfboffs]) > + (((FIXP_DBL)MAXVAL_DBL) >> channelCorrSF)) { + if (channelCorr[sfb + sfboffs] < FL2FXCONST_DBL(0.0f)) + channelCorr[sfb + sfboffs] = -(FIXP_DBL)MAXVAL_DBL; + else + channelCorr[sfb + sfboffs] = (FIXP_DBL)MAXVAL_DBL; + } else { + channelCorr[sfb + sfboffs] = channelCorr[sfb + sfboffs] + << channelCorrSF; + } + } + } + } + + /* for post processing: hrrErr is the error in terms of (too little) + * correlation weighted with the loudness of the SFB; SFBs with small + * hrrErr can be merged */ + if (hrrErr[sfb + sfboffs] == FL2FXCONST_DBL(1.0 / 8.0)) { + continue; + } + + hrrErr[sfb + sfboffs] = + fMultDiv2((FL2FXCONST_DBL(0.25f) - (channelCorr[sfb + sfboffs] >> 2)), + normSfbLoudness[sfb + sfboffs]); + + /* set IS mask/vector to 1, if correlation is high enough */ + if (fAbs(channelCorr[sfb + sfboffs]) >= isParams->corr_thresh) { + isMask[sfb + sfboffs] = 1; + } + } + } +} + +/***************************************************************************** + + functionname: FDKaacEnc_finalizeIntensityDecision + + description: Finalizes intensity decision + + input: isParams scale: none + hrrErr scale: none + realIsScale scale: LD_DATA_SHIFT + REAL_SCALE_SF + normSfbLoudness scale: none + + output: isMask scale: none + + returns: none + +*****************************************************************************/ +static void FDKaacEnc_finalizeIntensityDecision( + const FIXP_DBL *hrrErr, INT *isMask, const FIXP_DBL *realIsScale, + const FIXP_DBL *normSfbLoudness, const INTENSITY_PARAMETERS *isParams, + const INT sfbCnt, const INT sfbPerGroup, const INT maxSfbPerGroup) { + INT sfb, sfboffs, j; + FIXP_DBL isScaleLast = FL2FXCONST_DBL(0.0f); + INT isStartValueFound = 0; + + for (sfboffs = 0; sfboffs < sfbCnt; sfboffs += sfbPerGroup) { + INT startIsSfb = 0; + INT inIsBlock = 0; + INT currentIsSfbCount = 0; + FIXP_DBL overallHrrError = FL2FXCONST_DBL(0.0f); + FIXP_DBL isRegionLoudness = FL2FXCONST_DBL(0.0f); + + for (sfb = 0; sfb < maxSfbPerGroup; sfb++) { + if (isMask[sfboffs + sfb] == 1) { + if (currentIsSfbCount == 0) { + startIsSfb = sfboffs + sfb; + } + if (isStartValueFound == 0) { + isScaleLast = realIsScale[sfboffs + sfb]; + isStartValueFound = 1; + } + inIsBlock = 1; + currentIsSfbCount++; + overallHrrError += hrrErr[sfboffs + sfb] >> (MAX_SFB_PER_GROUP_SF - 3); + isRegionLoudness += + normSfbLoudness[sfboffs + sfb] >> MAX_SFB_PER_GROUP_SF; + } else { + /* based on correlation, IS should not be used + * -> use it anyway, if overall error is below threshold + * and if local error does not exceed threshold + * otherwise: check if there are enough IS SFBs + */ + if (inIsBlock) { + overallHrrError += + hrrErr[sfboffs + sfb] >> (MAX_SFB_PER_GROUP_SF - 3); + isRegionLoudness += + normSfbLoudness[sfboffs + sfb] >> MAX_SFB_PER_GROUP_SF; + + if ((hrrErr[sfboffs + sfb] < (isParams->local_error_thresh >> 3)) && + (overallHrrError < + (isParams->total_error_thresh >> MAX_SFB_PER_GROUP_SF))) { + currentIsSfbCount++; + /* overwrite correlation based decision */ + isMask[sfboffs + sfb] = 1; + } else { + inIsBlock = 0; + } + } + } + /* check for large direction deviation */ + if (inIsBlock) { + if (fAbs(isScaleLast - realIsScale[sfboffs + sfb]) < + (isParams->direction_deviation_thresh >> + (REAL_SCALE_SF + LD_DATA_SHIFT - + IS_DIRECTION_DEVIATION_THRESH_SF))) { + isScaleLast = realIsScale[sfboffs + sfb]; + } else { + isMask[sfboffs + sfb] = 0; + inIsBlock = 0; + currentIsSfbCount--; + } + } + + if (currentIsSfbCount > 0 && (!inIsBlock || sfb == maxSfbPerGroup - 1)) { + /* not enough SFBs -> do not use IS */ + if (currentIsSfbCount < isParams->min_is_sfbs || + (isRegionLoudness<isParams->is_region_min_loudness>> + MAX_SFB_PER_GROUP_SF)) { + for (j = startIsSfb; j <= sfboffs + sfb; j++) { + isMask[j] = 0; + } + isScaleLast = FL2FXCONST_DBL(0.0f); + isStartValueFound = 0; + for (j = 0; j < startIsSfb; j++) { + if (isMask[j] != 0) { + isScaleLast = realIsScale[j]; + isStartValueFound = 1; + } + } + } + currentIsSfbCount = 0; + overallHrrError = FL2FXCONST_DBL(0.0f); + isRegionLoudness = FL2FXCONST_DBL(0.0f); + } + } + } +} + +/***************************************************************************** + + functionname: FDKaacEnc_IntensityStereoProcessing + + description: Intensity stereo processing tool + + input: sfbEnergyLeft + sfbEnergyRight + mdctSpectrumLeft + mdctSpectrumRight + sfbThresholdLeft + sfbThresholdRight + sfbSpreadEnLeft + sfbSpreadEnRight + sfbEnergyLdDataLeft + sfbEnergyLdDataRight + + output: isBook + isScale + pnsData->pnsFlag + msDigest zeroed from start to sfbCnt + msMask zeroed from start to sfbCnt + mdctSpectrumRight zeroed where isBook!=0 + sfbEnergyRight zeroed where isBook!=0 + sfbSpreadEnRight zeroed where isBook!=0 + sfbThresholdRight zeroed where isBook!=0 + sfbEnergyLdDataRight FL2FXCONST_DBL(-1.0) where isBook!=0 + sfbThresholdLdDataRight FL2FXCONST_DBL(-0.515625f) where +isBook!=0 + + returns: none + +*****************************************************************************/ +void FDKaacEnc_IntensityStereoProcessing( + FIXP_DBL *sfbEnergyLeft, FIXP_DBL *sfbEnergyRight, + FIXP_DBL *mdctSpectrumLeft, FIXP_DBL *mdctSpectrumRight, + FIXP_DBL *sfbThresholdLeft, FIXP_DBL *sfbThresholdRight, + FIXP_DBL *sfbThresholdLdDataRight, FIXP_DBL *sfbSpreadEnLeft, + FIXP_DBL *sfbSpreadEnRight, FIXP_DBL *sfbEnergyLdDataLeft, + FIXP_DBL *sfbEnergyLdDataRight, INT *msDigest, INT *msMask, + const INT sfbCnt, const INT sfbPerGroup, const INT maxSfbPerGroup, + const INT *sfbOffset, const INT allowIS, INT *isBook, INT *isScale, + PNS_DATA *RESTRICT pnsData[2]) { + INT sfb, sfboffs, j; + FIXP_DBL scale; + FIXP_DBL lr; + FIXP_DBL hrrErr[MAX_GROUPED_SFB]; + FIXP_DBL normSfbLoudness[MAX_GROUPED_SFB]; + FIXP_DBL realIsScale[MAX_GROUPED_SFB]; + INTENSITY_PARAMETERS isParams; + INT isMask[MAX_GROUPED_SFB]; + + FDKmemclear((void *)isBook, sfbCnt * sizeof(INT)); + FDKmemclear((void *)isMask, sfbCnt * sizeof(INT)); + FDKmemclear((void *)realIsScale, sfbCnt * sizeof(FIXP_DBL)); + FDKmemclear((void *)isScale, sfbCnt * sizeof(INT)); + FDKmemclear((void *)hrrErr, sfbCnt * sizeof(FIXP_DBL)); + + if (!allowIS) return; + + FDKaacEnc_initIsParams(&isParams); + + /* compute / set the following values per SFB: + * - left/right ratio between channels + * - normalized loudness + * + loudness == average of energy in channels to 0.25 + * + normalization: division by sum of all SFB loudnesses + * - isMask (is set to 0 if channels are the same or one is 0) + */ + FDKaacEnc_prepareIntensityDecision( + sfbEnergyLeft, sfbEnergyRight, sfbEnergyLdDataLeft, sfbEnergyLdDataRight, + mdctSpectrumLeft, mdctSpectrumRight, &isParams, hrrErr, isMask, + realIsScale, normSfbLoudness, sfbCnt, sfbPerGroup, maxSfbPerGroup, + sfbOffset); + + FDKaacEnc_finalizeIntensityDecision(hrrErr, isMask, realIsScale, + normSfbLoudness, &isParams, sfbCnt, + sfbPerGroup, maxSfbPerGroup); + + for (sfb = 0; sfb < sfbCnt; sfb += sfbPerGroup) { + for (sfboffs = 0; sfboffs < maxSfbPerGroup; sfboffs++) { + INT sL, sR; + FIXP_DBL inv_n; + + msMask[sfb + sfboffs] = 0; + if (isMask[sfb + sfboffs] == 0) { + continue; + } + + if ((sfbEnergyLeft[sfb + sfboffs] < sfbThresholdLeft[sfb + sfboffs]) && + (fMult(FL2FXCONST_DBL(1.0f / 1.5f), sfbEnergyRight[sfb + sfboffs]) > + sfbThresholdRight[sfb + sfboffs])) { + continue; + } + /* NEW: if there is a big-enough IS region, switch off PNS */ + if (pnsData[0]) { + if (pnsData[0]->pnsFlag[sfb + sfboffs]) { + pnsData[0]->pnsFlag[sfb + sfboffs] = 0; + } + if (pnsData[1]->pnsFlag[sfb + sfboffs]) { + pnsData[1]->pnsFlag[sfb + sfboffs] = 0; + } + } + + inv_n = GetInvInt( + (sfbOffset[sfb + sfboffs + 1] - sfbOffset[sfb + sfboffs]) >> + 1); // scaled with 2 to compensate fMultDiv2() in subsequent loop + sL = calcSfbMaxScale(mdctSpectrumLeft, sfbOffset[sfb + sfboffs], + sfbOffset[sfb + sfboffs + 1]); + sR = calcSfbMaxScale(mdctSpectrumRight, sfbOffset[sfb + sfboffs], + sfbOffset[sfb + sfboffs + 1]); + + lr = FL2FXCONST_DBL(0.0f); + for (j = sfbOffset[sfb + sfboffs]; j < sfbOffset[sfb + sfboffs + 1]; j++) + lr += fMultDiv2( + fMultDiv2(mdctSpectrumLeft[j] << sL, mdctSpectrumRight[j] << sR), + inv_n); + lr = lr << 1; + + if (lr < FL2FXCONST_DBL(0.0f)) { + /* This means OUT OF phase intensity stereo, cf. standard */ + INT s0, s1, s2; + FIXP_DBL tmp, d, ed = FL2FXCONST_DBL(0.0f); + + s0 = fixMin(sL, sR); + for (j = sfbOffset[sfb + sfboffs]; j < sfbOffset[sfb + sfboffs + 1]; + j++) { + d = ((mdctSpectrumLeft[j] << s0) >> 1) - + ((mdctSpectrumRight[j] << s0) >> 1); + ed += fMultDiv2(d, d) >> (MDCT_SPEC_SF - 1); + } + msMask[sfb + sfboffs] = 1; + tmp = fDivNorm(sfbEnergyLeft[sfb + sfboffs], ed, &s1); + s2 = (s1) + (2 * s0) - 2 - MDCT_SPEC_SF; + if (s2 & 1) { + tmp = tmp >> 1; + s2 = s2 + 1; + } + s2 = (s2 >> 1) + 1; // +1 compensate fMultDiv2() in subsequent loop + s2 = fixMin(fixMax(s2, -(DFRACT_BITS - 1)), (DFRACT_BITS - 1)); + scale = sqrtFixp(tmp); + if (s2 < 0) { + s2 = -s2; + for (j = sfbOffset[sfb + sfboffs]; j < sfbOffset[sfb + sfboffs + 1]; + j++) { + mdctSpectrumLeft[j] = (fMultDiv2(mdctSpectrumLeft[j], scale) - + fMultDiv2(mdctSpectrumRight[j], scale)) >> + s2; + mdctSpectrumRight[j] = FL2FXCONST_DBL(0.0f); + } + } else { + for (j = sfbOffset[sfb + sfboffs]; j < sfbOffset[sfb + sfboffs + 1]; + j++) { + mdctSpectrumLeft[j] = (fMultDiv2(mdctSpectrumLeft[j], scale) - + fMultDiv2(mdctSpectrumRight[j], scale)) + << s2; + mdctSpectrumRight[j] = FL2FXCONST_DBL(0.0f); + } + } + } else { + /* This means IN phase intensity stereo, cf. standard */ + INT s0, s1, s2; + FIXP_DBL tmp, s, es = FL2FXCONST_DBL(0.0f); + + s0 = fixMin(sL, sR); + for (j = sfbOffset[sfb + sfboffs]; j < sfbOffset[sfb + sfboffs + 1]; + j++) { + s = ((mdctSpectrumLeft[j] << s0) >> 1) + + ((mdctSpectrumRight[j] << s0) >> 1); + es = fAddSaturate(es, fMultDiv2(s, s) >> + (MDCT_SPEC_SF - + 1)); // scaled 2*(mdctScale - s0 + 1) + MDCT_SPEC_SF + } + msMask[sfb + sfboffs] = 0; + tmp = fDivNorm(sfbEnergyLeft[sfb + sfboffs], es, &s1); + s2 = (s1) + (2 * s0) - 2 - MDCT_SPEC_SF; + if (s2 & 1) { + tmp = tmp >> 1; + s2 = s2 + 1; + } + s2 = (s2 >> 1) + 1; // +1 compensate fMultDiv2() in subsequent loop + s2 = fixMin(fixMax(s2, -(DFRACT_BITS - 1)), (DFRACT_BITS - 1)); + scale = sqrtFixp(tmp); + if (s2 < 0) { + s2 = -s2; + for (j = sfbOffset[sfb + sfboffs]; j < sfbOffset[sfb + sfboffs + 1]; + j++) { + mdctSpectrumLeft[j] = (fMultDiv2(mdctSpectrumLeft[j], scale) + + fMultDiv2(mdctSpectrumRight[j], scale)) >> + s2; + mdctSpectrumRight[j] = FL2FXCONST_DBL(0.0f); + } + } else { + for (j = sfbOffset[sfb + sfboffs]; j < sfbOffset[sfb + sfboffs + 1]; + j++) { + mdctSpectrumLeft[j] = (fMultDiv2(mdctSpectrumLeft[j], scale) + + fMultDiv2(mdctSpectrumRight[j], scale)) + << s2; + mdctSpectrumRight[j] = FL2FXCONST_DBL(0.0f); + } + } + } + + isBook[sfb + sfboffs] = CODE_BOOK_IS_IN_PHASE_NO; + + if (realIsScale[sfb + sfboffs] < FL2FXCONST_DBL(0.0f)) { + isScale[sfb + sfboffs] = + (INT)(((realIsScale[sfb + sfboffs] >> 1) - + FL2FXCONST_DBL( + 0.5f / (1 << (REAL_SCALE_SF + LD_DATA_SHIFT + 1)))) >> + (DFRACT_BITS - 1 - REAL_SCALE_SF - LD_DATA_SHIFT - 1)) + + 1; + } else { + isScale[sfb + sfboffs] = + (INT)(((realIsScale[sfb + sfboffs] >> 1) + + FL2FXCONST_DBL( + 0.5f / (1 << (REAL_SCALE_SF + LD_DATA_SHIFT + 1)))) >> + (DFRACT_BITS - 1 - REAL_SCALE_SF - LD_DATA_SHIFT - 1)); + } + + sfbEnergyRight[sfb + sfboffs] = FL2FXCONST_DBL(0.0f); + sfbEnergyLdDataRight[sfb + sfboffs] = FL2FXCONST_DBL(-1.0f); + sfbThresholdRight[sfb + sfboffs] = FL2FXCONST_DBL(0.0f); + sfbThresholdLdDataRight[sfb + sfboffs] = FL2FXCONST_DBL(-0.515625f); + sfbSpreadEnRight[sfb + sfboffs] = FL2FXCONST_DBL(0.0f); + + *msDigest = MS_SOME; + } + } +} diff --git a/fdk-aac/libAACenc/src/intensity.h b/fdk-aac/libAACenc/src/intensity.h new file mode 100644 index 0000000..70f23d5 --- /dev/null +++ b/fdk-aac/libAACenc/src/intensity.h @@ -0,0 +1,121 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): A. Horndasch (code originally from lwr and rtb) / Josef Hoepfl +(FDK) + + Description: intensity stereo prototype + +*******************************************************************************/ + +#ifndef INTENSITY_H +#define INTENSITY_H + +#include "aacenc_pns.h" +#include "common_fix.h" + +void FDKaacEnc_IntensityStereoProcessing( + FIXP_DBL *sfbEnergyLeft, FIXP_DBL *sfbEnergyRight, + FIXP_DBL *mdctSpectrumLeft, FIXP_DBL *mdctSpectrumRight, + FIXP_DBL *sfbThresholdLeft, FIXP_DBL *sfbThresholdRight, + FIXP_DBL *sfbThresholdLdDataRight, FIXP_DBL *sfbSpreadEnLeft, + FIXP_DBL *sfbSpreadEnRight, FIXP_DBL *sfbEnergyLdDataLeft, + FIXP_DBL *sfbEnergyLdDataRight, INT *msDigest, INT *msMask, + const INT sfbCnt, const INT sfbPerGroup, const INT maxSfbPerGroup, + const INT *sfbOffset, const INT allowIS, INT *isBook, INT *isScale, + PNS_DATA *RESTRICT pnsData[2]); + +#endif /* INTENSITY_H */ diff --git a/fdk-aac/libAACenc/src/interface.h b/fdk-aac/libAACenc/src/interface.h new file mode 100644 index 0000000..b1a31ef --- /dev/null +++ b/fdk-aac/libAACenc/src/interface.h @@ -0,0 +1,168 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M.Werner + + Description: Interface psychoaccoustic/quantizer + +*******************************************************************************/ + +#ifndef INTERFACE_H +#define INTERFACE_H + +#include "common_fix.h" +#include "FDK_audio.h" + +#include "psy_data.h" +#include "aacenc_tns.h" + +enum { MS_NONE = 0, MS_SOME = 1, MS_ALL = 2 }; + +enum { MS_ON = 1 }; + +struct TOOLSINFO { + INT msDigest; /* 0 = no MS; 1 = some MS, 2 = all MS */ + INT msMask[MAX_GROUPED_SFB]; +}; + +typedef struct { + INT sfbCnt; + INT sfbPerGroup; + INT maxSfbPerGroup; + INT lastWindowSequence; + INT windowShape; + INT groupingMask; + INT sfbOffsets[MAX_GROUPED_SFB + 1]; + + INT mdctScale; /* number of transform shifts */ + INT groupLen[MAX_NO_OF_GROUPS]; + + TNS_INFO tnsInfo; + INT noiseNrg[MAX_GROUPED_SFB]; + INT isBook[MAX_GROUPED_SFB]; + INT isScale[MAX_GROUPED_SFB]; + + /* memory located in QC_OUT_CHANNEL */ + FIXP_DBL *mdctSpectrum; + FIXP_DBL *sfbEnergy; + FIXP_DBL *sfbSpreadEnergy; + FIXP_DBL *sfbThresholdLdData; + FIXP_DBL *sfbMinSnrLdData; + FIXP_DBL *sfbEnergyLdData; + +} PSY_OUT_CHANNEL; + +typedef struct { + /* information specific to each channel */ + PSY_OUT_CHANNEL *psyOutChannel[(2)]; + + /* information shared by both channels */ + INT commonWindow; + struct TOOLSINFO toolsInfo; + +} PSY_OUT_ELEMENT; + +typedef struct { + PSY_OUT_ELEMENT *psyOutElement[((8))]; + PSY_OUT_CHANNEL *pPsyOutChannels[(8)]; + +} PSY_OUT; + +inline int isLowDelay(AUDIO_OBJECT_TYPE aot) { + return (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD); +} + +#endif /* INTERFACE_H */ diff --git a/fdk-aac/libAACenc/src/line_pe.cpp b/fdk-aac/libAACenc/src/line_pe.cpp new file mode 100644 index 0000000..47734e5 --- /dev/null +++ b/fdk-aac/libAACenc/src/line_pe.cpp @@ -0,0 +1,234 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M. Werner + + Description: Perceptual entropie module + +*******************************************************************************/ + +#include "line_pe.h" +#include "sf_estim.h" +#include "bit_cnt.h" + +#include "genericStds.h" + +static const FIXP_DBL C1LdData = + FL2FXCONST_DBL(3.0 / LD_DATA_SCALING); /* C1 = 3.0 = log(8.0)/log(2) */ +static const FIXP_DBL C2LdData = FL2FXCONST_DBL( + 1.3219281 / LD_DATA_SCALING); /* C2 = 1.3219281 = log(2.5)/log(2) */ +static const FIXP_DBL C3LdData = FL2FXCONST_DBL(0.5593573); /* 1-C2/C1 */ + +/* constants that do not change during successive pe calculations */ +void FDKaacEnc_prepareSfbPe(PE_CHANNEL_DATA *RESTRICT const peChanData, + const FIXP_DBL *RESTRICT const sfbEnergyLdData, + const FIXP_DBL *RESTRICT const sfbThresholdLdData, + const FIXP_DBL *RESTRICT const sfbFormFactorLdData, + const INT *RESTRICT const sfbOffset, + const INT sfbCnt, const INT sfbPerGroup, + const INT maxSfbPerGroup) { + INT sfbGrp, sfb; + INT sfbWidth; + FIXP_DBL avgFormFactorLdData; + const FIXP_DBL formFacScaling = + FL2FXCONST_DBL((float)FORM_FAC_SHIFT / LD_DATA_SCALING); + + for (sfbGrp = 0; sfbGrp < sfbCnt; sfbGrp += sfbPerGroup) { + for (sfb = 0; sfb < maxSfbPerGroup; sfb++) { + if ((FIXP_DBL)sfbEnergyLdData[sfbGrp + sfb] > + (FIXP_DBL)sfbThresholdLdData[sfbGrp + sfb]) { + sfbWidth = sfbOffset[sfbGrp + sfb + 1] - sfbOffset[sfbGrp + sfb]; + /* estimate number of active lines */ + avgFormFactorLdData = ((-sfbEnergyLdData[sfbGrp + sfb] >> 1) + + (CalcLdInt(sfbWidth) >> 1)) >> + 1; + peChanData->sfbNLines[sfbGrp + sfb] = (INT)CalcInvLdData( + (sfbFormFactorLdData[sfbGrp + sfb] + formFacScaling) + + avgFormFactorLdData); + /* Make sure sfbNLines is never greater than sfbWidth due to + * unaccuracies (e.g. sfbEnergyLdData[sfbGrp+sfb] = 0x80000000) */ + peChanData->sfbNLines[sfbGrp + sfb] = + fMin(sfbWidth, peChanData->sfbNLines[sfbGrp + sfb]); + } else { + peChanData->sfbNLines[sfbGrp + sfb] = 0; + } + } + } +} + +/* + formula for one sfb: + pe = n * ld(en/thr), if ld(en/thr) >= C1 + pe = n * (C2 + C3 * ld(en/thr)), if ld(en/thr) < C1 + n: estimated number of lines in sfb, + ld(x) = log(x)/log(2) + + constPart is sfbPe without the threshold part n*ld(thr) or n*C3*ld(thr) +*/ +void FDKaacEnc_calcSfbPe(PE_CHANNEL_DATA *RESTRICT const peChanData, + const FIXP_DBL *RESTRICT const sfbEnergyLdData, + const FIXP_DBL *RESTRICT const sfbThresholdLdData, + const INT sfbCnt, const INT sfbPerGroup, + const INT maxSfbPerGroup, + const INT *RESTRICT const isBook, + const INT *RESTRICT const isScale) { + INT sfbGrp, sfb, thisSfb; + INT nLines; + FIXP_DBL logDataRatio; + FIXP_DBL scaleLd = (FIXP_DBL)0; + INT lastValIs = 0; + + FIXP_DBL pe = 0; + FIXP_DBL constPart = 0; + FIXP_DBL nActiveLines = 0; + + FIXP_DBL tmpPe, tmpConstPart, tmpNActiveLines; + + for (sfbGrp = 0; sfbGrp < sfbCnt; sfbGrp += sfbPerGroup) { + for (sfb = 0; sfb < maxSfbPerGroup; sfb++) { + tmpPe = (FIXP_DBL)0; + tmpConstPart = (FIXP_DBL)0; + tmpNActiveLines = (FIXP_DBL)0; + + thisSfb = sfbGrp + sfb; + + if (sfbEnergyLdData[thisSfb] > sfbThresholdLdData[thisSfb]) { + logDataRatio = sfbEnergyLdData[thisSfb] - sfbThresholdLdData[thisSfb]; + nLines = peChanData->sfbNLines[thisSfb]; + + FIXP_DBL factor = nLines << (LD_DATA_SHIFT + PE_CONSTPART_SHIFT + 1); + if (logDataRatio >= C1LdData) { + /* scale sfbPe and sfbConstPart with PE_CONSTPART_SHIFT */ + tmpPe = fMultDiv2(logDataRatio, factor); + tmpConstPart = fMultDiv2(sfbEnergyLdData[thisSfb] + scaleLd, factor); + } else { + /* scale sfbPe and sfbConstPart with PE_CONSTPART_SHIFT */ + tmpPe = fMultDiv2( + ((FIXP_DBL)C2LdData + fMult(C3LdData, logDataRatio)), factor); + tmpConstPart = + fMultDiv2(((FIXP_DBL)C2LdData + + fMult(C3LdData, sfbEnergyLdData[thisSfb] + scaleLd)), + factor); + + nLines = fMultI(C3LdData, nLines); + } + tmpNActiveLines = (FIXP_DBL)nLines; + } else if (isBook[thisSfb]) { + /* provide for cost of scale factor for Intensity */ + INT delta = isScale[thisSfb] - lastValIs; + lastValIs = isScale[thisSfb]; + peChanData->sfbPe[thisSfb] = FDKaacEnc_bitCountScalefactorDelta(delta) + << PE_CONSTPART_SHIFT; + peChanData->sfbConstPart[thisSfb] = 0; + peChanData->sfbNActiveLines[thisSfb] = 0; + } + peChanData->sfbPe[thisSfb] = tmpPe; + peChanData->sfbConstPart[thisSfb] = tmpConstPart; + peChanData->sfbNActiveLines[thisSfb] = tmpNActiveLines; + + /* sum up peChanData values */ + pe += tmpPe; + constPart += tmpConstPart; + nActiveLines += tmpNActiveLines; + } + } + + /* correct scaled pe and constPart values */ + peChanData->pe = pe >> PE_CONSTPART_SHIFT; + peChanData->constPart = constPart >> PE_CONSTPART_SHIFT; + + peChanData->nActiveLines = nActiveLines; +} diff --git a/fdk-aac/libAACenc/src/line_pe.h b/fdk-aac/libAACenc/src/line_pe.h new file mode 100644 index 0000000..ecc2388 --- /dev/null +++ b/fdk-aac/libAACenc/src/line_pe.h @@ -0,0 +1,148 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M. Werner + + Description: Perceptual entropie module + +*******************************************************************************/ + +#ifndef LINE_PE_H +#define LINE_PE_H + +#include "common_fix.h" + +#include "psy_const.h" + +#define PE_CONSTPART_SHIFT FRACT_BITS + +typedef struct { + /* calculated by FDKaacEnc_prepareSfbPe */ + INT sfbNLines[MAX_GROUPED_SFB]; /* number of relevant lines in sfb */ + /* the rest is calculated by FDKaacEnc_calcSfbPe */ + INT sfbPe[MAX_GROUPED_SFB]; /* pe for each sfb */ + INT sfbConstPart[MAX_GROUPED_SFB]; /* constant part for each sfb */ + INT sfbNActiveLines[MAX_GROUPED_SFB]; /* number of active lines in sfb */ + INT pe; /* sum of sfbPe */ + INT constPart; /* sum of sfbConstPart */ + INT nActiveLines; /* sum of sfbNActiveLines */ +} PE_CHANNEL_DATA; + +typedef struct { + PE_CHANNEL_DATA peChannelData[(2)]; + INT pe; + INT constPart; + INT nActiveLines; + INT offset; +} PE_DATA; + +void FDKaacEnc_prepareSfbPe(PE_CHANNEL_DATA *RESTRICT const peChanData, + const FIXP_DBL *RESTRICT const sfbEnergyLdData, + const FIXP_DBL *RESTRICT const sfbThresholdLdData, + const FIXP_DBL *RESTRICT const sfbFormFactorLdData, + const INT *RESTRICT const sfbOffset, + const INT sfbCnt, const INT sfbPerGroup, + const INT maxSfbPerGroup); + +void FDKaacEnc_calcSfbPe(PE_CHANNEL_DATA *RESTRICT const peChanData, + const FIXP_DBL *RESTRICT const sfbEnergyLdData, + const FIXP_DBL *RESTRICT const sfbThresholdLdData, + const INT sfbCnt, const INT sfbPerGroup, + const INT maxSfbPerGroup, + const INT *RESTRICT const isBook, + const INT *RESTRICT const isScale); + +#endif diff --git a/fdk-aac/libAACenc/src/metadata_compressor.cpp b/fdk-aac/libAACenc/src/metadata_compressor.cpp new file mode 100644 index 0000000..bdac80a --- /dev/null +++ b/fdk-aac/libAACenc/src/metadata_compressor.cpp @@ -0,0 +1,1579 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M. Neusinger + + Description: Compressor for AAC Metadata Generator + +*******************************************************************************/ + +#include "metadata_compressor.h" +#include "channel_map.h" + +#define LOG2 0.69314718056f /* natural logarithm of 2 */ +#define ILOG2 1.442695041f /* 1/LOG2 */ +#define FIXP_ILOG2_DIV2 (FL2FXCONST_DBL(ILOG2 / 2)) + +/*----------------- defines ----------------------*/ + +#define MAX_DRC_CHANNELS (8) /*!< Max number of audio input channels. */ +#define DOWNMIX_SHIFT (3) /*!< Max 8 channel. */ +#define WEIGHTING_FILTER_SHIFT (2) /*!< Scaling used in weighting filter. */ + +#define METADATA_INT_BITS 10 +#define METADATA_LINT_BITS 20 +#define METADATA_INT_SCALE (INT64(1) << (METADATA_INT_BITS)) +#define METADATA_FRACT_BITS (DFRACT_BITS - 1 - METADATA_INT_BITS) +#define METADATA_FRACT_SCALE (INT64(1) << (METADATA_FRACT_BITS)) + +/** + * Enum for channel assignment. + */ +enum { L = 0, R = 1, C = 2, LFE = 3, LS = 4, RS = 5, S = 6, LS2 = 7, RS2 = 8 }; + +/*--------------- structure definitions --------------------*/ + +/** + * Structure holds weighting filter filter states. + */ +struct WEIGHTING_STATES { + FIXP_DBL x1; + FIXP_DBL x2; + FIXP_DBL y1; + FIXP_DBL y2; +}; + +/** + * Dynamic Range Control compressor structure. + */ +struct DRC_COMP { + FIXP_DBL maxBoostThr[2]; /*!< Max boost threshold. */ + FIXP_DBL boostThr[2]; /*!< Boost threshold. */ + FIXP_DBL earlyCutThr[2]; /*!< Early cut threshold. */ + FIXP_DBL cutThr[2]; /*!< Cut threshold. */ + FIXP_DBL maxCutThr[2]; /*!< Max cut threshold. */ + + FIXP_DBL boostFac[2]; /*!< Precalculated factor for boost compression. */ + FIXP_DBL + earlyCutFac[2]; /*!< Precalculated factor for early cut compression. */ + FIXP_DBL cutFac[2]; /*!< Precalculated factor for cut compression. */ + + FIXP_DBL maxBoost[2]; /*!< Maximum boost. */ + FIXP_DBL maxCut[2]; /*!< Maximum cut. */ + FIXP_DBL maxEarlyCut[2]; /*!< Maximum early cut. */ + + FIXP_DBL fastAttack[2]; /*!< Fast attack coefficient. */ + FIXP_DBL fastDecay[2]; /*!< Fast release coefficient. */ + FIXP_DBL slowAttack[2]; /*!< Slow attack coefficient. */ + FIXP_DBL slowDecay[2]; /*!< Slow release coefficient. */ + UINT holdOff[2]; /*!< Hold time in blocks. */ + + FIXP_DBL attackThr[2]; /*!< Slow/fast attack threshold. */ + FIXP_DBL decayThr[2]; /*!< Slow/fast release threshold. */ + + DRC_PROFILE profile[2]; /*!< DRC profile. */ + INT blockLength; /*!< Block length in samples. */ + UINT sampleRate; /*!< Sample rate. */ + CHANNEL_MODE chanConfig; /*!< Channel configuration. */ + + UCHAR useWeighting; /*!< Use weighting filter. */ + + UINT channels; /*!< Number of channels. */ + UINT fullChannels; /*!< Number of full range channels. */ + INT channelIdx[9]; /*!< Offsets of interleaved channel samples (L, R, C, LFE, + Ls, Rs, S, Ls2, Rs2). */ + + FIXP_DBL smoothLevel[2]; /*!< level smoothing states */ + FIXP_DBL smoothGain[2]; /*!< gain smoothing states */ + UINT holdCnt[2]; /*!< hold counter */ + + FIXP_DBL limGain[2]; /*!< limiter gain */ + FIXP_DBL limDecay; /*!< limiter decay (linear) */ + FIXP_DBL prevPeak[2]; /*!< max peak of previous block (stereo/mono)*/ + + WEIGHTING_STATES + filter[MAX_DRC_CHANNELS]; /*!< array holds weighting filter states */ +}; + +/*---------------- constants -----------------------*/ + +/** + * Profile tables. + */ +static const FIXP_DBL tabMaxBoostThr[] = { + (FIXP_DBL)(-(43 << METADATA_FRACT_BITS)), + (FIXP_DBL)(-(53 << METADATA_FRACT_BITS)), + (FIXP_DBL)(-(55 << METADATA_FRACT_BITS)), + (FIXP_DBL)(-(65 << METADATA_FRACT_BITS)), + (FIXP_DBL)(-(50 << METADATA_FRACT_BITS)), + (FIXP_DBL)(-(40 << METADATA_FRACT_BITS))}; +static const FIXP_DBL tabBoostThr[] = { + (FIXP_DBL)(-(31 << METADATA_FRACT_BITS)), + (FIXP_DBL)(-(41 << METADATA_FRACT_BITS)), + (FIXP_DBL)(-(31 << METADATA_FRACT_BITS)), + (FIXP_DBL)(-(41 << METADATA_FRACT_BITS)), + (FIXP_DBL)(-(31 << METADATA_FRACT_BITS)), + (FIXP_DBL)(-(31 << METADATA_FRACT_BITS))}; +static const FIXP_DBL tabEarlyCutThr[] = { + (FIXP_DBL)(-(26 << METADATA_FRACT_BITS)), + (FIXP_DBL)(-(21 << METADATA_FRACT_BITS)), + (FIXP_DBL)(-(26 << METADATA_FRACT_BITS)), + (FIXP_DBL)(-(21 << METADATA_FRACT_BITS)), + (FIXP_DBL)(-(26 << METADATA_FRACT_BITS)), + (FIXP_DBL)(-(20 << METADATA_FRACT_BITS))}; +static const FIXP_DBL tabCutThr[] = {(FIXP_DBL)(-(16 << METADATA_FRACT_BITS)), + (FIXP_DBL)(-(11 << METADATA_FRACT_BITS)), + (FIXP_DBL)(-(16 << METADATA_FRACT_BITS)), + (FIXP_DBL)(-(21 << METADATA_FRACT_BITS)), + (FIXP_DBL)(-(16 << METADATA_FRACT_BITS)), + (FIXP_DBL)(-(10 << METADATA_FRACT_BITS))}; +static const FIXP_DBL tabMaxCutThr[] = { + (FIXP_DBL)(4 << METADATA_FRACT_BITS), (FIXP_DBL)(9 << METADATA_FRACT_BITS), + (FIXP_DBL)(4 << METADATA_FRACT_BITS), (FIXP_DBL)(9 << METADATA_FRACT_BITS), + (FIXP_DBL)(4 << METADATA_FRACT_BITS), (FIXP_DBL)(4 << METADATA_FRACT_BITS)}; +static const FIXP_DBL tabBoostRatio[] = { + FL2FXCONST_DBL(((1.f / 2.f) - 1.f)), FL2FXCONST_DBL(((1.f / 2.f) - 1.f)), + FL2FXCONST_DBL(((1.f / 2.f) - 1.f)), FL2FXCONST_DBL(((1.f / 2.f) - 1.f)), + FL2FXCONST_DBL(((1.f / 5.f) - 1.f)), FL2FXCONST_DBL(((1.f / 5.f) - 1.f))}; +static const FIXP_DBL tabEarlyCutRatio[] = { + FL2FXCONST_DBL(((1.f / 2.f) - 1.f)), FL2FXCONST_DBL(((1.f / 2.f) - 1.f)), + FL2FXCONST_DBL(((1.f / 2.f) - 1.f)), FL2FXCONST_DBL(((1.f / 1.f) - 1.f)), + FL2FXCONST_DBL(((1.f / 2.f) - 1.f)), FL2FXCONST_DBL(((1.f / 2.f) - 1.f))}; +static const FIXP_DBL tabCutRatio[] = { + FL2FXCONST_DBL(((1.f / 20.f) - 1.f)), FL2FXCONST_DBL(((1.f / 20.f) - 1.f)), + FL2FXCONST_DBL(((1.f / 20.f) - 1.f)), FL2FXCONST_DBL(((1.f / 2.f) - 1.f)), + FL2FXCONST_DBL(((1.f / 20.f) - 1.f)), FL2FXCONST_DBL(((1.f / 20.f) - 1.f))}; +static const FIXP_DBL tabMaxBoost[] = {(FIXP_DBL)(6 << METADATA_FRACT_BITS), + (FIXP_DBL)(6 << METADATA_FRACT_BITS), + (FIXP_DBL)(12 << METADATA_FRACT_BITS), + (FIXP_DBL)(12 << METADATA_FRACT_BITS), + (FIXP_DBL)(15 << METADATA_FRACT_BITS), + (FIXP_DBL)(15 << METADATA_FRACT_BITS)}; +static const FIXP_DBL tabMaxCut[] = {(FIXP_DBL)(24 << METADATA_FRACT_BITS), + (FIXP_DBL)(24 << METADATA_FRACT_BITS), + (FIXP_DBL)(24 << METADATA_FRACT_BITS), + (FIXP_DBL)(15 << METADATA_FRACT_BITS), + (FIXP_DBL)(24 << METADATA_FRACT_BITS), + (FIXP_DBL)(24 << METADATA_FRACT_BITS)}; +static const FIXP_DBL tabFastAttack[] = { + FL2FXCONST_DBL((10.f / 1000.f) / METADATA_INT_SCALE), + FL2FXCONST_DBL((10.f / 1000.f) / METADATA_INT_SCALE), + FL2FXCONST_DBL((10.f / 1000.f) / METADATA_INT_SCALE), + FL2FXCONST_DBL((10.f / 1000.f) / METADATA_INT_SCALE), + FL2FXCONST_DBL((10.f / 1000.f) / METADATA_INT_SCALE), + FL2FXCONST_DBL((0.f / 1000.f) / METADATA_INT_SCALE)}; +static const FIXP_DBL tabFastDecay[] = { + FL2FXCONST_DBL((1000.f / 1000.f) / METADATA_INT_SCALE), + FL2FXCONST_DBL((1000.f / 1000.f) / METADATA_INT_SCALE), + FL2FXCONST_DBL((1000.f / 1000.f) / METADATA_INT_SCALE), + FL2FXCONST_DBL((1000.f / 1000.f) / METADATA_INT_SCALE), + FL2FXCONST_DBL((200.f / 1000.f) / METADATA_INT_SCALE), + FL2FXCONST_DBL((0.f / 1000.f) / METADATA_INT_SCALE)}; +static const FIXP_DBL tabSlowAttack[] = { + FL2FXCONST_DBL((100.f / 1000.f) / METADATA_INT_SCALE), + FL2FXCONST_DBL((100.f / 1000.f) / METADATA_INT_SCALE), + FL2FXCONST_DBL((100.f / 1000.f) / METADATA_INT_SCALE), + FL2FXCONST_DBL((100.f / 1000.f) / METADATA_INT_SCALE), + FL2FXCONST_DBL((100.f / 1000.f) / METADATA_INT_SCALE), + FL2FXCONST_DBL((0.f / 1000.f) / METADATA_INT_SCALE)}; +static const FIXP_DBL tabSlowDecay[] = { + FL2FXCONST_DBL((3000.f / 1000.f) / METADATA_INT_SCALE), + FL2FXCONST_DBL((3000.f / 1000.f) / METADATA_INT_SCALE), + FL2FXCONST_DBL((10000.f / 1000.f) / METADATA_INT_SCALE), + FL2FXCONST_DBL((3000.f / 1000.f) / METADATA_INT_SCALE), + FL2FXCONST_DBL((1000.f / 1000.f) / METADATA_INT_SCALE), + FL2FXCONST_DBL((0.f / 1000.f) / METADATA_INT_SCALE)}; + +static const INT tabHoldOff[] = {10, 10, 10, 10, 10, 0}; + +static const FIXP_DBL tabAttackThr[] = {(FIXP_DBL)(15 << METADATA_FRACT_BITS), + (FIXP_DBL)(15 << METADATA_FRACT_BITS), + (FIXP_DBL)(15 << METADATA_FRACT_BITS), + (FIXP_DBL)(15 << METADATA_FRACT_BITS), + (FIXP_DBL)(10 << METADATA_FRACT_BITS), + (FIXP_DBL)(0 << METADATA_FRACT_BITS)}; +static const FIXP_DBL tabDecayThr[] = {(FIXP_DBL)(20 << METADATA_FRACT_BITS), + (FIXP_DBL)(20 << METADATA_FRACT_BITS), + (FIXP_DBL)(20 << METADATA_FRACT_BITS), + (FIXP_DBL)(20 << METADATA_FRACT_BITS), + (FIXP_DBL)(10 << METADATA_FRACT_BITS), + (FIXP_DBL)(0 << METADATA_FRACT_BITS)}; + +/** + * Weighting filter coefficients (biquad bandpass). + */ +static const FIXP_DBL b0 = FL2FXCONST_DBL(0.53050662f); /* b1 = 0, b2 = -b0 */ +static const FIXP_DBL a1 = FL2FXCONST_DBL(-0.95237983f), + a2 = FL2FXCONST_DBL(-0.02248836f); /* a0 = 1 */ + +/*------------- function definitions ----------------*/ + +/** + * \brief Calculate scaling factor for denoted processing block. + * + * \param blockLength Length of processing block. + * + * \return shiftFactor + */ +static UINT getShiftFactor(const UINT length) { + UINT ldN; + for (ldN = 1; (((UINT)1) << ldN) < length; ldN++) + ; + + return ldN; +} + +/** + * \brief Sum up fixpoint values with best possible accuracy. + * + * \param value1 First input value. + * \param q1 Scaling factor of first input value. + * \param pValue2 Pointer to second input value, will be modified on + * return. + * \param pQ2 Pointer to second scaling factor, will be modified on + * return. + * + * \return void + */ +static void fixpAdd(const FIXP_DBL value1, const int q1, + FIXP_DBL* const pValue2, int* const pQ2) { + const int headroom1 = fNormz(fixp_abs(value1)) - 1; + const int headroom2 = fNormz(fixp_abs(*pValue2)) - 1; + int resultScale = fixMax(q1 - headroom1, (*pQ2) - headroom2); + + if ((value1 != FL2FXCONST_DBL(0.f)) && (*pValue2 != FL2FXCONST_DBL(0.f))) { + resultScale++; + } + + *pValue2 = scaleValue(value1, q1 - resultScale) + + scaleValue(*pValue2, (*pQ2) - resultScale); + *pQ2 = (*pValue2 != (FIXP_DBL)0) ? resultScale : DFRACT_BITS - 1; +} + +/** + * \brief Function for converting time constant to filter coefficient. + * + * \param t Time constant. + * \param sampleRate Sampling rate in Hz. + * \param blockLength Length of processing block in samples per channel. + * + * \return result = 1.0 - exp(-1.0/((t) * (f))) + */ +static FIXP_DBL tc2Coeff(const FIXP_DBL t, const INT sampleRate, + const INT blockLength) { + FIXP_DBL sampleRateFract; + FIXP_DBL blockLengthFract; + FIXP_DBL f, product; + FIXP_DBL exponent, result; + INT e_res; + + /* f = sampleRate/blockLength */ + sampleRateFract = + (FIXP_DBL)(sampleRate << (DFRACT_BITS - 1 - METADATA_LINT_BITS)); + blockLengthFract = + (FIXP_DBL)(blockLength << (DFRACT_BITS - 1 - METADATA_LINT_BITS)); + f = fDivNorm(sampleRateFract, blockLengthFract, &e_res); + f = scaleValue(f, e_res - METADATA_INT_BITS); /* convert to METADATA_FRACT */ + + /* product = t*f */ + product = fMultNorm(t, f, &e_res); + product = scaleValue( + product, e_res + METADATA_INT_BITS); /* convert to METADATA_FRACT */ + + /* exponent = (-1.0/((t) * (f))) */ + exponent = fDivNorm(METADATA_FRACT_SCALE, product, &e_res); + exponent = scaleValue( + exponent, e_res - METADATA_INT_BITS); /* convert to METADATA_FRACT */ + + /* exponent * ld(e) */ + exponent = fMult(exponent, FIXP_ILOG2_DIV2) << 1; /* e^(x) = 2^(x*ld(e)) */ + + /* exp(-1.0/((t) * (f))) */ + result = f2Pow(-exponent, DFRACT_BITS - 1 - METADATA_FRACT_BITS, &e_res); + + /* result = 1.0 - exp(-1.0/((t) * (f))) */ + result = (FIXP_DBL)MAXVAL_DBL - scaleValue(result, e_res); + + return result; +} + +static void findPeakLevels(HDRC_COMP drcComp, const INT_PCM* const inSamples, + const FIXP_DBL clev, const FIXP_DBL slev, + const FIXP_DBL ext_leva, const FIXP_DBL ext_levb, + const FIXP_DBL lfe_lev, const FIXP_DBL dmxGain5, + const FIXP_DBL dmxGain2, FIXP_DBL peak[2]) { + int i, c; + FIXP_DBL tmp = FL2FXCONST_DBL(0.f); + INT_PCM maxSample = 0; + + /* find peak level */ + peak[0] = peak[1] = FL2FXCONST_DBL(0.f); + for (i = 0; i < drcComp->blockLength; i++) { + const INT_PCM* pSamples = &inSamples[i * drcComp->channels]; + + /* single channels */ + for (c = 0; c < (int)drcComp->channels; c++) { + maxSample = fMax(maxSample, (INT_PCM)fAbs(pSamples[c])); + } + } + peak[0] = fixMax(peak[0], FX_PCM2FX_DBL(maxSample) >> DOWNMIX_SHIFT); + + /* 7.1/6.1 to 5.1 downmixes */ + if (drcComp->fullChannels > 5) { + for (i = 0; i < drcComp->blockLength; i++) { + const INT_PCM* pSamples = &inSamples[i * drcComp->channels]; + + /* channel 1 (L, Ls,...) */ + tmp = FL2FXCONST_DBL(0.f); + switch (drcComp->chanConfig) { + case MODE_6_1: + tmp += + fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[4]]) >> + (DOWNMIX_SHIFT - 1); /* Ls */ + tmp += + fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[6]]) >> + (DOWNMIX_SHIFT - 1); /* Cs */ + break; + case MODE_7_1_BACK: + case MODE_7_1_REAR_SURROUND: + tmp += + fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[4]]) >> + (DOWNMIX_SHIFT - 1); /* Ls */ + tmp += + fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >> + (DOWNMIX_SHIFT - 1); /* Lrs / Lss */ + break; + case MODE_1_2_2_2_1: + case MODE_7_1_FRONT_CENTER: + tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[0]]) >> + DOWNMIX_SHIFT); /* L */ + tmp += + fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >> + (DOWNMIX_SHIFT - 1); /* Lc */ + break; + case MODE_7_1_TOP_FRONT: + tmp += + fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[0]]) >> + (DOWNMIX_SHIFT - 1); /* L */ + tmp += + fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >> + (DOWNMIX_SHIFT - 1); /* Lvh */ + break; + default: + break; + } + peak[0] = fixMax(peak[0], fixp_abs(tmp)); + + /* channel 2 (R, Rs,...) */ + tmp = FL2FXCONST_DBL(0.f); + switch (drcComp->chanConfig) { + case MODE_6_1: + tmp += + fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[5]]) >> + (DOWNMIX_SHIFT - 1); /* Rs */ + tmp += + fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[6]]) >> + (DOWNMIX_SHIFT - 1); /* Cs */ + break; + case MODE_7_1_BACK: + case MODE_7_1_REAR_SURROUND: + tmp += + fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[5]]) >> + (DOWNMIX_SHIFT - 1); /* Rs */ + tmp += + fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >> + (DOWNMIX_SHIFT - 1); /* Rrs / Rss */ + break; + case MODE_1_2_2_2_1: + case MODE_7_1_FRONT_CENTER: + tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[1]]) >> + DOWNMIX_SHIFT); /* R */ + tmp += + fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >> + (DOWNMIX_SHIFT - 1); /* Rc */ + break; + case MODE_7_1_TOP_FRONT: + tmp += + fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[1]]) >> + (DOWNMIX_SHIFT - 1); /* R */ + tmp += + fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >> + (DOWNMIX_SHIFT - 1); /* Rvh */ + break; + default: + break; + } + peak[0] = fixMax(peak[0], fixp_abs(tmp)); + + /* channel 3 (C) */ + tmp = FL2FXCONST_DBL(0.f); + switch (drcComp->chanConfig) { + case MODE_1_2_2_2_1: + case MODE_7_1_FRONT_CENTER: + tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >> + DOWNMIX_SHIFT); /* C */ + tmp += + fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >> + (DOWNMIX_SHIFT - 1); /* Lc */ + tmp += + fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >> + (DOWNMIX_SHIFT - 1); /* Rc */ + break; + default: + break; + } + peak[0] = fixMax(peak[0], fixp_abs(tmp)); + + } /* for (blocklength) */ + + /* take downmix gain into accout */ + peak[0] = fMult(dmxGain5, peak[0]) + << (DFRACT_BITS - 1 - METADATA_FRACT_BITS); + } + + /* 7.1 / 5.1 to stereo downmixes */ + if (drcComp->fullChannels > 2) { + /* Lt/Rt downmix */ + for (i = 0; i < drcComp->blockLength; i++) { + const INT_PCM* pSamples = &inSamples[i * drcComp->channels]; + + /* Lt */ + tmp = FL2FXCONST_DBL(0.f); + if (drcComp->channelIdx[LS] >= 0) + tmp -= fMultDiv2(FL2FXCONST_DBL(0.707f), + (FIXP_PCM)pSamples[drcComp->channelIdx[LS]]) >> + (DOWNMIX_SHIFT - 1); /* Ls */ + if (drcComp->channelIdx[LS2] >= 0) + tmp -= fMultDiv2(FL2FXCONST_DBL(0.707f), + (FIXP_PCM)pSamples[drcComp->channelIdx[LS2]]) >> + (DOWNMIX_SHIFT - 1); /* Ls2 */ + if (drcComp->channelIdx[RS] >= 0) + tmp -= fMultDiv2(FL2FXCONST_DBL(0.707f), + (FIXP_PCM)pSamples[drcComp->channelIdx[RS]]) >> + (DOWNMIX_SHIFT - 1); /* Rs */ + if (drcComp->channelIdx[RS2] >= 0) + tmp -= fMultDiv2(FL2FXCONST_DBL(0.707f), + (FIXP_PCM)pSamples[drcComp->channelIdx[RS2]]) >> + (DOWNMIX_SHIFT - 1); /* Rs2 */ + if ((drcComp->channelIdx[LS] >= 0) && (drcComp->channelIdx[LS2] >= 0)) + tmp = fMult(FL2FXCONST_DBL(0.707f), tmp); /* 7.1ch */ + if (drcComp->channelIdx[S] >= 0) + tmp -= fMultDiv2(FL2FXCONST_DBL(0.707f), + (FIXP_PCM)pSamples[drcComp->channelIdx[S]]) >> + (DOWNMIX_SHIFT - 1); /* S */ + if (drcComp->channelIdx[C] >= 0) + tmp += fMultDiv2(FL2FXCONST_DBL(0.707f), + (FIXP_PCM)pSamples[drcComp->channelIdx[C]]) >> + (DOWNMIX_SHIFT - 1); /* C */ + tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[L]]) >> + DOWNMIX_SHIFT); /* L */ + + /* apply scaling of downmix gains */ + /* only for positive values only, as legacy decoders might not know this + * parameter */ + if (dmxGain2 > FL2FXCONST_DBL(0.f)) { + if (drcComp->fullChannels > 5) { + tmp = fMult(dmxGain5, tmp) << (DFRACT_BITS - 1 - METADATA_FRACT_BITS); + } + tmp = fMult(dmxGain2, tmp) << (DFRACT_BITS - 1 - METADATA_FRACT_BITS); + } + peak[0] = fixMax(peak[0], fixp_abs(tmp)); + + /* Rt */ + tmp = FL2FXCONST_DBL(0.f); + if (drcComp->channelIdx[LS] >= 0) + tmp += fMultDiv2(FL2FXCONST_DBL(0.707f), + (FIXP_PCM)pSamples[drcComp->channelIdx[LS]]) >> + (DOWNMIX_SHIFT - 1); /* Ls */ + if (drcComp->channelIdx[LS2] >= 0) + tmp += fMultDiv2(FL2FXCONST_DBL(0.707f), + (FIXP_PCM)pSamples[drcComp->channelIdx[LS2]]) >> + (DOWNMIX_SHIFT - 1); /* Ls2 */ + if (drcComp->channelIdx[RS] >= 0) + tmp += fMultDiv2(FL2FXCONST_DBL(0.707f), + (FIXP_PCM)pSamples[drcComp->channelIdx[RS]]) >> + (DOWNMIX_SHIFT - 1); /* Rs */ + if (drcComp->channelIdx[RS2] >= 0) + tmp += fMultDiv2(FL2FXCONST_DBL(0.707f), + (FIXP_PCM)pSamples[drcComp->channelIdx[RS2]]) >> + (DOWNMIX_SHIFT - 1); /* Rs2 */ + if ((drcComp->channelIdx[RS] >= 0) && (drcComp->channelIdx[RS2] >= 0)) + tmp = fMult(FL2FXCONST_DBL(0.707f), tmp); /* 7.1ch */ + if (drcComp->channelIdx[S] >= 0) + tmp += fMultDiv2(FL2FXCONST_DBL(0.707f), + (FIXP_PCM)pSamples[drcComp->channelIdx[S]]) >> + (DOWNMIX_SHIFT - 1); /* S */ + if (drcComp->channelIdx[C] >= 0) + tmp += fMultDiv2(FL2FXCONST_DBL(0.707f), + (FIXP_PCM)pSamples[drcComp->channelIdx[C]]) >> + (DOWNMIX_SHIFT - 1); /* C */ + tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[R]]) >> + DOWNMIX_SHIFT); /* R */ + + /* apply scaling of downmix gains */ + /* only for positive values only, as legacy decoders might not know this + * parameter */ + if (dmxGain2 > FL2FXCONST_DBL(0.f)) { + if (drcComp->fullChannels > 5) { + tmp = fMult(dmxGain5, tmp) << (DFRACT_BITS - 1 - METADATA_FRACT_BITS); + } + tmp = fMult(dmxGain2, tmp) << (DFRACT_BITS - 1 - METADATA_FRACT_BITS); + } + peak[0] = fixMax(peak[0], fixp_abs(tmp)); + } + + /* Lo/Ro downmix */ + for (i = 0; i < drcComp->blockLength; i++) { + const INT_PCM* pSamples = &inSamples[i * drcComp->channels]; + + /* Lo */ + tmp = FL2FXCONST_DBL(0.f); + switch (drcComp->chanConfig) { + case MODE_6_1: + tmp += fMultDiv2(fMult(slev, ext_leva), + (FIXP_PCM)pSamples[drcComp->channelIdx[4]]) >> + (DOWNMIX_SHIFT - 1); /* Ls */ + tmp += fMultDiv2(fMult(slev, ext_levb), + (FIXP_PCM)pSamples[drcComp->channelIdx[6]]) >> + (DOWNMIX_SHIFT - 1); /* Cs */ + tmp += fMultDiv2(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >> + (DOWNMIX_SHIFT - 1); /* C */ + tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[0]]) >> + DOWNMIX_SHIFT); /* L */ + tmp += + fMultDiv2(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> + (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ + break; + case MODE_7_1_BACK: + case MODE_7_1_REAR_SURROUND: + tmp += fMultDiv2(fMult(slev, ext_leva), + (FIXP_PCM)pSamples[drcComp->channelIdx[4]]) >> + (DOWNMIX_SHIFT - 1); /* Ls */ + tmp += fMultDiv2(fMult(slev, ext_levb), + (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >> + (DOWNMIX_SHIFT - 1); /* Lrs / Lss*/ + tmp += fMultDiv2(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >> + (DOWNMIX_SHIFT - 1); /* C */ + tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[0]]) >> + DOWNMIX_SHIFT); /* L */ + tmp += + fMultDiv2(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> + (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ + break; + case MODE_1_2_2_2_1: + case MODE_7_1_FRONT_CENTER: + tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[0]]) >> + DOWNMIX_SHIFT); /* L */ + tmp += + fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >> + (DOWNMIX_SHIFT - 1); /* Lc */ + tmp += fMultDiv2(fMult(ext_leva, clev), + (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >> + (DOWNMIX_SHIFT - 1); /* Lc - second path*/ + tmp += fMultDiv2(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >> + (DOWNMIX_SHIFT - 1); /* C */ + tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[4]]) >> + (DOWNMIX_SHIFT - 1); /* Ls */ + tmp += + fMultDiv2(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> + (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ + break; + case MODE_7_1_TOP_FRONT: + tmp += + fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[0]]) >> + (DOWNMIX_SHIFT - 1); /* L */ + tmp += + fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >> + (DOWNMIX_SHIFT - 1); /* Lvh */ + tmp += fMultDiv2(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >> + (DOWNMIX_SHIFT - 1); /* C */ + tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[4]]) >> + (DOWNMIX_SHIFT - 1); /* Ls */ + tmp += + fMultDiv2(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> + (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ + break; + default: + if (drcComp->channelIdx[LS] >= 0) + tmp += + fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[LS]]) >> + (DOWNMIX_SHIFT - 1); /* Ls */ + if (drcComp->channelIdx[LS2] >= 0) + tmp += + fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[LS2]]) >> + (DOWNMIX_SHIFT - 1); /* Ls2 */ + if ((drcComp->channelIdx[LS] >= 0) && (drcComp->channelIdx[LS2] >= 0)) + tmp = fMult(FL2FXCONST_DBL(0.707f), tmp); /* 7.1ch */ + if (drcComp->channelIdx[S] >= 0) + tmp += + fMultDiv2(slev, + fMult(FL2FXCONST_DBL(0.7f), + (FIXP_PCM)pSamples[drcComp->channelIdx[S]])) >> + (DOWNMIX_SHIFT - 1); /* S */ + if (drcComp->channelIdx[C] >= 0) + tmp += + fMultDiv2(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[C]]) >> + (DOWNMIX_SHIFT - 1); /* C */ + if (drcComp->channelIdx[3] >= 0) + tmp += fMultDiv2(lfe_lev, + (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> + (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ + tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[L]]) >> + DOWNMIX_SHIFT); /* L */ + break; + } + + /* apply scaling of downmix gains */ + /* only for positive values only, as legacy decoders might not know this + * parameter */ + if (dmxGain2 > FL2FXCONST_DBL(0.f)) { + if (drcComp->fullChannels > 5) { + tmp = fMult(dmxGain5, tmp) << (DFRACT_BITS - 1 - METADATA_FRACT_BITS); + } + tmp = fMult(dmxGain2, tmp) << (DFRACT_BITS - 1 - METADATA_FRACT_BITS); + } + peak[0] = fixMax(peak[0], fixp_abs(tmp)); + + /* Ro */ + tmp = FL2FXCONST_DBL(0.f); + switch (drcComp->chanConfig) { + case MODE_6_1: + tmp += fMultDiv2(fMult(slev, ext_leva), + (FIXP_PCM)pSamples[drcComp->channelIdx[5]]) >> + (DOWNMIX_SHIFT - 1); /* Rs */ + tmp += fMultDiv2(fMult(slev, ext_levb), + (FIXP_PCM)pSamples[drcComp->channelIdx[6]]) >> + (DOWNMIX_SHIFT - 1); /* Cs */ + tmp += fMultDiv2(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >> + (DOWNMIX_SHIFT - 1); /* C */ + tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[1]]) >> + DOWNMIX_SHIFT); /* R */ + tmp += + fMultDiv2(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> + (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ + break; + case MODE_7_1_BACK: + case MODE_7_1_REAR_SURROUND: + tmp += fMultDiv2(fMult(slev, ext_leva), + (FIXP_PCM)pSamples[drcComp->channelIdx[5]]) >> + (DOWNMIX_SHIFT - 1); /* Rs */ + tmp += fMultDiv2(fMult(slev, ext_levb), + (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >> + (DOWNMIX_SHIFT - 1); /* Rrs / Rss*/ + tmp += fMultDiv2(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >> + (DOWNMIX_SHIFT - 1); /* C */ + tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[1]]) >> + DOWNMIX_SHIFT); /* R */ + tmp += + fMultDiv2(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> + (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ + break; + case MODE_1_2_2_2_1: + case MODE_7_1_FRONT_CENTER: + tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[1]]) >> + DOWNMIX_SHIFT); /* R */ + tmp += + fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >> + (DOWNMIX_SHIFT - 1); /* Rc */ + tmp += fMultDiv2(fMult(ext_leva, clev), + (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >> + (DOWNMIX_SHIFT - 1); /* Rc - second path*/ + tmp += fMultDiv2(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >> + (DOWNMIX_SHIFT - 1); /* C */ + tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[5]]) >> + (DOWNMIX_SHIFT - 1); /* Rs */ + tmp += + fMultDiv2(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> + (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ + break; + case MODE_7_1_TOP_FRONT: + tmp += + fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[1]]) >> + (DOWNMIX_SHIFT - 1); /* R */ + tmp += + fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >> + (DOWNMIX_SHIFT - 1); /* Rvh */ + tmp += fMultDiv2(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >> + (DOWNMIX_SHIFT - 1); /* C */ + tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[5]]) >> + (DOWNMIX_SHIFT - 1); /* Rs */ + tmp += + fMultDiv2(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> + (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ + break; + default: + if (drcComp->channelIdx[RS] >= 0) + tmp += + fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[RS]]) >> + (DOWNMIX_SHIFT - 1); /* Rs */ + if (drcComp->channelIdx[RS2] >= 0) + tmp += + fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[RS2]]) >> + (DOWNMIX_SHIFT - 1); /* Rs2 */ + if ((drcComp->channelIdx[RS] >= 0) && (drcComp->channelIdx[RS2] >= 0)) + tmp = fMult(FL2FXCONST_DBL(0.707f), tmp); /* 7.1ch */ + if (drcComp->channelIdx[S] >= 0) + tmp += + fMultDiv2(slev, + fMult(FL2FXCONST_DBL(0.7f), + (FIXP_PCM)pSamples[drcComp->channelIdx[S]])) >> + (DOWNMIX_SHIFT - 1); /* S */ + if (drcComp->channelIdx[C] >= 0) + tmp += + fMultDiv2(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[C]]) >> + (DOWNMIX_SHIFT - 1); /* C */ + if (drcComp->channelIdx[3] >= 0) + tmp += fMultDiv2(lfe_lev, + (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> + (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ + tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[R]]) >> + DOWNMIX_SHIFT); /* R */ + } + + /* apply scaling of downmix gains */ + /* only for positive values only, as legacy decoders might not know this + * parameter */ + if (dmxGain2 > FL2FXCONST_DBL(0.f)) { + if (drcComp->fullChannels > 5) { + tmp = fMult(dmxGain5, tmp) << (DFRACT_BITS - 1 - METADATA_FRACT_BITS); + } + tmp = fMult(dmxGain2, tmp) << (DFRACT_BITS - 1 - METADATA_FRACT_BITS); + } + peak[0] = fixMax(peak[0], fixp_abs(tmp)); + } + } + + peak[1] = fixMax(peak[0], peak[1]); + + /* Mono Downmix - for comp_val only */ + if (drcComp->fullChannels > 1) { + for (i = 0; i < drcComp->blockLength; i++) { + const INT_PCM* pSamples = &inSamples[i * drcComp->channels]; + + tmp = FL2FXCONST_DBL(0.f); + switch (drcComp->chanConfig) { + case MODE_6_1: + tmp += fMultDiv2(fMult(slev, ext_leva), + (FIXP_PCM)pSamples[drcComp->channelIdx[4]]) >> + (DOWNMIX_SHIFT - 1); /* Ls */ + tmp += fMultDiv2(fMult(slev, ext_leva), + (FIXP_PCM)pSamples[drcComp->channelIdx[5]]) >> + (DOWNMIX_SHIFT - 1); /* Rs */ + tmp += fMult(fMult(slev, ext_levb), + (FIXP_PCM)pSamples[drcComp->channelIdx[6]]) >> + (DOWNMIX_SHIFT - 1); /* Cs */ + tmp += fMult(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >> + (DOWNMIX_SHIFT - 1); /* C */ + tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[0]]) >> + DOWNMIX_SHIFT); /* L */ + tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[1]]) >> + DOWNMIX_SHIFT); /* R */ + tmp += fMult(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> + (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ + break; + case MODE_7_1_BACK: + case MODE_7_1_REAR_SURROUND: + tmp += fMultDiv2(fMult(slev, ext_leva), + (FIXP_PCM)pSamples[drcComp->channelIdx[4]]) >> + (DOWNMIX_SHIFT - 1); /* Ls */ + tmp += fMultDiv2(fMult(slev, ext_leva), + (FIXP_PCM)pSamples[drcComp->channelIdx[5]]) >> + (DOWNMIX_SHIFT - 1); /* Rs */ + tmp += fMultDiv2(fMult(slev, ext_levb), + (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >> + (DOWNMIX_SHIFT - 1); /* Lrs / Lss*/ + tmp += fMultDiv2(fMult(slev, ext_levb), + (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >> + (DOWNMIX_SHIFT - 1); /* Rrs / Rss*/ + tmp += fMult(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >> + (DOWNMIX_SHIFT - 1); /* C */ + tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[0]]) >> + DOWNMIX_SHIFT); /* L */ + tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[1]]) >> + DOWNMIX_SHIFT); /* R */ + tmp += fMult(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> + (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ + break; + case MODE_1_2_2_2_1: + case MODE_7_1_FRONT_CENTER: + tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[0]]) >> + DOWNMIX_SHIFT); /* L */ + tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[1]]) >> + DOWNMIX_SHIFT); /* R */ + tmp += + fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >> + (DOWNMIX_SHIFT - 1); /* Lc */ + tmp += + fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >> + (DOWNMIX_SHIFT - 1); /* Rc */ + tmp += fMultDiv2(fMult(ext_leva, clev), + (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >> + (DOWNMIX_SHIFT - 1); /* Lc - second path*/ + tmp += fMultDiv2(fMult(ext_leva, clev), + (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >> + (DOWNMIX_SHIFT - 1); /* Rc - second path*/ + tmp += fMult(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >> + (DOWNMIX_SHIFT - 1); /* C */ + tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[4]]) >> + (DOWNMIX_SHIFT - 1); /* Ls */ + tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[5]]) >> + (DOWNMIX_SHIFT - 1); /* Rs */ + tmp += fMult(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> + (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ + break; + case MODE_7_1_TOP_FRONT: + tmp += + fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[0]]) >> + (DOWNMIX_SHIFT - 1); /* L */ + tmp += + fMultDiv2(ext_leva, (FIXP_PCM)pSamples[drcComp->channelIdx[1]]) >> + (DOWNMIX_SHIFT - 1); /* R */ + tmp += + fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[7]]) >> + (DOWNMIX_SHIFT - 1); /* Lvh */ + tmp += + fMultDiv2(ext_levb, (FIXP_PCM)pSamples[drcComp->channelIdx[8]]) >> + (DOWNMIX_SHIFT - 1); /* Rvh */ + tmp += fMult(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[2]]) >> + (DOWNMIX_SHIFT - 1); /* C */ + tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[4]]) >> + (DOWNMIX_SHIFT - 1); /* Ls */ + tmp += fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[5]]) >> + (DOWNMIX_SHIFT - 1); /* Rs */ + tmp += fMult(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> + (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ + break; + default: + if (drcComp->channelIdx[LS] >= 0) + tmp += + fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[LS]]) >> + (DOWNMIX_SHIFT - 1); /* Ls */ + if (drcComp->channelIdx[LS2] >= 0) + tmp += + fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[LS2]]) >> + (DOWNMIX_SHIFT - 1); /* Ls2 */ + if (drcComp->channelIdx[RS] >= 0) + tmp += + fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[RS]]) >> + (DOWNMIX_SHIFT - 1); /* Rs */ + if (drcComp->channelIdx[RS2] >= 0) + tmp += + fMultDiv2(slev, (FIXP_PCM)pSamples[drcComp->channelIdx[RS2]]) >> + (DOWNMIX_SHIFT - 1); /* Rs2 */ + if ((drcComp->channelIdx[LS] >= 0) && (drcComp->channelIdx[LS2] >= 0)) + tmp = fMult(FL2FXCONST_DBL(0.707f), tmp); /* 7.1ch */ + /*if ((drcComp->channelIdx[RS] >= 0) && (drcComp->channelIdx[RS2] >= 0)) tmp *=0.707f;*/ /* 7.1ch */ + if (drcComp->channelIdx[S] >= 0) + tmp += + fMultDiv2(slev, + fMult(FL2FXCONST_DBL(0.7f), + (FIXP_PCM)pSamples[drcComp->channelIdx[S]])) >> + (DOWNMIX_SHIFT - 1); /* S */ + if (drcComp->channelIdx[C] >= 0) + tmp += fMult(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[C]]) >> + (DOWNMIX_SHIFT - 1); /* C (2*clev) */ + if (drcComp->channelIdx[3] >= 0) + tmp += fMult(lfe_lev, (FIXP_PCM)pSamples[drcComp->channelIdx[3]]) >> + (DOWNMIX_SHIFT - 1 - LFE_LEV_SCALE); /* LFE */ + tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[L]]) >> + DOWNMIX_SHIFT); /* L */ + tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[R]]) >> + DOWNMIX_SHIFT); /* R */ + } + + /* apply scaling of downmix gains */ + /* only for positive values only, as legacy decoders might not know this + * parameter */ + if (dmxGain2 > FL2FXCONST_DBL(0.f)) { + if (drcComp->fullChannels > 5) { + tmp = fMult(dmxGain5, tmp) << (DFRACT_BITS - 1 - METADATA_FRACT_BITS); + } + tmp = fMult(dmxGain2, tmp) << (DFRACT_BITS - 1 - METADATA_FRACT_BITS); + } + peak[1] = fixMax(peak[1], fixp_abs(tmp)); + } + } +} + +INT FDK_DRC_Generator_Open(HDRC_COMP* phDrcComp) { + INT err = 0; + HDRC_COMP hDcComp = NULL; + + if (phDrcComp == NULL) { + err = -1; + goto bail; + } + + /* allocate memory */ + hDcComp = (HDRC_COMP)FDKcalloc(1, sizeof(DRC_COMP)); + + if (hDcComp == NULL) { + err = -1; + goto bail; + } + + FDKmemclear(hDcComp, sizeof(DRC_COMP)); + + /* Return drc compressor instance */ + *phDrcComp = hDcComp; + return err; +bail: + FDK_DRC_Generator_Close(&hDcComp); + return err; +} + +INT FDK_DRC_Generator_Close(HDRC_COMP* phDrcComp) { + if (phDrcComp == NULL) { + return -1; + } + if (*phDrcComp != NULL) { + FDKfree(*phDrcComp); + *phDrcComp = NULL; + } + return 0; +} + +INT FDK_DRC_Generator_Initialize(HDRC_COMP drcComp, + const DRC_PROFILE profileLine, + const DRC_PROFILE profileRF, + const INT blockLength, const UINT sampleRate, + const CHANNEL_MODE channelMode, + const CHANNEL_ORDER channelOrder, + const UCHAR useWeighting) { + int i; + CHANNEL_MAPPING channelMapping; + + drcComp->limDecay = + FL2FXCONST_DBL(((0.006f / 256) * blockLength) / METADATA_INT_SCALE); + + /* Save parameters. */ + drcComp->blockLength = blockLength; + drcComp->sampleRate = sampleRate; + drcComp->chanConfig = channelMode; + drcComp->useWeighting = useWeighting; + + if (FDK_DRC_Generator_setDrcProfile(drcComp, profileLine, profileRF) != + 0) { /* expects initialized blockLength and sampleRate */ + return (-1); + } + + /* Set number of channels and channel offsets. */ + if (FDKaacEnc_InitChannelMapping(channelMode, channelOrder, + &channelMapping) != AAC_ENC_OK) { + return (-2); + } + + for (i = 0; i < 9; i++) drcComp->channelIdx[i] = -1; + + switch (channelMode) { + case MODE_1: /* mono */ + drcComp->channelIdx[C] = channelMapping.elInfo[0].ChannelIndex[0]; + break; + case MODE_2: /* stereo */ + drcComp->channelIdx[L] = channelMapping.elInfo[0].ChannelIndex[0]; + drcComp->channelIdx[R] = channelMapping.elInfo[0].ChannelIndex[1]; + break; + case MODE_1_2: /* 3ch */ + drcComp->channelIdx[L] = channelMapping.elInfo[1].ChannelIndex[0]; + drcComp->channelIdx[R] = channelMapping.elInfo[1].ChannelIndex[1]; + drcComp->channelIdx[C] = channelMapping.elInfo[0].ChannelIndex[0]; + break; + case MODE_1_2_1: /* 4ch */ + drcComp->channelIdx[L] = channelMapping.elInfo[1].ChannelIndex[0]; + drcComp->channelIdx[R] = channelMapping.elInfo[1].ChannelIndex[1]; + drcComp->channelIdx[C] = channelMapping.elInfo[0].ChannelIndex[0]; + drcComp->channelIdx[S] = channelMapping.elInfo[2].ChannelIndex[0]; + break; + case MODE_1_2_2: /* 5ch */ + drcComp->channelIdx[L] = channelMapping.elInfo[1].ChannelIndex[0]; + drcComp->channelIdx[R] = channelMapping.elInfo[1].ChannelIndex[1]; + drcComp->channelIdx[C] = channelMapping.elInfo[0].ChannelIndex[0]; + drcComp->channelIdx[LS] = channelMapping.elInfo[2].ChannelIndex[0]; + drcComp->channelIdx[RS] = channelMapping.elInfo[2].ChannelIndex[1]; + break; + case MODE_1_2_2_1: /* 5.1 ch */ + drcComp->channelIdx[L] = channelMapping.elInfo[1].ChannelIndex[0]; + drcComp->channelIdx[R] = channelMapping.elInfo[1].ChannelIndex[1]; + drcComp->channelIdx[C] = channelMapping.elInfo[0].ChannelIndex[0]; + drcComp->channelIdx[LFE] = channelMapping.elInfo[3].ChannelIndex[0]; + drcComp->channelIdx[LS] = channelMapping.elInfo[2].ChannelIndex[0]; + drcComp->channelIdx[RS] = channelMapping.elInfo[2].ChannelIndex[1]; + break; + case MODE_1_2_2_2_1: /* 7.1 ch */ + case MODE_7_1_FRONT_CENTER: + drcComp->channelIdx[L] = channelMapping.elInfo[2].ChannelIndex[0]; /* l */ + drcComp->channelIdx[R] = channelMapping.elInfo[2].ChannelIndex[1]; /* r */ + drcComp->channelIdx[C] = channelMapping.elInfo[0].ChannelIndex[0]; /* c */ + drcComp->channelIdx[LFE] = + channelMapping.elInfo[4].ChannelIndex[0]; /* lfe */ + drcComp->channelIdx[LS] = + channelMapping.elInfo[3].ChannelIndex[0]; /* ls */ + drcComp->channelIdx[RS] = + channelMapping.elInfo[3].ChannelIndex[1]; /* rs */ + drcComp->channelIdx[LS2] = + channelMapping.elInfo[1].ChannelIndex[0]; /* lc */ + drcComp->channelIdx[RS2] = + channelMapping.elInfo[1].ChannelIndex[1]; /* rc */ + break; + case MODE_7_1_BACK: + case MODE_7_1_REAR_SURROUND: + drcComp->channelIdx[L] = channelMapping.elInfo[1].ChannelIndex[0]; /* l */ + drcComp->channelIdx[R] = channelMapping.elInfo[1].ChannelIndex[1]; /* r */ + drcComp->channelIdx[C] = channelMapping.elInfo[0].ChannelIndex[0]; /* c */ + drcComp->channelIdx[LFE] = + channelMapping.elInfo[4].ChannelIndex[0]; /* lfe */ + drcComp->channelIdx[LS] = + channelMapping.elInfo[3].ChannelIndex[0]; /* lrear */ + drcComp->channelIdx[RS] = + channelMapping.elInfo[3].ChannelIndex[1]; /* rrear */ + drcComp->channelIdx[LS2] = + channelMapping.elInfo[2].ChannelIndex[0]; /* ls */ + drcComp->channelIdx[RS2] = + channelMapping.elInfo[2].ChannelIndex[1]; /* rs */ + break; + case MODE_6_1: + drcComp->channelIdx[L] = channelMapping.elInfo[1].ChannelIndex[0]; /* l */ + drcComp->channelIdx[R] = channelMapping.elInfo[1].ChannelIndex[1]; /* r */ + drcComp->channelIdx[C] = channelMapping.elInfo[0].ChannelIndex[0]; /* c */ + drcComp->channelIdx[LFE] = + channelMapping.elInfo[4].ChannelIndex[0]; /* lfe */ + drcComp->channelIdx[LS] = + channelMapping.elInfo[2].ChannelIndex[0]; /* ls */ + drcComp->channelIdx[RS] = + channelMapping.elInfo[2].ChannelIndex[1]; /* rs */ + drcComp->channelIdx[S] = channelMapping.elInfo[3].ChannelIndex[0]; /* s */ + break; + case MODE_7_1_TOP_FRONT: + drcComp->channelIdx[L] = channelMapping.elInfo[1].ChannelIndex[0]; /* l */ + drcComp->channelIdx[R] = channelMapping.elInfo[1].ChannelIndex[1]; /* r */ + drcComp->channelIdx[C] = channelMapping.elInfo[0].ChannelIndex[0]; /* c */ + drcComp->channelIdx[LFE] = + channelMapping.elInfo[3].ChannelIndex[0]; /* lfe */ + drcComp->channelIdx[LS] = + channelMapping.elInfo[2].ChannelIndex[0]; /* ls */ + drcComp->channelIdx[RS] = + channelMapping.elInfo[2].ChannelIndex[1]; /* rs */ + drcComp->channelIdx[LS2] = + channelMapping.elInfo[4].ChannelIndex[0]; /* lvh2 */ + drcComp->channelIdx[RS2] = + channelMapping.elInfo[4].ChannelIndex[1]; /* rvh2 */ + break; + default: + return (-1); + } + + drcComp->fullChannels = channelMapping.nChannelsEff; + drcComp->channels = channelMapping.nChannels; + + /* Init states. */ + drcComp->smoothLevel[0] = drcComp->smoothLevel[1] = + (FIXP_DBL)(-(135 << METADATA_FRACT_BITS)); + + FDKmemclear(drcComp->smoothGain, sizeof(drcComp->smoothGain)); + FDKmemclear(drcComp->holdCnt, sizeof(drcComp->holdCnt)); + FDKmemclear(drcComp->limGain, sizeof(drcComp->limGain)); + FDKmemclear(drcComp->prevPeak, sizeof(drcComp->prevPeak)); + FDKmemclear(drcComp->filter, sizeof(drcComp->filter)); + + return (0); +} + +INT FDK_DRC_Generator_setDrcProfile(HDRC_COMP drcComp, + const DRC_PROFILE profileLine, + const DRC_PROFILE profileRF) { + int profileIdx, i; + + drcComp->profile[0] = profileLine; + drcComp->profile[1] = profileRF; + + for (i = 0; i < 2; i++) { + /* get profile index */ + switch (drcComp->profile[i]) { + case DRC_NONE: + case DRC_NOT_PRESENT: + case DRC_FILMSTANDARD: + profileIdx = 0; + break; + case DRC_FILMLIGHT: + profileIdx = 1; + break; + case DRC_MUSICSTANDARD: + profileIdx = 2; + break; + case DRC_MUSICLIGHT: + profileIdx = 3; + break; + case DRC_SPEECH: + profileIdx = 4; + break; + case DRC_DELAY_TEST: + profileIdx = 5; + break; + default: + return (-1); + } + + /* get parameters for selected profile */ + if (profileIdx >= 0) { + drcComp->maxBoostThr[i] = tabMaxBoostThr[profileIdx]; + drcComp->boostThr[i] = tabBoostThr[profileIdx]; + drcComp->earlyCutThr[i] = tabEarlyCutThr[profileIdx]; + drcComp->cutThr[i] = tabCutThr[profileIdx]; + drcComp->maxCutThr[i] = tabMaxCutThr[profileIdx]; + + drcComp->boostFac[i] = tabBoostRatio[profileIdx]; + drcComp->earlyCutFac[i] = tabEarlyCutRatio[profileIdx]; + drcComp->cutFac[i] = tabCutRatio[profileIdx]; + + drcComp->maxBoost[i] = tabMaxBoost[profileIdx]; + drcComp->maxCut[i] = tabMaxCut[profileIdx]; + drcComp->maxEarlyCut[i] = + -fMult((drcComp->cutThr[i] - drcComp->earlyCutThr[i]), + drcComp->earlyCutFac[i]); /* no scaling after mult needed, + earlyCutFac is in FIXP_DBL */ + + drcComp->fastAttack[i] = tc2Coeff( + tabFastAttack[profileIdx], drcComp->sampleRate, drcComp->blockLength); + drcComp->fastDecay[i] = tc2Coeff( + tabFastDecay[profileIdx], drcComp->sampleRate, drcComp->blockLength); + drcComp->slowAttack[i] = tc2Coeff( + tabSlowAttack[profileIdx], drcComp->sampleRate, drcComp->blockLength); + drcComp->slowDecay[i] = tc2Coeff( + tabSlowDecay[profileIdx], drcComp->sampleRate, drcComp->blockLength); + drcComp->holdOff[i] = tabHoldOff[profileIdx] * 256 / drcComp->blockLength; + + drcComp->attackThr[i] = tabAttackThr[profileIdx]; + drcComp->decayThr[i] = tabDecayThr[profileIdx]; + } + + drcComp->smoothGain[i] = FL2FXCONST_DBL(0.f); + } + return (0); +} + +INT FDK_DRC_Generator_Calc(HDRC_COMP drcComp, const INT_PCM* const inSamples, + const UINT inSamplesBufSize, const INT dialnorm, + const INT drc_TargetRefLevel, + const INT comp_TargetRefLevel, const FIXP_DBL clev, + const FIXP_DBL slev, const FIXP_DBL ext_leva, + const FIXP_DBL ext_levb, const FIXP_DBL lfe_lev, + const INT dmxGain5, const INT dmxGain2, + INT* const pDynrng, INT* const pCompr) { + int i, c; + FIXP_DBL peak[2]; + + /************************************************************************** + * compressor + **************************************************************************/ + if ((drcComp->profile[0] != DRC_NONE) || (drcComp->profile[1] != DRC_NONE)) { + /* Calc loudness level */ + FIXP_DBL level_b = FL2FXCONST_DBL(0.f); + int level_e = DFRACT_BITS - 1; + + /* Increase energy time resolution with shorter processing blocks. 16 is an + * empiric value. */ + const int granuleLength = fixMin(16, drcComp->blockLength); + + if (drcComp->useWeighting) { + FIXP_DBL x1, x2, y, y1, y2; + /* sum of filter coefficients about 2.5 -> squared value is 6.25 + WEIGHTING_FILTER_SHIFT is 2 -> scaling about 16, therefore reduce + granuleShift by 1. + */ + const int granuleShift = getShiftFactor(granuleLength) - 1; + + for (c = 0; c < (int)drcComp->channels; c++) { + const INT_PCM* pSamples = inSamples + c * inSamplesBufSize; + + if (c == drcComp->channelIdx[LFE]) { + continue; /* skip LFE */ + } + + /* get filter states */ + x1 = drcComp->filter[c].x1; + x2 = drcComp->filter[c].x2; + y1 = drcComp->filter[c].y1; + y2 = drcComp->filter[c].y2; + + i = 0; + + do { + int offset = i; + FIXP_DBL accu = FL2FXCONST_DBL(0.f); + + for (i = offset; + i < fixMin(offset + granuleLength, drcComp->blockLength); i++) { + /* apply weighting filter */ + FIXP_DBL x = + FX_PCM2FX_DBL((FIXP_PCM)pSamples[i]) >> WEIGHTING_FILTER_SHIFT; + + /* y = b0 * (x - x2) - a1 * y1 - a2 * y2; */ + y = fMult(b0, x - x2) - fMult(a1, y1) - fMult(a2, y2); + + x2 = x1; + x1 = x; + y2 = y1; + y1 = y; + + accu += fPow2Div2(y) >> (granuleShift - 1); /* partial energy */ + } /* i */ + + fixpAdd(accu, granuleShift + 2 * WEIGHTING_FILTER_SHIFT, &level_b, + &level_e); /* sup up partial energies */ + + } while (i < drcComp->blockLength); + + /* save filter states */ + drcComp->filter[c].x1 = x1; + drcComp->filter[c].x2 = x2; + drcComp->filter[c].y1 = y1; + drcComp->filter[c].y2 = y2; + } /* c */ + } /* weighting */ + else { + const int granuleShift = getShiftFactor(granuleLength); + + for (c = 0; c < (int)drcComp->channels; c++) { + const INT_PCM* pSamples = inSamples + c * inSamplesBufSize; + + if ((int)c == drcComp->channelIdx[LFE]) { + continue; /* skip LFE */ + } + + i = 0; + + do { + int offset = i; + FIXP_DBL accu = FL2FXCONST_DBL(0.f); + + for (i = offset; + i < fixMin(offset + granuleLength, drcComp->blockLength); i++) { + /* partial energy */ + accu += fPow2Div2((FIXP_PCM)pSamples[i]) >> (granuleShift - 1); + } /* i */ + + fixpAdd(accu, granuleShift, &level_b, + &level_e); /* sup up partial energies */ + + } while (i < drcComp->blockLength); + } + } /* weighting */ + + /* + * Convert to dBFS, apply dialnorm + */ + /* level scaling */ + + /* descaled level in ld64 representation */ + FIXP_DBL ldLevel = + CalcLdData(level_b) + + (FIXP_DBL)((level_e - 12) << (DFRACT_BITS - 1 - LD_DATA_SHIFT)) - + CalcLdData((FIXP_DBL)(drcComp->blockLength << (DFRACT_BITS - 1 - 12))); + + /* if (level < 1e-10) level = 1e-10f; */ + ldLevel = + fMax(ldLevel, FL2FXCONST_DBL(-0.51905126482615036685473741085772f)); + + /* level = 10 * log(level)/log(10) + 3; + * = 10*log(2)/log(10) * ld(level) + 3; + * = 10 * 0.30102999566398119521373889472449 * ld(level) + 3 + * = 10 * (0.30102999566398119521373889472449 * ld(level) + 0.3) + * = 10 * (0.30102999566398119521373889472449 * ld64(level) + 0.3/64) + * * 64 + * + * additional scaling with METADATA_FRACT_BITS: + * = 10 * (0.30102999566398119521373889472449 * ld64(level) + 0.3/64) + * * 64 * 2^(METADATA_FRACT_BITS) = 10 * (0.30102999566398119521373889472449 + * * ld64(level) + 0.3/64) * 2^(METADATA_FRACT_BITS+LD_DATA_SHIFT) = + * 10*2^(METADATA_FRACT_BITS+LD_DATA_SHIFT) * ( + * 0.30102999566398119521373889472449 * ld64(level) + 0.3/64 ) + * */ + FIXP_DBL level = fMult( + (FIXP_DBL)(10 << (METADATA_FRACT_BITS + LD_DATA_SHIFT)), + fMult(FL2FXCONST_DBL(0.30102999566398119521373889472449f), ldLevel) + + (FIXP_DBL)(FL2FXCONST_DBL(0.3f) >> LD_DATA_SHIFT)); + + /* level -= dialnorm + 31 */ /* this is fixed to Dolby-ReferenceLevel as + compressor profiles are defined relative to + this */ + level -= ((FIXP_DBL)(dialnorm << (METADATA_FRACT_BITS - 16)) + + (FIXP_DBL)(31 << METADATA_FRACT_BITS)); + + for (i = 0; i < 2; i++) { + if (drcComp->profile[i] == DRC_NONE) { + /* no compression */ + drcComp->smoothGain[i] = FL2FXCONST_DBL(0.f); + } else { + FIXP_DBL gain, alpha, lvl2smthlvl; + + /* calc static gain */ + if (level <= drcComp->maxBoostThr[i]) { + /* max boost */ + gain = drcComp->maxBoost[i]; + } else if (level < drcComp->boostThr[i]) { + /* boost range */ + gain = fMult((level - drcComp->boostThr[i]), drcComp->boostFac[i]); + } else if (level <= drcComp->earlyCutThr[i]) { + /* null band */ + gain = FL2FXCONST_DBL(0.f); + } else if (level <= drcComp->cutThr[i]) { + /* early cut range */ + gain = + fMult((level - drcComp->earlyCutThr[i]), drcComp->earlyCutFac[i]); + } else if (level < drcComp->maxCutThr[i]) { + /* cut range */ + gain = fMult((level - drcComp->cutThr[i]), drcComp->cutFac[i]) - + drcComp->maxEarlyCut[i]; + } else { + /* max cut */ + gain = -drcComp->maxCut[i]; + } + + /* choose time constant */ + lvl2smthlvl = level - drcComp->smoothLevel[i]; + if (gain < drcComp->smoothGain[i]) { + /* attack */ + if (lvl2smthlvl > drcComp->attackThr[i]) { + /* fast attack */ + alpha = drcComp->fastAttack[i]; + } else { + /* slow attack */ + alpha = drcComp->slowAttack[i]; + } + } else { + /* release */ + if (lvl2smthlvl < -drcComp->decayThr[i]) { + /* fast release */ + alpha = drcComp->fastDecay[i]; + } else { + /* slow release */ + alpha = drcComp->slowDecay[i]; + } + } + + /* smooth gain & level */ + if ((gain < drcComp->smoothGain[i]) || + (drcComp->holdCnt[i] == + 0)) { /* hold gain unless we have an attack or hold + period is over */ + FIXP_DBL accu; + + /* drcComp->smoothLevel[i] = (1-alpha) * drcComp->smoothLevel[i] + + * alpha * level; */ + accu = fMult(((FIXP_DBL)MAXVAL_DBL - alpha), drcComp->smoothLevel[i]); + accu += fMult(alpha, level); + drcComp->smoothLevel[i] = accu; + + /* drcComp->smoothGain[i] = (1-alpha) * drcComp->smoothGain[i] + + * alpha * gain; */ + accu = fMult(((FIXP_DBL)MAXVAL_DBL - alpha), drcComp->smoothGain[i]); + accu += fMult(alpha, gain); + drcComp->smoothGain[i] = accu; + } + + /* hold counter */ + if (drcComp->holdCnt[i]) { + drcComp->holdCnt[i]--; + } + if (gain < drcComp->smoothGain[i]) { + drcComp->holdCnt[i] = drcComp->holdOff[i]; + } + } /* profile != DRC_NONE */ + } /* for i=1..2 */ + } else { + /* no compression */ + drcComp->smoothGain[0] = FL2FXCONST_DBL(0.f); + drcComp->smoothGain[1] = FL2FXCONST_DBL(0.f); + } + + /************************************************************************** + * limiter + **************************************************************************/ + + findPeakLevels(drcComp, inSamples, clev, slev, ext_leva, ext_levb, lfe_lev, + (FIXP_DBL)((LONG)(dmxGain5) << (METADATA_FRACT_BITS - 16)), + (FIXP_DBL)((LONG)(dmxGain2) << (METADATA_FRACT_BITS - 16)), + peak); + + for (i = 0; i < 2; i++) { + FIXP_DBL tmp = drcComp->prevPeak[i]; + drcComp->prevPeak[i] = peak[i]; + peak[i] = fixMax(peak[i], tmp); + + /* + * Convert to dBFS, apply dialnorm + */ + /* descaled peak in ld64 representation */ + FIXP_DBL ld_peak = + CalcLdData(peak[i]) + + (FIXP_DBL)((LONG)DOWNMIX_SHIFT << (DFRACT_BITS - 1 - LD_DATA_SHIFT)); + + /* if (peak < 1e-6) level = 1e-6f; */ + ld_peak = + fMax(ld_peak, FL2FXCONST_DBL(-0.31143075889569022011284244651463f)); + + /* peak[i] = 20 * log(peak[i])/log(10) + 0.2f + + * (drcComp->smoothGain[i]*2^METADATA_FRACT_BITS) peak[i] = 20 * + * log(2)/log(10) * ld(peak[i]) + 0.2f + + * (drcComp->smoothGain[i]*2^METADATA_FRACT_BITS) peak[i] = 10 * + * 2*0.30102999566398119521373889472449 * ld(peak[i]) + 0.2f + + * (drcComp->smoothGain[i]*2^METADATA_FRACT_BITS) + * + * additional scaling with METADATA_FRACT_BITS: + * peak[i] = (10 * 2*0.30102999566398119521373889472449 * ld64(peak[i]) * 64 + * + 0.2f + + * (drcComp->smoothGain[i]*2^METADATA_FRACT_BITS))*2^(-METADATA_FRACT_BITS) + * peak[i] = 10*2^(METADATA_FRACT_BITS+LD_DATA_SHIFT) * + * 2*0.30102999566398119521373889472449 * ld64(peak[i]) + * + 0.2f*2^(-METADATA_FRACT_BITS) + drcComp->smoothGain[i] + */ + peak[i] = fMult( + (FIXP_DBL)(10 << (METADATA_FRACT_BITS + LD_DATA_SHIFT)), + fMult(FL2FX_DBL(2 * 0.30102999566398119521373889472449f), ld_peak)); + peak[i] += + (FL2FX_DBL(0.5f) >> METADATA_INT_BITS); /* add a little bit headroom */ + peak[i] += drcComp->smoothGain[i]; + } + + /* peak -= dialnorm + 31; */ /* this is Dolby style only */ + peak[0] -= (FIXP_DBL)((dialnorm - drc_TargetRefLevel) + << (METADATA_FRACT_BITS - + 16)); /* peak[0] -= dialnorm - drc_TargetRefLevel */ + + /* peak += 11; */ + /* this is Dolby style only */ /* RF mode output is 11dB higher */ + /*peak += comp_TargetRefLevel - drc_TargetRefLevel;*/ + peak[1] -= + (FIXP_DBL)((dialnorm - comp_TargetRefLevel) + << (METADATA_FRACT_BITS - + 16)); /* peak[1] -= dialnorm - comp_TargetRefLevel */ + + /* limiter gain */ + drcComp->limGain[0] += drcComp->limDecay; /* linear limiter release */ + drcComp->limGain[0] = fixMin(drcComp->limGain[0], -peak[0]); + + drcComp->limGain[1] += 2 * drcComp->limDecay; /* linear limiter release */ + drcComp->limGain[1] = fixMin(drcComp->limGain[1], -peak[1]); + + /*************************************************************************/ + + /* apply limiting, return DRC gains*/ + { + FIXP_DBL tmp; + + tmp = drcComp->smoothGain[0]; + if (drcComp->limGain[0] < FL2FXCONST_DBL(0.f)) { + tmp += drcComp->limGain[0]; + } + *pDynrng = (LONG)scaleValue(tmp, -(METADATA_FRACT_BITS - 16)); + + tmp = drcComp->smoothGain[1]; + if (drcComp->limGain[1] < FL2FXCONST_DBL(0.f)) { + tmp += drcComp->limGain[1]; + } + *pCompr = (LONG)scaleValue(tmp, -(METADATA_FRACT_BITS - 16)); + } + + return 0; +} + +DRC_PROFILE FDK_DRC_Generator_getDrcProfile(const HDRC_COMP drcComp) { + return drcComp->profile[0]; +} + +DRC_PROFILE FDK_DRC_Generator_getCompProfile(const HDRC_COMP drcComp) { + return drcComp->profile[1]; +} diff --git a/fdk-aac/libAACenc/src/metadata_compressor.h b/fdk-aac/libAACenc/src/metadata_compressor.h new file mode 100644 index 0000000..1d0aa42 --- /dev/null +++ b/fdk-aac/libAACenc/src/metadata_compressor.h @@ -0,0 +1,255 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M. Neusinger + + Description: Compressor for AAC Metadata Generator + +*******************************************************************************/ + +#ifndef METADATA_COMPRESSOR_H +#define METADATA_COMPRESSOR_H + +#include "FDK_audio.h" +#include "common_fix.h" + +#include "aacenc.h" + +#define LFE_LEV_SCALE 2 + +/** + * DRC compression profiles. + */ +typedef enum DRC_PROFILE { + DRC_NONE = 0, + DRC_FILMSTANDARD = 1, + DRC_FILMLIGHT = 2, + DRC_MUSICSTANDARD = 3, + DRC_MUSICLIGHT = 4, + DRC_SPEECH = 5, + DRC_DELAY_TEST = 6, + DRC_NOT_PRESENT = -2 + +} DRC_PROFILE; + +/** + * DRC Compressor handle. + */ +typedef struct DRC_COMP DRC_COMP, *HDRC_COMP; + +/** + * \brief Open a DRC Compressor instance. + * + * Allocate memory for a compressor instance. + * + * \param phDrcComp A pointer to a compressor handle. Initialized on + * return. + * + * \return + * - 0, on succes. + * - unequal 0, on failure. + */ +INT FDK_DRC_Generator_Open(HDRC_COMP *phDrcComp); + +/** + * \brief Close the DRC Compressor instance. + * + * Deallocate instance and free whole memory. + * + * \param phDrcComp Pointer to the compressor handle to be + * deallocated. + * + * \return + * - 0, on succes. + * - unequal 0, on failure. + */ +INT FDK_DRC_Generator_Close(HDRC_COMP *phDrcComp); + +/** + * \brief Configure DRC Compressor. + * + * \param drcComp Compressor handle. + * \param profileLine DRC profile for line mode. + * \param profileRF DRC profile for RF mode. + * \param blockLength Length of processing block in samples per + * channel. + * \param sampleRate Sampling rate in Hz. + * \param channelMode Channel configuration. + * \param channelOrder Channel order, MPEG or WAV. + * \param useWeighting Use weighting filter for loudness calculation + * + * \return + * - 0, on success, + * - unequal 0, on failure + */ +INT FDK_DRC_Generator_Initialize(HDRC_COMP drcComp, + const DRC_PROFILE profileLine, + const DRC_PROFILE profileRF, + const INT blockLength, const UINT sampleRate, + const CHANNEL_MODE channelMode, + const CHANNEL_ORDER channelOrder, + const UCHAR useWeighting); + +/** + * \brief Calculate DRC Compressor Gain. + * + * \param drcComp Compressor handle. + * \param inSamples Pointer to interleaved input audio samples. + * \param inSamplesBufSize Size of inSamples for one channel. + * \param dialnorm Dialog Level in dB (typically -31...-1). + * \param drc_TargetRefLevel + * \param comp_TargetRefLevel + * \param clev Downmix center mix factor (typically 0.707, + * 0.595 or 0.5) + * \param slev Downmix surround mix factor (typically 0.707, + * 0.5, or 0) + * \param ext_leva Downmix gain factor A + * \param ext_levb Downmix gain factor B + * \param lfe_lev LFE gain factor + * \param dmxGain5 Gain factor for downmix to 5 channels + * \param dmxGain2 Gain factor for downmix to 2 channels + * \param dynrng Pointer to variable receiving line mode DRC gain + * in dB + * \param compr Pointer to variable receiving RF mode DRC gain + * in dB + * + * \return + * - 0, on success, + * - unequal 0, on failure + */ +INT FDK_DRC_Generator_Calc(HDRC_COMP drcComp, const INT_PCM *const inSamples, + const UINT inSamplesBufSize, const INT dialnorm, + const INT drc_TargetRefLevel, + const INT comp_TargetRefLevel, const FIXP_DBL clev, + const FIXP_DBL slev, const FIXP_DBL ext_leva, + const FIXP_DBL ext_levb, const FIXP_DBL lfe_lev, + const INT dmxGain5, const INT dmxGain2, + INT *const dynrng, INT *const compr); + +/** + * \brief Configure DRC Compressor Profile. + * + * \param drcComp Compressor handle. + * \param profileLine DRC profile for line mode. + * \param profileRF DRC profile for RF mode. + * + * \return + * - 0, on success, + * - unequal 0, on failure + */ +INT FDK_DRC_Generator_setDrcProfile(HDRC_COMP drcComp, + const DRC_PROFILE profileLine, + const DRC_PROFILE profileRF); + +/** + * \brief Get DRC profile for line mode. + * + * \param drcComp Compressor handle. + * + * \return Current Profile. + */ +DRC_PROFILE FDK_DRC_Generator_getDrcProfile(const HDRC_COMP drcComp); + +/** + * \brief Get DRC profile for RF mode. + * + * \param drcComp Compressor handle. + * + * \return Current Profile. + */ +DRC_PROFILE FDK_DRC_Generator_getCompProfile(const HDRC_COMP drcComp); + +#endif /* METADATA_COMPRESSOR_H */ diff --git a/fdk-aac/libAACenc/src/metadata_main.cpp b/fdk-aac/libAACenc/src/metadata_main.cpp new file mode 100644 index 0000000..ada4502 --- /dev/null +++ b/fdk-aac/libAACenc/src/metadata_main.cpp @@ -0,0 +1,1191 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): V. Bacigalupo + + Description: Metadata Encoder library interface functions + +*******************************************************************************/ + +#include "metadata_main.h" +#include "metadata_compressor.h" +#include "FDK_bitstream.h" +#include "FDK_audio.h" +#include "genericStds.h" + +/*----------------- defines ----------------------*/ +#define MAX_DRC_BANDS (1 << 4) +#define MAX_DRC_FRAMELEN (2 * 1024) +#define MAX_DELAY_FRAMES (3) + +/*--------------- structure definitions --------------------*/ + +typedef struct AAC_METADATA { + /* MPEG: Dynamic Range Control */ + struct { + UCHAR prog_ref_level_present; + SCHAR prog_ref_level; + + UCHAR dyn_rng_sgn[MAX_DRC_BANDS]; + UCHAR dyn_rng_ctl[MAX_DRC_BANDS]; + + UCHAR drc_bands_present; + UCHAR drc_band_incr; + UCHAR drc_band_top[MAX_DRC_BANDS]; + UCHAR drc_interpolation_scheme; + AACENC_METADATA_DRC_PROFILE drc_profile; + INT drc_TargetRefLevel; /* used for Limiter */ + + /* excluded channels */ + UCHAR excluded_chns_present; + UCHAR exclude_mask[2]; /* MAX_NUMBER_CHANNELS/8 */ + } mpegDrc; + + /* ETSI: addtl ancillary data */ + struct { + /* Heavy Compression */ + UCHAR compression_on; /* flag, if compression value should be written */ + UCHAR compression_value; /* compression value */ + AACENC_METADATA_DRC_PROFILE comp_profile; + INT comp_TargetRefLevel; /* used for Limiter */ + INT timecode_coarse_status; + INT timecode_fine_status; + + UCHAR extAncDataStatus; + + struct { + UCHAR ext_downmix_lvl_status; + UCHAR ext_downmix_gain_status; + UCHAR ext_lfe_downmix_status; + UCHAR + ext_dmix_a_idx; /* extended downmix level (0..7, according to table) + */ + UCHAR + ext_dmix_b_idx; /* extended downmix level (0..7, according to table) + */ + UCHAR dmx_gain_5_sgn; + UCHAR dmx_gain_5_idx; + UCHAR dmx_gain_2_sgn; + UCHAR dmx_gain_2_idx; + UCHAR ext_dmix_lfe_idx; /* extended downmix level for lfe (0..15, + according to table) */ + + } extAncData; + + } etsiAncData; + + SCHAR centerMixLevel; /* center downmix level (0...7, according to table) */ + SCHAR + surroundMixLevel; /* surround downmix level (0...7, according to table) */ + UCHAR WritePCEMixDwnIdx; /* flag */ + UCHAR DmxLvl_On; /* flag */ + + UCHAR dolbySurroundMode; + UCHAR drcPresentationMode; + + UCHAR + metadataMode; /* indicate meta data mode in current frame (delay line) */ + +} AAC_METADATA; + +typedef struct FDK_METADATA_ENCODER { + INT metadataMode; + HDRC_COMP hDrcComp; + AACENC_MetaData submittedMetaData; + + INT nAudioDataDelay; /* Additional delay to round up to next frame border (in + samples) */ + INT nMetaDataDelay; /* Meta data delay (in frames) */ + INT nChannels; + CHANNEL_MODE channelMode; + + INT_PCM* pAudioDelayBuffer; + + AAC_METADATA metaDataBuffer[MAX_DELAY_FRAMES]; + INT metaDataDelayIdx; + + UCHAR drcInfoPayload[12]; + UCHAR drcDsePayload[8]; + + INT matrix_mixdown_idx; + + AACENC_EXT_PAYLOAD exPayload[2]; + INT nExtensions; + + UINT maxChannels; /* Maximum number of audio channels to be supported. */ + + INT finalizeMetaData; /* Delay switch off by one frame and write default + configuration to finalize the metadata setup. */ + INT initializeMetaData; /* Fill up delay line with first meta data info. This + is required to have meta data already in first + frame. */ +} FDK_METADATA_ENCODER; + +/*---------------- constants -----------------------*/ +static const AACENC_MetaData defaultMetaDataSetup = { + AACENC_METADATA_DRC_NONE, /* drc_profile */ + AACENC_METADATA_DRC_NOT_PRESENT, /* comp_profile */ + -(31 << 16), /* drc_TargetRefLevel */ + -(23 << 16), /* comp_TargetRefLevel */ + 0, /* prog_ref_level_present */ + -(23 << 16), /* prog_ref_level */ + 0, /* PCE_mixdown_idx_present */ + 0, /* ETSI_DmxLvl_present */ + 0, /* centerMixLevel */ + 0, /* surroundMixLevel */ + 0, /* dolbySurroundMode */ + 0, /* drcPresentationMode */ + {0, 0, 0, 0, 0, 0, 0, 0, 0} /* ExtMetaData */ +}; + +static const FIXP_DBL dmxTable[8] = { + ((FIXP_DBL)MAXVAL_DBL), FL2FXCONST_DBL(0.841f), FL2FXCONST_DBL(0.707f), + FL2FXCONST_DBL(0.596f), FL2FXCONST_DBL(0.500f), FL2FXCONST_DBL(0.422f), + FL2FXCONST_DBL(0.355f), FL2FXCONST_DBL(0.000f)}; + +#define FL2DMXLFE(a) FL2FXCONST_DBL((a) / (1 << LFE_LEV_SCALE)) +static const FIXP_DBL dmxLfeTable[16] = { + FL2DMXLFE(3.162f), FL2DMXLFE(2.000f), FL2DMXLFE(1.679f), FL2DMXLFE(1.413f), + FL2DMXLFE(1.189f), FL2DMXLFE(1.000f), FL2DMXLFE(0.841f), FL2DMXLFE(0.707f), + FL2DMXLFE(0.596f), FL2DMXLFE(0.500f), FL2DMXLFE(0.316f), FL2DMXLFE(0.178f), + FL2DMXLFE(0.100f), FL2DMXLFE(0.032f), FL2DMXLFE(0.010f), FL2DMXLFE(0.000f)}; + +static const UCHAR surmix2matrix_mixdown_idx[8] = {0, 0, 0, 1, 1, 2, 2, 3}; + +/*--------------- function declarations --------------------*/ +static FDK_METADATA_ERROR WriteMetadataPayload( + const HANDLE_FDK_METADATA_ENCODER hMetaData, + const AAC_METADATA* const pMetadata); + +static INT WriteDynamicRangeInfoPayload(const AAC_METADATA* const pMetadata, + UCHAR* const pExtensionPayload); + +static INT WriteEtsiAncillaryDataPayload(const AAC_METADATA* const pMetadata, + UCHAR* const pExtensionPayload); + +static FDK_METADATA_ERROR CompensateAudioDelay( + HANDLE_FDK_METADATA_ENCODER hMetaDataEnc, INT_PCM* const pAudioSamples, + const UINT audioSamplesBufSize, const INT nAudioSamples); + +static FDK_METADATA_ERROR LoadSubmittedMetadata( + const AACENC_MetaData* const hMetadata, const INT nChannels, + const INT metadataMode, AAC_METADATA* const pAacMetaData); + +static FDK_METADATA_ERROR ProcessCompressor(AAC_METADATA* pMetadata, + HDRC_COMP hDrcComp, + const INT_PCM* const pSamples, + const UINT samplesBufSize, + const INT nSamples); + +/*------------- function definitions ----------------*/ + +static DRC_PROFILE convertProfile(AACENC_METADATA_DRC_PROFILE aacProfile) { + DRC_PROFILE drcProfile = DRC_NONE; + + switch (aacProfile) { + case AACENC_METADATA_DRC_NONE: + drcProfile = DRC_NONE; + break; + case AACENC_METADATA_DRC_FILMSTANDARD: + drcProfile = DRC_FILMSTANDARD; + break; + case AACENC_METADATA_DRC_FILMLIGHT: + drcProfile = DRC_FILMLIGHT; + break; + case AACENC_METADATA_DRC_MUSICSTANDARD: + drcProfile = DRC_MUSICSTANDARD; + break; + case AACENC_METADATA_DRC_MUSICLIGHT: + drcProfile = DRC_MUSICLIGHT; + break; + case AACENC_METADATA_DRC_SPEECH: + drcProfile = DRC_SPEECH; + break; + case AACENC_METADATA_DRC_NOT_PRESENT: + drcProfile = DRC_NOT_PRESENT; + break; + default: + drcProfile = DRC_NONE; + break; + } + return drcProfile; +} + +/* convert dialog normalization to program reference level */ +/* NOTE: this only is correct, if the decoder target level is set to -31dB for + * line mode / -20dB for RF mode */ +static UCHAR dialnorm2progreflvl(const INT d) { + return ((UCHAR)fMax(0, fMin((-d + (1 << 13)) >> 14, 127))); +} + +/* convert program reference level to dialog normalization */ +static INT progreflvl2dialnorm(const UCHAR p) { + return -((INT)(p << (16 - 2))); +} + +/* encode downmix levels to Downmixing_levels_MPEG4 */ +static SCHAR encodeDmxLvls(const SCHAR cmixlev, const SCHAR surmixlev) { + SCHAR dmxLvls = 0; + dmxLvls |= 0x80 | (cmixlev << 4); /* center_mix_level_on */ + dmxLvls |= 0x08 | surmixlev; /* surround_mix_level_on */ + + return dmxLvls; +} + +/* encode AAC DRC gain (ISO/IEC 14496-3:2005 4.5.2.7) */ +static void encodeDynrng(INT gain, UCHAR* const dyn_rng_ctl, + UCHAR* const dyn_rng_sgn) { + if (gain < 0) { + *dyn_rng_sgn = 1; + gain = -gain; + } else { + *dyn_rng_sgn = 0; + } + gain = fMin(gain, (127 << 14)); + + *dyn_rng_ctl = (UCHAR)((gain + (1 << 13)) >> 14); +} + +/* decode AAC DRC gain (ISO/IEC 14496-3:2005 4.5.2.7) */ +static INT decodeDynrng(const UCHAR dyn_rng_ctl, const UCHAR dyn_rng_sgn) { + INT tmp = ((INT)dyn_rng_ctl << (16 - 2)); + if (dyn_rng_sgn) tmp = -tmp; + + return tmp; +} + +/* encode AAC compression value (ETSI TS 101 154 page 99) */ +static UCHAR encodeCompr(INT gain) { + UCHAR x, y; + INT tmp; + + /* tmp = (int)((48.164f - gain) / 6.0206f * 15 + 0.5f); */ + tmp = ((3156476 - gain) * 15 + 197283) / 394566; + + if (tmp >= 240) { + return 0xFF; + } else if (tmp < 0) { + return 0; + } else { + x = tmp / 15; + y = tmp % 15; + } + + return (x << 4) | y; +} + +/* decode AAC compression value (ETSI TS 101 154 page 99) */ +static INT decodeCompr(const UCHAR compr) { + INT gain; + SCHAR x = compr >> 4; /* 4 MSB of compr */ + UCHAR y = (compr & 0x0F); /* 4 LSB of compr */ + + /* gain = (INT)((48.164f - 6.0206f * x - 0.4014f * y) ); */ + gain = (INT)( + scaleValue((FIXP_DBL)(((LONG)FL2FXCONST_DBL(6.0206f / 128.f) * (8 - x) - + (LONG)FL2FXCONST_DBL(0.4014f / 128.f) * y)), + -(DFRACT_BITS - 1 - 7 - 16))); + + return gain; +} + +FDK_METADATA_ERROR FDK_MetadataEnc_Open(HANDLE_FDK_METADATA_ENCODER* phMetaData, + const UINT maxChannels) { + FDK_METADATA_ERROR err = METADATA_OK; + HANDLE_FDK_METADATA_ENCODER hMetaData = NULL; + + if (phMetaData == NULL) { + err = METADATA_INVALID_HANDLE; + goto bail; + } + + /* allocate memory */ + if (NULL == (hMetaData = (HANDLE_FDK_METADATA_ENCODER)FDKcalloc( + 1, sizeof(FDK_METADATA_ENCODER)))) { + err = METADATA_MEMORY_ERROR; + goto bail; + } + FDKmemclear(hMetaData, sizeof(FDK_METADATA_ENCODER)); + + if (NULL == (hMetaData->pAudioDelayBuffer = (INT_PCM*)FDKcalloc( + maxChannels * MAX_DRC_FRAMELEN, sizeof(INT_PCM)))) { + err = METADATA_MEMORY_ERROR; + goto bail; + } + FDKmemclear(hMetaData->pAudioDelayBuffer, + maxChannels * MAX_DRC_FRAMELEN * sizeof(INT_PCM)); + hMetaData->maxChannels = maxChannels; + + /* Allocate DRC Compressor. */ + if (FDK_DRC_Generator_Open(&hMetaData->hDrcComp) != 0) { + err = METADATA_MEMORY_ERROR; + goto bail; + } + hMetaData->channelMode = MODE_UNKNOWN; + + /* Return metadata instance */ + *phMetaData = hMetaData; + + return err; + +bail: + FDK_MetadataEnc_Close(&hMetaData); + return err; +} + +FDK_METADATA_ERROR FDK_MetadataEnc_Close( + HANDLE_FDK_METADATA_ENCODER* phMetaData) { + FDK_METADATA_ERROR err = METADATA_OK; + + if (phMetaData == NULL) { + err = METADATA_INVALID_HANDLE; + goto bail; + } + + if (*phMetaData != NULL) { + FDK_DRC_Generator_Close(&(*phMetaData)->hDrcComp); + FDKfree((*phMetaData)->pAudioDelayBuffer); + FDKfree(*phMetaData); + *phMetaData = NULL; + } +bail: + return err; +} + +FDK_METADATA_ERROR FDK_MetadataEnc_Init( + HANDLE_FDK_METADATA_ENCODER hMetaData, const INT resetStates, + const INT metadataMode, const INT audioDelay, const UINT frameLength, + const UINT sampleRate, const UINT nChannels, const CHANNEL_MODE channelMode, + const CHANNEL_ORDER channelOrder) { + FDK_METADATA_ERROR err = METADATA_OK; + int i, nFrames, delay; + + if (hMetaData == NULL) { + err = METADATA_INVALID_HANDLE; + goto bail; + } + + /* Determine values for delay compensation. */ + for (nFrames = 0, delay = audioDelay - (INT)frameLength; delay > 0; + delay -= (INT)frameLength, nFrames++) + ; + + if ((nChannels > (8)) || (nChannels > hMetaData->maxChannels) || + ((-delay) > MAX_DRC_FRAMELEN) || nFrames >= MAX_DELAY_FRAMES) { + err = METADATA_INIT_ERROR; + goto bail; + } + + /* Initialize with default setup. */ + FDKmemcpy(&hMetaData->submittedMetaData, &defaultMetaDataSetup, + sizeof(AACENC_MetaData)); + + hMetaData->finalizeMetaData = + 0; /* finalize meta data only while on/off switching, else disabled */ + hMetaData->initializeMetaData = + 0; /* fill up meta data delay line only at a reset otherwise disabled */ + + /* Reset delay lines. */ + if (resetStates || (hMetaData->nAudioDataDelay != -delay) || + (hMetaData->channelMode != channelMode)) { + if (resetStates || (hMetaData->channelMode == MODE_UNKNOWN)) { + /* clear delay buffer */ + FDKmemclear(hMetaData->pAudioDelayBuffer, + hMetaData->maxChannels * MAX_DRC_FRAMELEN * sizeof(INT_PCM)); + } else { + /* if possible, keep static audio channels for seamless channel + * reconfiguration */ + FDK_channelMapDescr mapDescrPrev, mapDescr; + int c, src[2] = {-1, -1}, dst[2] = {-1, -1}; + + if (channelOrder == CH_ORDER_WG4) { + FDK_chMapDescr_init(&mapDescrPrev, FDK_mapInfoTabWg4, + FDK_mapInfoTabLenWg4, 0); + FDK_chMapDescr_init(&mapDescr, FDK_mapInfoTabWg4, + FDK_mapInfoTabLenWg4, 0); + } else { + FDK_chMapDescr_init(&mapDescrPrev, NULL, 0, + (channelOrder == CH_ORDER_MPEG) ? 1 : 0); + FDK_chMapDescr_init(&mapDescr, NULL, 0, + (channelOrder == CH_ORDER_MPEG) ? 1 : 0); + } + + switch (channelMode) { + case MODE_1: + if ((INT)nChannels != 2) { + /* preserve center channel */ + src[0] = FDK_chMapDescr_getMapValue(&mapDescrPrev, 0, + hMetaData->channelMode); + dst[0] = FDK_chMapDescr_getMapValue(&mapDescr, 0, channelMode); + } + break; + case MODE_2: + case MODE_1_2: + case MODE_1_2_1: + case MODE_1_2_2: + case MODE_1_2_2_1: + if (hMetaData->nChannels >= 2) { + /* preserve left/right channel */ + src[0] = FDK_chMapDescr_getMapValue( + &mapDescrPrev, ((hMetaData->channelMode == 2) ? 0 : 1), + hMetaData->channelMode); + src[1] = FDK_chMapDescr_getMapValue( + &mapDescrPrev, ((hMetaData->channelMode == 2) ? 1 : 2), + hMetaData->channelMode); + dst[0] = FDK_chMapDescr_getMapValue( + &mapDescr, ((channelMode == 2) ? 0 : 1), channelMode); + dst[1] = FDK_chMapDescr_getMapValue( + &mapDescr, ((channelMode == 2) ? 1 : 2), channelMode); + } + break; + default:; + } + C_ALLOC_SCRATCH_START(scratch_audioDelayBuffer, INT_PCM, (8)); + FDKmemclear(scratch_audioDelayBuffer, (8) * sizeof(INT_PCM)); + + i = (hMetaData->nChannels > (INT)nChannels) + ? 0 + : hMetaData->nAudioDataDelay - 1; + do { + for (c = 0; c < 2; c++) { + if (src[c] != -1 && dst[c] != -1) { + scratch_audioDelayBuffer[dst[c]] = + hMetaData->pAudioDelayBuffer[i * hMetaData->nChannels + src[c]]; + } + } + FDKmemcpy(&hMetaData->pAudioDelayBuffer[i * nChannels], + scratch_audioDelayBuffer, nChannels * sizeof(INT_PCM)); + i += (hMetaData->nChannels > (INT)nChannels) ? 1 : -1; + } while ((i < hMetaData->nAudioDataDelay) && (i >= 0)); + + C_ALLOC_SCRATCH_END(scratch_audioDelayBuffer, INT_PCM, (8)); + } + FDKmemclear(hMetaData->metaDataBuffer, sizeof(hMetaData->metaDataBuffer)); + hMetaData->metaDataDelayIdx = 0; + hMetaData->initializeMetaData = + 1; /* fill up delay line with first meta data info */ + } else { + /* Enable meta data. */ + if ((hMetaData->metadataMode == 0) && (metadataMode != 0)) { + /* disable meta data in all delay lines */ + for (i = 0; + i < (int)(sizeof(hMetaData->metaDataBuffer) / sizeof(AAC_METADATA)); + i++) { + LoadSubmittedMetadata(&hMetaData->submittedMetaData, nChannels, 0, + &hMetaData->metaDataBuffer[i]); + } + } + + /* Disable meta data.*/ + if ((hMetaData->metadataMode != 0) && (metadataMode == 0)) { + hMetaData->finalizeMetaData = hMetaData->metadataMode; + } + } + + /* Initialize delay. */ + hMetaData->nAudioDataDelay = -delay; + hMetaData->nMetaDataDelay = nFrames; + hMetaData->nChannels = nChannels; + hMetaData->channelMode = channelMode; + hMetaData->metadataMode = metadataMode; + + /* Initialize compressor. */ + if ((metadataMode == 1) || (metadataMode == 2)) { + if (FDK_DRC_Generator_Initialize(hMetaData->hDrcComp, DRC_NONE, DRC_NONE, + frameLength, sampleRate, channelMode, + channelOrder, 1) != 0) { + err = METADATA_INIT_ERROR; + } + } +bail: + return err; +} + +static FDK_METADATA_ERROR ProcessCompressor(AAC_METADATA* pMetadata, + HDRC_COMP hDrcComp, + const INT_PCM* const pSamples, + const UINT samplesBufSize, + const INT nSamples) { + FDK_METADATA_ERROR err = METADATA_OK; + + INT dynrng, compr; + INT dmxGain5, dmxGain2; + DRC_PROFILE profileDrc; + DRC_PROFILE profileComp; + + if ((pMetadata == NULL) || (hDrcComp == NULL)) { + err = METADATA_INVALID_HANDLE; + return err; + } + + profileDrc = convertProfile(pMetadata->mpegDrc.drc_profile); + profileComp = convertProfile(pMetadata->etsiAncData.comp_profile); + + /* first, check if profile is same as last frame + * otherwise, update setup */ + if ((profileDrc != FDK_DRC_Generator_getDrcProfile(hDrcComp)) || + (profileComp != FDK_DRC_Generator_getCompProfile(hDrcComp))) { + FDK_DRC_Generator_setDrcProfile(hDrcComp, profileDrc, profileComp); + } + + /* Sanity check */ + if (profileComp == DRC_NONE) { + pMetadata->etsiAncData.compression_value = 0x80; /* to ensure no external + values will be written + if not configured */ + } + + /* in case of embedding external values, copy this now (limiter may overwrite + * them) */ + dynrng = decodeDynrng(pMetadata->mpegDrc.dyn_rng_ctl[0], + pMetadata->mpegDrc.dyn_rng_sgn[0]); + compr = decodeCompr(pMetadata->etsiAncData.compression_value); + + dmxGain5 = decodeDynrng(pMetadata->etsiAncData.extAncData.dmx_gain_5_idx, + pMetadata->etsiAncData.extAncData.dmx_gain_5_sgn); + dmxGain2 = decodeDynrng(pMetadata->etsiAncData.extAncData.dmx_gain_2_idx, + pMetadata->etsiAncData.extAncData.dmx_gain_2_sgn); + + /* Call compressor */ + if (FDK_DRC_Generator_Calc( + hDrcComp, pSamples, samplesBufSize, + progreflvl2dialnorm(pMetadata->mpegDrc.prog_ref_level), + pMetadata->mpegDrc.drc_TargetRefLevel, + pMetadata->etsiAncData.comp_TargetRefLevel, + dmxTable[pMetadata->centerMixLevel], + dmxTable[pMetadata->surroundMixLevel], + dmxTable[pMetadata->etsiAncData.extAncData.ext_dmix_a_idx], + dmxTable[pMetadata->etsiAncData.extAncData.ext_dmix_b_idx], + pMetadata->etsiAncData.extAncData.ext_lfe_downmix_status + ? dmxLfeTable[pMetadata->etsiAncData.extAncData.ext_dmix_lfe_idx] + : (FIXP_DBL)0, + dmxGain5, dmxGain2, &dynrng, &compr) != 0) { + err = METADATA_ENCODE_ERROR; + goto bail; + } + + /* Write DRC values */ + pMetadata->mpegDrc.drc_band_incr = 0; + encodeDynrng(dynrng, pMetadata->mpegDrc.dyn_rng_ctl, + pMetadata->mpegDrc.dyn_rng_sgn); + pMetadata->etsiAncData.compression_value = encodeCompr(compr); + +bail: + return err; +} + +FDK_METADATA_ERROR FDK_MetadataEnc_Process( + HANDLE_FDK_METADATA_ENCODER hMetaDataEnc, INT_PCM* const pAudioSamples, + const UINT audioSamplesBufSize, const INT nAudioSamples, + const AACENC_MetaData* const pMetadata, + AACENC_EXT_PAYLOAD** ppMetaDataExtPayload, UINT* nMetaDataExtensions, + INT* matrix_mixdown_idx) { + FDK_METADATA_ERROR err = METADATA_OK; + int metaDataDelayWriteIdx, metaDataDelayReadIdx, metadataMode; + + /* Where to write new meta data info */ + metaDataDelayWriteIdx = hMetaDataEnc->metaDataDelayIdx; + + /* How to write the data */ + metadataMode = hMetaDataEnc->metadataMode; + + /* Compensate meta data delay. */ + hMetaDataEnc->metaDataDelayIdx++; + if (hMetaDataEnc->metaDataDelayIdx > hMetaDataEnc->nMetaDataDelay) + hMetaDataEnc->metaDataDelayIdx = 0; + + /* Where to read pending meta data info from. */ + metaDataDelayReadIdx = hMetaDataEnc->metaDataDelayIdx; + + /* Submit new data if available. */ + if (pMetadata != NULL) { + FDKmemcpy(&hMetaDataEnc->submittedMetaData, pMetadata, + sizeof(AACENC_MetaData)); + } + + /* Write one additional frame with default configuration of meta data. Ensure + * defined behaviour on decoder side. */ + if ((hMetaDataEnc->finalizeMetaData != 0) && + (hMetaDataEnc->metadataMode == 0)) { + FDKmemcpy(&hMetaDataEnc->submittedMetaData, &defaultMetaDataSetup, + sizeof(AACENC_MetaData)); + metadataMode = hMetaDataEnc->finalizeMetaData; + hMetaDataEnc->finalizeMetaData = 0; + } + + /* Get last submitted data. */ + if ((err = LoadSubmittedMetadata( + &hMetaDataEnc->submittedMetaData, hMetaDataEnc->nChannels, + metadataMode, + &hMetaDataEnc->metaDataBuffer[metaDataDelayWriteIdx])) != + METADATA_OK) { + goto bail; + } + + /* Calculate compressor if necessary and updata meta data info */ + if ((hMetaDataEnc->metaDataBuffer[metaDataDelayWriteIdx].metadataMode == 1) || + (hMetaDataEnc->metaDataBuffer[metaDataDelayWriteIdx].metadataMode == 2)) { + if ((err = ProcessCompressor( + &hMetaDataEnc->metaDataBuffer[metaDataDelayWriteIdx], + hMetaDataEnc->hDrcComp, pAudioSamples, audioSamplesBufSize, + nAudioSamples)) != METADATA_OK) { + /* Get last submitted data again. */ + LoadSubmittedMetadata( + &hMetaDataEnc->submittedMetaData, hMetaDataEnc->nChannels, + metadataMode, &hMetaDataEnc->metaDataBuffer[metaDataDelayWriteIdx]); + } + } + + /* Fill up delay line with initial meta data info.*/ + if ((hMetaDataEnc->initializeMetaData != 0) && + (hMetaDataEnc->metadataMode != 0)) { + int i; + for (i = 0; + i < (int)(sizeof(hMetaDataEnc->metaDataBuffer) / sizeof(AAC_METADATA)); + i++) { + if (i != metaDataDelayWriteIdx) { + FDKmemcpy(&hMetaDataEnc->metaDataBuffer[i], + &hMetaDataEnc->metaDataBuffer[metaDataDelayWriteIdx], + sizeof(hMetaDataEnc->metaDataBuffer[metaDataDelayWriteIdx])); + } + } + hMetaDataEnc->initializeMetaData = 0; + } + + /* Convert Meta Data side info to bitstream data. */ + FDK_ASSERT(metaDataDelayReadIdx < MAX_DELAY_FRAMES); + if ((err = WriteMetadataPayload( + hMetaDataEnc, + &hMetaDataEnc->metaDataBuffer[metaDataDelayReadIdx])) != + METADATA_OK) { + goto bail; + } + + /* Assign meta data to output */ + *ppMetaDataExtPayload = hMetaDataEnc->exPayload; + *nMetaDataExtensions = hMetaDataEnc->nExtensions; + *matrix_mixdown_idx = hMetaDataEnc->matrix_mixdown_idx; + +bail: + /* Compensate audio delay, reset err status. */ + err = CompensateAudioDelay(hMetaDataEnc, pAudioSamples, audioSamplesBufSize, + nAudioSamples / hMetaDataEnc->nChannels); + + return err; +} + +static FDK_METADATA_ERROR CompensateAudioDelay( + HANDLE_FDK_METADATA_ENCODER hMetaDataEnc, INT_PCM* const pAudioSamples, + const UINT audioSamplesBufSize, const INT nAudioSamples) { + FDK_METADATA_ERROR err = METADATA_OK; + + if (hMetaDataEnc->nAudioDataDelay) { + C_ALLOC_SCRATCH_START(scratch_audioDelayBuffer, INT_PCM, 1024); + + for (int c = 0; c < hMetaDataEnc->nChannels; c++) { + int M = 1024; + INT_PCM* pAudioSamples2 = pAudioSamples + c * audioSamplesBufSize; + int delayIdx = hMetaDataEnc->nAudioDataDelay; + + do { + M = fMin(M, delayIdx); + delayIdx -= M; + + FDKmemcpy(&scratch_audioDelayBuffer[0], + &pAudioSamples2[(nAudioSamples - M)], sizeof(INT_PCM) * M); + FDKmemmove(&pAudioSamples2[M], &pAudioSamples2[0], + sizeof(INT_PCM) * (nAudioSamples - M)); + FDKmemcpy( + &pAudioSamples2[0], + &hMetaDataEnc->pAudioDelayBuffer[delayIdx + + c * hMetaDataEnc->nAudioDataDelay], + sizeof(INT_PCM) * M); + FDKmemcpy( + &hMetaDataEnc->pAudioDelayBuffer[delayIdx + + c * hMetaDataEnc->nAudioDataDelay], + &scratch_audioDelayBuffer[0], sizeof(INT_PCM) * M); + + } while (delayIdx > 0); + } + + C_ALLOC_SCRATCH_END(scratch_audioDelayBuffer, INT_PCM, 1024); + } + + return err; +} + +/*----------------------------------------------------------------------------- + + functionname: WriteMetadataPayload + description: fills anc data and extension payload + returns: Error status + + ------------------------------------------------------------------------------*/ +static FDK_METADATA_ERROR WriteMetadataPayload( + const HANDLE_FDK_METADATA_ENCODER hMetaData, + const AAC_METADATA* const pMetadata) { + FDK_METADATA_ERROR err = METADATA_OK; + + if ((hMetaData == NULL) || (pMetadata == NULL)) { + err = METADATA_INVALID_HANDLE; + goto bail; + } + + hMetaData->nExtensions = 0; + hMetaData->matrix_mixdown_idx = -1; + + if (pMetadata->metadataMode != 0) { + /* AAC-DRC */ + if ((pMetadata->metadataMode == 1) || (pMetadata->metadataMode == 2)) { + hMetaData->exPayload[hMetaData->nExtensions].pData = + hMetaData->drcInfoPayload; + hMetaData->exPayload[hMetaData->nExtensions].dataType = EXT_DYNAMIC_RANGE; + hMetaData->exPayload[hMetaData->nExtensions].associatedChElement = -1; + + hMetaData->exPayload[hMetaData->nExtensions].dataSize = + WriteDynamicRangeInfoPayload( + pMetadata, hMetaData->exPayload[hMetaData->nExtensions].pData); + + hMetaData->nExtensions++; + } /* pMetadata->metadataMode==1 || pMetadata->metadataMode==2 */ + + /* Matrix Mixdown Coefficient in PCE */ + if (pMetadata->WritePCEMixDwnIdx) { + hMetaData->matrix_mixdown_idx = + surmix2matrix_mixdown_idx[pMetadata->surroundMixLevel]; + } + + /* ETSI TS 101 154 (DVB) - MPEG4 ancillary_data() */ + if ((pMetadata->metadataMode == 2) || + (pMetadata->metadataMode == 3)) /* MP4_METADATA_MPEG_ETSI */ + { + hMetaData->exPayload[hMetaData->nExtensions].pData = + hMetaData->drcDsePayload; + hMetaData->exPayload[hMetaData->nExtensions].dataType = EXT_DATA_ELEMENT; + hMetaData->exPayload[hMetaData->nExtensions].associatedChElement = -1; + + hMetaData->exPayload[hMetaData->nExtensions].dataSize = + WriteEtsiAncillaryDataPayload( + pMetadata, hMetaData->exPayload[hMetaData->nExtensions].pData); + + hMetaData->nExtensions++; + } /* metadataMode==2 || metadataMode==3 */ + + } /* metadataMode != 0 */ + +bail: + return err; +} + +static INT WriteDynamicRangeInfoPayload(const AAC_METADATA* const pMetadata, + UCHAR* const pExtensionPayload) { + const INT pce_tag_present = 0; /* yet fixed setting! */ + const INT prog_ref_lev_res_bits = 0; + INT i, drc_num_bands = 1; + + FDK_BITSTREAM bsWriter; + FDKinitBitStream(&bsWriter, pExtensionPayload, 16, 0, BS_WRITER); + + /* dynamic_range_info() */ + FDKwriteBits(&bsWriter, pce_tag_present, 1); /* pce_tag_present */ + if (pce_tag_present) { + FDKwriteBits(&bsWriter, 0x0, 4); /* pce_instance_tag */ + FDKwriteBits(&bsWriter, 0x0, 4); /* drc_tag_reserved_bits */ + } + + /* Exclude channels */ + FDKwriteBits(&bsWriter, (pMetadata->mpegDrc.excluded_chns_present) ? 1 : 0, + 1); /* excluded_chns_present*/ + + /* Multiband DRC */ + FDKwriteBits(&bsWriter, (pMetadata->mpegDrc.drc_bands_present) ? 1 : 0, + 1); /* drc_bands_present */ + if (pMetadata->mpegDrc.drc_bands_present) { + FDKwriteBits(&bsWriter, pMetadata->mpegDrc.drc_band_incr, + 4); /* drc_band_incr */ + FDKwriteBits(&bsWriter, pMetadata->mpegDrc.drc_interpolation_scheme, + 4); /* drc_interpolation_scheme */ + drc_num_bands += pMetadata->mpegDrc.drc_band_incr; + for (i = 0; i < drc_num_bands; i++) { + FDKwriteBits(&bsWriter, pMetadata->mpegDrc.drc_band_top[i], + 8); /* drc_band_top */ + } + } + + /* Program Reference Level */ + FDKwriteBits(&bsWriter, pMetadata->mpegDrc.prog_ref_level_present, + 1); /* prog_ref_level_present */ + if (pMetadata->mpegDrc.prog_ref_level_present) { + FDKwriteBits(&bsWriter, pMetadata->mpegDrc.prog_ref_level, + 7); /* prog_ref_level */ + FDKwriteBits(&bsWriter, prog_ref_lev_res_bits, + 1); /* prog_ref_level_reserved_bits */ + } + + /* DRC Values */ + for (i = 0; i < drc_num_bands; i++) { + FDKwriteBits(&bsWriter, (pMetadata->mpegDrc.dyn_rng_sgn[i]) ? 1 : 0, + 1); /* dyn_rng_sgn[ */ + FDKwriteBits(&bsWriter, pMetadata->mpegDrc.dyn_rng_ctl[i], + 7); /* dyn_rng_ctl */ + } + + /* return number of valid bits in extension payload. */ + return FDKgetValidBits(&bsWriter); +} + +static INT WriteEtsiAncillaryDataPayload(const AAC_METADATA* const pMetadata, + UCHAR* const pExtensionPayload) { + FDK_BITSTREAM bsWriter; + FDKinitBitStream(&bsWriter, pExtensionPayload, 16, 0, BS_WRITER); + + /* ancillary_data_sync */ + FDKwriteBits(&bsWriter, 0xBC, 8); + + /* bs_info */ + FDKwriteBits(&bsWriter, 0x3, 2); /* mpeg_audio_type */ + FDKwriteBits(&bsWriter, pMetadata->dolbySurroundMode, + 2); /* dolby_surround_mode */ + FDKwriteBits(&bsWriter, pMetadata->drcPresentationMode, + 2); /* DRC presentation mode */ + FDKwriteBits(&bsWriter, 0x0, 1); /* stereo_downmix_mode */ + FDKwriteBits(&bsWriter, 0x0, 1); /* reserved */ + + /* ancillary_data_status */ + FDKwriteBits(&bsWriter, 0, 3); /* 3 bit Reserved, set to "0" */ + FDKwriteBits(&bsWriter, (pMetadata->DmxLvl_On) ? 1 : 0, + 1); /* downmixing_levels_MPEG4_status */ + FDKwriteBits(&bsWriter, pMetadata->etsiAncData.extAncDataStatus, + 1); /* ext_anc_data_status */ + FDKwriteBits(&bsWriter, (pMetadata->etsiAncData.compression_on) ? 1 : 0, + 1); /* audio_coding_mode_and_compression status */ + FDKwriteBits(&bsWriter, + (pMetadata->etsiAncData.timecode_coarse_status) ? 1 : 0, + 1); /* coarse_grain_timecode_status */ + FDKwriteBits(&bsWriter, (pMetadata->etsiAncData.timecode_fine_status) ? 1 : 0, + 1); /* fine_grain_timecode_status */ + + /* downmixing_levels_MPEG4_status */ + if (pMetadata->DmxLvl_On) { + FDKwriteBits( + &bsWriter, + encodeDmxLvls(pMetadata->centerMixLevel, pMetadata->surroundMixLevel), + 8); + } + + /* audio_coding_mode_and_compression_status */ + if (pMetadata->etsiAncData.compression_on) { + FDKwriteBits(&bsWriter, 0x01, 8); /* audio coding mode */ + FDKwriteBits(&bsWriter, pMetadata->etsiAncData.compression_value, + 8); /* compression value */ + } + + /* grain-timecode coarse/fine */ + if (pMetadata->etsiAncData.timecode_coarse_status) { + FDKwriteBits(&bsWriter, 0x0, 16); /* not yet supported */ + } + + if (pMetadata->etsiAncData.timecode_fine_status) { + FDKwriteBits(&bsWriter, 0x0, 16); /* not yet supported */ + } + + /* extended ancillary data structure */ + if (pMetadata->etsiAncData.extAncDataStatus) { + /* ext_ancillary_data_status */ + FDKwriteBits(&bsWriter, 0, 1); /* Reserved, set to "0" */ + FDKwriteBits(&bsWriter, + pMetadata->etsiAncData.extAncData.ext_downmix_lvl_status, + 1); /* ext_downmixing_levels_status */ + FDKwriteBits(&bsWriter, + pMetadata->etsiAncData.extAncData.ext_downmix_gain_status, + 1); /* ext_downmixing_global_gains_status */ + FDKwriteBits(&bsWriter, + pMetadata->etsiAncData.extAncData.ext_lfe_downmix_status, + 1); /* ext_downmixing_lfe_level_status" */ + FDKwriteBits(&bsWriter, 0, 4); /* Reserved, set to "0" */ + + /* ext_downmixing_levels */ + if (pMetadata->etsiAncData.extAncData.ext_downmix_lvl_status) { + FDKwriteBits(&bsWriter, pMetadata->etsiAncData.extAncData.ext_dmix_a_idx, + 3); /* dmix_a_idx */ + FDKwriteBits(&bsWriter, pMetadata->etsiAncData.extAncData.ext_dmix_b_idx, + 3); /* dmix_b_idx */ + FDKwriteBits(&bsWriter, 0, 2); /* Reserved, set to "0" */ + } + + /* ext_downmixing_gains */ + if (pMetadata->etsiAncData.extAncData.ext_downmix_gain_status) { + FDKwriteBits(&bsWriter, pMetadata->etsiAncData.extAncData.dmx_gain_5_sgn, + 1); /* dmx_gain_5_sign */ + FDKwriteBits(&bsWriter, pMetadata->etsiAncData.extAncData.dmx_gain_5_idx, + 6); /* dmx_gain_5_idx */ + FDKwriteBits(&bsWriter, 0, 1); /* Reserved, set to "0" */ + FDKwriteBits(&bsWriter, pMetadata->etsiAncData.extAncData.dmx_gain_2_sgn, + 1); /* dmx_gain_2_sign */ + FDKwriteBits(&bsWriter, pMetadata->etsiAncData.extAncData.dmx_gain_2_idx, + 6); /* dmx_gain_2_idx */ + FDKwriteBits(&bsWriter, 0, 1); /* Reserved, set to "0" */ + } + + if (pMetadata->etsiAncData.extAncData.ext_lfe_downmix_status) { + FDKwriteBits(&bsWriter, + pMetadata->etsiAncData.extAncData.ext_dmix_lfe_idx, + 4); /* dmix_lfe_idx */ + FDKwriteBits(&bsWriter, 0, 4); /* Reserved, set to "0" */ + } + } + + return FDKgetValidBits(&bsWriter); +} + +static FDK_METADATA_ERROR LoadSubmittedMetadata( + const AACENC_MetaData* const hMetadata, const INT nChannels, + const INT metadataMode, AAC_METADATA* const pAacMetaData) { + FDK_METADATA_ERROR err = METADATA_OK; + + if (pAacMetaData == NULL) { + err = METADATA_INVALID_HANDLE; + } else { + /* init struct */ + FDKmemclear(pAacMetaData, sizeof(AAC_METADATA)); + + if (hMetadata != NULL) { + /* convert data */ + pAacMetaData->mpegDrc.drc_profile = hMetadata->drc_profile; + pAacMetaData->etsiAncData.comp_profile = hMetadata->comp_profile; + pAacMetaData->mpegDrc.drc_TargetRefLevel = hMetadata->drc_TargetRefLevel; + pAacMetaData->etsiAncData.comp_TargetRefLevel = + hMetadata->comp_TargetRefLevel; + pAacMetaData->mpegDrc.prog_ref_level_present = + hMetadata->prog_ref_level_present; + pAacMetaData->mpegDrc.prog_ref_level = + dialnorm2progreflvl(hMetadata->prog_ref_level); + + pAacMetaData->centerMixLevel = hMetadata->centerMixLevel; + pAacMetaData->surroundMixLevel = hMetadata->surroundMixLevel; + pAacMetaData->WritePCEMixDwnIdx = hMetadata->PCE_mixdown_idx_present; + pAacMetaData->DmxLvl_On = hMetadata->ETSI_DmxLvl_present; + + pAacMetaData->etsiAncData.compression_on = + (hMetadata->comp_profile == AACENC_METADATA_DRC_NOT_PRESENT ? 0 : 1); + + if (pAacMetaData->mpegDrc.drc_profile == + AACENC_METADATA_DRC_NOT_PRESENT) { + pAacMetaData->mpegDrc.drc_profile = + AACENC_METADATA_DRC_NONE; /* MPEG DRC gains are + always present in BS + syntax */ + /* we should give a warning, but ErrorHandler does not support this */ + } + + if (nChannels == 2) { + pAacMetaData->dolbySurroundMode = + hMetadata->dolbySurroundMode; /* dolby_surround_mode */ + } else { + pAacMetaData->dolbySurroundMode = 0; + } + + pAacMetaData->drcPresentationMode = hMetadata->drcPresentationMode; + /* override external values if DVB DRC presentation mode is given */ + if (pAacMetaData->drcPresentationMode == 1) { + pAacMetaData->mpegDrc.drc_TargetRefLevel = + fMax(-(31 << 16), pAacMetaData->mpegDrc.drc_TargetRefLevel); + pAacMetaData->etsiAncData.comp_TargetRefLevel = fMax( + -(20 << 16), + pAacMetaData->etsiAncData.comp_TargetRefLevel); /* implies -23dB */ + } + if (pAacMetaData->drcPresentationMode == 2) { + pAacMetaData->mpegDrc.drc_TargetRefLevel = + fMax(-(23 << 16), pAacMetaData->mpegDrc.drc_TargetRefLevel); + pAacMetaData->etsiAncData.comp_TargetRefLevel = + fMax(-(23 << 16), pAacMetaData->etsiAncData.comp_TargetRefLevel); + } + if (pAacMetaData->etsiAncData.comp_profile == + AACENC_METADATA_DRC_NOT_PRESENT) { + /* DVB defines to revert to Light DRC if heavy is not present */ + if (pAacMetaData->drcPresentationMode != 0) { + /* we exclude the "not indicated" mode as this requires the user to + * define desired levels anyway */ + pAacMetaData->mpegDrc.drc_TargetRefLevel = + fMax(pAacMetaData->etsiAncData.comp_TargetRefLevel, + pAacMetaData->mpegDrc.drc_TargetRefLevel); + } + } + + pAacMetaData->etsiAncData.timecode_coarse_status = + 0; /* not yet supported - attention: Update + GetEstMetadataBytesPerFrame() if enable this! */ + pAacMetaData->etsiAncData.timecode_fine_status = + 0; /* not yet supported - attention: Update + GetEstMetadataBytesPerFrame() if enable this! */ + + /* extended ancillary data */ + pAacMetaData->etsiAncData.extAncDataStatus = + ((hMetadata->ExtMetaData.extAncDataEnable == 1) ? 1 : 0); + + if (pAacMetaData->etsiAncData.extAncDataStatus) { + pAacMetaData->etsiAncData.extAncData.ext_downmix_lvl_status = + (hMetadata->ExtMetaData.extDownmixLevelEnable ? 1 : 0); + pAacMetaData->etsiAncData.extAncData.ext_downmix_gain_status = + (hMetadata->ExtMetaData.dmxGainEnable ? 1 : 0); + pAacMetaData->etsiAncData.extAncData.ext_lfe_downmix_status = + (hMetadata->ExtMetaData.lfeDmxEnable ? 1 : 0); + + pAacMetaData->etsiAncData.extAncData.ext_dmix_a_idx = + hMetadata->ExtMetaData.extDownmixLevel_A; + pAacMetaData->etsiAncData.extAncData.ext_dmix_b_idx = + hMetadata->ExtMetaData.extDownmixLevel_B; + + if (pAacMetaData->etsiAncData.extAncData.ext_downmix_gain_status) { + encodeDynrng(hMetadata->ExtMetaData.dmxGain5, + &(pAacMetaData->etsiAncData.extAncData.dmx_gain_5_idx), + &(pAacMetaData->etsiAncData.extAncData.dmx_gain_5_sgn)); + encodeDynrng(hMetadata->ExtMetaData.dmxGain2, + &(pAacMetaData->etsiAncData.extAncData.dmx_gain_2_idx), + &(pAacMetaData->etsiAncData.extAncData.dmx_gain_2_sgn)); + } else { + encodeDynrng(1 << 16, + &(pAacMetaData->etsiAncData.extAncData.dmx_gain_5_idx), + &(pAacMetaData->etsiAncData.extAncData.dmx_gain_5_sgn)); + encodeDynrng(1 << 16, + &(pAacMetaData->etsiAncData.extAncData.dmx_gain_2_idx), + &(pAacMetaData->etsiAncData.extAncData.dmx_gain_2_sgn)); + } + + if (pAacMetaData->etsiAncData.extAncData.ext_lfe_downmix_status) { + pAacMetaData->etsiAncData.extAncData.ext_dmix_lfe_idx = + hMetadata->ExtMetaData.lfeDmxLevel; + } else { + pAacMetaData->etsiAncData.extAncData.ext_dmix_lfe_idx = + 15; /* -inf dB */ + } + } else { + pAacMetaData->etsiAncData.extAncData.ext_downmix_lvl_status = 0; + pAacMetaData->etsiAncData.extAncData.ext_downmix_gain_status = 0; + pAacMetaData->etsiAncData.extAncData.ext_lfe_downmix_status = 0; + + pAacMetaData->etsiAncData.extAncData.ext_dmix_a_idx = 7; /* -inf dB */ + pAacMetaData->etsiAncData.extAncData.ext_dmix_b_idx = 7; /* -inf dB */ + + encodeDynrng(1 << 16, + &(pAacMetaData->etsiAncData.extAncData.dmx_gain_5_idx), + &(pAacMetaData->etsiAncData.extAncData.dmx_gain_5_sgn)); + encodeDynrng(1 << 16, + &(pAacMetaData->etsiAncData.extAncData.dmx_gain_2_idx), + &(pAacMetaData->etsiAncData.extAncData.dmx_gain_2_sgn)); + + pAacMetaData->etsiAncData.extAncData.ext_dmix_lfe_idx = + 15; /* -inf dB */ + } + + pAacMetaData->metadataMode = metadataMode; + } else { + pAacMetaData->metadataMode = 0; /* there is no configuration available */ + } + } + + return err; +} + +INT FDK_MetadataEnc_GetDelay(HANDLE_FDK_METADATA_ENCODER hMetadataEnc) { + INT delay = 0; + + if (hMetadataEnc != NULL) { + delay = hMetadataEnc->nAudioDataDelay; + } + + return delay; +} diff --git a/fdk-aac/libAACenc/src/metadata_main.h b/fdk-aac/libAACenc/src/metadata_main.h new file mode 100644 index 0000000..d872c77 --- /dev/null +++ b/fdk-aac/libAACenc/src/metadata_main.h @@ -0,0 +1,226 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): V. Bacigalupo + + Description: Metadata Encoder library interface functions + +*******************************************************************************/ + +#ifndef METADATA_MAIN_H +#define METADATA_MAIN_H + +/* Includes ******************************************************************/ +#include "aacenc_lib.h" +#include "aacenc.h" + +/* Defines *******************************************************************/ + +/* Data Types ****************************************************************/ + +typedef enum { + METADATA_OK = 0x0000, /*!< No error happened. All fine. */ + METADATA_INVALID_HANDLE = + 0x0020, /*!< Handle passed to function call was invalid. */ + METADATA_MEMORY_ERROR = 0x0021, /*!< Memory allocation failed. */ + METADATA_INIT_ERROR = 0x0040, /*!< General initialization error. */ + METADATA_ENCODE_ERROR = + 0x0060 /*!< The encoding process was interrupted by an unexpected error. + */ + +} FDK_METADATA_ERROR; + +/** + * Meta Data handle. + */ +typedef struct FDK_METADATA_ENCODER *HANDLE_FDK_METADATA_ENCODER; + +/** + * \brief Open a Meta Data instance. + * + * \param phMetadataEnc A pointer to a Meta Data handle to be allocated. + * Initialized on return. + * \param maxChannels Maximum number of supported audio channels. + * + * \return + * - METADATA_OK, on succes. + * - METADATA_INVALID_HANDLE, METADATA_MEMORY_ERROR, on failure. + */ +FDK_METADATA_ERROR FDK_MetadataEnc_Open( + HANDLE_FDK_METADATA_ENCODER *phMetadataEnc, const UINT maxChannels); + +/** + * \brief Initialize a Meta Data instance. + * + * \param hMetadataEnc Meta Data handle. + * \param resetStates Indication for full reset of all states. + * \param metadataMode Configures meta data output format (0,1,2,3). + * \param audioDelay Delay cause by the audio encoder. + * \param frameLength Number of samples to be processes within one + * frame. + * \param sampleRate Sampling rat in Hz of audio input signal. + * \param nChannels Number of audio input channels. + * \param channelMode Channel configuration which is used by the + * encoder. + * \param channelOrder Channel order of the input data. (WAV, MPEG) + * + * \return + * - METADATA_OK, on succes. + * - METADATA_INVALID_HANDLE, METADATA_INIT_ERROR, on failure. + */ +FDK_METADATA_ERROR FDK_MetadataEnc_Init( + HANDLE_FDK_METADATA_ENCODER hMetadataEnc, const INT resetStates, + const INT metadataMode, const INT audioDelay, const UINT frameLength, + const UINT sampleRate, const UINT nChannels, const CHANNEL_MODE channelMode, + const CHANNEL_ORDER channelOrder); + +/** + * \brief Calculate Meta Data processing. + * + * This function treats all step necessary for meta data processing. + * - Receive new meta data and make usable. + * - Calculate DRC compressor and extract meta data info. + * - Make meta data available for extern use. + * - Apply audio data and meta data delay compensation. + * + * \param hMetadataEnc Meta Data handle. + * \param pAudioSamples Pointer to audio input data. Existing function + * overwrites audio data with delayed audio samples. + * \param nAudioSamples Number of input audio samples to be prcessed. + * \param pMetadata Pointer to Metat Data input. + * \param ppMetaDataExtPayload Pointer to extension payload array. Filled on + * return. + * \param nMetaDataExtensions Pointer to variable to describe number of + * available extension payloads. Filled on return. + * \param matrix_mixdown_idx Pointer to variable for matrix mixdown + * coefficient. Filled on return. + * + * \return + * - METADATA_OK, on succes. + * - METADATA_INVALID_HANDLE, METADATA_ENCODE_ERROR, on failure. + */ +FDK_METADATA_ERROR FDK_MetadataEnc_Process( + HANDLE_FDK_METADATA_ENCODER hMetadataEnc, INT_PCM *const pAudioSamples, + const UINT audioSamplesBufSize, const INT nAudioSamples, + const AACENC_MetaData *const pMetadata, + AACENC_EXT_PAYLOAD **ppMetaDataExtPayload, UINT *nMetaDataExtensions, + INT *matrix_mixdown_idx); + +/** + * \brief Close the Meta Data instance. + * + * Deallocate instance and free whole memory. + * + * \param phMetaData Pointer to the Meta Data handle to be + * deallocated. + * + * \return + * - METADATA_OK, on succes. + * - METADATA_INVALID_HANDLE, on failure. + */ +FDK_METADATA_ERROR FDK_MetadataEnc_Close( + HANDLE_FDK_METADATA_ENCODER *phMetaData); + +/** + * \brief Get Meta Data Encoder delay. + * + * \param hMetadataEnc Meta Data Encoder handle. + * + * \return Delay caused by Meta Data module. + */ +INT FDK_MetadataEnc_GetDelay(HANDLE_FDK_METADATA_ENCODER hMetadataEnc); + +#endif /* METADATA_MAIN_H */ diff --git a/fdk-aac/libAACenc/src/mps_main.cpp b/fdk-aac/libAACenc/src/mps_main.cpp new file mode 100644 index 0000000..1048228 --- /dev/null +++ b/fdk-aac/libAACenc/src/mps_main.cpp @@ -0,0 +1,529 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): Markus Lohwasser + + Description: Mpeg Surround library interface functions + +*******************************************************************************/ + +/* Includes ******************************************************************/ +#include "mps_main.h" +#include "sacenc_lib.h" + +/* Data Types ****************************************************************/ +struct MPS_ENCODER { + HANDLE_MP4SPACE_ENCODER hSacEncoder; + + AUDIO_OBJECT_TYPE audioObjectType; + + FDK_bufDescr inBufDesc; + FDK_bufDescr outBufDesc; + SACENC_InArgs inargs; + SACENC_OutArgs outargs; + + void *pInBuffer[1]; + UINT pInBufferSize[1]; + UINT pInBufferElSize[1]; + UINT pInBufferType[1]; + + void *pOutBuffer[2]; + UINT pOutBufferSize[2]; + UINT pOutBufferElSize[2]; + UINT pOutBufferType[2]; + + UCHAR sacOutBuffer[1024]; /* Worst case memory consumption for ELDv2: 768 + bytes => 6144 bits (Core + SBR + MPS) */ +}; + +struct MPS_CONFIG_TAB { + AUDIO_OBJECT_TYPE audio_object_type; + CHANNEL_MODE channel_mode; + ULONG sbr_ratio; + ULONG sampling_rate; + ULONG bitrate_min; + ULONG bitrate_max; +}; + +/* Constants *****************************************************************/ +static const MPS_CONFIG_TAB mpsConfigTab[] = { + {AOT_ER_AAC_ELD, MODE_212, 0, 16000, 16000, 39999}, + {AOT_ER_AAC_ELD, MODE_212, 0, 22050, 16000, 49999}, + {AOT_ER_AAC_ELD, MODE_212, 0, 24000, 16000, 61999}, + {AOT_ER_AAC_ELD, MODE_212, 0, 32000, 20000, 84999}, + {AOT_ER_AAC_ELD, MODE_212, 0, 44100, 50000, 192000}, + {AOT_ER_AAC_ELD, MODE_212, 0, 48000, 62000, 192000}, + + {AOT_ER_AAC_ELD, MODE_212, 1, 16000, 18000, 31999}, + {AOT_ER_AAC_ELD, MODE_212, 1, 22050, 18000, 31999}, + {AOT_ER_AAC_ELD, MODE_212, 1, 24000, 20000, 64000}, + + {AOT_ER_AAC_ELD, MODE_212, 2, 32000, 18000, 64000}, + {AOT_ER_AAC_ELD, MODE_212, 2, 44100, 21000, 64000}, + {AOT_ER_AAC_ELD, MODE_212, 2, 48000, 26000, 64000} + +}; + +/* Function / Class Declarations *********************************************/ + +/* Function / Class Definition ***********************************************/ +static INT FDK_MpegsEnc_WriteFrameHeader(HANDLE_MPS_ENCODER hMpsEnc, + UCHAR *const pOutputBuffer, + const int outputBufferSize); + +MPS_ENCODER_ERROR FDK_MpegsEnc_Open(HANDLE_MPS_ENCODER *phMpsEnc) { + MPS_ENCODER_ERROR error = MPS_ENCODER_OK; + HANDLE_MPS_ENCODER hMpsEnc = NULL; + + if (phMpsEnc == NULL) { + error = MPS_ENCODER_INVALID_HANDLE; + goto bail; + } + + if (NULL == + (hMpsEnc = (HANDLE_MPS_ENCODER)FDKcalloc(1, sizeof(MPS_ENCODER)))) { + error = MPS_ENCODER_MEMORY_ERROR; + goto bail; + } + FDKmemclear(hMpsEnc, sizeof(MPS_ENCODER)); + + if (SACENC_OK != FDK_sacenc_open(&hMpsEnc->hSacEncoder)) { + error = MPS_ENCODER_MEMORY_ERROR; + goto bail; + } + + /* Return mps encoder instance */ + *phMpsEnc = hMpsEnc; + +bail: + if (error != MPS_ENCODER_OK) { + FDK_MpegsEnc_Close(&hMpsEnc); + } + return error; +} + +MPS_ENCODER_ERROR FDK_MpegsEnc_Close(HANDLE_MPS_ENCODER *phMpsEnc) { + MPS_ENCODER_ERROR error = MPS_ENCODER_OK; + + if (phMpsEnc == NULL) { + error = MPS_ENCODER_INVALID_HANDLE; + goto bail; + } + + if (*phMpsEnc != NULL) { + FDK_sacenc_close(&(*phMpsEnc)->hSacEncoder); + FDKfree(*phMpsEnc); + *phMpsEnc = NULL; + } +bail: + return error; +} + +MPS_ENCODER_ERROR FDK_MpegsEnc_Init(HANDLE_MPS_ENCODER hMpsEnc, + const AUDIO_OBJECT_TYPE audioObjectType, + const UINT samplingrate, const UINT bitrate, + const UINT sbrRatio, const UINT framelength, + const UINT inputBufferSizePerChannel, + const UINT coreCoderDelay) { + MPS_ENCODER_ERROR error = MPS_ENCODER_OK; + const UINT fs_low = 27713; /* low MPS sampling frequencies */ + const UINT fs_high = 55426; /* high MPS sampling frequencies */ + UINT nTimeSlots = 0, nQmfBandsLd = 0; + + if (hMpsEnc == NULL) { + error = MPS_ENCODER_INVALID_HANDLE; + goto bail; + } + + /* Combine MPS with SBR only if the number of QMF band fits together.*/ + switch (sbrRatio) { + case 1: /* downsampled sbr - 32 QMF bands required */ + if (!(samplingrate < fs_low)) { + error = MPS_ENCODER_INIT_ERROR; + goto bail; + } + break; + case 2: /* dualrate - 64 QMF bands required */ + if (!((samplingrate >= fs_low) && (samplingrate < fs_high))) { + error = MPS_ENCODER_INIT_ERROR; + goto bail; + } + break; + case 0: + default:; /* time interface - no samplingrate restriction */ + } + + /* 32 QMF-Bands ( fs < 27713 ) + * 64 QMF-Bands ( 27713 >= fs <= 55426 ) + * 128 QMF-Bands ( fs > 55426 ) + */ + nQmfBandsLd = + (samplingrate < fs_low) ? 5 : ((samplingrate > fs_high) ? 7 : 6); + nTimeSlots = framelength >> nQmfBandsLd; + + /* check if number of qmf bands is usable for given framelength */ + if (framelength != (nTimeSlots << nQmfBandsLd)) { + error = MPS_ENCODER_INIT_ERROR; + goto bail; + } + + /* is given bitrate intended to be supported */ + if ((INT)bitrate != FDK_MpegsEnc_GetClosestBitRate(audioObjectType, MODE_212, + samplingrate, sbrRatio, + bitrate)) { + error = MPS_ENCODER_INIT_ERROR; + goto bail; + } + + /* init SAC library */ + switch (audioObjectType) { + case AOT_ER_AAC_ELD: { + const UINT noInterFrameCoding = 0; + + if ((SACENC_OK != + FDK_sacenc_setParam(hMpsEnc->hSacEncoder, SACENC_LOWDELAY, + (noInterFrameCoding == 1) ? 1 : 2)) || + (SACENC_OK != FDK_sacenc_setParam(hMpsEnc->hSacEncoder, + SACENC_ENC_MODE, SACENC_212)) || + (SACENC_OK != FDK_sacenc_setParam(hMpsEnc->hSacEncoder, + SACENC_SAMPLERATE, samplingrate)) || + (SACENC_OK != FDK_sacenc_setParam(hMpsEnc->hSacEncoder, + SACENC_FRAME_TIME_SLOTS, + nTimeSlots)) || + (SACENC_OK != FDK_sacenc_setParam(hMpsEnc->hSacEncoder, + SACENC_PARAM_BANDS, + SACENC_BANDS_15)) || + (SACENC_OK != + FDK_sacenc_setParam(hMpsEnc->hSacEncoder, SACENC_TIME_DOM_DMX, 2)) || + (SACENC_OK != + FDK_sacenc_setParam(hMpsEnc->hSacEncoder, SACENC_COARSE_QUANT, 0)) || + (SACENC_OK != FDK_sacenc_setParam(hMpsEnc->hSacEncoder, + SACENC_QUANT_MODE, + SACENC_QUANTMODE_FINE)) || + (SACENC_OK != FDK_sacenc_setParam(hMpsEnc->hSacEncoder, + SACENC_TIME_ALIGNMENT, 0)) || + (SACENC_OK != FDK_sacenc_setParam(hMpsEnc->hSacEncoder, + SACENC_INDEPENDENCY_FACTOR, 20))) { + error = MPS_ENCODER_INIT_ERROR; + goto bail; + } + break; + } + default: + error = MPS_ENCODER_INIT_ERROR; + goto bail; + } + + if (SACENC_OK != FDK_sacenc_init(hMpsEnc->hSacEncoder, coreCoderDelay)) { + error = MPS_ENCODER_INIT_ERROR; + } + + hMpsEnc->audioObjectType = audioObjectType; + + hMpsEnc->inBufDesc.ppBase = (void **)&hMpsEnc->pInBuffer; + hMpsEnc->inBufDesc.pBufSize = hMpsEnc->pInBufferSize; + hMpsEnc->inBufDesc.pEleSize = hMpsEnc->pInBufferElSize; + hMpsEnc->inBufDesc.pBufType = hMpsEnc->pInBufferType; + hMpsEnc->inBufDesc.numBufs = 1; + + hMpsEnc->outBufDesc.ppBase = (void **)&hMpsEnc->pOutBuffer; + hMpsEnc->outBufDesc.pBufSize = hMpsEnc->pOutBufferSize; + hMpsEnc->outBufDesc.pEleSize = hMpsEnc->pOutBufferElSize; + hMpsEnc->outBufDesc.pBufType = hMpsEnc->pOutBufferType; + hMpsEnc->outBufDesc.numBufs = 2; + + hMpsEnc->pInBuffer[0] = NULL; + hMpsEnc->pInBufferSize[0] = 0; + hMpsEnc->pInBufferElSize[0] = sizeof(INT_PCM); + hMpsEnc->pInBufferType[0] = (FDK_BUF_TYPE_INPUT | FDK_BUF_TYPE_PCM_DATA); + + hMpsEnc->pOutBuffer[0] = NULL; + hMpsEnc->pOutBufferSize[0] = 0; + hMpsEnc->pOutBufferElSize[0] = sizeof(INT_PCM); + hMpsEnc->pOutBufferType[0] = (FDK_BUF_TYPE_OUTPUT | FDK_BUF_TYPE_PCM_DATA); + + hMpsEnc->pOutBuffer[1] = NULL; + hMpsEnc->pOutBufferSize[1] = 0; + hMpsEnc->pOutBufferElSize[1] = sizeof(UCHAR); + hMpsEnc->pOutBufferType[1] = (FDK_BUF_TYPE_OUTPUT | FDK_BUF_TYPE_BS_DATA); + + hMpsEnc->inargs.isInputInterleaved = 0; + hMpsEnc->inargs.inputBufferSizePerChannel = inputBufferSizePerChannel; + +bail: + return error; +} + +MPS_ENCODER_ERROR FDK_MpegsEnc_Process(HANDLE_MPS_ENCODER hMpsEnc, + INT_PCM *const pAudioSamples, + const INT nAudioSamples, + AACENC_EXT_PAYLOAD *pMpsExtPayload) { + MPS_ENCODER_ERROR error = MPS_ENCODER_OK; + + if (hMpsEnc == NULL) { + error = MPS_ENCODER_INVALID_HANDLE; + } else { + int sacHeaderFlag = 1; + int sacOutBufferOffset = 0; + + /* In case of eld the ssc is explicit and doesn't need to be inband */ + if (hMpsEnc->audioObjectType == AOT_ER_AAC_ELD) { + sacHeaderFlag = 0; + } + + /* 4 bits nibble after extension type */ + hMpsEnc->sacOutBuffer[0] = (sacHeaderFlag == 0) ? 0x3 : 0x7; + sacOutBufferOffset += 1; + + if (sacHeaderFlag) { + sacOutBufferOffset += FDK_MpegsEnc_WriteFrameHeader( + hMpsEnc, &hMpsEnc->sacOutBuffer[sacOutBufferOffset], + sizeof(hMpsEnc->sacOutBuffer) - sacOutBufferOffset); + } + + /* Register input and output buffer. */ + hMpsEnc->pInBuffer[0] = (void *)pAudioSamples; + hMpsEnc->inargs.nInputSamples = nAudioSamples; + + hMpsEnc->pOutBuffer[0] = (void *)pAudioSamples; + hMpsEnc->pOutBufferSize[0] = sizeof(INT_PCM) * nAudioSamples / 2; + + hMpsEnc->pOutBuffer[1] = (void *)&hMpsEnc->sacOutBuffer[sacOutBufferOffset]; + hMpsEnc->pOutBufferSize[1] = + sizeof(hMpsEnc->sacOutBuffer) - sacOutBufferOffset; + + /* encode SAC frame */ + if (SACENC_OK != FDK_sacenc_encode(hMpsEnc->hSacEncoder, + &hMpsEnc->inBufDesc, + &hMpsEnc->outBufDesc, &hMpsEnc->inargs, + &hMpsEnc->outargs)) { + error = MPS_ENCODER_ENCODE_ERROR; + goto bail; + } + + /* export MPS payload */ + pMpsExtPayload->pData = (UCHAR *)hMpsEnc->sacOutBuffer; + pMpsExtPayload->dataSize = + hMpsEnc->outargs.nOutputBits + 8 * (sacOutBufferOffset - 1); + pMpsExtPayload->dataType = EXT_LDSAC_DATA; + pMpsExtPayload->associatedChElement = -1; + } + +bail: + return error; +} + +INT FDK_MpegsEnc_WriteSpatialSpecificConfig(HANDLE_MPS_ENCODER hMpsEnc, + HANDLE_FDK_BITSTREAM hBs) { + INT sscBits = 0; + + if (NULL != hMpsEnc) { + MP4SPACEENC_INFO mp4SpaceEncoderInfo; + FDK_sacenc_getInfo(hMpsEnc->hSacEncoder, &mp4SpaceEncoderInfo); + + if (hBs != NULL) { + int i; + int writtenBits = 0; + for (i = 0; i<mp4SpaceEncoderInfo.pSscBuf->nSscSizeBits>> 3; i++) { + FDKwriteBits(hBs, mp4SpaceEncoderInfo.pSscBuf->pSsc[i], 8); + writtenBits += 8; + } + FDKwriteBits(hBs, mp4SpaceEncoderInfo.pSscBuf->pSsc[i], + mp4SpaceEncoderInfo.pSscBuf->nSscSizeBits - writtenBits); + } /* hBS */ + + sscBits = mp4SpaceEncoderInfo.pSscBuf->nSscSizeBits; + + } /* valid hMpsEnc */ + + return sscBits; +} + +static INT FDK_MpegsEnc_WriteFrameHeader(HANDLE_MPS_ENCODER hMpsEnc, + UCHAR *const pOutputBuffer, + const int outputBufferSize) { + const int sacTimeAlignFlag = 0; + + /* Initialize variables */ + int numBits = 0; + + if ((NULL != hMpsEnc) && (NULL != pOutputBuffer)) { + UINT alignAnchor, cnt; + FDK_BITSTREAM Bs; + FDKinitBitStream(&Bs, pOutputBuffer, outputBufferSize, 0, BS_WRITER); + + /* Calculate SSC length information */ + cnt = (FDK_MpegsEnc_WriteSpatialSpecificConfig(hMpsEnc, NULL) + 7) >> 3; + + /* Write SSC */ + FDKwriteBits(&Bs, sacTimeAlignFlag, 1); + + if (cnt < 127) { + FDKwriteBits(&Bs, cnt, 7); + } else { + FDKwriteBits(&Bs, 127, 7); + FDKwriteBits(&Bs, cnt - 127, 16); + } + + alignAnchor = FDKgetValidBits(&Bs); + FDK_MpegsEnc_WriteSpatialSpecificConfig(hMpsEnc, &Bs); + FDKbyteAlign(&Bs, alignAnchor); /* bsFillBits */ + + if (sacTimeAlignFlag) { + FDK_ASSERT(1); /* time alignment not supported */ + } + + numBits = FDKgetValidBits(&Bs); + } /* valid handle */ + + return ((numBits + 7) >> 3); +} + +INT FDK_MpegsEnc_GetClosestBitRate(const AUDIO_OBJECT_TYPE audioObjectType, + const CHANNEL_MODE channelMode, + const UINT samplingrate, const UINT sbrRatio, + const UINT bitrate) { + unsigned int i; + int targetBitrate = -1; + + for (i = 0; i < sizeof(mpsConfigTab) / sizeof(MPS_CONFIG_TAB); i++) { + if ((mpsConfigTab[i].audio_object_type == audioObjectType) && + (mpsConfigTab[i].channel_mode == channelMode) && + (mpsConfigTab[i].sbr_ratio == sbrRatio) && + (mpsConfigTab[i].sampling_rate == samplingrate)) { + targetBitrate = fMin(fMax(bitrate, mpsConfigTab[i].bitrate_min), + mpsConfigTab[i].bitrate_max); + } + } + + return targetBitrate; +} + +INT FDK_MpegsEnc_GetDelay(HANDLE_MPS_ENCODER hMpsEnc) { + INT delay = 0; + + if (NULL != hMpsEnc) { + MP4SPACEENC_INFO mp4SpaceEncoderInfo; + FDK_sacenc_getInfo(hMpsEnc->hSacEncoder, &mp4SpaceEncoderInfo); + delay = mp4SpaceEncoderInfo.nCodecDelay; + } + + return delay; +} + +INT FDK_MpegsEnc_GetDecDelay(HANDLE_MPS_ENCODER hMpsEnc) { + INT delay = 0; + + if (NULL != hMpsEnc) { + MP4SPACEENC_INFO mp4SpaceEncoderInfo; + FDK_sacenc_getInfo(hMpsEnc->hSacEncoder, &mp4SpaceEncoderInfo); + delay = mp4SpaceEncoderInfo.nDecoderDelay; + } + + return delay; +} + +MPS_ENCODER_ERROR FDK_MpegsEnc_GetLibInfo(LIB_INFO *info) { + MPS_ENCODER_ERROR error = MPS_ENCODER_OK; + + if (NULL == info) { + error = MPS_ENCODER_INVALID_HANDLE; + } else if (SACENC_OK != FDK_sacenc_getLibInfo(info)) { + error = MPS_ENCODER_INIT_ERROR; + } + + return error; +} diff --git a/fdk-aac/libAACenc/src/mps_main.h b/fdk-aac/libAACenc/src/mps_main.h new file mode 100644 index 0000000..f56678a --- /dev/null +++ b/fdk-aac/libAACenc/src/mps_main.h @@ -0,0 +1,270 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): Markus Lohwasser + + Description: Mpeg Surround library interface functions + +*******************************************************************************/ + +#ifndef MPS_MAIN_H +#define MPS_MAIN_H + +/* Includes ******************************************************************/ +#include "aacenc.h" +#include "FDK_audio.h" +#include "machine_type.h" + +/* Defines *******************************************************************/ +typedef enum { + MPS_ENCODER_OK = 0x0000, /*!< No error happened. All fine. */ + MPS_ENCODER_INVALID_HANDLE = + 0x0020, /*!< Handle passed to function call was invalid. */ + MPS_ENCODER_MEMORY_ERROR = 0x0021, /*!< Memory allocation failed. */ + MPS_ENCODER_INIT_ERROR = 0x0040, /*!< General initialization error. */ + MPS_ENCODER_ENCODE_ERROR = + 0x0060 /*!< The encoding process was interrupted by an unexpected error. + */ + +} MPS_ENCODER_ERROR; + +/* Data Types ****************************************************************/ + +/** + * MPEG Surround Encoder interface handle. + */ +typedef struct MPS_ENCODER MPS_ENCODER, *HANDLE_MPS_ENCODER; + +/* Function / Class Declarations *********************************************/ + +/** + * \brief Open a Mpeg Surround Encoder instance. + * + * \phMpsEnc A pointer to a MPS handle to be allocated. + * Initialized on return. + * + * \return + * - MPS_ENCODER_OK, on succes. + * - MPS_ENCODER_INVALID_HANDLE, MPS_ENCODER_MEMORY_ERROR, on failure. + */ +MPS_ENCODER_ERROR FDK_MpegsEnc_Open(HANDLE_MPS_ENCODER *phMpsEnc); + +/** + * \brief Close the Mpeg Surround Encoder instance. + * + * Deallocate instance and free whole memory. + * + * \param phMpsEnc Pointer to the MPS handle to be deallocated. + * + * \return + * - MPS_ENCODER_OK, on succes. + * - MPS_ENCODER_INVALID_HANDLE, on failure. + */ +MPS_ENCODER_ERROR FDK_MpegsEnc_Close(HANDLE_MPS_ENCODER *phMpsEnc); + +/** + * \brief Initialize a Mpeg Surround Encoder instance. + * + * \param hMpsEnc MPS Encoder handle. + * \param audioObjectType Audio object type. + * \param samplingrate Sampling rate in Hz of audio input signal. + * \param bitrate Encder target bitrate. + * \param sbrRatio SBR sampling rate ratio. + * \param framelength Number of samples to be processes within one + * frame. + * \param inputBufferSizePerChannel Size of input buffer per channel. + * \param coreCoderDelay Core coder delay. + * + * \return + * - MPS_ENCODER_OK, on succes. + * - MPS_ENCODER_INVALID_HANDLE, MPS_ENCODER_ENCODE_ERROR, on failure. + */ +MPS_ENCODER_ERROR FDK_MpegsEnc_Init(HANDLE_MPS_ENCODER hMpsEnc, + const AUDIO_OBJECT_TYPE audioObjectType, + const UINT samplingrate, const UINT bitrate, + const UINT sbrRatio, const UINT framelength, + const UINT inputBufferSizePerChannel, + const UINT coreCoderDelay); + +/** + * \brief Calculate Mpeg Surround processing. + * + * This fuction applies the MPS processing. The MPS side info will be written to + * extension payload. The input audio data will be overwritten by the calculated + * downmix. + * + * \param hMpsEnc MPS Encoder handle. + * \param pAudioSamples Pointer to audio input/output data. + * \param nAudioSamples Number of input audio samples to be prcessed. + * \param pMpsExtPayload Pointer to extension payload to be filled on + * return. + * + * \return + * - MPS_ENCODER_OK, on succes. + * - MPS_ENCODER_INVALID_HANDLE, MPS_ENCODER_ENCODE_ERROR, on failure. + */ +MPS_ENCODER_ERROR FDK_MpegsEnc_Process(HANDLE_MPS_ENCODER hMpsEnc, + INT_PCM *const pAudioSamples, + const INT nAudioSamples, + AACENC_EXT_PAYLOAD *pMpsExtPayload); + +/** + * \brief Write Spatial Specific Config. + * + * This function can be called via call back from the transport library to write + * the Spatial Specific Config to given bitstream buffer. + * + * \param hMpsEnc MPS Encoder handle. + * \param hBs Bitstream buffer handle. + * + * \return Number of written bits. + */ +INT FDK_MpegsEnc_WriteSpatialSpecificConfig(HANDLE_MPS_ENCODER hMpsEnc, + HANDLE_FDK_BITSTREAM hBs); + +/** + * \brief Get closest valid bitrate supported by given config. + * + * \param audioObjectType Audio object type. + * \param channelMode Encoder channel mode. + * \param samplingrate Sampling rate in Hz of audio input signal. + * \param sbrRatio SBR sampling rate ratio. + * \param bitrate The desired target bitrate. + * + * \return Closest valid bitrate to given bitrate.. + */ +INT FDK_MpegsEnc_GetClosestBitRate(const AUDIO_OBJECT_TYPE audioObjectType, + const CHANNEL_MODE channelMode, + const UINT samplingrate, const UINT sbrRatio, + const UINT bitrate); + +/** + * \brief Get codec delay. + * + * This function returns delay of the whole en-/decoded signal, including + * corecoder delay. + * + * \param hMpsEnc MPS Encoder handle. + * + * \return Codec delay in samples. + */ +INT FDK_MpegsEnc_GetDelay(HANDLE_MPS_ENCODER hMpsEnc); + +/** + * \brief Get Mpeg Surround Decoder delay. + * + * This function returns delay of the Mpeg Surround decoder. + * + * \param hMpsEnc MPS Encoder handle. + * + * \return Mpeg Surround Decoder delay in samples. + */ +INT FDK_MpegsEnc_GetDecDelay(HANDLE_MPS_ENCODER hMpsEnc); + +/** + * \brief Get information about encoder library build. + * + * Fill a given LIB_INFO structure with library version information. + * + * \param info Pointer to an allocated LIB_INFO struct. + * + * \return + * - MPS_ENCODER_OK, on succes. + * - MPS_ENCODER_INVALID_HANDLE, MPS_ENCODER_INIT_ERROR, on failure. + */ +MPS_ENCODER_ERROR FDK_MpegsEnc_GetLibInfo(LIB_INFO *info); + +#endif /* MPS_MAIN_H */ diff --git a/fdk-aac/libAACenc/src/ms_stereo.cpp b/fdk-aac/libAACenc/src/ms_stereo.cpp new file mode 100644 index 0000000..6a121b2 --- /dev/null +++ b/fdk-aac/libAACenc/src/ms_stereo.cpp @@ -0,0 +1,295 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M.Werner + + Description: MS stereo processing + +*******************************************************************************/ + +#include "ms_stereo.h" + +#include "psy_const.h" + +/* static const float scaleMinThres = 1.0f; */ /* 0.75f for 3db boost */ + +void FDKaacEnc_MsStereoProcessing(PSY_DATA *RESTRICT psyData[(2)], + PSY_OUT_CHANNEL *psyOutChannel[2], + const INT *isBook, INT *msDigest, /* output */ + INT *msMask, /* output */ + const INT allowMS, const INT sfbCnt, + const INT sfbPerGroup, + const INT maxSfbPerGroup, + const INT *sfbOffset) { + FIXP_DBL *sfbEnergyLeft = + psyData[0]->sfbEnergy.Long; /* modified where msMask==1 */ + FIXP_DBL *sfbEnergyRight = + psyData[1]->sfbEnergy.Long; /* modified where msMask==1 */ + const FIXP_DBL *sfbEnergyMid = psyData[0]->sfbEnergyMS.Long; + const FIXP_DBL *sfbEnergySide = psyData[1]->sfbEnergyMS.Long; + FIXP_DBL *sfbThresholdLeft = + psyData[0]->sfbThreshold.Long; /* modified where msMask==1 */ + FIXP_DBL *sfbThresholdRight = + psyData[1]->sfbThreshold.Long; /* modified where msMask==1 */ + + FIXP_DBL *sfbSpreadEnLeft = psyData[0]->sfbSpreadEnergy.Long; + FIXP_DBL *sfbSpreadEnRight = psyData[1]->sfbSpreadEnergy.Long; + + FIXP_DBL *sfbEnergyLeftLdData = + psyOutChannel[0]->sfbEnergyLdData; /* modified where msMask==1 */ + FIXP_DBL *sfbEnergyRightLdData = + psyOutChannel[1]->sfbEnergyLdData; /* modified where msMask==1 */ + FIXP_DBL *sfbEnergyMidLdData = psyData[0]->sfbEnergyMSLdData; + FIXP_DBL *sfbEnergySideLdData = psyData[1]->sfbEnergyMSLdData; + FIXP_DBL *sfbThresholdLeftLdData = + psyOutChannel[0]->sfbThresholdLdData; /* modified where msMask==1 */ + FIXP_DBL *sfbThresholdRightLdData = + psyOutChannel[1]->sfbThresholdLdData; /* modified where msMask==1 */ + + FIXP_DBL *mdctSpectrumLeft = + psyData[0]->mdctSpectrum; /* modified where msMask==1 */ + FIXP_DBL *mdctSpectrumRight = + psyData[1]->mdctSpectrum; /* modified where msMask==1 */ + + INT sfb, sfboffs, j; /* loop counters */ + FIXP_DBL pnlrLdData, pnmsLdData; + FIXP_DBL minThresholdLdData; + FIXP_DBL minThreshold; + INT useMS; + + INT msMaskTrueSomewhere = 0; /* to determine msDigest */ + INT numMsMaskFalse = + 0; /* number of non-intensity bands where L/R coding is used */ + + for (sfb = 0; sfb < sfbCnt; sfb += sfbPerGroup) { + for (sfboffs = 0; sfboffs < maxSfbPerGroup; sfboffs++) { + if ((isBook == NULL) ? 1 : (isBook[sfb + sfboffs] == 0)) { + FIXP_DBL tmp; + + /* + minThreshold=min(sfbThresholdLeft[sfb+sfboffs], + sfbThresholdRight[sfb+sfboffs])*scaleMinThres; pnlr = + (sfbThresholdLeft[sfb+sfboffs]/ + max(sfbEnergyLeft[sfb+sfboffs],sfbThresholdLeft[sfb+sfboffs]))* + (sfbThresholdRight[sfb+sfboffs]/ + max(sfbEnergyRight[sfb+sfboffs],sfbThresholdRight[sfb+sfboffs])); + pnms = + (minThreshold/max(sfbEnergyMid[sfb+sfboffs],minThreshold))* + (minThreshold/max(sfbEnergySide[sfb+sfboffs],minThreshold)); + useMS = (pnms > pnlr); + */ + + /* we assume that scaleMinThres == 1.0f and we can drop it */ + minThresholdLdData = fixMin(sfbThresholdLeftLdData[sfb + sfboffs], + sfbThresholdRightLdData[sfb + sfboffs]); + + /* pnlrLdData = sfbThresholdLeftLdData[sfb+sfboffs] - + max(sfbEnergyLeftLdData[sfb+sfboffs], + sfbThresholdLeftLdData[sfb+sfboffs]) + + sfbThresholdRightLdData[sfb+sfboffs] - + max(sfbEnergyRightLdData[sfb+sfboffs], + sfbThresholdRightLdData[sfb+sfboffs]); */ + tmp = fixMax(sfbEnergyLeftLdData[sfb + sfboffs], + sfbThresholdLeftLdData[sfb + sfboffs]); + pnlrLdData = (sfbThresholdLeftLdData[sfb + sfboffs] >> 1) - (tmp >> 1); + pnlrLdData = pnlrLdData + (sfbThresholdRightLdData[sfb + sfboffs] >> 1); + tmp = fixMax(sfbEnergyRightLdData[sfb + sfboffs], + sfbThresholdRightLdData[sfb + sfboffs]); + pnlrLdData = pnlrLdData - (tmp >> 1); + + /* pnmsLdData = minThresholdLdData - + max(sfbEnergyMidLdData[sfb+sfboffs], minThresholdLdData) + + minThresholdLdData - max(sfbEnergySideLdData[sfb+sfboffs], + minThresholdLdData); */ + tmp = fixMax(sfbEnergyMidLdData[sfb + sfboffs], minThresholdLdData); + pnmsLdData = minThresholdLdData - (tmp >> 1); + tmp = fixMax(sfbEnergySideLdData[sfb + sfboffs], minThresholdLdData); + pnmsLdData = pnmsLdData - (tmp >> 1); + useMS = ((allowMS != 0) && (pnmsLdData > pnlrLdData)) ? 1 : 0; + + if (useMS) { + msMask[sfb + sfboffs] = 1; + msMaskTrueSomewhere = 1; + for (j = sfbOffset[sfb + sfboffs]; j < sfbOffset[sfb + sfboffs + 1]; + j++) { + FIXP_DBL specL, specR; + specL = mdctSpectrumLeft[j] >> 1; + specR = mdctSpectrumRight[j] >> 1; + mdctSpectrumLeft[j] = specL + specR; + mdctSpectrumRight[j] = specL - specR; + } + minThreshold = fixMin(sfbThresholdLeft[sfb + sfboffs], + sfbThresholdRight[sfb + sfboffs]); + sfbThresholdLeft[sfb + sfboffs] = sfbThresholdRight[sfb + sfboffs] = + minThreshold; + sfbThresholdLeftLdData[sfb + sfboffs] = + sfbThresholdRightLdData[sfb + sfboffs] = minThresholdLdData; + sfbEnergyLeft[sfb + sfboffs] = sfbEnergyMid[sfb + sfboffs]; + sfbEnergyRight[sfb + sfboffs] = sfbEnergySide[sfb + sfboffs]; + sfbEnergyLeftLdData[sfb + sfboffs] = + sfbEnergyMidLdData[sfb + sfboffs]; + sfbEnergyRightLdData[sfb + sfboffs] = + sfbEnergySideLdData[sfb + sfboffs]; + + sfbSpreadEnLeft[sfb + sfboffs] = sfbSpreadEnRight[sfb + sfboffs] = + fixMin(sfbSpreadEnLeft[sfb + sfboffs], + sfbSpreadEnRight[sfb + sfboffs]) >> + 1; + + } else { + msMask[sfb + sfboffs] = 0; + numMsMaskFalse++; + } /* useMS */ + } /* isBook */ + else { + /* keep mDigest from IS module */ + if (msMask[sfb + sfboffs]) { + msMaskTrueSomewhere = 1; + } + /* prohibit MS_MASK_ALL in combination with IS */ + numMsMaskFalse = 9; + } /* isBook */ + } /* sfboffs */ + } /* sfb */ + + if (msMaskTrueSomewhere == 1) { + if ((numMsMaskFalse == 0) || + ((numMsMaskFalse < maxSfbPerGroup) && (numMsMaskFalse < 9))) { + *msDigest = SI_MS_MASK_ALL; + /* loop through M/S bands; if msMask==0, set it to 1 and apply M/S */ + for (sfb = 0; sfb < sfbCnt; sfb += sfbPerGroup) { + for (sfboffs = 0; sfboffs < maxSfbPerGroup; sfboffs++) { + if (((isBook == NULL) ? 1 : (isBook[sfb + sfboffs] == 0)) && + (msMask[sfb + sfboffs] == 0)) { + msMask[sfb + sfboffs] = 1; + /* apply M/S coding */ + for (j = sfbOffset[sfb + sfboffs]; j < sfbOffset[sfb + sfboffs + 1]; + j++) { + FIXP_DBL specL, specR; + specL = mdctSpectrumLeft[j] >> 1; + specR = mdctSpectrumRight[j] >> 1; + mdctSpectrumLeft[j] = specL + specR; + mdctSpectrumRight[j] = specL - specR; + } + minThreshold = fixMin(sfbThresholdLeft[sfb + sfboffs], + sfbThresholdRight[sfb + sfboffs]); + sfbThresholdLeft[sfb + sfboffs] = sfbThresholdRight[sfb + sfboffs] = + minThreshold; + minThresholdLdData = fixMin(sfbThresholdLeftLdData[sfb + sfboffs], + sfbThresholdRightLdData[sfb + sfboffs]); + sfbThresholdLeftLdData[sfb + sfboffs] = + sfbThresholdRightLdData[sfb + sfboffs] = minThresholdLdData; + sfbEnergyLeft[sfb + sfboffs] = sfbEnergyMid[sfb + sfboffs]; + sfbEnergyRight[sfb + sfboffs] = sfbEnergySide[sfb + sfboffs]; + sfbEnergyLeftLdData[sfb + sfboffs] = + sfbEnergyMidLdData[sfb + sfboffs]; + sfbEnergyRightLdData[sfb + sfboffs] = + sfbEnergySideLdData[sfb + sfboffs]; + + sfbSpreadEnLeft[sfb + sfboffs] = sfbSpreadEnRight[sfb + sfboffs] = + fixMin(sfbSpreadEnLeft[sfb + sfboffs], + sfbSpreadEnRight[sfb + sfboffs]) >> + 1; + } + } + } + } else { + *msDigest = SI_MS_MASK_SOME; + } + } else { + *msDigest = SI_MS_MASK_NONE; + } +} diff --git a/fdk-aac/libAACenc/src/ms_stereo.h b/fdk-aac/libAACenc/src/ms_stereo.h new file mode 100644 index 0000000..a202307 --- /dev/null +++ b/fdk-aac/libAACenc/src/ms_stereo.h @@ -0,0 +1,117 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M.Werner + + Description: MS stereo processing + +*******************************************************************************/ + +#ifndef MS_STEREO_H +#define MS_STEREO_H + +#include "interface.h" + +void FDKaacEnc_MsStereoProcessing(PSY_DATA *RESTRICT psyData[(2)], + PSY_OUT_CHANNEL *psyOutChannel[2], + const INT *isBook, INT *msDigest, /* output */ + INT *msMask, /* output */ + const INT allowMS, const INT sfbCnt, + const INT sfbPerGroup, + const INT maxSfbPerGroup, + const INT *sfbOffset); + +#endif /* MS_STEREO_H */ diff --git a/fdk-aac/libAACenc/src/noisedet.cpp b/fdk-aac/libAACenc/src/noisedet.cpp new file mode 100644 index 0000000..c984304 --- /dev/null +++ b/fdk-aac/libAACenc/src/noisedet.cpp @@ -0,0 +1,235 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M. Lohwasser + + Description: noisedet.c + Routines for Noise Detection + +*******************************************************************************/ + +#include "noisedet.h" + +#include "aacenc_pns.h" +#include "pnsparam.h" + +/***************************************************************************** + + functionname: FDKaacEnc_fuzzyIsSmaller + description: Fuzzy value calculation for "testVal is smaller than refVal" + returns: fuzzy value + input: test and ref Value, + low and high Lim + output: return fuzzy value + +*****************************************************************************/ +static FIXP_SGL FDKaacEnc_fuzzyIsSmaller(FIXP_DBL testVal, FIXP_DBL refVal, + FIXP_DBL loLim, FIXP_DBL hiLim) { + if (refVal <= FL2FXCONST_DBL(0.0)) + return (FL2FXCONST_SGL(0.0f)); + else if (testVal >= fMult((hiLim >> 1) + (loLim >> 1), refVal)) + return (FL2FXCONST_SGL(0.0f)); + else + return ((FIXP_SGL)MAXVAL_SGL); +} + +/***************************************************************************** + + functionname: FDKaacEnc_noiseDetect + description: detect tonal sfb's; two tests + Powerdistribution: + sfb splittet in four regions, + compare the energy in all sections + PsychTonality: + compare tonality from chaosmeasure with reftonality + returns: + input: spectrum of one large mdct + number of sfb's + pointer to offset of sfb's + pointer to noiseFuzzyMeasure (modified) + noiseparams struct + pointer to sfb energies + pointer to tonality calculated in chaosmeasure + output: noiseFuzzy Measure + +*****************************************************************************/ + +void FDKaacEnc_noiseDetect(FIXP_DBL *RESTRICT mdctSpectrum, + INT *RESTRICT sfbMaxScaleSpec, INT sfbActive, + const INT *RESTRICT sfbOffset, + FIXP_SGL *RESTRICT noiseFuzzyMeasure, + NOISEPARAMS *np, FIXP_SGL *RESTRICT sfbtonality) + +{ + int i, k, sfb, sfbWidth; + FIXP_SGL fuzzy, fuzzyTotal; + FIXP_DBL refVal, testVal; + + /***** Start detection phase *****/ + /* Start noise detection for each band based on a number of checks */ + for (sfb = 0; sfb < sfbActive; sfb++) { + fuzzyTotal = (FIXP_SGL)MAXVAL_SGL; + sfbWidth = sfbOffset[sfb + 1] - sfbOffset[sfb]; + + /* Reset output for lower bands or too small bands */ + if (sfb < np->startSfb || sfbWidth < np->minSfbWidth) { + noiseFuzzyMeasure[sfb] = FL2FXCONST_SGL(0.0f); + continue; + } + + if ((np->detectionAlgorithmFlags & USE_POWER_DISTRIBUTION) && + (fuzzyTotal > FL2FXCONST_SGL(0.5f))) { + FIXP_DBL fhelp1, fhelp2, fhelp3, fhelp4, maxVal, minVal; + INT leadingBits = fixMax( + 0, (sfbMaxScaleSpec[sfb] - + 3)); /* max sfbWidth = 96/4 ; 2^5=32 => 5/2 = 3 (spc*spc) */ + + /* check power distribution in four regions */ + fhelp1 = fhelp2 = fhelp3 = fhelp4 = FL2FXCONST_DBL(0.0f); + k = sfbWidth >> 2; /* Width of a quarter band */ + + for (i = sfbOffset[sfb]; i < sfbOffset[sfb] + k; i++) { + fhelp1 = fPow2AddDiv2(fhelp1, mdctSpectrum[i] << leadingBits); + fhelp2 = fPow2AddDiv2(fhelp2, mdctSpectrum[i + k] << leadingBits); + fhelp3 = fPow2AddDiv2(fhelp3, mdctSpectrum[i + 2 * k] << leadingBits); + fhelp4 = fPow2AddDiv2(fhelp4, mdctSpectrum[i + 3 * k] << leadingBits); + } + + /* get max into fhelp: */ + maxVal = fixMax(fhelp1, fhelp2); + maxVal = fixMax(maxVal, fhelp3); + maxVal = fixMax(maxVal, fhelp4); + + /* get min into fhelp1: */ + minVal = fixMin(fhelp1, fhelp2); + minVal = fixMin(minVal, fhelp3); + minVal = fixMin(minVal, fhelp4); + + /* Normalize min and max Val */ + leadingBits = CountLeadingBits(maxVal); + testVal = maxVal << leadingBits; + refVal = minVal << leadingBits; + + /* calculate fuzzy value for power distribution */ + testVal = fMultDiv2(testVal, np->powDistPSDcurve[sfb]); + + fuzzy = FDKaacEnc_fuzzyIsSmaller( + testVal, /* 1/2 * maxValue * PSDcurve */ + refVal, /* 1 * minValue */ + FL2FXCONST_DBL(0.495), /* 1/2 * loLim (0.99f/2) */ + FL2FXCONST_DBL(0.505)); /* 1/2 * hiLim (1.01f/2) */ + + fuzzyTotal = fixMin(fuzzyTotal, fuzzy); + } + + if ((np->detectionAlgorithmFlags & USE_PSYCH_TONALITY) && + (fuzzyTotal > FL2FXCONST_SGL(0.5f))) { + /* Detection with tonality-value of psych. acoustic (here: 1 is tonal!)*/ + + testVal = FX_SGL2FX_DBL(sfbtonality[sfb]) >> 1; /* 1/2 * sfbTonality */ + refVal = np->refTonality; + + fuzzy = FDKaacEnc_fuzzyIsSmaller( + testVal, refVal, FL2FXCONST_DBL(0.45f), /* 1/2 * loLim (0.9f/2) */ + FL2FXCONST_DBL(0.55f)); /* 1/2 * hiLim (1.1f/2) */ + + fuzzyTotal = fixMin(fuzzyTotal, fuzzy); + } + + /* Output of final result */ + noiseFuzzyMeasure[sfb] = fuzzyTotal; + } +} diff --git a/fdk-aac/libAACenc/src/noisedet.h b/fdk-aac/libAACenc/src/noisedet.h new file mode 100644 index 0000000..478701f --- /dev/null +++ b/fdk-aac/libAACenc/src/noisedet.h @@ -0,0 +1,116 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M. Lohwasser + + Description: noisedet.h + +*******************************************************************************/ + +#ifndef NOISEDET_H +#define NOISEDET_H + +#include "common_fix.h" + +#include "pnsparam.h" +#include "psy_data.h" + +void FDKaacEnc_noiseDetect(FIXP_DBL *mdctSpectrum, INT *sfbMaxScaleSpec, + INT sfbActive, const INT *sfbOffset, + FIXP_SGL noiseFuzzyMeasure[], NOISEPARAMS *np, + FIXP_SGL *sfbtonality); + +#endif /* NOISEDET_H */ diff --git a/fdk-aac/libAACenc/src/pns_func.h b/fdk-aac/libAACenc/src/pns_func.h new file mode 100644 index 0000000..88f4586 --- /dev/null +++ b/fdk-aac/libAACenc/src/pns_func.h @@ -0,0 +1,138 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M. Lohwasser + + Description: pns_func.h + +*******************************************************************************/ + +#ifndef PNS_FUNC_H +#define PNS_FUNC_H + +#include "common_fix.h" +#include "aacenc_pns.h" +#include "psy_data.h" + +AAC_ENCODER_ERROR FDKaacEnc_InitPnsConfiguration( + PNS_CONFIG *pnsConf, INT bitRate, INT sampleRate, INT usePns, INT sfbCnt, + const INT *sfbOffset, const INT numChan, const INT isLC); + +void FDKaacEnc_PnsDetect(PNS_CONFIG *pnsConf, PNS_DATA *pnsData, + const INT lastWindowSequence, const INT sfbActive, + const INT maxSfbPerGroup, FIXP_DBL *sfbThresholdLdData, + const INT *sfbOffset, FIXP_DBL *mdctSpectrum, + INT *sfbMaxScaleSpec, FIXP_SGL *sfbtonality, + int tnsOrder, INT tnsPredictionGain, INT tnsActive, + FIXP_DBL *sfbEnergyLdData, INT *noiseNrg); + +void FDKaacEnc_CodePnsChannel(const INT sfbActive, PNS_CONFIG *pnsConf, + INT *pnsFlag, FIXP_DBL *sfbEnergy, INT *noiseNrg, + FIXP_DBL *sfbThreshold); + +void FDKaacEnc_PreProcessPnsChannelPair( + const INT sfbActive, FIXP_DBL *sfbEnergyLeft, FIXP_DBL *sfbEnergyRight, + FIXP_DBL *sfbEnergyLeftLD, FIXP_DBL *sfbEnergyRightLD, + FIXP_DBL *sfbEnergyMid, PNS_CONFIG *pnsConfLeft, PNS_DATA *pnsDataLeft, + PNS_DATA *pnsDataRight); + +void FDKaacEnc_PostProcessPnsChannelPair(const INT sfbActive, + PNS_CONFIG *pnsConf, + PNS_DATA *pnsDataLeft, + PNS_DATA *pnsDataRight, INT *msMask, + INT *msDigest); + +#endif /* PNS_FUNC_H */ diff --git a/fdk-aac/libAACenc/src/pnsparam.cpp b/fdk-aac/libAACenc/src/pnsparam.cpp new file mode 100644 index 0000000..a6aab06 --- /dev/null +++ b/fdk-aac/libAACenc/src/pnsparam.cpp @@ -0,0 +1,574 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M.Lohwasser + + Description: PNS parameters depending on bitrate and bandwidth + +*******************************************************************************/ + +#include "pnsparam.h" + +#include "psy_configuration.h" + +typedef struct { + SHORT startFreq; + /* Parameters for detection */ + FIXP_SGL refPower; + FIXP_SGL refTonality; + SHORT tnsGainThreshold; /* scaled by TNS_PREDGAIN_SCALE (=1000) */ + SHORT tnsPNSGainThreshold; /* scaled by TNS_PREDGAIN_SCALE (=1000) */ + FIXP_SGL gapFillThr; + SHORT minSfbWidth; + USHORT detectionAlgorithmFlags; +} PNS_INFO_TAB; + +typedef struct { + ULONG brFrom; + ULONG brTo; + UCHAR S16000; + UCHAR S22050; + UCHAR S24000; + UCHAR S32000; + UCHAR S44100; + UCHAR S48000; +} AUTO_PNS_TAB; + +static const AUTO_PNS_TAB levelTable_mono[] = { + { + 0, + 11999, + 0, + 1, + 1, + 1, + 1, + 1, + }, + { + 12000, + 19999, + 0, + 1, + 1, + 1, + 1, + 1, + }, + { + 20000, + 28999, + 0, + 2, + 1, + 1, + 1, + 1, + }, + { + 29000, + 40999, + 0, + 4, + 4, + 4, + 2, + 2, + }, + { + 41000, + 55999, + 0, + 9, + 9, + 7, + 7, + 7, + }, + { + 56000, + 61999, + 0, + 0, + 0, + 0, + 9, + 9, + }, + { + 62000, + 75999, + 0, + 0, + 0, + 0, + 0, + 0, + }, + { + 76000, + 92999, + 0, + 0, + 0, + 0, + 0, + 0, + }, + { + 93000, + 999999, + 0, + 0, + 0, + 0, + 0, + 0, + }, +}; + +static const AUTO_PNS_TAB levelTable_stereo[] = { + { + 0, + 11999, + 0, + 1, + 1, + 1, + 1, + 1, + }, + { + 12000, + 19999, + 0, + 3, + 1, + 1, + 1, + 1, + }, + { + 20000, + 28999, + 0, + 3, + 3, + 3, + 2, + 2, + }, + { + 29000, + 40999, + 0, + 7, + 6, + 6, + 5, + 5, + }, + { + 41000, + 55999, + 0, + 9, + 9, + 7, + 7, + 7, + }, + { + 56000, + 79999, + 0, + 0, + 0, + 0, + 0, + 0, + }, + { + 80000, + 99999, + 0, + 0, + 0, + 0, + 0, + 0, + }, + { + 100000, + 999999, + 0, + 0, + 0, + 0, + 0, + 0, + }, +}; + +static const PNS_INFO_TAB pnsInfoTab[] = { + /*0 pns off */ + /*1*/ {4000, FL2FXCONST_SGL(0.04), FL2FXCONST_SGL(0.06), 1150, 1200, + FL2FXCONST_SGL(0.02), 8, + USE_POWER_DISTRIBUTION | USE_PSYCH_TONALITY | USE_TNS_GAIN_THR | + USE_TNS_PNS /*| JUST_LONG_WINDOW*/}, + /*2*/ + {4000, FL2FXCONST_SGL(0.04), FL2FXCONST_SGL(0.07), 1130, 1300, + FL2FXCONST_SGL(0.05), 8, + USE_POWER_DISTRIBUTION | USE_PSYCH_TONALITY | USE_TNS_GAIN_THR | + USE_TNS_PNS /*| JUST_LONG_WINDOW*/}, + /*3*/ + {4100, FL2FXCONST_SGL(0.04), FL2FXCONST_SGL(0.07), 1100, 1400, + FL2FXCONST_SGL(0.10), 8, + USE_POWER_DISTRIBUTION | USE_PSYCH_TONALITY | USE_TNS_GAIN_THR | + USE_TNS_PNS /*| JUST_LONG_WINDOW*/}, + /*4*/ + {4100, FL2FXCONST_SGL(0.03), FL2FXCONST_SGL(0.10), 1100, 1400, + FL2FXCONST_SGL(0.15), 8, + USE_POWER_DISTRIBUTION | USE_PSYCH_TONALITY | USE_TNS_GAIN_THR | + USE_TNS_PNS /*| JUST_LONG_WINDOW*/}, + /*5*/ + {4300, FL2FXCONST_SGL(0.03), FL2FXCONST_SGL(0.10), 1100, 1400, + FL2FXCONST_SGL(0.15), 8, + USE_POWER_DISTRIBUTION | USE_PSYCH_TONALITY | USE_TNS_GAIN_THR | + USE_TNS_PNS | JUST_LONG_WINDOW}, + /*6*/ + {5000, FL2FXCONST_SGL(0.03), FL2FXCONST_SGL(0.10), 1100, 1400, + FL2FXCONST_SGL(0.25), 8, + USE_POWER_DISTRIBUTION | USE_PSYCH_TONALITY | USE_TNS_GAIN_THR | + USE_TNS_PNS | JUST_LONG_WINDOW}, + /*7*/ + {5500, FL2FXCONST_SGL(0.03), FL2FXCONST_SGL(0.12), 1100, 1400, + FL2FXCONST_SGL(0.35), 8, + USE_POWER_DISTRIBUTION | USE_PSYCH_TONALITY | USE_TNS_GAIN_THR | + USE_TNS_PNS | JUST_LONG_WINDOW}, + /*8*/ + {6000, FL2FXCONST_SGL(0.03), FL2FXCONST_SGL(0.12), 1080, 1400, + FL2FXCONST_SGL(0.40), 8, + USE_POWER_DISTRIBUTION | USE_PSYCH_TONALITY | USE_TNS_GAIN_THR | + USE_TNS_PNS | JUST_LONG_WINDOW}, + /*9*/ + {6000, FL2FXCONST_SGL(0.03), FL2FXCONST_SGL(0.14), 1070, 1400, + FL2FXCONST_SGL(0.45), 8, + USE_POWER_DISTRIBUTION | USE_PSYCH_TONALITY | USE_TNS_GAIN_THR | + USE_TNS_PNS | JUST_LONG_WINDOW}, +}; + +static const AUTO_PNS_TAB levelTable_lowComplexity[] = { + { + 0, + 27999, + 0, + 0, + 0, + 0, + 0, + 0, + }, + { + 28000, + 31999, + 0, + 2, + 2, + 2, + 2, + 2, + }, + { + 32000, + 47999, + 0, + 3, + 3, + 3, + 3, + 3, + }, + { + 48000, + 48000, + 0, + 4, + 4, + 4, + 4, + 4, + }, + { + 48001, + 999999, + 0, + 0, + 0, + 0, + 0, + 0, + }, +}; +/* conversion of old LC tuning tables to new (LD enc) structure (only entries + * which are actually used were converted) */ +static const PNS_INFO_TAB pnsInfoTab_lowComplexity[] = { + /*0 pns off */ + /* DEFAULT parameter set */ + /*1*/ {4100, FL2FXCONST_SGL(0.03), FL2FXCONST_SGL(0.16), 1100, 1400, + FL2FXCONST_SGL(0.5), 16, + USE_POWER_DISTRIBUTION | USE_PSYCH_TONALITY | USE_TNS_GAIN_THR | + USE_TNS_PNS | JUST_LONG_WINDOW}, + /*2*/ + {4100, FL2FXCONST_SGL(0.05), FL2FXCONST_SGL(0.10), 1410, 1400, + FL2FXCONST_SGL(0.5), 16, + USE_POWER_DISTRIBUTION | USE_PSYCH_TONALITY | USE_TNS_GAIN_THR | + USE_TNS_PNS | JUST_LONG_WINDOW}, + /*3*/ + {4100, FL2FXCONST_SGL(0.05), FL2FXCONST_SGL(0.10), 1100, 1400, + FL2FXCONST_SGL(0.5), 16, + USE_POWER_DISTRIBUTION | USE_PSYCH_TONALITY | USE_TNS_GAIN_THR | + USE_TNS_PNS | JUST_LONG_WINDOW}, + /* LOWSUBST -> PNS is used less often than with DEFAULT parameter set (for + br: 48000 - 79999) */ + /*4*/ + {4100, FL2FXCONST_SGL(0.20), FL2FXCONST_SGL(0.10), 1410, 1400, + FL2FXCONST_SGL(0.5), 16, + USE_POWER_DISTRIBUTION | USE_PSYCH_TONALITY | USE_TNS_GAIN_THR | + USE_TNS_PNS | JUST_LONG_WINDOW}, +}; + +/**************************************************************************** + function to look up used pns level +****************************************************************************/ +int FDKaacEnc_lookUpPnsUse(int bitRate, int sampleRate, int numChan, + const int isLC) { + int hUsePns = 0, size, i; + const AUTO_PNS_TAB *levelTable; + + if (isLC) { + levelTable = &levelTable_lowComplexity[0]; + size = sizeof(levelTable_lowComplexity); + } else { /* (E)LD */ + levelTable = (numChan > 1) ? &levelTable_stereo[0] : &levelTable_mono[0]; + size = (numChan > 1) ? sizeof(levelTable_stereo) : sizeof(levelTable_mono); + } + + for (i = 0; i < (int)(size / sizeof(AUTO_PNS_TAB)); i++) { + if (((ULONG)bitRate >= levelTable[i].brFrom) && + ((ULONG)bitRate <= levelTable[i].brTo)) + break; + } + + /* sanity check */ + if ((int)(sizeof(pnsInfoTab) / sizeof(PNS_INFO_TAB)) < i) { + return (PNS_TABLE_ERROR); + } + + switch (sampleRate) { + case 16000: + hUsePns = levelTable[i].S16000; + break; + case 22050: + hUsePns = levelTable[i].S22050; + break; + case 24000: + hUsePns = levelTable[i].S24000; + break; + case 32000: + hUsePns = levelTable[i].S32000; + break; + case 44100: + hUsePns = levelTable[i].S44100; + break; + case 48000: + hUsePns = levelTable[i].S48000; + break; + default: + if (isLC) { + hUsePns = levelTable[i].S48000; + } + break; + } + + return (hUsePns); +} + +/***************************************************************************** + + functionname: FDKaacEnc_GetPnsParam + description: Gets PNS parameters depending on bitrate and bandwidth or + bitsPerLine + returns: error status + input: Noiseparams struct, bitrate, sampling rate, + number of sfb's, pointer to sfb offset + output: PNS parameters + +*****************************************************************************/ +AAC_ENCODER_ERROR FDKaacEnc_GetPnsParam(NOISEPARAMS *np, INT bitRate, + INT sampleRate, INT sfbCnt, + const INT *sfbOffset, INT *usePns, + INT numChan, const INT isLC) { + int i, hUsePns; + const PNS_INFO_TAB *pnsInfo; + + if (*usePns <= 0) return AAC_ENC_OK; + + if (isLC) { + np->detectionAlgorithmFlags = IS_LOW_COMPLEXITY; + + pnsInfo = pnsInfoTab_lowComplexity; + + /* new pns params */ + hUsePns = FDKaacEnc_lookUpPnsUse(bitRate, sampleRate, numChan, isLC); + if (hUsePns == 0) { + *usePns = 0; + return AAC_ENC_OK; + } + + if (hUsePns == PNS_TABLE_ERROR) { + return AAC_ENC_PNS_TABLE_ERROR; + } + + /* select correct row of tuning table */ + pnsInfo += hUsePns - 1; + + } else { + np->detectionAlgorithmFlags = 0; + pnsInfo = pnsInfoTab; + + /* new pns params */ + hUsePns = FDKaacEnc_lookUpPnsUse(bitRate, sampleRate, numChan, isLC); + if (hUsePns == 0) { + *usePns = 0; + return AAC_ENC_OK; + } + if (hUsePns == PNS_TABLE_ERROR) return AAC_ENC_PNS_TABLE_ERROR; + + /* select correct row of tuning table */ + pnsInfo += hUsePns - 1; + } + + np->startSfb = FDKaacEnc_FreqToBandWidthRounding( + pnsInfo->startFreq, sampleRate, sfbCnt, sfbOffset); + + np->detectionAlgorithmFlags |= pnsInfo->detectionAlgorithmFlags; + + np->refPower = FX_SGL2FX_DBL(pnsInfo->refPower); + np->refTonality = FX_SGL2FX_DBL(pnsInfo->refTonality); + np->tnsGainThreshold = pnsInfo->tnsGainThreshold; + np->tnsPNSGainThreshold = pnsInfo->tnsPNSGainThreshold; + np->minSfbWidth = pnsInfo->minSfbWidth; + + np->gapFillThr = + pnsInfo->gapFillThr; /* for LC it is always FL2FXCONST_SGL(0.5) */ + + /* assuming a constant dB/Hz slope in the signal's PSD curve, + the detection threshold needs to be corrected for the width of the band */ + + for (i = 0; i < (sfbCnt - 1); i++) { + INT qtmp, sfbWidth; + FIXP_DBL tmp; + + sfbWidth = sfbOffset[i + 1] - sfbOffset[i]; + + tmp = fPow(np->refPower, 0, sfbWidth, DFRACT_BITS - 1 - 5, &qtmp); + np->powDistPSDcurve[i] = (FIXP_SGL)((LONG)(scaleValue(tmp, qtmp) >> 16)); + } + np->powDistPSDcurve[sfbCnt] = np->powDistPSDcurve[sfbCnt - 1]; + + return AAC_ENC_OK; +} diff --git a/fdk-aac/libAACenc/src/pnsparam.h b/fdk-aac/libAACenc/src/pnsparam.h new file mode 100644 index 0000000..c37738a --- /dev/null +++ b/fdk-aac/libAACenc/src/pnsparam.h @@ -0,0 +1,149 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M. Lohwasser + + Description: PNS parameters depending on bitrate and bandwidth + +*******************************************************************************/ + +#ifndef PNSPARAM_H +#define PNSPARAM_H + +#include "aacenc.h" +#include "common_fix.h" +#include "psy_const.h" + +#define NUM_PNSINFOTAB 4 +#define PNS_TABLE_ERROR -1 + +/* detection algorithm flags */ +#define USE_POWER_DISTRIBUTION (1 << 0) +#define USE_PSYCH_TONALITY (1 << 1) +#define USE_TNS_GAIN_THR (1 << 2) +#define USE_TNS_PNS (1 << 3) +#define JUST_LONG_WINDOW (1 << 4) +/* additional algorithm flags */ +#define IS_LOW_COMPLEXITY (1 << 5) + +typedef struct { + /* PNS start band */ + short startSfb; + + /* detection algorithm flags */ + USHORT detectionAlgorithmFlags; + + /* Parameters for detection */ + FIXP_DBL refPower; + FIXP_DBL refTonality; + INT tnsGainThreshold; + INT tnsPNSGainThreshold; + INT minSfbWidth; + FIXP_SGL powDistPSDcurve[MAX_GROUPED_SFB]; + FIXP_SGL gapFillThr; +} NOISEPARAMS; + +int FDKaacEnc_lookUpPnsUse(int bitRate, int sampleRate, int numChan, + const int isLC); + +/****** Definition of prototypes ******/ + +AAC_ENCODER_ERROR FDKaacEnc_GetPnsParam(NOISEPARAMS *np, INT bitRate, + INT sampleRate, INT sfbCnt, + const INT *sfbOffset, INT *usePns, + INT numChan, const INT isLC); + +#endif /* PNSPARAM_H */ diff --git a/fdk-aac/libAACenc/src/pre_echo_control.cpp b/fdk-aac/libAACenc/src/pre_echo_control.cpp new file mode 100644 index 0000000..3d5d153 --- /dev/null +++ b/fdk-aac/libAACenc/src/pre_echo_control.cpp @@ -0,0 +1,176 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M.Werner + + Description: Pre echo control + +*******************************************************************************/ + +#include "pre_echo_control.h" +#include "psy_configuration.h" + +void FDKaacEnc_InitPreEchoControl(FIXP_DBL *RESTRICT pbThresholdNm1, + INT *calcPreEcho, INT numPb, + FIXP_DBL *RESTRICT sfbPcmQuantThreshold, + INT *mdctScalenm1) { + *mdctScalenm1 = PCM_QUANT_THR_SCALE >> 1; + + FDKmemcpy(pbThresholdNm1, sfbPcmQuantThreshold, numPb * sizeof(FIXP_DBL)); + + *calcPreEcho = 1; +} + +void FDKaacEnc_PreEchoControl(FIXP_DBL *RESTRICT pbThresholdNm1, + INT calcPreEcho, INT numPb, + INT maxAllowedIncreaseFactor, + FIXP_SGL minRemainingThresholdFactor, + FIXP_DBL *RESTRICT pbThreshold, INT mdctScale, + INT *mdctScalenm1) { + int i; + FIXP_DBL tmpThreshold1, tmpThreshold2; + int scaling; + + /* If lastWindowSequence in previous frame was start- or stop-window, + skip preechocontrol calculation */ + if (calcPreEcho == 0) { + /* copy thresholds to internal memory */ + FDKmemcpy(pbThresholdNm1, pbThreshold, numPb * sizeof(FIXP_DBL)); + *mdctScalenm1 = mdctScale; + return; + } + + if (mdctScale > *mdctScalenm1) { + /* if current thresholds are downscaled more than the ones from the last + * block */ + scaling = 2 * (mdctScale - *mdctScalenm1); + for (i = 0; i < numPb; i++) { + /* multiplication with return data type fract ist equivalent to int + * multiplication */ + FDK_ASSERT(scaling >= 0); + tmpThreshold1 = maxAllowedIncreaseFactor * (pbThresholdNm1[i] >> scaling); + tmpThreshold2 = fMult(minRemainingThresholdFactor, pbThreshold[i]); + + FIXP_DBL tmp = pbThreshold[i]; + + /* copy thresholds to internal memory */ + pbThresholdNm1[i] = tmp; + + tmp = fixMin(tmp, tmpThreshold1); + pbThreshold[i] = fixMax(tmp, tmpThreshold2); + } + } else { + /* if thresholds of last block are more downscaled than the current ones */ + scaling = 2 * (*mdctScalenm1 - mdctScale); + for (i = 0; i < numPb; i++) { + /* multiplication with return data type fract ist equivalent to int + * multiplication */ + tmpThreshold1 = (maxAllowedIncreaseFactor >> 1) * pbThresholdNm1[i]; + tmpThreshold2 = fMult(minRemainingThresholdFactor, pbThreshold[i]); + + /* copy thresholds to internal memory */ + pbThresholdNm1[i] = pbThreshold[i]; + + FDK_ASSERT(scaling >= 0); + if ((pbThreshold[i] >> (scaling + 1)) > tmpThreshold1) { + pbThreshold[i] = tmpThreshold1 << (scaling + 1); + } + pbThreshold[i] = fixMax(pbThreshold[i], tmpThreshold2); + } + } + + *mdctScalenm1 = mdctScale; +} diff --git a/fdk-aac/libAACenc/src/pre_echo_control.h b/fdk-aac/libAACenc/src/pre_echo_control.h new file mode 100644 index 0000000..688efdb --- /dev/null +++ b/fdk-aac/libAACenc/src/pre_echo_control.h @@ -0,0 +1,118 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M.Werner + + Description: Pre echo control + +*******************************************************************************/ + +#ifndef PRE_ECHO_CONTROL_H +#define PRE_ECHO_CONTROL_H + +#include "common_fix.h" + +void FDKaacEnc_InitPreEchoControl(FIXP_DBL *pbThresholdnm1, INT *calcPreEcho, + INT numPb, FIXP_DBL *sfbPcmQuantThreshold, + INT *mdctScalenm1); + +void FDKaacEnc_PreEchoControl(FIXP_DBL *pbThresholdNm1, INT calcPreEcho, + INT numPb, INT maxAllowedIncreaseFactor, + FIXP_SGL minRemainingThresholdFactor, + FIXP_DBL *pbThreshold, INT mdctScale, + INT *mdctScalenm1); + +#endif diff --git a/fdk-aac/libAACenc/src/psy_configuration.cpp b/fdk-aac/libAACenc/src/psy_configuration.cpp new file mode 100644 index 0000000..b444b58 --- /dev/null +++ b/fdk-aac/libAACenc/src/psy_configuration.cpp @@ -0,0 +1,801 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M.Werner + + Description: Psychoaccoustic configuration + +*******************************************************************************/ + +#include "psy_configuration.h" +#include "adj_thr.h" +#include "aacEnc_rom.h" + +#include "genericStds.h" + +#include "FDK_trigFcts.h" + +typedef struct { + LONG sampleRate; + const SFB_PARAM_LONG *paramLong; + const SFB_PARAM_SHORT *paramShort; +} SFB_INFO_TAB; + +static const SFB_INFO_TAB sfbInfoTab[] = { + {8000, &p_FDKaacEnc_8000_long_1024, &p_FDKaacEnc_8000_short_128}, + {11025, &p_FDKaacEnc_11025_long_1024, &p_FDKaacEnc_11025_short_128}, + {12000, &p_FDKaacEnc_12000_long_1024, &p_FDKaacEnc_12000_short_128}, + {16000, &p_FDKaacEnc_16000_long_1024, &p_FDKaacEnc_16000_short_128}, + {22050, &p_FDKaacEnc_22050_long_1024, &p_FDKaacEnc_22050_short_128}, + {24000, &p_FDKaacEnc_24000_long_1024, &p_FDKaacEnc_24000_short_128}, + {32000, &p_FDKaacEnc_32000_long_1024, &p_FDKaacEnc_32000_short_128}, + {44100, &p_FDKaacEnc_44100_long_1024, &p_FDKaacEnc_44100_short_128}, + {48000, &p_FDKaacEnc_48000_long_1024, &p_FDKaacEnc_48000_short_128}, + {64000, &p_FDKaacEnc_64000_long_1024, &p_FDKaacEnc_64000_short_128}, + {88200, &p_FDKaacEnc_88200_long_1024, &p_FDKaacEnc_88200_short_128}, + {96000, &p_FDKaacEnc_96000_long_1024, &p_FDKaacEnc_96000_short_128} + +}; + + + +const SFB_PARAM_LONG p_FDKaacEnc_8000_long_960 = { + 40, + { 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 12, 16, + 16, 16, 16, 16, 16, 16, 20, 20, 20, 20, 24, 24, 24, 28, + 28, 32, 36, 36, 40, 44, 48, 52, 56, 60, 64, 16 } +}; +const SFB_PARAM_SHORT p_FDKaacEnc_8000_short_120 = { + 15, + { 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 16, 20, 12 } +}; + +const SFB_PARAM_LONG p_FDKaacEnc_11025_long_960 = { + 42, + { 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 12, 12, 12, + 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24, + 24, 28, 28, 32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64 } +}; +const SFB_PARAM_SHORT p_FDKaacEnc_11025_short_120 = { + 15, + { 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 12 } +}; + +const SFB_PARAM_LONG p_FDKaacEnc_12000_long_960 = { + 42, + { 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 12, 12, 12, + 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24, + 24, 28, 28, 32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64 } +}; +const SFB_PARAM_SHORT p_FDKaacEnc_12000_short_120 = { + 15, + { 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 12 } +}; + +const SFB_PARAM_LONG p_FDKaacEnc_16000_long_960 = { + 42, + { 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 8, 12, 12, 12, + 12, 12, 12, 12, 12, 12, 16, 16, 16, 16, 20, 20, 20, 24, + 24, 28, 28, 32, 36, 40, 40, 44, 48, 52, 56, 60, 64, 64 } +}; +const SFB_PARAM_SHORT p_FDKaacEnc_16000_short_120 = { + 15, + { 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 12, 12, 16, 20, 12 } +}; + +const SFB_PARAM_LONG p_FDKaacEnc_22050_long_960 = { + 46, + { 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, + 8, 8, 8, 8, 8, 8, 8, 12, 12, 12, 12, 16, 16, 16, + 20, 20, 24, 24, 28, 28, 32, 36, 36, 40, 44, 48, 52, 52, + 64, 64, 64, 64 } +}; +const SFB_PARAM_SHORT p_FDKaacEnc_22050_short_120 = { + 15, + { 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 12 } +}; + +const SFB_PARAM_LONG p_FDKaacEnc_24000_long_960 = { + 46, + { 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, + 8, 8, 8, 8, 8, 8, 8, 12, 12, 12, 12, 16, 16, 16, + 20, 20, 24, 24, 28, 28, 32, 36, 36, 40, 44, 48, 52, 52, + 64, 64, 64, 64 } +}; +const SFB_PARAM_SHORT p_FDKaacEnc_24000_short_120 = { + 15, + { 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 16, 16, 12 } +}; + +const SFB_PARAM_LONG p_FDKaacEnc_32000_long_960 = { + 49, + { 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 8, 8, 12, 12, 12, + 12, 16, 16, 20, 20, 24, 24, 28, 28, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, + 32, 32, 32, 32, 32, 32, 32, 32, 32 } +}; + +const SFB_PARAM_SHORT p_FDKaacEnc_32000_short_120 = { + 14, + { 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 8 } +}; + +const SFB_PARAM_LONG p_FDKaacEnc_44100_long_960 = { + 49, + { 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, + 8, 8, 8, 12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, + 28, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, + 32, 32, 32, 32, 32, 32, 32 } +}; + +const SFB_PARAM_SHORT p_FDKaacEnc_44100_short_120 = { + 14, + { 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 8 } +}; + +const SFB_PARAM_LONG p_FDKaacEnc_48000_long_960 = { + 49, + { 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, + 8, 8, 8, 12, 12, 12, 12, 16, 16, 20, 20, 24, 24, 28, + 28, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32, + 32, 32, 32, 32, 32, 32, 32 } +}; +const SFB_PARAM_SHORT p_FDKaacEnc_48000_short_120 = { + 14, + { 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, 12, 16, 16, 8 } +}; + +const SFB_PARAM_LONG p_FDKaacEnc_64000_long_960 = { + 46, + { 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 12, 12, + 12, 16, 16, 16, 20, 24, 24, 28, 36, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, 40, + 40, 40, 40, 40, 40, 16 } +}; +const SFB_PARAM_SHORT p_FDKaacEnc_64000_short_120 = { + 12, + { 4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 28 } +}; + +const SFB_PARAM_LONG p_FDKaacEnc_88200_long_960 = { + 40, + { 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 12, + 12, 12, 12, 12, 16, 16, 24, 28, 36, 44, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64 } +}; +const SFB_PARAM_SHORT p_FDKaacEnc_88200_short_120 = { + 12, + { 4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 28 } +}; + +const SFB_PARAM_LONG p_FDKaacEnc_96000_long_960 = { + 40, + { 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, 12, + 12, 12, 12, 12, 16, 16, 24, 28, 36, 44, 64, 64, 64, 64, 64, 64, 64, 64, 64, 64 } +}; +const SFB_PARAM_SHORT p_FDKaacEnc_96000_short_120 = { + 12, + { 4, 4, 4, 4, 4, 4, 8, 8, 8, 16, 28, 28 } +}; + + +static const SFB_INFO_TAB sfbInfoTab960[] = { + { 8000, &p_FDKaacEnc_8000_long_960, &p_FDKaacEnc_8000_short_120}, + {11025, &p_FDKaacEnc_11025_long_960, &p_FDKaacEnc_11025_short_120}, + {12000, &p_FDKaacEnc_12000_long_960, &p_FDKaacEnc_12000_short_120}, + {16000, &p_FDKaacEnc_16000_long_960, &p_FDKaacEnc_16000_short_120}, + {22050, &p_FDKaacEnc_22050_long_960, &p_FDKaacEnc_22050_short_120}, + {24000, &p_FDKaacEnc_24000_long_960, &p_FDKaacEnc_24000_short_120}, + {32000, &p_FDKaacEnc_32000_long_960, &p_FDKaacEnc_32000_short_120}, + {44100, &p_FDKaacEnc_44100_long_960, &p_FDKaacEnc_44100_short_120}, + {48000, &p_FDKaacEnc_48000_long_960, &p_FDKaacEnc_48000_short_120}, + {64000, &p_FDKaacEnc_64000_long_960, &p_FDKaacEnc_64000_short_120}, + {88200, &p_FDKaacEnc_88200_long_960, &p_FDKaacEnc_88200_short_120}, + {96000, &p_FDKaacEnc_96000_long_960, &p_FDKaacEnc_96000_short_120}, +}; + + +/* 22050 and 24000 Hz */ +static const SFB_PARAM_LONG p_22050_long_512 = { + 31, {4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, 12, + 12, 16, 20, 24, 28, 32, 32, 32, 32, 32, 32, 32, 32, 32, 32}}; + +/* 32000 Hz */ +static const SFB_PARAM_LONG p_32000_long_512 = { + 37, + {4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, 8, + 12, 12, 12, 12, 16, 16, 16, 20, 24, 24, 28, 32, 32, 32, 32, 32, 32, 32}}; + +/* 44100 Hz */ +static const SFB_PARAM_LONG p_44100_long_512 = { + 36, {4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, + 8, 8, 12, 12, 12, 12, 16, 20, 24, 28, 32, 32, 32, 32, 32, 32, 32, 52}}; + +static const SFB_INFO_TAB sfbInfoTabLD512[] = { + {8000, &p_22050_long_512, NULL}, {11025, &p_22050_long_512, NULL}, + {12000, &p_22050_long_512, NULL}, {16000, &p_22050_long_512, NULL}, + {22050, &p_22050_long_512, NULL}, {24000, &p_22050_long_512, NULL}, + {32000, &p_32000_long_512, NULL}, {44100, &p_44100_long_512, NULL}, + {48000, &p_44100_long_512, NULL}, {64000, &p_44100_long_512, NULL}, + {88200, &p_44100_long_512, NULL}, {96000, &p_44100_long_512, NULL}, + {128000, &p_44100_long_512, NULL}, {176400, &p_44100_long_512, NULL}, + {192000, &p_44100_long_512, NULL}, {256000, &p_44100_long_512, NULL}, + {352800, &p_44100_long_512, NULL}, {384000, &p_44100_long_512, NULL}, +}; + +/* 22050 and 24000 Hz */ +static const SFB_PARAM_LONG p_22050_long_480 = { + 30, {4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 12, + 12, 12, 16, 20, 24, 28, 32, 32, 32, 32, 32, 32, 32, 32, 32}}; + +/* 32000 Hz */ +static const SFB_PARAM_LONG p_32000_long_480 = { + 37, {4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, + 8, 8, 8, 12, 12, 12, 16, 16, 20, 24, 32, 32, 32, 32, 32, 32, 32, 32}}; + +/* 44100 Hz */ +static const SFB_PARAM_LONG p_44100_long_480 = { + 35, {4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 8, 8, 8, 8, + 8, 12, 12, 12, 12, 12, 16, 16, 24, 28, 32, 32, 32, 32, 32, 32, 48}}; + +static const SFB_INFO_TAB sfbInfoTabLD480[] = { + {8000, &p_22050_long_480, NULL}, {11025, &p_22050_long_480, NULL}, + {12000, &p_22050_long_480, NULL}, {16000, &p_22050_long_480, NULL}, + {22050, &p_22050_long_480, NULL}, {24000, &p_22050_long_480, NULL}, + {32000, &p_32000_long_480, NULL}, {44100, &p_44100_long_480, NULL}, + {48000, &p_44100_long_480, NULL}, {64000, &p_44100_long_480, NULL}, + {88200, &p_44100_long_480, NULL}, {96000, &p_44100_long_480, NULL}, + {128000, &p_44100_long_480, NULL}, {176400, &p_44100_long_480, NULL}, + {192000, &p_44100_long_480, NULL}, {256000, &p_44100_long_480, NULL}, + {352800, &p_44100_long_480, NULL}, {384000, &p_44100_long_480, NULL}, +}; + +/* Fixed point precision definitions */ +#define Q_BARCVAL (25) + +AAC_ENCODER_ERROR FDKaacEnc_initSfbTable(const LONG sampleRate, + const INT blockType, + const INT granuleLength, + INT *const sfbOffset, + INT *const sfbCnt) { + INT i, specStartOffset = 0; + INT granuleLengthWindow = granuleLength; + const UCHAR *sfbWidth = NULL; + const SFB_INFO_TAB *sfbInfo = NULL; + int size; + + /* + select table + */ + switch (granuleLength) { + case 1024: + sfbInfo = sfbInfoTab; + size = (INT)(sizeof(sfbInfoTab) / sizeof(SFB_INFO_TAB)); + break; + case 960: + sfbInfo = sfbInfoTab960; + size = (INT)(sizeof(sfbInfoTab960)/sizeof(SFB_INFO_TAB)); + break; + case 512: + sfbInfo = sfbInfoTabLD512; + size = sizeof(sfbInfoTabLD512); + break; + case 480: + sfbInfo = sfbInfoTabLD480; + size = sizeof(sfbInfoTabLD480); + break; + default: + return AAC_ENC_INVALID_FRAME_LENGTH; + } + + for (i = 0; i < size; i++) { + if (sfbInfo[i].sampleRate == sampleRate) { + switch (blockType) { + case LONG_WINDOW: + case START_WINDOW: + case STOP_WINDOW: + sfbWidth = sfbInfo[i].paramLong->sfbWidth; + *sfbCnt = sfbInfo[i].paramLong->sfbCnt; + break; + case SHORT_WINDOW: + sfbWidth = sfbInfo[i].paramShort->sfbWidth; + *sfbCnt = sfbInfo[i].paramShort->sfbCnt; + granuleLengthWindow /= TRANS_FAC; + break; + } + break; + } + } + if (i == size) { + return AAC_ENC_UNSUPPORTED_SAMPLINGRATE; + } + + /* + calc sfb offsets + */ + for (i = 0; i < *sfbCnt; i++) { + sfbOffset[i] = specStartOffset; + specStartOffset += sfbWidth[i]; + if (specStartOffset >= granuleLengthWindow) { + i++; + break; + } + } + *sfbCnt = fixMin(i, *sfbCnt); + sfbOffset[*sfbCnt] = fixMin(specStartOffset, granuleLengthWindow); + return AAC_ENC_OK; +} + +/***************************************************************************** + + functionname: FDKaacEnc_BarcLineValue + description: Calculates barc value for one frequency line + returns: barc value of line + input: number of lines in transform, index of line to check, Fs + output: + +*****************************************************************************/ +static FIXP_DBL FDKaacEnc_BarcLineValue(INT noOfLines, INT fftLine, + LONG samplingFreq) { + FIXP_DBL FOURBY3EM4 = (FIXP_DBL)0x45e7b273; /* 4.0/3 * 0.0001 in q43 */ + FIXP_DBL PZZZ76 = (FIXP_DBL)0x639d5e4a; /* 0.00076 in q41 */ + FIXP_DBL ONE3P3 = (FIXP_DBL)0x35333333; /* 13.3 in q26 */ + FIXP_DBL THREEP5 = (FIXP_DBL)0x1c000000; /* 3.5 in q27 */ + FIXP_DBL INV480 = (FIXP_DBL)0x44444444; // 1/480 in q39 + + FIXP_DBL center_freq, x1, x2; + FIXP_DBL bvalFFTLine, atan1, atan2; + + /* Theoritical maximum of center_freq (samp_freq*0.5) is 96khz * 0.5 = 48000 + */ + /* Theoritical maximum of x1 is 1.3333333e-4f * center_freq = 6.4, can keep in + * q28 */ + /* Theoritical maximum of x2 is 0.00076f * center_freq = 36.48, can keep in + * q25 */ + + center_freq = fftLine * samplingFreq; /* q11 or q8 */ + + + switch (noOfLines) { + case 1024: + center_freq = center_freq << 2; /* q13 */ + break; + case 960: + center_freq = fMult(center_freq, INV480) << 3; + break; + case 128: + center_freq = center_freq << 5; /* q13 */ + break; + case 120: + center_freq = fMult(center_freq, INV480) << 6; + break; + case 512: + center_freq = (fftLine * samplingFreq) << 3; // q13 + break; + case 480: + center_freq = fMult(center_freq, INV480) << 4; // q13 + break; + default: + center_freq = (FIXP_DBL)0; + } + + + x1 = fMult(center_freq, FOURBY3EM4); /* q13 * q43 - (DFRACT_BITS-1) = q25 */ + x2 = fMult(center_freq, PZZZ76) + << 2; /* q13 * q41 - (DFRACT_BITS-1) + 2 = q25 */ + + atan1 = fixp_atan(x1); + atan2 = fixp_atan(x2); + + /* q25 (q26 * q30 - (DFRACT_BITS-1)) + q25 (q27 * q30 * q30) */ + bvalFFTLine = fMult(ONE3P3, atan2) + fMult(THREEP5, fMult(atan1, atan1)); + return (bvalFFTLine); +} + +/* + do not consider energies below a certain input signal level, + i.e. of -96dB or 1 bit at 16 bit PCM resolution, + might need to be configurable to e.g. 24 bit PCM Input or a lower + resolution for low bit rates +*/ +static void FDKaacEnc_InitMinPCMResolution(int numPb, int *pbOffset, + FIXP_DBL *sfbPCMquantThreshold) { +/* PCM_QUANT_NOISE = FDKpow(10.0f, - 20.f / 10.0f) * ABS_LOW * NORM_PCM_ENERGY * + * FDKpow(2,PCM_QUANT_THR_SCALE) */ +#define PCM_QUANT_NOISE ((FIXP_DBL)0x00547062) + + for (int i = 0; i < numPb; i++) { + sfbPCMquantThreshold[i] = (pbOffset[i + 1] - pbOffset[i]) * PCM_QUANT_NOISE; + } +} + +static FIXP_DBL getMaskFactor(const FIXP_DBL dbVal_fix, const INT dbVal_e, + const FIXP_DBL ten_fix, const INT ten_e) { + INT q_msk; + FIXP_DBL mask_factor; + + mask_factor = fPow(ten_fix, DFRACT_BITS - 1 - ten_e, -dbVal_fix, + DFRACT_BITS - 1 - dbVal_e, &q_msk); + q_msk = fixMin(DFRACT_BITS - 1, fixMax(-(DFRACT_BITS - 1), q_msk)); + + if ((q_msk > 0) && (mask_factor > (FIXP_DBL)MAXVAL_DBL >> q_msk)) { + mask_factor = (FIXP_DBL)MAXVAL_DBL; + } else { + mask_factor = scaleValue(mask_factor, q_msk); + } + + return (mask_factor); +} + +static void FDKaacEnc_initSpreading(INT numPb, FIXP_DBL *pbBarcValue, + FIXP_DBL *pbMaskLoFactor, + FIXP_DBL *pbMaskHiFactor, + FIXP_DBL *pbMaskLoFactorSprEn, + FIXP_DBL *pbMaskHiFactorSprEn, + const LONG bitrate, const INT blockType) + +{ + INT i; + FIXP_DBL MASKLOWSPREN, MASKHIGHSPREN; + + FIXP_DBL MASKHIGH = (FIXP_DBL)0x30000000; /* 1.5 in q29 */ + FIXP_DBL MASKLOW = (FIXP_DBL)0x60000000; /* 3.0 in q29 */ + FIXP_DBL MASKLOWSPRENLONG = (FIXP_DBL)0x60000000; /* 3.0 in q29 */ + FIXP_DBL MASKHIGHSPRENLONG = (FIXP_DBL)0x40000000; /* 2.0 in q29 */ + FIXP_DBL MASKHIGHSPRENLONGLOWBR = (FIXP_DBL)0x30000000; /* 1.5 in q29 */ + FIXP_DBL MASKLOWSPRENSHORT = (FIXP_DBL)0x40000000; /* 2.0 in q29 */ + FIXP_DBL MASKHIGHSPRENSHORT = (FIXP_DBL)0x30000000; /* 1.5 in q29 */ + FIXP_DBL TEN = (FIXP_DBL)0x50000000; /* 10.0 in q27 */ + + if (blockType != SHORT_WINDOW) { + MASKLOWSPREN = MASKLOWSPRENLONG; + MASKHIGHSPREN = + (bitrate > 20000) ? MASKHIGHSPRENLONG : MASKHIGHSPRENLONGLOWBR; + } else { + MASKLOWSPREN = MASKLOWSPRENSHORT; + MASKHIGHSPREN = MASKHIGHSPRENSHORT; + } + + for (i = 0; i < numPb; i++) { + if (i > 0) { + pbMaskHiFactor[i] = getMaskFactor( + fMult(MASKHIGH, (pbBarcValue[i] - pbBarcValue[i - 1])), 23, TEN, 27); + + pbMaskLoFactor[i - 1] = getMaskFactor( + fMult(MASKLOW, (pbBarcValue[i] - pbBarcValue[i - 1])), 23, TEN, 27); + + pbMaskHiFactorSprEn[i] = getMaskFactor( + fMult(MASKHIGHSPREN, (pbBarcValue[i] - pbBarcValue[i - 1])), 23, TEN, + 27); + + pbMaskLoFactorSprEn[i - 1] = getMaskFactor( + fMult(MASKLOWSPREN, (pbBarcValue[i] - pbBarcValue[i - 1])), 23, TEN, + 27); + } else { + pbMaskHiFactor[i] = (FIXP_DBL)0; + pbMaskLoFactor[numPb - 1] = (FIXP_DBL)0; + pbMaskHiFactorSprEn[i] = (FIXP_DBL)0; + pbMaskLoFactorSprEn[numPb - 1] = (FIXP_DBL)0; + } + } +} + +static void FDKaacEnc_initBarcValues(INT numPb, INT *pbOffset, INT numLines, + INT samplingFrequency, FIXP_DBL *pbBval) { + INT i; + FIXP_DBL MAX_BARC = (FIXP_DBL)0x30000000; /* 24.0 in q25 */ + + for (i = 0; i < numPb; i++) { + FIXP_DBL v1, v2, cur_bark; + v1 = FDKaacEnc_BarcLineValue(numLines, pbOffset[i], samplingFrequency); + v2 = FDKaacEnc_BarcLineValue(numLines, pbOffset[i + 1], samplingFrequency); + cur_bark = (v1 >> 1) + (v2 >> 1); + pbBval[i] = fixMin(cur_bark, MAX_BARC); + } +} + +static void FDKaacEnc_initMinSnr(const LONG bitrate, const LONG samplerate, + const INT numLines, const INT *sfbOffset, + const INT sfbActive, const INT blockType, + FIXP_DBL *sfbMinSnrLdData) { + INT sfb; + + /* Fix conversion variables */ + INT qbfac, qperwin, qdiv, qpeprt_const, qpeprt; + INT qtmp, qsnr, sfbWidth; + + FIXP_DBL MAX_BARC = (FIXP_DBL)0x30000000; /* 24.0 in q25 */ + FIXP_DBL MAX_BARCP1 = (FIXP_DBL)0x32000000; /* 25.0 in q25 */ + FIXP_DBL BITS2PEFAC = (FIXP_DBL)0x4b851eb8; /* 1.18 in q30 */ + FIXP_DBL PERS2P4 = (FIXP_DBL)0x624dd2f2; /* 0.024 in q36 */ + FIXP_DBL ONEP5 = (FIXP_DBL)0x60000000; /* 1.5 in q30 */ + FIXP_DBL MAX_SNR = (FIXP_DBL)0x33333333; /* 0.8 in q30 */ + FIXP_DBL MIN_SNR = (FIXP_DBL)0x003126e9; /* 0.003 in q30 */ + + FIXP_DBL barcFactor, pePerWindow, pePart, barcWidth; + FIXP_DBL pePart_const, tmp, snr, one_qsnr, one_point5; + + /* relative number of active barks */ + barcFactor = fDivNorm(fixMin(FDKaacEnc_BarcLineValue( + numLines, sfbOffset[sfbActive], samplerate), + MAX_BARC), + MAX_BARCP1, &qbfac); + + qbfac = DFRACT_BITS - 1 - qbfac; + + pePerWindow = fDivNorm(bitrate, samplerate, &qperwin); + qperwin = DFRACT_BITS - 1 - qperwin; + pePerWindow = fMult(pePerWindow, BITS2PEFAC); + qperwin = qperwin + 30 - (DFRACT_BITS - 1); + pePerWindow = fMult(pePerWindow, PERS2P4); + qperwin = qperwin + 36 - (DFRACT_BITS - 1); + + switch (numLines) { + case 1024: + qperwin = qperwin - 10; + break; + case 128: + qperwin = qperwin - 7; + break; + case 512: + qperwin = qperwin - 9; + break; + case 480: + qperwin = qperwin - 9; + pePerWindow = fMult(pePerWindow, FL2FXCONST_DBL(480.f / 512.f)); + break; + case 960: + pePerWindow = fMult(pePerWindow, FL2FXCONST_DBL(960.f/1024.f)); + qperwin = qperwin - 10; + break; + case 120: + pePerWindow = fMult(pePerWindow, FL2FXCONST_DBL(120.f/128.f)); + qperwin = qperwin - 7; + break; + } + + /* for short blocks it is assumed that more bits are available */ + if (blockType == SHORT_WINDOW) { + pePerWindow = fMult(pePerWindow, ONEP5); + qperwin = qperwin + 30 - (DFRACT_BITS - 1); + } + pePart_const = fDivNorm(pePerWindow, barcFactor, &qdiv); + qpeprt_const = qperwin - qbfac + DFRACT_BITS - 1 - qdiv; + + for (sfb = 0; sfb < sfbActive; sfb++) { + barcWidth = + FDKaacEnc_BarcLineValue(numLines, sfbOffset[sfb + 1], samplerate) - + FDKaacEnc_BarcLineValue(numLines, sfbOffset[sfb], samplerate); + + /* adapt to sfb bands */ + pePart = fMult(pePart_const, barcWidth); + qpeprt = qpeprt_const + 25 - (DFRACT_BITS - 1); + + /* pe -> snr calculation */ + sfbWidth = (sfbOffset[sfb + 1] - sfbOffset[sfb]); + pePart = fDivNorm(pePart, sfbWidth, &qdiv); + qpeprt += DFRACT_BITS - 1 - qdiv; + + tmp = f2Pow(pePart, DFRACT_BITS - 1 - qpeprt, &qtmp); + qtmp = DFRACT_BITS - 1 - qtmp; + + /* Subtract 1.5 */ + qsnr = fixMin(qtmp, 30); + tmp = tmp >> (qtmp - qsnr); + + if ((30 + 1 - qsnr) > (DFRACT_BITS - 1)) + one_point5 = (FIXP_DBL)0; + else + one_point5 = (FIXP_DBL)(ONEP5 >> (30 + 1 - qsnr)); + + snr = (tmp >> 1) - (one_point5); + qsnr -= 1; + + /* max(snr, 1.0) */ + if (qsnr > 0) + one_qsnr = (FIXP_DBL)(1 << qsnr); + else + one_qsnr = (FIXP_DBL)0; + + snr = fixMax(one_qsnr, snr); + + /* 1/snr */ + snr = fDivNorm(one_qsnr, snr, &qsnr); + qsnr = DFRACT_BITS - 1 - qsnr; + snr = (qsnr > 30) ? (snr >> (qsnr - 30)) : snr; + + /* upper limit is -1 dB */ + snr = (snr > MAX_SNR) ? MAX_SNR : snr; + + /* lower limit is -25 dB */ + snr = (snr < MIN_SNR) ? MIN_SNR : snr; + snr = snr << 1; + + sfbMinSnrLdData[sfb] = CalcLdData(snr); + } +} + +AAC_ENCODER_ERROR FDKaacEnc_InitPsyConfiguration(INT bitrate, INT samplerate, + INT bandwidth, INT blocktype, + INT granuleLength, INT useIS, + INT useMS, + PSY_CONFIGURATION *psyConf, + FB_TYPE filterbank) { + AAC_ENCODER_ERROR ErrorStatus; + INT sfb; + FIXP_DBL sfbBarcVal[MAX_SFB]; + const INT frameLengthLong = granuleLength; + const INT frameLengthShort = granuleLength / TRANS_FAC; + INT downscaleFactor = 1; + + switch (granuleLength) { + case 256: + case 240: + downscaleFactor = 2; + break; + case 128: + case 120: + downscaleFactor = 4; + break; + default: + downscaleFactor = 1; + break; + } + + FDKmemclear(psyConf, sizeof(PSY_CONFIGURATION)); + psyConf->granuleLength = granuleLength; + psyConf->filterbank = filterbank; + + psyConf->allowIS = (useIS) && ((bitrate / bandwidth) < 5); + psyConf->allowMS = useMS; + + /* init sfb table */ + ErrorStatus = FDKaacEnc_initSfbTable(samplerate * downscaleFactor, blocktype, + granuleLength * downscaleFactor, + psyConf->sfbOffset, &psyConf->sfbCnt); + + if (ErrorStatus != AAC_ENC_OK) return ErrorStatus; + + /* calculate barc values for each pb */ + FDKaacEnc_initBarcValues(psyConf->sfbCnt, psyConf->sfbOffset, + psyConf->sfbOffset[psyConf->sfbCnt], samplerate, + sfbBarcVal); + + FDKaacEnc_InitMinPCMResolution(psyConf->sfbCnt, psyConf->sfbOffset, + psyConf->sfbPcmQuantThreshold); + + /* calculate spreading function */ + FDKaacEnc_initSpreading(psyConf->sfbCnt, sfbBarcVal, + psyConf->sfbMaskLowFactor, psyConf->sfbMaskHighFactor, + psyConf->sfbMaskLowFactorSprEn, + psyConf->sfbMaskHighFactorSprEn, bitrate, blocktype); + + /* init ratio */ + + psyConf->maxAllowedIncreaseFactor = 2; /* integer */ + psyConf->minRemainingThresholdFactor = (FIXP_SGL)0x0148; + /* FL2FXCONST_SGL(0.01f); */ /* fract */ + + psyConf->clipEnergy = + (FIXP_DBL)0x773593ff; /* FL2FXCONST_DBL(1.0e9*NORM_PCM_ENERGY); */ + + if (blocktype != SHORT_WINDOW) { + psyConf->lowpassLine = + (INT)((2 * bandwidth * frameLengthLong) / samplerate); + psyConf->lowpassLineLFE = LFE_LOWPASS_LINE; + } else { + psyConf->lowpassLine = + (INT)((2 * bandwidth * frameLengthShort) / samplerate); + psyConf->lowpassLineLFE = 0; /* LFE only in lonf blocks */ + /* psyConf->clipEnergy /= (TRANS_FAC * TRANS_FAC); */ + psyConf->clipEnergy >>= 6; + } + + for (sfb = 0; sfb < psyConf->sfbCnt; sfb++) { + if (psyConf->sfbOffset[sfb] >= psyConf->lowpassLine) break; + } + psyConf->sfbActive = fMax(sfb, 1); + + for (sfb = 0; sfb < psyConf->sfbCnt; sfb++) { + if (psyConf->sfbOffset[sfb] >= psyConf->lowpassLineLFE) break; + } + psyConf->sfbActiveLFE = sfb; + psyConf->sfbActive = fMax(psyConf->sfbActive, psyConf->sfbActiveLFE); + + /* calculate minSnr */ + FDKaacEnc_initMinSnr(bitrate, samplerate * downscaleFactor, + psyConf->sfbOffset[psyConf->sfbCnt], psyConf->sfbOffset, + psyConf->sfbActive, blocktype, psyConf->sfbMinSnrLdData); + + return AAC_ENC_OK; +} diff --git a/fdk-aac/libAACenc/src/psy_configuration.h b/fdk-aac/libAACenc/src/psy_configuration.h new file mode 100644 index 0000000..52b2887 --- /dev/null +++ b/fdk-aac/libAACenc/src/psy_configuration.h @@ -0,0 +1,171 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M.Werner + + Description: Psychoaccoustic configuration + +*******************************************************************************/ + +#ifndef PSY_CONFIGURATION_H +#define PSY_CONFIGURATION_H + +#include "aacenc.h" +#include "common_fix.h" + +#include "psy_const.h" +#include "aacenc_tns.h" +#include "aacenc_pns.h" + +#define THR_SHIFTBITS 4 +#define PCM_QUANT_THR_SCALE 16 +#define BITS_PER_LINE_SHIFT 3 + +#define C_RATIO \ + (FIXP_DBL)0x02940a10 /* FL2FXCONST_DBL(0.001258925f) << THR_SHIFTBITS; */ /* pow(10.0f, -(29.0f/10.0f)) */ + +typedef struct { + INT sfbCnt; /* number of existing sf bands */ + INT sfbActive; /* number of sf bands containing energy after lowpass */ + INT sfbActiveLFE; + INT sfbOffset[MAX_SFB + 1]; + + INT filterbank; /* LC, LD or ELD */ + + FIXP_DBL sfbPcmQuantThreshold[MAX_SFB]; + + INT maxAllowedIncreaseFactor; /* preecho control */ + FIXP_SGL minRemainingThresholdFactor; + + INT lowpassLine; + INT lowpassLineLFE; + FIXP_DBL clipEnergy; /* for level dependend tmn */ + + FIXP_DBL sfbMaskLowFactor[MAX_SFB]; + FIXP_DBL sfbMaskHighFactor[MAX_SFB]; + + FIXP_DBL sfbMaskLowFactorSprEn[MAX_SFB]; + FIXP_DBL sfbMaskHighFactorSprEn[MAX_SFB]; + + FIXP_DBL sfbMinSnrLdData[MAX_SFB]; /* minimum snr (formerly known as bmax) */ + + TNS_CONFIG tnsConf; + PNS_CONFIG pnsConf; + + INT granuleLength; + INT allowIS; + INT allowMS; +} PSY_CONFIGURATION; + +typedef struct { + UCHAR sfbCnt; /* Number of scalefactor bands */ + UCHAR sfbWidth[MAX_SFB_LONG]; /* Width of scalefactor bands for long blocks */ +} SFB_PARAM_LONG; + +typedef struct { + UCHAR sfbCnt; /* Number of scalefactor bands */ + UCHAR + sfbWidth[MAX_SFB_SHORT]; /* Width of scalefactor bands for short blocks */ +} SFB_PARAM_SHORT; + +AAC_ENCODER_ERROR FDKaacEnc_InitPsyConfiguration(INT bitrate, INT samplerate, + INT bandwidth, INT blocktype, + INT granuleLength, INT useIS, + INT useMS, + PSY_CONFIGURATION *psyConf, + FB_TYPE filterbank); + +#endif /* PSY_CONFIGURATION_H */ diff --git a/fdk-aac/libAACenc/src/psy_const.h b/fdk-aac/libAACenc/src/psy_const.h new file mode 100644 index 0000000..c3f3f64 --- /dev/null +++ b/fdk-aac/libAACenc/src/psy_const.h @@ -0,0 +1,169 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M.Werner + + Description: Global psychoaccoustic constants + +*******************************************************************************/ + +#ifndef PSY_CONST_H +#define PSY_CONST_H + +#define TRUE 1 +#define FALSE 0 + +#define TRANS_FAC 8 /* encoder short long ratio */ + +#define FRAME_LEN_LONG_960 (960) +#define FRAME_MAXLEN_SHORT ((1024) / TRANS_FAC) +#define FRAME_LEN_SHORT_128 ((1024) / TRANS_FAC) +#define FRAME_LEN_SHORT_120 (FRAME_LEN_LONG_960 / TRANS_FAC) + +/* Filterbank type*/ +enum FB_TYPE { FB_LC = 0, FB_LD = 1, FB_ELD = 2 }; + +/* Block types */ +#define N_BLOCKTYPES 6 +enum { + LONG_WINDOW = 0, + START_WINDOW, + SHORT_WINDOW, + STOP_WINDOW, + _LOWOV_WINDOW, /* Do not use this block type out side of block_switch.cpp */ + WRONG_WINDOW +}; + +/* Window shapes */ +enum { + SINE_WINDOW = 0, + KBD_WINDOW = 1, + LOL_WINDOW = 2 /* Low OverLap window shape for AAC-LD */ +}; + +/* + MS stuff +*/ +enum { SI_MS_MASK_NONE = 0, SI_MS_MASK_SOME = 1, SI_MS_MASK_ALL = 2 }; + +#define MAX_NO_OF_GROUPS 4 +#define MAX_SFB_LONG \ + 51 /* 51 for a memory optimized implementation, maybe 64 for convenient \ + debugging */ +#define MAX_SFB_SHORT \ + 15 /* 15 for a memory optimized implementation, maybe 16 for convenient \ + debugging */ + +#define MAX_SFB \ + (MAX_SFB_SHORT > MAX_SFB_LONG ? MAX_SFB_SHORT : MAX_SFB_LONG) /* = 51 */ +#define MAX_GROUPED_SFB \ + (MAX_NO_OF_GROUPS * MAX_SFB_SHORT > MAX_SFB_LONG \ + ? MAX_NO_OF_GROUPS * MAX_SFB_SHORT \ + : MAX_SFB_LONG) /* = 60 */ + +#define MAX_INPUT_BUFFER_SIZE (2 * (1024)) /* 2048 */ + +#define PCM_LEVEL 1.0f +#define NORM_PCM (PCM_LEVEL / 32768.0f) +#define NORM_PCM_ENERGY (NORM_PCM * NORM_PCM) +#define LOG_NORM_PCM -15 + +#define TNS_PREDGAIN_SCALE (1000) + +#define LFE_LOWPASS_LINE 12 +#define LFE_LOWPASS_LINE_MIN 4 + +#endif /* PSY_CONST_H */ diff --git a/fdk-aac/libAACenc/src/psy_data.h b/fdk-aac/libAACenc/src/psy_data.h new file mode 100644 index 0000000..fc04734 --- /dev/null +++ b/fdk-aac/libAACenc/src/psy_data.h @@ -0,0 +1,169 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M.Werner + + Description: Psychoaccoustic data + +*******************************************************************************/ + +#ifndef PSY_DATA_H +#define PSY_DATA_H + +#include "block_switch.h" +#include "mdct.h" + +/* Be careful with MAX_SFB_LONG as length of the .Long arrays. + * sfbEnergy.Long and sfbEnergyMS.Long and sfbThreshold.Long are used as a + * temporary storage for the regrouped short energies and thresholds between + * FDKaacEnc_groupShortData() and BuildInterface() in FDKaacEnc_psyMain(). The + * space required for this is MAX_GROUPED_SFB ( = MAX_NO_OF_GROUPS*MAX_SFB_SHORT + * ). However, this is not important if unions are used (which is not possible + * with pfloat). */ + +typedef shouldBeUnion { + FIXP_DBL Long[MAX_GROUPED_SFB]; + FIXP_DBL Short[TRANS_FAC][MAX_SFB_SHORT]; +} +SFB_THRESHOLD; + +typedef shouldBeUnion { + FIXP_DBL Long[MAX_GROUPED_SFB]; + FIXP_DBL Short[TRANS_FAC][MAX_SFB_SHORT]; +} +SFB_ENERGY; + +typedef shouldBeUnion { + FIXP_DBL Long[MAX_GROUPED_SFB]; + FIXP_DBL Short[TRANS_FAC][MAX_SFB_SHORT]; +} +SFB_LD_ENERGY; + +typedef shouldBeUnion { + INT Long[MAX_GROUPED_SFB]; + INT Short[TRANS_FAC][MAX_SFB_SHORT]; +} +SFB_MAX_SCALE; + +typedef struct { + INT_PCM* psyInputBuffer; + FIXP_DBL overlapAddBuffer[3 * 512 / 2]; + + mdct_t mdctPers; /* MDCT persistent data */ + BLOCK_SWITCHING_CONTROL blockSwitchingControl; /* block switching */ + FIXP_DBL sfbThresholdnm1[MAX_SFB]; /* FDKaacEnc_PreEchoControl */ + INT mdctScalenm1; /* scale of last block's mdct (FDKaacEnc_PreEchoControl) */ + INT calcPreEcho; + INT isLFE; +} PSY_STATIC; + +typedef struct { + FIXP_DBL* mdctSpectrum; + SFB_THRESHOLD sfbThreshold; /* adapt */ + SFB_ENERGY sfbEnergy; /* sfb energies */ + SFB_LD_ENERGY sfbEnergyLdData; /* sfb energies in ldData format */ + SFB_MAX_SCALE sfbMaxScaleSpec; + SFB_ENERGY sfbEnergyMS; /* mid/side sfb energies */ + FIXP_DBL sfbEnergyMSLdData[MAX_GROUPED_SFB]; /* mid/side sfb energies in + ldData format */ + SFB_ENERGY sfbSpreadEnergy; + INT mdctScale; /* exponent of data in mdctSpectrum */ + INT groupedSfbOffset[MAX_GROUPED_SFB + 1]; + INT sfbActive; + INT lowpassLine; +} PSY_DATA; + +#endif /* PSY_DATA_H */ diff --git a/fdk-aac/libAACenc/src/psy_main.cpp b/fdk-aac/libAACenc/src/psy_main.cpp new file mode 100644 index 0000000..f6345e4 --- /dev/null +++ b/fdk-aac/libAACenc/src/psy_main.cpp @@ -0,0 +1,1348 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M.Werner + + Description: Psychoaccoustic major function block + +*******************************************************************************/ + +#include "psy_const.h" + +#include "block_switch.h" +#include "transform.h" +#include "spreading.h" +#include "pre_echo_control.h" +#include "band_nrg.h" +#include "psy_configuration.h" +#include "psy_data.h" +#include "ms_stereo.h" +#include "interface.h" +#include "psy_main.h" +#include "grp_data.h" +#include "tns_func.h" +#include "pns_func.h" +#include "tonality.h" +#include "aacEnc_ram.h" +#include "intensity.h" + +/* blending to reduce gibbs artifacts */ +#define FADE_OUT_LEN 6 +static const FIXP_DBL fadeOutFactor[FADE_OUT_LEN] = { + 1840644096, 1533870080, 1227096064, 920322048, 613548032, 306774016}; + +/* forward definitions */ + +/***************************************************************************** + + functionname: FDKaacEnc_PsyNew + description: allocates memory for psychoacoustic + returns: an error code + input: pointer to a psych handle + +*****************************************************************************/ +AAC_ENCODER_ERROR FDKaacEnc_PsyNew(PSY_INTERNAL **phpsy, const INT nElements, + const INT nChannels, UCHAR *dynamic_RAM) { + AAC_ENCODER_ERROR ErrorStatus; + PSY_INTERNAL *hPsy; + INT i; + + hPsy = GetRam_aacEnc_PsyInternal(); + *phpsy = hPsy; + if (hPsy == NULL) { + ErrorStatus = AAC_ENC_NO_MEMORY; + goto bail; + } + + for (i = 0; i < nElements; i++) { + /* PSY_ELEMENT */ + hPsy->psyElement[i] = GetRam_aacEnc_PsyElement(i); + if (hPsy->psyElement[i] == NULL) { + ErrorStatus = AAC_ENC_NO_MEMORY; + goto bail; + } + } + + for (i = 0; i < nChannels; i++) { + /* PSY_STATIC */ + hPsy->pStaticChannels[i] = GetRam_aacEnc_PsyStatic(i); + if (hPsy->pStaticChannels[i] == NULL) { + ErrorStatus = AAC_ENC_NO_MEMORY; + goto bail; + } + /* AUDIO INPUT BUFFER */ + hPsy->pStaticChannels[i]->psyInputBuffer = GetRam_aacEnc_PsyInputBuffer(i); + if (hPsy->pStaticChannels[i]->psyInputBuffer == NULL) { + ErrorStatus = AAC_ENC_NO_MEMORY; + goto bail; + } + } + + /* reusable psych memory */ + hPsy->psyDynamic = GetRam_aacEnc_PsyDynamic(0, dynamic_RAM); + + return AAC_ENC_OK; + +bail: + FDKaacEnc_PsyClose(phpsy, NULL); + + return ErrorStatus; +} + +/***************************************************************************** + + functionname: FDKaacEnc_PsyOutNew + description: allocates memory for psyOut struc + returns: an error code + input: pointer to a psych handle + +*****************************************************************************/ +AAC_ENCODER_ERROR FDKaacEnc_PsyOutNew(PSY_OUT **phpsyOut, const INT nElements, + const INT nChannels, const INT nSubFrames, + UCHAR *dynamic_RAM) { + AAC_ENCODER_ERROR ErrorStatus; + int n, i; + int elInc = 0, chInc = 0; + + for (n = 0; n < nSubFrames; n++) { + phpsyOut[n] = GetRam_aacEnc_PsyOut(n); + + if (phpsyOut[n] == NULL) { + ErrorStatus = AAC_ENC_NO_MEMORY; + goto bail; + } + + for (i = 0; i < nChannels; i++) { + phpsyOut[n]->pPsyOutChannels[i] = GetRam_aacEnc_PsyOutChannel(chInc++); + if (NULL == phpsyOut[n]->pPsyOutChannels[i]) { + ErrorStatus = AAC_ENC_NO_MEMORY; + goto bail; + } + } + + for (i = 0; i < nElements; i++) { + phpsyOut[n]->psyOutElement[i] = GetRam_aacEnc_PsyOutElements(elInc++); + if (phpsyOut[n]->psyOutElement[i] == NULL) { + ErrorStatus = AAC_ENC_NO_MEMORY; + goto bail; + } + } + } /* nSubFrames */ + + return AAC_ENC_OK; + +bail: + FDKaacEnc_PsyClose(NULL, phpsyOut); + return ErrorStatus; +} + +AAC_ENCODER_ERROR FDKaacEnc_psyInitStates(PSY_INTERNAL *hPsy, + PSY_STATIC *psyStatic, + AUDIO_OBJECT_TYPE audioObjectType) { + /* init input buffer */ + FDKmemclear(psyStatic->psyInputBuffer, + MAX_INPUT_BUFFER_SIZE * sizeof(INT_PCM)); + + FDKaacEnc_InitBlockSwitching(&psyStatic->blockSwitchingControl, + isLowDelay(audioObjectType)); + + return AAC_ENC_OK; +} + +AAC_ENCODER_ERROR FDKaacEnc_psyInit(PSY_INTERNAL *hPsy, PSY_OUT **phpsyOut, + const INT nSubFrames, + const INT nMaxChannels, + const AUDIO_OBJECT_TYPE audioObjectType, + CHANNEL_MAPPING *cm) { + AAC_ENCODER_ERROR ErrorStatus = AAC_ENC_OK; + int i, ch, n, chInc = 0, resetChannels = 3; + + if ((nMaxChannels > 2) && (cm->nChannels == 2)) { + chInc = 1; + FDKaacEnc_psyInitStates(hPsy, hPsy->pStaticChannels[0], audioObjectType); + } + + if ((nMaxChannels == 2)) { + resetChannels = 0; + } + + for (i = 0; i < cm->nElements; i++) { + for (ch = 0; ch < cm->elInfo[i].nChannelsInEl; ch++) { + hPsy->psyElement[i]->psyStatic[ch] = hPsy->pStaticChannels[chInc]; + if (cm->elInfo[i].elType != ID_LFE) { + if (chInc >= resetChannels) { + FDKaacEnc_psyInitStates(hPsy, hPsy->psyElement[i]->psyStatic[ch], + audioObjectType); + } + mdct_init(&(hPsy->psyElement[i]->psyStatic[ch]->mdctPers), NULL, 0); + hPsy->psyElement[i]->psyStatic[ch]->isLFE = 0; + } else { + hPsy->psyElement[i]->psyStatic[ch]->isLFE = 1; + } + chInc++; + } + } + + for (n = 0; n < nSubFrames; n++) { + chInc = 0; + for (i = 0; i < cm->nElements; i++) { + for (ch = 0; ch < cm->elInfo[i].nChannelsInEl; ch++) { + phpsyOut[n]->psyOutElement[i]->psyOutChannel[ch] = + phpsyOut[n]->pPsyOutChannels[chInc++]; + } + } + } + + return ErrorStatus; +} + +/***************************************************************************** + + functionname: FDKaacEnc_psyMainInit + description: initializes psychoacoustic + returns: an error code + +*****************************************************************************/ + +AAC_ENCODER_ERROR FDKaacEnc_psyMainInit( + PSY_INTERNAL *hPsy, AUDIO_OBJECT_TYPE audioObjectType, CHANNEL_MAPPING *cm, + INT sampleRate, INT granuleLength, INT bitRate, INT tnsMask, INT bandwidth, + INT usePns, INT useIS, INT useMS, UINT syntaxFlags, ULONG initFlags) { + AAC_ENCODER_ERROR ErrorStatus; + int i, ch; + int channelsEff = cm->nChannelsEff; + int tnsChannels = 0; + FB_TYPE filterBank; + + switch (FDKaacEnc_GetMonoStereoMode(cm->encMode)) { + /* ... and map to tnsChannels */ + case EL_MODE_MONO: + tnsChannels = 1; + break; + case EL_MODE_STEREO: + tnsChannels = 2; + break; + default: + tnsChannels = 0; + } + + switch (audioObjectType) { + default: + filterBank = FB_LC; + break; + case AOT_ER_AAC_LD: + filterBank = FB_LD; + break; + case AOT_ER_AAC_ELD: + filterBank = FB_ELD; + break; + } + + hPsy->granuleLength = granuleLength; + + ErrorStatus = FDKaacEnc_InitPsyConfiguration( + bitRate / channelsEff, sampleRate, bandwidth, LONG_WINDOW, + hPsy->granuleLength, useIS, useMS, &(hPsy->psyConf[0]), filterBank); + if (ErrorStatus != AAC_ENC_OK) return ErrorStatus; + + ErrorStatus = FDKaacEnc_InitTnsConfiguration( + (bitRate * tnsChannels) / channelsEff, sampleRate, tnsChannels, + LONG_WINDOW, hPsy->granuleLength, isLowDelay(audioObjectType), + (syntaxFlags & AC_SBR_PRESENT) ? 1 : 0, &(hPsy->psyConf[0].tnsConf), + &hPsy->psyConf[0], (INT)(tnsMask & 2), (INT)(tnsMask & 8)); + + if (ErrorStatus != AAC_ENC_OK) return ErrorStatus; + + if (granuleLength > 512) { + ErrorStatus = FDKaacEnc_InitPsyConfiguration( + bitRate / channelsEff, sampleRate, bandwidth, SHORT_WINDOW, + hPsy->granuleLength, useIS, useMS, &hPsy->psyConf[1], filterBank); + + if (ErrorStatus != AAC_ENC_OK) return ErrorStatus; + + ErrorStatus = FDKaacEnc_InitTnsConfiguration( + (bitRate * tnsChannels) / channelsEff, sampleRate, tnsChannels, + SHORT_WINDOW, hPsy->granuleLength, isLowDelay(audioObjectType), + (syntaxFlags & AC_SBR_PRESENT) ? 1 : 0, &hPsy->psyConf[1].tnsConf, + &hPsy->psyConf[1], (INT)(tnsMask & 1), (INT)(tnsMask & 4)); + + if (ErrorStatus != AAC_ENC_OK) return ErrorStatus; + } + + for (i = 0; i < cm->nElements; i++) { + for (ch = 0; ch < cm->elInfo[i].nChannelsInEl; ch++) { + if (initFlags) { + /* reset states */ + FDKaacEnc_psyInitStates(hPsy, hPsy->psyElement[i]->psyStatic[ch], + audioObjectType); + } + + FDKaacEnc_InitPreEchoControl( + hPsy->psyElement[i]->psyStatic[ch]->sfbThresholdnm1, + &hPsy->psyElement[i]->psyStatic[ch]->calcPreEcho, + hPsy->psyConf[0].sfbCnt, hPsy->psyConf[0].sfbPcmQuantThreshold, + &hPsy->psyElement[i]->psyStatic[ch]->mdctScalenm1); + } + } + + ErrorStatus = FDKaacEnc_InitPnsConfiguration( + &hPsy->psyConf[0].pnsConf, bitRate / channelsEff, sampleRate, usePns, + hPsy->psyConf[0].sfbCnt, hPsy->psyConf[0].sfbOffset, + cm->elInfo[0].nChannelsInEl, (hPsy->psyConf[0].filterbank == FB_LC)); + if (ErrorStatus != AAC_ENC_OK) return ErrorStatus; + + if (granuleLength > 512) { + ErrorStatus = FDKaacEnc_InitPnsConfiguration( + &hPsy->psyConf[1].pnsConf, bitRate / channelsEff, sampleRate, usePns, + hPsy->psyConf[1].sfbCnt, hPsy->psyConf[1].sfbOffset, + cm->elInfo[1].nChannelsInEl, (hPsy->psyConf[1].filterbank == FB_LC)); + if (ErrorStatus != AAC_ENC_OK) return ErrorStatus; + } + + return ErrorStatus; +} + +/***************************************************************************** + + functionname: FDKaacEnc_psyMain + description: psychoacoustic + returns: an error code + + This function assumes that enough input data is in the modulo buffer. + +*****************************************************************************/ +AAC_ENCODER_ERROR FDKaacEnc_psyMain(INT channels, PSY_ELEMENT *psyElement, + PSY_DYNAMIC *psyDynamic, + PSY_CONFIGURATION *psyConf, + PSY_OUT_ELEMENT *RESTRICT psyOutElement, + INT_PCM *pInput, const UINT inputBufSize, + INT *chIdx, INT totalChannels) { + const INT commonWindow = 1; + INT maxSfbPerGroup[(2)]; + INT mdctSpectrum_e; + INT ch; /* counts through channels */ + INT w; /* counts through windows */ + INT sfb; /* counts through scalefactor bands */ + INT line; /* counts through lines */ + + PSY_CONFIGURATION *RESTRICT hPsyConfLong = &psyConf[0]; + PSY_CONFIGURATION *RESTRICT hPsyConfShort = &psyConf[1]; + PSY_OUT_CHANNEL **RESTRICT psyOutChannel = psyOutElement->psyOutChannel; + FIXP_SGL sfbTonality[(2)][MAX_SFB_LONG]; + + PSY_STATIC **RESTRICT psyStatic = psyElement->psyStatic; + + PSY_DATA *RESTRICT psyData[(2)]; + TNS_DATA *RESTRICT tnsData[(2)]; + PNS_DATA *RESTRICT pnsData[(2)]; + + INT zeroSpec = TRUE; /* means all spectral lines are zero */ + + INT blockSwitchingOffset; + + PSY_CONFIGURATION *RESTRICT hThisPsyConf[(2)]; + INT windowLength[(2)]; + INT nWindows[(2)]; + INT wOffset; + + INT maxSfb[(2)]; + INT *pSfbMaxScaleSpec[(2)]; + FIXP_DBL *pSfbEnergy[(2)]; + FIXP_DBL *pSfbSpreadEnergy[(2)]; + FIXP_DBL *pSfbEnergyLdData[(2)]; + FIXP_DBL *pSfbEnergyMS[(2)]; + FIXP_DBL *pSfbThreshold[(2)]; + + INT isShortWindow[(2)]; + + /* number of incoming time samples to be processed */ + const INT nTimeSamples = psyConf->granuleLength; + + switch (hPsyConfLong->filterbank) { + case FB_LC: + blockSwitchingOffset = + nTimeSamples + (9 * nTimeSamples / (2 * TRANS_FAC)); + break; + case FB_LD: + case FB_ELD: + blockSwitchingOffset = nTimeSamples; + break; + default: + return AAC_ENC_UNSUPPORTED_FILTERBANK; + } + + for (ch = 0; ch < channels; ch++) { + psyData[ch] = &psyDynamic->psyData[ch]; + tnsData[ch] = &psyDynamic->tnsData[ch]; + pnsData[ch] = &psyDynamic->pnsData[ch]; + + psyData[ch]->mdctSpectrum = psyOutChannel[ch]->mdctSpectrum; + } + + /* block switching */ + if (hPsyConfLong->filterbank != FB_ELD) { + int err; + + for (ch = 0; ch < channels; ch++) { + C_ALLOC_SCRATCH_START(pTimeSignal, INT_PCM, (1024)) + + /* copy input data and use for block switching */ + FDKmemcpy(pTimeSignal, pInput + chIdx[ch] * inputBufSize, + nTimeSamples * sizeof(INT_PCM)); + + FDKaacEnc_BlockSwitching(&psyStatic[ch]->blockSwitchingControl, + nTimeSamples, psyStatic[ch]->isLFE, pTimeSignal); + + /* fill up internal input buffer, to 2xframelength samples */ + FDKmemcpy(psyStatic[ch]->psyInputBuffer + blockSwitchingOffset, + pTimeSignal, + (2 * nTimeSamples - blockSwitchingOffset) * sizeof(INT_PCM)); + + C_ALLOC_SCRATCH_END(pTimeSignal, INT_PCM, (1024)) + } + + /* synch left and right block type */ + err = FDKaacEnc_SyncBlockSwitching( + &psyStatic[0]->blockSwitchingControl, + (channels > 1) ? &psyStatic[1]->blockSwitchingControl : NULL, channels, + commonWindow); + + if (err) { + return AAC_ENC_UNSUPPORTED_AOT; /* mixed up LC and LD */ + } + + } else { + for (ch = 0; ch < channels; ch++) { + /* copy input data and use for block switching */ + FDKmemcpy(psyStatic[ch]->psyInputBuffer + blockSwitchingOffset, + pInput + chIdx[ch] * inputBufSize, + nTimeSamples * sizeof(INT_PCM)); + } + } + + for (ch = 0; ch < channels; ch++) + isShortWindow[ch] = + (psyStatic[ch]->blockSwitchingControl.lastWindowSequence == + SHORT_WINDOW); + + /* set parameters according to window length */ + for (ch = 0; ch < channels; ch++) { + if (isShortWindow[ch]) { + hThisPsyConf[ch] = hPsyConfShort; + windowLength[ch] = psyConf->granuleLength / TRANS_FAC; + nWindows[ch] = TRANS_FAC; + maxSfb[ch] = MAX_SFB_SHORT; + + pSfbMaxScaleSpec[ch] = psyData[ch]->sfbMaxScaleSpec.Short[0]; + pSfbEnergy[ch] = psyData[ch]->sfbEnergy.Short[0]; + pSfbSpreadEnergy[ch] = psyData[ch]->sfbSpreadEnergy.Short[0]; + pSfbEnergyLdData[ch] = psyData[ch]->sfbEnergyLdData.Short[0]; + pSfbEnergyMS[ch] = psyData[ch]->sfbEnergyMS.Short[0]; + pSfbThreshold[ch] = psyData[ch]->sfbThreshold.Short[0]; + + } else { + hThisPsyConf[ch] = hPsyConfLong; + windowLength[ch] = psyConf->granuleLength; + nWindows[ch] = 1; + maxSfb[ch] = MAX_GROUPED_SFB; + + pSfbMaxScaleSpec[ch] = psyData[ch]->sfbMaxScaleSpec.Long; + pSfbEnergy[ch] = psyData[ch]->sfbEnergy.Long; + pSfbSpreadEnergy[ch] = psyData[ch]->sfbSpreadEnergy.Long; + pSfbEnergyLdData[ch] = psyData[ch]->sfbEnergyLdData.Long; + pSfbEnergyMS[ch] = psyData[ch]->sfbEnergyMS.Long; + pSfbThreshold[ch] = psyData[ch]->sfbThreshold.Long; + } + } + + /* Transform and get mdctScaling for all channels and windows. */ + for (ch = 0; ch < channels; ch++) { + /* update number of active bands */ + if (psyStatic[ch]->isLFE) { + psyData[ch]->sfbActive = hThisPsyConf[ch]->sfbActiveLFE; + psyData[ch]->lowpassLine = hThisPsyConf[ch]->lowpassLineLFE; + } else { + psyData[ch]->sfbActive = hThisPsyConf[ch]->sfbActive; + psyData[ch]->lowpassLine = hThisPsyConf[ch]->lowpassLine; + } + + if (hThisPsyConf[ch]->filterbank == FB_ELD) { + if (FDKaacEnc_Transform_Real_Eld( + psyStatic[ch]->psyInputBuffer, psyData[ch]->mdctSpectrum, + psyStatic[ch]->blockSwitchingControl.lastWindowSequence, + psyStatic[ch]->blockSwitchingControl.windowShape, + &psyStatic[ch]->blockSwitchingControl.lastWindowShape, + nTimeSamples, &mdctSpectrum_e, hThisPsyConf[ch]->filterbank, + psyStatic[ch]->overlapAddBuffer) != 0) { + return AAC_ENC_UNSUPPORTED_FILTERBANK; + } + } else { + if (FDKaacEnc_Transform_Real( + psyStatic[ch]->psyInputBuffer, psyData[ch]->mdctSpectrum, + psyStatic[ch]->blockSwitchingControl.lastWindowSequence, + psyStatic[ch]->blockSwitchingControl.windowShape, + &psyStatic[ch]->blockSwitchingControl.lastWindowShape, + &psyStatic[ch]->mdctPers, nTimeSamples, &mdctSpectrum_e, + hThisPsyConf[ch]->filterbank) != 0) { + return AAC_ENC_UNSUPPORTED_FILTERBANK; + } + } + + for (w = 0; w < nWindows[ch]; w++) { + wOffset = w * windowLength[ch]; + + /* Low pass / highest sfb */ + FDKmemclear( + &psyData[ch]->mdctSpectrum[psyData[ch]->lowpassLine + wOffset], + (windowLength[ch] - psyData[ch]->lowpassLine) * sizeof(FIXP_DBL)); + + if ((hPsyConfLong->filterbank != FB_LC) && + (psyData[ch]->lowpassLine >= FADE_OUT_LEN)) { + /* Do blending to reduce gibbs artifacts */ + for (int i = 0; i < FADE_OUT_LEN; i++) { + psyData[ch]->mdctSpectrum[psyData[ch]->lowpassLine + wOffset - + FADE_OUT_LEN + i] = + fMult(psyData[ch]->mdctSpectrum[psyData[ch]->lowpassLine + + wOffset - FADE_OUT_LEN + i], + fadeOutFactor[i]); + } + } + + /* Check for zero spectrum. These loops will usually terminate very, very + * early. */ + for (line = 0; (line < psyData[ch]->lowpassLine) && (zeroSpec == TRUE); + line++) { + if (psyData[ch]->mdctSpectrum[line + wOffset] != (FIXP_DBL)0) { + zeroSpec = FALSE; + break; + } + } + + } /* w loop */ + + psyData[ch]->mdctScale = mdctSpectrum_e; + + /* rotate internal time samples */ + FDKmemmove(psyStatic[ch]->psyInputBuffer, + psyStatic[ch]->psyInputBuffer + nTimeSamples, + nTimeSamples * sizeof(INT_PCM)); + + /* ... and get remaining samples from input buffer */ + FDKmemcpy(psyStatic[ch]->psyInputBuffer + nTimeSamples, + pInput + (2 * nTimeSamples - blockSwitchingOffset) + + chIdx[ch] * inputBufSize, + (blockSwitchingOffset - nTimeSamples) * sizeof(INT_PCM)); + + } /* ch */ + + /* Do some rescaling to get maximum possible accuracy for energies */ + if (zeroSpec == FALSE) { + /* Calc possible spectrum leftshift for each sfb (1 means: 1 bit left shift + * is possible without overflow) */ + INT minSpecShift = MAX_SHIFT_DBL; + INT nrgShift = MAX_SHIFT_DBL; + INT finalShift = MAX_SHIFT_DBL; + FIXP_DBL currNrg = 0; + FIXP_DBL maxNrg = 0; + + for (ch = 0; ch < channels; ch++) { + for (w = 0; w < nWindows[ch]; w++) { + wOffset = w * windowLength[ch]; + FDKaacEnc_CalcSfbMaxScaleSpec( + psyData[ch]->mdctSpectrum + wOffset, hThisPsyConf[ch]->sfbOffset, + pSfbMaxScaleSpec[ch] + w * maxSfb[ch], psyData[ch]->sfbActive); + + for (sfb = 0; sfb < psyData[ch]->sfbActive; sfb++) + minSpecShift = fixMin(minSpecShift, + (pSfbMaxScaleSpec[ch] + w * maxSfb[ch])[sfb]); + } + } + + /* Calc possible energy leftshift for each sfb (1 means: 1 bit left shift is + * possible without overflow) */ + for (ch = 0; ch < channels; ch++) { + for (w = 0; w < nWindows[ch]; w++) { + wOffset = w * windowLength[ch]; + currNrg = FDKaacEnc_CheckBandEnergyOptim( + psyData[ch]->mdctSpectrum + wOffset, + pSfbMaxScaleSpec[ch] + w * maxSfb[ch], hThisPsyConf[ch]->sfbOffset, + psyData[ch]->sfbActive, pSfbEnergy[ch] + w * maxSfb[ch], + pSfbEnergyLdData[ch] + w * maxSfb[ch], minSpecShift - 4); + + maxNrg = fixMax(maxNrg, currNrg); + } + } + + if (maxNrg != (FIXP_DBL)0) { + nrgShift = (CountLeadingBits(maxNrg) >> 1) + (minSpecShift - 4); + } + + /* 2check: Hasn't this decision to be made for both channels? */ + /* For short windows 1 additional bit headroom is necessary to prevent + * overflows when summing up energies in FDKaacEnc_groupShortData() */ + if (isShortWindow[0]) nrgShift--; + + /* both spectrum and energies mustn't overflow */ + finalShift = fixMin(minSpecShift, nrgShift); + + /* do not shift more than 3 bits more to the left than signal without + * blockfloating point would be to avoid overflow of scaled PCM quantization + * thresholds */ + if (finalShift > psyData[0]->mdctScale + 3) + finalShift = psyData[0]->mdctScale + 3; + + FDK_ASSERT(finalShift >= 0); /* right shift is not allowed */ + + /* correct sfbEnergy and sfbEnergyLdData with new finalShift */ + FIXP_DBL ldShift = finalShift * FL2FXCONST_DBL(2.0 / 64); + for (ch = 0; ch < channels; ch++) { + INT maxSfb_ch = maxSfb[ch]; + INT w_maxSfb_ch = 0; + for (w = 0; w < nWindows[ch]; w++) { + for (sfb = 0; sfb < psyData[ch]->sfbActive; sfb++) { + INT scale = fixMax(0, (pSfbMaxScaleSpec[ch] + w_maxSfb_ch)[sfb] - 4); + scale = fixMin((scale - finalShift) << 1, DFRACT_BITS - 1); + if (scale >= 0) + (pSfbEnergy[ch] + w_maxSfb_ch)[sfb] >>= (scale); + else + (pSfbEnergy[ch] + w_maxSfb_ch)[sfb] <<= (-scale); + (pSfbThreshold[ch] + w_maxSfb_ch)[sfb] = + fMult((pSfbEnergy[ch] + w_maxSfb_ch)[sfb], C_RATIO); + (pSfbEnergyLdData[ch] + w_maxSfb_ch)[sfb] += ldShift; + } + w_maxSfb_ch += maxSfb_ch; + } + } + + if (finalShift != 0) { + for (ch = 0; ch < channels; ch++) { + INT wLen = windowLength[ch]; + INT lowpassLine = psyData[ch]->lowpassLine; + wOffset = 0; + FIXP_DBL *mdctSpectrum = &psyData[ch]->mdctSpectrum[0]; + for (w = 0; w < nWindows[ch]; w++) { + FIXP_DBL *spectrum = &mdctSpectrum[wOffset]; + for (line = 0; line < lowpassLine; line++) { + spectrum[line] <<= finalShift; + } + wOffset += wLen; + + /* update sfbMaxScaleSpec */ + for (sfb = 0; sfb < psyData[ch]->sfbActive; sfb++) + (pSfbMaxScaleSpec[ch] + w * maxSfb[ch])[sfb] -= finalShift; + } + /* update mdctScale */ + psyData[ch]->mdctScale -= finalShift; + } + } + + } else { + /* all spectral lines are zero */ + for (ch = 0; ch < channels; ch++) { + psyData[ch]->mdctScale = + 0; /* otherwise mdctScale would be for example 7 and PCM quantization + * thresholds would be shifted 14 bits to the right causing some of + * them to become 0 (which causes problems later) */ + /* clear sfbMaxScaleSpec */ + for (w = 0; w < nWindows[ch]; w++) { + for (sfb = 0; sfb < psyData[ch]->sfbActive; sfb++) { + (pSfbMaxScaleSpec[ch] + w * maxSfb[ch])[sfb] = 0; + (pSfbEnergy[ch] + w * maxSfb[ch])[sfb] = (FIXP_DBL)0; + (pSfbEnergyLdData[ch] + w * maxSfb[ch])[sfb] = FL2FXCONST_DBL(-1.0f); + (pSfbThreshold[ch] + w * maxSfb[ch])[sfb] = (FIXP_DBL)0; + } + } + } + } + + /* Advance psychoacoustics: Tonality and TNS */ + if ((channels >= 1) && (psyStatic[0]->isLFE)) { + tnsData[0]->dataRaw.Long.subBlockInfo.tnsActive[HIFILT] = 0; + tnsData[0]->dataRaw.Long.subBlockInfo.tnsActive[LOFILT] = 0; + } else { + for (ch = 0; ch < channels; ch++) { + if (!isShortWindow[ch]) { + /* tonality */ + FDKaacEnc_CalculateFullTonality( + psyData[ch]->mdctSpectrum, pSfbMaxScaleSpec[ch], + pSfbEnergyLdData[ch], sfbTonality[ch], psyData[ch]->sfbActive, + hThisPsyConf[ch]->sfbOffset, hThisPsyConf[ch]->pnsConf.usePns); + } + } /* ch */ + + if (hPsyConfLong->tnsConf.tnsActive || hPsyConfShort->tnsConf.tnsActive) { + INT tnsActive[TRANS_FAC] = {0}; + INT nrgScaling[2] = {0, 0}; + INT tnsSpecShift = 0; + + for (ch = 0; ch < channels; ch++) { + for (w = 0; w < nWindows[ch]; w++) { + wOffset = w * windowLength[ch]; + /* TNS */ + FDKaacEnc_TnsDetect( + tnsData[ch], &hThisPsyConf[ch]->tnsConf, + &psyOutChannel[ch]->tnsInfo, hThisPsyConf[ch]->sfbCnt, + psyData[ch]->mdctSpectrum + wOffset, w, + psyStatic[ch]->blockSwitchingControl.lastWindowSequence); + } + } + + if (channels == 2) { + FDKaacEnc_TnsSync( + tnsData[1], tnsData[0], &psyOutChannel[1]->tnsInfo, + &psyOutChannel[0]->tnsInfo, + + psyStatic[1]->blockSwitchingControl.lastWindowSequence, + psyStatic[0]->blockSwitchingControl.lastWindowSequence, + &hThisPsyConf[1]->tnsConf); + } + + if (channels >= 1) { + FDK_ASSERT(1 == commonWindow); /* all checks for TNS do only work for + common windows (which is always set)*/ + for (w = 0; w < nWindows[0]; w++) { + if (isShortWindow[0]) + tnsActive[w] = + tnsData[0]->dataRaw.Short.subBlockInfo[w].tnsActive[HIFILT] || + tnsData[0]->dataRaw.Short.subBlockInfo[w].tnsActive[LOFILT] || + tnsData[channels - 1] + ->dataRaw.Short.subBlockInfo[w] + .tnsActive[HIFILT] || + tnsData[channels - 1] + ->dataRaw.Short.subBlockInfo[w] + .tnsActive[LOFILT]; + else + tnsActive[w] = + tnsData[0]->dataRaw.Long.subBlockInfo.tnsActive[HIFILT] || + tnsData[0]->dataRaw.Long.subBlockInfo.tnsActive[LOFILT] || + tnsData[channels - 1] + ->dataRaw.Long.subBlockInfo.tnsActive[HIFILT] || + tnsData[channels - 1] + ->dataRaw.Long.subBlockInfo.tnsActive[LOFILT]; + } + } + + for (ch = 0; ch < channels; ch++) { + if (tnsActive[0] && !isShortWindow[ch]) { + /* Scale down spectrum if tns is active in one of the two channels + * with same lastWindowSequence */ + /* first part of threshold calculation; it's not necessary to update + * sfbMaxScaleSpec */ + INT shift = 1; + for (sfb = 0; sfb < hThisPsyConf[ch]->lowpassLine; sfb++) { + psyData[ch]->mdctSpectrum[sfb] = + psyData[ch]->mdctSpectrum[sfb] >> shift; + } + + /* update thresholds */ + for (sfb = 0; sfb < psyData[ch]->sfbActive; sfb++) { + pSfbThreshold[ch][sfb] >>= (2 * shift); + } + + psyData[ch]->mdctScale += shift; /* update mdctScale */ + + /* calc sfbEnergies after tnsEncode again ! */ + } + } + + for (ch = 0; ch < channels; ch++) { + for (w = 0; w < nWindows[ch]; w++) { + wOffset = w * windowLength[ch]; + FDKaacEnc_TnsEncode( + &psyOutChannel[ch]->tnsInfo, tnsData[ch], + hThisPsyConf[ch]->sfbCnt, &hThisPsyConf[ch]->tnsConf, + hThisPsyConf[ch]->sfbOffset[psyData[ch]->sfbActive], + /*hThisPsyConf[ch]->lowpassLine*/ /* filter stops + before that + line ! */ + psyData[ch]->mdctSpectrum + + wOffset, + w, psyStatic[ch]->blockSwitchingControl.lastWindowSequence); + + if (tnsActive[w]) { + /* Calc sfb-bandwise mdct-energies for left and right channel again, + */ + /* if tns active in current channel or in one channel with same + * lastWindowSequence left and right */ + FDKaacEnc_CalcSfbMaxScaleSpec(psyData[ch]->mdctSpectrum + wOffset, + hThisPsyConf[ch]->sfbOffset, + pSfbMaxScaleSpec[ch] + w * maxSfb[ch], + psyData[ch]->sfbActive); + } + } + } + + for (ch = 0; ch < channels; ch++) { + for (w = 0; w < nWindows[ch]; w++) { + if (tnsActive[w]) { + if (isShortWindow[ch]) { + FDKaacEnc_CalcBandEnergyOptimShort( + psyData[ch]->mdctSpectrum + w * windowLength[ch], + pSfbMaxScaleSpec[ch] + w * maxSfb[ch], + hThisPsyConf[ch]->sfbOffset, psyData[ch]->sfbActive, + pSfbEnergy[ch] + w * maxSfb[ch]); + } else { + nrgScaling[ch] = /* with tns, energy calculation can overflow; -> + scaling */ + FDKaacEnc_CalcBandEnergyOptimLong( + psyData[ch]->mdctSpectrum, pSfbMaxScaleSpec[ch], + hThisPsyConf[ch]->sfbOffset, psyData[ch]->sfbActive, + pSfbEnergy[ch], pSfbEnergyLdData[ch]); + tnsSpecShift = + fixMax(tnsSpecShift, nrgScaling[ch]); /* nrgScaling is set + only if nrg would + have an overflow */ + } + } /* if tnsActive */ + } + } /* end channel loop */ + + /* adapt scaling to prevent nrg overflow, only for long blocks */ + for (ch = 0; ch < channels; ch++) { + if ((tnsSpecShift != 0) && !isShortWindow[ch]) { + /* scale down spectrum, nrg's and thresholds, if there was an overflow + * in sfbNrg calculation after tns */ + for (line = 0; line < hThisPsyConf[ch]->lowpassLine; line++) { + psyData[ch]->mdctSpectrum[line] >>= tnsSpecShift; + } + INT scale = (tnsSpecShift - nrgScaling[ch]) << 1; + for (sfb = 0; sfb < psyData[ch]->sfbActive; sfb++) { + pSfbEnergyLdData[ch][sfb] -= + scale * FL2FXCONST_DBL(1.0 / LD_DATA_SCALING); + pSfbEnergy[ch][sfb] >>= scale; + pSfbThreshold[ch][sfb] >>= (tnsSpecShift << 1); + } + psyData[ch]->mdctScale += tnsSpecShift; /* update mdctScale; not + necessary to update + sfbMaxScaleSpec */ + } + } /* end channel loop */ + + } /* TNS active */ + else { + /* In case of disable TNS, reset its dynamic data. Some of its elements is + * required in PNS detection below. */ + FDKmemclear(psyDynamic->tnsData, sizeof(psyDynamic->tnsData)); + } + } /* !isLFE */ + + /* Advance thresholds */ + for (ch = 0; ch < channels; ch++) { + INT headroom; + + FIXP_DBL clipEnergy; + INT energyShift = psyData[ch]->mdctScale * 2; + INT clipNrgShift = energyShift - THR_SHIFTBITS; + if (isShortWindow[ch]) + headroom = 6; + else + headroom = 0; + + if (clipNrgShift >= 0) + clipEnergy = hThisPsyConf[ch]->clipEnergy >> clipNrgShift; + else if (clipNrgShift >= -headroom) + clipEnergy = hThisPsyConf[ch]->clipEnergy << -clipNrgShift; + else + clipEnergy = (FIXP_DBL)MAXVAL_DBL; + + for (w = 0; w < nWindows[ch]; w++) { + INT i; + /* limit threshold to avoid clipping */ + for (i = 0; i < psyData[ch]->sfbActive; i++) { + *(pSfbThreshold[ch] + w * maxSfb[ch] + i) = + fixMin(*(pSfbThreshold[ch] + w * maxSfb[ch] + i), clipEnergy); + } + + /* spreading */ + FDKaacEnc_SpreadingMax(psyData[ch]->sfbActive, + hThisPsyConf[ch]->sfbMaskLowFactor, + hThisPsyConf[ch]->sfbMaskHighFactor, + pSfbThreshold[ch] + w * maxSfb[ch]); + + /* PCM quantization threshold */ + energyShift += PCM_QUANT_THR_SCALE; + if (energyShift >= 0) { + energyShift = fixMin(DFRACT_BITS - 1, energyShift); + for (i = 0; i < psyData[ch]->sfbActive; i++) { + *(pSfbThreshold[ch] + w * maxSfb[ch] + i) = fixMax( + *(pSfbThreshold[ch] + w * maxSfb[ch] + i) >> THR_SHIFTBITS, + (hThisPsyConf[ch]->sfbPcmQuantThreshold[i] >> energyShift)); + } + } else { + energyShift = fixMin(DFRACT_BITS - 1, -energyShift); + for (i = 0; i < psyData[ch]->sfbActive; i++) { + *(pSfbThreshold[ch] + w * maxSfb[ch] + i) = fixMax( + *(pSfbThreshold[ch] + w * maxSfb[ch] + i) >> THR_SHIFTBITS, + (hThisPsyConf[ch]->sfbPcmQuantThreshold[i] << energyShift)); + } + } + + if (!psyStatic[ch]->isLFE) { + /* preecho control */ + if (psyStatic[ch]->blockSwitchingControl.lastWindowSequence == + STOP_WINDOW) { + /* prevent FDKaacEnc_PreEchoControl from comparing stop + thresholds with short thresholds */ + for (i = 0; i < psyData[ch]->sfbActive; i++) { + psyStatic[ch]->sfbThresholdnm1[i] = (FIXP_DBL)MAXVAL_DBL; + } + + psyStatic[ch]->mdctScalenm1 = 0; + psyStatic[ch]->calcPreEcho = 0; + } + + FDKaacEnc_PreEchoControl( + psyStatic[ch]->sfbThresholdnm1, psyStatic[ch]->calcPreEcho, + psyData[ch]->sfbActive, hThisPsyConf[ch]->maxAllowedIncreaseFactor, + hThisPsyConf[ch]->minRemainingThresholdFactor, + pSfbThreshold[ch] + w * maxSfb[ch], psyData[ch]->mdctScale, + &psyStatic[ch]->mdctScalenm1); + + psyStatic[ch]->calcPreEcho = 1; + + if (psyStatic[ch]->blockSwitchingControl.lastWindowSequence == + START_WINDOW) { + /* prevent FDKaacEnc_PreEchoControl in next frame to compare start + thresholds with short thresholds */ + for (i = 0; i < psyData[ch]->sfbActive; i++) { + psyStatic[ch]->sfbThresholdnm1[i] = (FIXP_DBL)MAXVAL_DBL; + } + + psyStatic[ch]->mdctScalenm1 = 0; + psyStatic[ch]->calcPreEcho = 0; + } + } + + /* spread energy to avoid hole detection */ + FDKmemcpy(pSfbSpreadEnergy[ch] + w * maxSfb[ch], + pSfbEnergy[ch] + w * maxSfb[ch], + psyData[ch]->sfbActive * sizeof(FIXP_DBL)); + + FDKaacEnc_SpreadingMax(psyData[ch]->sfbActive, + hThisPsyConf[ch]->sfbMaskLowFactorSprEn, + hThisPsyConf[ch]->sfbMaskHighFactorSprEn, + pSfbSpreadEnergy[ch] + w * maxSfb[ch]); + } + } + + /* Calc bandwise energies for mid and side channel. Do it only if 2 channels + * exist */ + if (channels == 2) { + for (w = 0; w < nWindows[1]; w++) { + wOffset = w * windowLength[1]; + FDKaacEnc_CalcBandNrgMSOpt( + psyData[0]->mdctSpectrum + wOffset, + psyData[1]->mdctSpectrum + wOffset, + pSfbMaxScaleSpec[0] + w * maxSfb[0], + pSfbMaxScaleSpec[1] + w * maxSfb[1], hThisPsyConf[1]->sfbOffset, + psyData[0]->sfbActive, pSfbEnergyMS[0] + w * maxSfb[0], + pSfbEnergyMS[1] + w * maxSfb[1], + (psyStatic[1]->blockSwitchingControl.lastWindowSequence != + SHORT_WINDOW), + psyData[0]->sfbEnergyMSLdData, psyData[1]->sfbEnergyMSLdData); + } + } + + /* group short data (maxSfb[ch] for short blocks is determined here) */ + for (ch = 0; ch < channels; ch++) { + if (isShortWindow[ch]) { + int sfbGrp; + int noSfb = psyStatic[ch]->blockSwitchingControl.noOfGroups * + hPsyConfShort->sfbCnt; + /* At this point, energies and thresholds are copied/regrouped from the + * ".Short" to the ".Long" arrays */ + FDKaacEnc_groupShortData( + psyData[ch]->mdctSpectrum, &psyData[ch]->sfbThreshold, + &psyData[ch]->sfbEnergy, &psyData[ch]->sfbEnergyMS, + &psyData[ch]->sfbSpreadEnergy, hPsyConfShort->sfbCnt, + psyData[ch]->sfbActive, hPsyConfShort->sfbOffset, + hPsyConfShort->sfbMinSnrLdData, psyData[ch]->groupedSfbOffset, + &maxSfbPerGroup[ch], psyOutChannel[ch]->sfbMinSnrLdData, + psyStatic[ch]->blockSwitchingControl.noOfGroups, + psyStatic[ch]->blockSwitchingControl.groupLen, + psyConf[1].granuleLength); + + /* calculate ldData arrays (short values are in .Long-arrays after + * FDKaacEnc_groupShortData) */ + for (sfbGrp = 0; sfbGrp < noSfb; sfbGrp += hPsyConfShort->sfbCnt) { + LdDataVector(&psyData[ch]->sfbEnergy.Long[sfbGrp], + &psyOutChannel[ch]->sfbEnergyLdData[sfbGrp], + psyData[ch]->sfbActive); + } + + /* calc sfbThrld and set Values smaller 2^-31 to 2^-33*/ + for (sfbGrp = 0; sfbGrp < noSfb; sfbGrp += hPsyConfShort->sfbCnt) { + LdDataVector(&psyData[ch]->sfbThreshold.Long[sfbGrp], + &psyOutChannel[ch]->sfbThresholdLdData[sfbGrp], + psyData[ch]->sfbActive); + for (sfb = 0; sfb < psyData[ch]->sfbActive; sfb++) { + psyOutChannel[ch]->sfbThresholdLdData[sfbGrp + sfb] = + fixMax(psyOutChannel[ch]->sfbThresholdLdData[sfbGrp + sfb], + FL2FXCONST_DBL(-0.515625f)); + } + } + + if (channels == 2) { + for (sfbGrp = 0; sfbGrp < noSfb; sfbGrp += hPsyConfShort->sfbCnt) { + LdDataVector(&psyData[ch]->sfbEnergyMS.Long[sfbGrp], + &psyData[ch]->sfbEnergyMSLdData[sfbGrp], + psyData[ch]->sfbActive); + } + } + + FDKmemcpy(psyOutChannel[ch]->sfbOffsets, psyData[ch]->groupedSfbOffset, + (MAX_GROUPED_SFB + 1) * sizeof(INT)); + + } else { + int i; + /* maxSfb[ch] for long blocks */ + for (sfb = psyData[ch]->sfbActive - 1; sfb >= 0; sfb--) { + for (line = hPsyConfLong->sfbOffset[sfb + 1] - 1; + line >= hPsyConfLong->sfbOffset[sfb]; line--) { + if (psyData[ch]->mdctSpectrum[line] != FL2FXCONST_SGL(0.0f)) break; + } + if (line > hPsyConfLong->sfbOffset[sfb]) break; + } + maxSfbPerGroup[ch] = sfb + 1; + maxSfbPerGroup[ch] = + fixMax(fixMin(5, psyData[ch]->sfbActive), maxSfbPerGroup[ch]); + + /* sfbNrgLdData is calculated in FDKaacEnc_advancePsychLong, copy in + * psyOut structure */ + FDKmemcpy(psyOutChannel[ch]->sfbEnergyLdData, + psyData[ch]->sfbEnergyLdData.Long, + psyData[ch]->sfbActive * sizeof(FIXP_DBL)); + + FDKmemcpy(psyOutChannel[ch]->sfbOffsets, hPsyConfLong->sfbOffset, + (MAX_GROUPED_SFB + 1) * sizeof(INT)); + + /* sfbMinSnrLdData modified in adjust threshold, copy necessary */ + FDKmemcpy(psyOutChannel[ch]->sfbMinSnrLdData, + hPsyConfLong->sfbMinSnrLdData, + psyData[ch]->sfbActive * sizeof(FIXP_DBL)); + + /* sfbEnergyMSLdData ist already calculated in FDKaacEnc_CalcBandNrgMSOpt; + * only in long case */ + + /* calc sfbThrld and set Values smaller 2^-31 to 2^-33*/ + LdDataVector(psyData[ch]->sfbThreshold.Long, + psyOutChannel[ch]->sfbThresholdLdData, + psyData[ch]->sfbActive); + for (i = 0; i < psyData[ch]->sfbActive; i++) { + psyOutChannel[ch]->sfbThresholdLdData[i] = + fixMax(psyOutChannel[ch]->sfbThresholdLdData[i], + FL2FXCONST_DBL(-0.515625f)); + } + } + } + + /* + Intensity parameter intialization. + */ + for (ch = 0; ch < channels; ch++) { + FDKmemclear(psyOutChannel[ch]->isBook, MAX_GROUPED_SFB * sizeof(INT)); + FDKmemclear(psyOutChannel[ch]->isScale, MAX_GROUPED_SFB * sizeof(INT)); + } + + for (ch = 0; ch < channels; ch++) { + INT win = (isShortWindow[ch] ? 1 : 0); + if (!psyStatic[ch]->isLFE) { + /* PNS Decision */ + FDKaacEnc_PnsDetect( + &(psyConf[0].pnsConf), pnsData[ch], + psyStatic[ch]->blockSwitchingControl.lastWindowSequence, + psyData[ch]->sfbActive, + maxSfbPerGroup[ch], /* count of Sfb which are not zero. */ + psyOutChannel[ch]->sfbThresholdLdData, psyConf[win].sfbOffset, + psyData[ch]->mdctSpectrum, psyData[ch]->sfbMaxScaleSpec.Long, + sfbTonality[ch], psyOutChannel[ch]->tnsInfo.order[0][0], + tnsData[ch]->dataRaw.Long.subBlockInfo.predictionGain[HIFILT], + tnsData[ch]->dataRaw.Long.subBlockInfo.tnsActive[HIFILT], + psyOutChannel[ch]->sfbEnergyLdData, psyOutChannel[ch]->noiseNrg); + } /* !isLFE */ + } /* ch */ + + /* + stereo Processing + */ + if (channels == 2) { + psyOutElement->toolsInfo.msDigest = MS_NONE; + psyOutElement->commonWindow = commonWindow; + if (psyOutElement->commonWindow) + maxSfbPerGroup[0] = maxSfbPerGroup[1] = + fixMax(maxSfbPerGroup[0], maxSfbPerGroup[1]); + if (psyStatic[0]->blockSwitchingControl.lastWindowSequence != + SHORT_WINDOW) { + /* PNS preprocessing depending on ms processing: PNS not in Short Window! + */ + FDKaacEnc_PreProcessPnsChannelPair( + psyData[0]->sfbActive, (&psyData[0]->sfbEnergy)->Long, + (&psyData[1]->sfbEnergy)->Long, psyOutChannel[0]->sfbEnergyLdData, + psyOutChannel[1]->sfbEnergyLdData, psyData[0]->sfbEnergyMS.Long, + &(psyConf[0].pnsConf), pnsData[0], pnsData[1]); + + FDKaacEnc_IntensityStereoProcessing( + psyData[0]->sfbEnergy.Long, psyData[1]->sfbEnergy.Long, + psyData[0]->mdctSpectrum, psyData[1]->mdctSpectrum, + psyData[0]->sfbThreshold.Long, psyData[1]->sfbThreshold.Long, + psyOutChannel[1]->sfbThresholdLdData, + psyData[0]->sfbSpreadEnergy.Long, psyData[1]->sfbSpreadEnergy.Long, + psyOutChannel[0]->sfbEnergyLdData, psyOutChannel[1]->sfbEnergyLdData, + &psyOutElement->toolsInfo.msDigest, psyOutElement->toolsInfo.msMask, + psyConf[0].sfbCnt, psyConf[0].sfbCnt, maxSfbPerGroup[0], + psyConf[0].sfbOffset, + psyConf[0].allowIS && psyOutElement->commonWindow, + psyOutChannel[1]->isBook, psyOutChannel[1]->isScale, pnsData); + + FDKaacEnc_MsStereoProcessing( + psyData, psyOutChannel, psyOutChannel[1]->isBook, + &psyOutElement->toolsInfo.msDigest, psyOutElement->toolsInfo.msMask, + psyConf[0].allowMS, psyData[0]->sfbActive, psyData[0]->sfbActive, + maxSfbPerGroup[0], psyOutChannel[0]->sfbOffsets); + + /* PNS postprocessing */ + FDKaacEnc_PostProcessPnsChannelPair( + psyData[0]->sfbActive, &(psyConf[0].pnsConf), pnsData[0], pnsData[1], + psyOutElement->toolsInfo.msMask, &psyOutElement->toolsInfo.msDigest); + + } else { + FDKaacEnc_IntensityStereoProcessing( + psyData[0]->sfbEnergy.Long, psyData[1]->sfbEnergy.Long, + psyData[0]->mdctSpectrum, psyData[1]->mdctSpectrum, + psyData[0]->sfbThreshold.Long, psyData[1]->sfbThreshold.Long, + psyOutChannel[1]->sfbThresholdLdData, + psyData[0]->sfbSpreadEnergy.Long, psyData[1]->sfbSpreadEnergy.Long, + psyOutChannel[0]->sfbEnergyLdData, psyOutChannel[1]->sfbEnergyLdData, + &psyOutElement->toolsInfo.msDigest, psyOutElement->toolsInfo.msMask, + psyStatic[0]->blockSwitchingControl.noOfGroups * + hPsyConfShort->sfbCnt, + psyConf[1].sfbCnt, maxSfbPerGroup[0], psyData[0]->groupedSfbOffset, + psyConf[0].allowIS && psyOutElement->commonWindow, + psyOutChannel[1]->isBook, psyOutChannel[1]->isScale, pnsData); + + /* it's OK to pass the ".Long" arrays here. They contain grouped short + * data since FDKaacEnc_groupShortData() */ + FDKaacEnc_MsStereoProcessing( + psyData, psyOutChannel, psyOutChannel[1]->isBook, + &psyOutElement->toolsInfo.msDigest, psyOutElement->toolsInfo.msMask, + psyConf[1].allowMS, + psyStatic[0]->blockSwitchingControl.noOfGroups * + hPsyConfShort->sfbCnt, + hPsyConfShort->sfbCnt, maxSfbPerGroup[0], + psyOutChannel[0]->sfbOffsets); + } + } /* (channels == 2) */ + + /* + PNS Coding + */ + for (ch = 0; ch < channels; ch++) { + if (psyStatic[ch]->isLFE) { + /* no PNS coding */ + for (sfb = 0; sfb < psyData[ch]->sfbActive; sfb++) { + psyOutChannel[ch]->noiseNrg[sfb] = NO_NOISE_PNS; + } + } else { + FDKaacEnc_CodePnsChannel( + psyData[ch]->sfbActive, &(hThisPsyConf[ch]->pnsConf), + pnsData[ch]->pnsFlag, psyData[ch]->sfbEnergyLdData.Long, + psyOutChannel[ch]->noiseNrg, /* this is the energy that will be + written to the bitstream */ + psyOutChannel[ch]->sfbThresholdLdData); + } + } + + /* + build output + */ + for (ch = 0; ch < channels; ch++) { + INT mask; + int grp; + psyOutChannel[ch]->maxSfbPerGroup = maxSfbPerGroup[ch]; + psyOutChannel[ch]->mdctScale = psyData[ch]->mdctScale; + if (isShortWindow[ch] == 0) { + psyOutChannel[ch]->sfbCnt = hPsyConfLong->sfbActive; + psyOutChannel[ch]->sfbPerGroup = hPsyConfLong->sfbActive; + psyOutChannel[ch]->lastWindowSequence = + psyStatic[ch]->blockSwitchingControl.lastWindowSequence; + psyOutChannel[ch]->windowShape = + psyStatic[ch]->blockSwitchingControl.windowShape; + } else { + INT sfbCnt = psyStatic[ch]->blockSwitchingControl.noOfGroups * + hPsyConfShort->sfbCnt; + + psyOutChannel[ch]->sfbCnt = sfbCnt; + psyOutChannel[ch]->sfbPerGroup = hPsyConfShort->sfbCnt; + psyOutChannel[ch]->lastWindowSequence = SHORT_WINDOW; + psyOutChannel[ch]->windowShape = SINE_WINDOW; + } + /* generate grouping mask */ + mask = 0; + for (grp = 0; grp < psyStatic[ch]->blockSwitchingControl.noOfGroups; + grp++) { + int j; + mask <<= 1; + for (j = 1; j < psyStatic[ch]->blockSwitchingControl.groupLen[grp]; j++) { + mask = (mask << 1) | 1; + } + } + psyOutChannel[ch]->groupingMask = mask; + + /* build interface */ + FDKmemcpy(psyOutChannel[ch]->groupLen, + psyStatic[ch]->blockSwitchingControl.groupLen, + MAX_NO_OF_GROUPS * sizeof(INT)); + FDKmemcpy(psyOutChannel[ch]->sfbEnergy, (&psyData[ch]->sfbEnergy)->Long, + MAX_GROUPED_SFB * sizeof(FIXP_DBL)); + FDKmemcpy(psyOutChannel[ch]->sfbSpreadEnergy, + (&psyData[ch]->sfbSpreadEnergy)->Long, + MAX_GROUPED_SFB * sizeof(FIXP_DBL)); + // FDKmemcpy(psyOutChannel[ch]->mdctSpectrum, + // psyData[ch]->mdctSpectrum, (1024)*sizeof(FIXP_DBL)); + } + + return AAC_ENC_OK; +} + +void FDKaacEnc_PsyClose(PSY_INTERNAL **phPsyInternal, PSY_OUT **phPsyOut) { + int n, i; + + if (phPsyInternal != NULL) { + PSY_INTERNAL *hPsyInternal = *phPsyInternal; + + if (hPsyInternal) { + for (i = 0; i < (8); i++) { + if (hPsyInternal->pStaticChannels[i]) { + if (hPsyInternal->pStaticChannels[i]->psyInputBuffer) + FreeRam_aacEnc_PsyInputBuffer( + &hPsyInternal->pStaticChannels[i] + ->psyInputBuffer); /* AUDIO INPUT BUFFER */ + + FreeRam_aacEnc_PsyStatic( + &hPsyInternal->pStaticChannels[i]); /* PSY_STATIC */ + } + } + + for (i = 0; i < ((8)); i++) { + if (hPsyInternal->psyElement[i]) + FreeRam_aacEnc_PsyElement( + &hPsyInternal->psyElement[i]); /* PSY_ELEMENT */ + } + + FreeRam_aacEnc_PsyInternal(phPsyInternal); + } + } + + if (phPsyOut != NULL) { + for (n = 0; n < (1); n++) { + if (phPsyOut[n]) { + for (i = 0; i < (8); i++) { + if (phPsyOut[n]->pPsyOutChannels[i]) + FreeRam_aacEnc_PsyOutChannel( + &phPsyOut[n]->pPsyOutChannels[i]); /* PSY_OUT_CHANNEL */ + } + + for (i = 0; i < ((8)); i++) { + if (phPsyOut[n]->psyOutElement[i]) + FreeRam_aacEnc_PsyOutElements( + &phPsyOut[n]->psyOutElement[i]); /* PSY_OUT_ELEMENTS */ + } + + FreeRam_aacEnc_PsyOut(&phPsyOut[n]); + } + } + } +} diff --git a/fdk-aac/libAACenc/src/psy_main.h b/fdk-aac/libAACenc/src/psy_main.h new file mode 100644 index 0000000..7cc01a3 --- /dev/null +++ b/fdk-aac/libAACenc/src/psy_main.h @@ -0,0 +1,161 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M.Werner + + Description: Psychoaccoustic major function block + +*******************************************************************************/ + +#ifndef PSY_MAIN_H +#define PSY_MAIN_H + +#include "psy_configuration.h" +#include "qc_data.h" +#include "aacenc_pns.h" + +/* + psych internal +*/ +typedef struct { + PSY_STATIC *psyStatic[(2)]; + +} PSY_ELEMENT; + +typedef struct { + PSY_DATA psyData[(2)]; + TNS_DATA tnsData[(2)]; + PNS_DATA pnsData[(2)]; + +} PSY_DYNAMIC; + +typedef struct { + PSY_CONFIGURATION psyConf[2]; /* LONG / SHORT */ + PSY_ELEMENT *psyElement[((8))]; + PSY_STATIC *pStaticChannels[(8)]; + PSY_DYNAMIC *psyDynamic; + INT granuleLength; + +} PSY_INTERNAL; + +AAC_ENCODER_ERROR FDKaacEnc_PsyNew(PSY_INTERNAL **phpsy, const INT nElements, + const INT nChannels, UCHAR *dynamic_RAM); + +AAC_ENCODER_ERROR FDKaacEnc_PsyOutNew(PSY_OUT **phpsyOut, const INT nElements, + const INT nChannels, const INT nSubFrames, + UCHAR *dynamic_RAM); + +AAC_ENCODER_ERROR FDKaacEnc_psyInit(PSY_INTERNAL *hPsy, PSY_OUT **phpsyOut, + const INT nSubFrames, + const INT nMaxChannels, + const AUDIO_OBJECT_TYPE audioObjectType, + CHANNEL_MAPPING *cm); + +AAC_ENCODER_ERROR FDKaacEnc_psyMainInit( + PSY_INTERNAL *hPsy, AUDIO_OBJECT_TYPE audioObjectType, CHANNEL_MAPPING *cm, + INT sampleRate, INT granuleLength, INT bitRate, INT tnsMask, INT bandwidth, + INT usePns, INT useIS, INT useMS, UINT syntaxFlags, ULONG initFlags); + +AAC_ENCODER_ERROR FDKaacEnc_psyMain(INT channels, PSY_ELEMENT *psyElement, + PSY_DYNAMIC *psyDynamic, + PSY_CONFIGURATION *psyConf, + PSY_OUT_ELEMENT *psyOutElement, + INT_PCM *pInput, const UINT inputBufSize, + INT *chIdx, INT totalChannels); + +void FDKaacEnc_PsyClose(PSY_INTERNAL **phPsyInternal, PSY_OUT **phPsyOut); + +#endif /* PSY_MAIN_H */ diff --git a/fdk-aac/libAACenc/src/qc_data.h b/fdk-aac/libAACenc/src/qc_data.h new file mode 100644 index 0000000..6e671ed --- /dev/null +++ b/fdk-aac/libAACenc/src/qc_data.h @@ -0,0 +1,299 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M. Werner + + Description: Quantizing & coding data + +*******************************************************************************/ + +#ifndef QC_DATA_H +#define QC_DATA_H + +#include "aacenc.h" +#include "psy_const.h" +#include "dyn_bits.h" +#include "adj_thr_data.h" +#include "line_pe.h" +#include "FDK_audio.h" +#include "interface.h" + +typedef enum { + QCDATA_BR_MODE_INVALID = -1, + QCDATA_BR_MODE_CBR = 0, /* Constant bit rate, given average bitrate */ + QCDATA_BR_MODE_VBR_1 = 1, /* Variable bit rate, very low */ + QCDATA_BR_MODE_VBR_2 = 2, /* Variable bit rate, low */ + QCDATA_BR_MODE_VBR_3 = 3, /* Variable bit rate, medium */ + QCDATA_BR_MODE_VBR_4 = 4, /* Variable bit rate, high */ + QCDATA_BR_MODE_VBR_5 = 5, /* Variable bit rate, very high */ + QCDATA_BR_MODE_FF = 6, /* Fixed frame mode. */ + QCDATA_BR_MODE_SFR = 7 /* Superframe mode. */ + +} QCDATA_BR_MODE; + +typedef struct { + MP4_ELEMENT_ID elType; + INT instanceTag; + INT nChannelsInEl; + INT ChannelIndex[2]; + FIXP_DBL relativeBits; +} ELEMENT_INFO; + +typedef struct { + CHANNEL_MODE encMode; + INT nChannels; + INT nChannelsEff; + INT nElements; + ELEMENT_INFO elInfo[((8))]; +} CHANNEL_MAPPING; + +typedef struct { + INT paddingRest; +} PADDING; + +/* Quantizing & coding stage */ + +struct QC_INIT { + CHANNEL_MAPPING *channelMapping; + INT sceCpe; /* not used yet */ + INT maxBits; /* maximum number of bits in reservoir */ + INT averageBits; /* average number of bits we should use */ + INT bitRes; + INT sampleRate; /* output sample rate */ + INT isLowDelay; /* if set, calc bits2PE factor depending on samplerate */ + INT staticBits; /* Bits per frame consumed by transport layers. */ + QCDATA_BR_MODE bitrateMode; + INT meanPe; + INT chBitrate; /* Bitrate/channel */ + INT invQuant; + INT maxIterations; /* Maximum number of allowed iterations before + FDKaacEnc_crashRecovery() is applied. */ + FIXP_DBL maxBitFac; + INT bitrate; + INT nSubFrames; /* helper variable */ + INT minBits; /* minimal number of bits in one frame*/ + AACENC_BITRES_MODE bitResMode; /* 0: full bitreservoir, 1: reduced + bitreservoir, 2: disabled bitreservoir */ + INT bitDistributionMode; /* Configure element-wise execution or execution over + all elements for the pe-dependent + threshold-adaption */ + + PADDING padding; +}; + +typedef struct { + FIXP_DBL mdctSpectrum[(1024)]; + + SHORT quantSpec[(1024)]; + + UINT maxValueInSfb[MAX_GROUPED_SFB]; + INT scf[MAX_GROUPED_SFB]; + INT globalGain; + SECTION_DATA sectionData; + + FIXP_DBL sfbFormFactorLdData[MAX_GROUPED_SFB]; + + FIXP_DBL sfbThresholdLdData[MAX_GROUPED_SFB]; + FIXP_DBL sfbMinSnrLdData[MAX_GROUPED_SFB]; + FIXP_DBL sfbEnergyLdData[MAX_GROUPED_SFB]; + FIXP_DBL sfbEnergy[MAX_GROUPED_SFB]; + FIXP_DBL sfbWeightedEnergyLdData[MAX_GROUPED_SFB]; + + FIXP_DBL sfbEnFacLd[MAX_GROUPED_SFB]; + + FIXP_DBL sfbSpreadEnergy[MAX_GROUPED_SFB]; + +} QC_OUT_CHANNEL; + +typedef struct { + EXT_PAYLOAD_TYPE type; /* type of the extension payload */ + INT nPayloadBits; /* size of the payload */ + UCHAR *pPayload; /* pointer to payload */ + +} QC_OUT_EXTENSION; + +typedef struct { + INT staticBitsUsed; /* for verification purposes */ + INT dynBitsUsed; /* for verification purposes */ + + INT extBitsUsed; /* bit consumption of extended fill elements */ + INT nExtensions; /* number of extension payloads for this element */ + QC_OUT_EXTENSION extension[(1)]; /* reffering extension payload */ + + INT grantedDynBits; + + INT grantedPe; + INT grantedPeCorr; + + PE_DATA peData; + + QC_OUT_CHANNEL *qcOutChannel[(2)]; + + UCHAR + *dynMem_Ah_Flag; /* pointer to dynamic buffer used by AhFlag in function + FDKaacEnc_adaptThresholdsToPe() */ + UCHAR + *dynMem_Thr_Exp; /* pointer to dynamic buffer used by ThrExp in function + FDKaacEnc_adaptThresholdsToPe() */ + UCHAR *dynMem_SfbNActiveLinesLdData; /* pointer to dynamic buffer used by + sfbNActiveLinesLdData in function + FDKaacEnc_correctThresh() */ + +} QC_OUT_ELEMENT; + +typedef struct { + QC_OUT_ELEMENT *qcElement[((8))]; + QC_OUT_CHANNEL *pQcOutChannels[(8)]; + QC_OUT_EXTENSION extension[(2 + 2)]; /* global extension payload */ + INT nExtensions; /* number of extension payloads for this AU */ + INT maxDynBits; /* maximal allowed dynamic bits in frame */ + INT grantedDynBits; /* granted dynamic bits in frame */ + INT totFillBits; /* fill bits */ + INT elementExtBits; /* element associated extension payload bits, e.g. sbr, + drc ... */ + INT globalExtBits; /* frame/au associated extension payload bits (anc data + ...) */ + INT staticBits; /* aac side info bits */ + + INT totalNoRedPe; + INT totalGrantedPeCorr; + + INT usedDynBits; /* number of dynamic bits in use */ + INT alignBits; /* AU alignment bits */ + INT totalBits; /* sum of static, dyn, sbr, fill, align and dse bits */ + +} QC_OUT; + +typedef struct { + INT chBitrateEl; /* channel bitrate in element + (totalbitrate*el_relativeBits/el_channels) */ + INT maxBitsEl; /* used in crash recovery */ + INT bitResLevelEl; /* update bitreservoir level in each call of + FDKaacEnc_QCMain */ + INT maxBitResBitsEl; /* nEffChannels*6144 - averageBitsInFrame */ + FIXP_DBL relativeBitsEl; /* Bits relative to total Bits*/ +} ELEMENT_BITS; + +typedef struct { + /* this is basically struct QC_INIT */ + + INT globHdrBits; + INT maxBitsPerFrame; /* maximal allowed bits per frame, 6144*nChannelsEff */ + INT minBitsPerFrame; /* minimal allowd bits per fram, superframing - DRM */ + INT nElements; + QCDATA_BR_MODE bitrateMode; + AACENC_BITRES_MODE bitResMode; /* 0: full bitreservoir, 1: reduced + bitreservoir, 2: disabled bitreservoir */ + INT bitResTot; + INT bitResTotMax; + INT maxIterations; /* Maximum number of allowed iterations before + FDKaacEnc_crashRecovery() is applied. */ + INT invQuant; + + FIXP_DBL vbrQualFactor; + FIXP_DBL maxBitFac; + + PADDING padding; + + ELEMENT_BITS *elementBits[((8))]; + BITCNTR_STATE *hBitCounter; + ADJ_THR_STATE *hAdjThr; + + INT dZoneQuantEnable; /* enable dead zone quantizer */ + +} QC_STATE; + +#endif /* QC_DATA_H */ diff --git a/fdk-aac/libAACenc/src/qc_main.cpp b/fdk-aac/libAACenc/src/qc_main.cpp new file mode 100644 index 0000000..0bf234c --- /dev/null +++ b/fdk-aac/libAACenc/src/qc_main.cpp @@ -0,0 +1,1555 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M. Werner + + Description: Quantizing & coding + +*******************************************************************************/ + +#include "qc_main.h" +#include "quantize.h" +#include "interface.h" +#include "adj_thr.h" +#include "sf_estim.h" +#include "bit_cnt.h" +#include "dyn_bits.h" +#include "channel_map.h" +#include "aacEnc_ram.h" + +#include "genericStds.h" + +#define AACENC_DZQ_BR_THR 32000 /* Dead zone quantizer bitrate threshold */ + +typedef struct { + QCDATA_BR_MODE bitrateMode; + LONG vbrQualFactor; +} TAB_VBR_QUAL_FACTOR; + +static const TAB_VBR_QUAL_FACTOR tableVbrQualFactor[] = { + {QCDATA_BR_MODE_VBR_1, + FL2FXCONST_DBL(0.160f)}, /* Approx. 32 - 48 (AC-LC), 32 - 56 + (AAC-LD/ELD) kbps/channel */ + {QCDATA_BR_MODE_VBR_2, + FL2FXCONST_DBL(0.148f)}, /* Approx. 40 - 56 (AC-LC), 40 - 64 + (AAC-LD/ELD) kbps/channel */ + {QCDATA_BR_MODE_VBR_3, + FL2FXCONST_DBL(0.135f)}, /* Approx. 48 - 64 (AC-LC), 48 - 72 + (AAC-LD/ELD) kbps/channel */ + {QCDATA_BR_MODE_VBR_4, + FL2FXCONST_DBL(0.111f)}, /* Approx. 64 - 80 (AC-LC), 64 - 88 + (AAC-LD/ELD) kbps/channel */ + {QCDATA_BR_MODE_VBR_5, + FL2FXCONST_DBL(0.070f)} /* Approx. 96 - 120 (AC-LC), 112 - 144 + (AAC-LD/ELD) kbps/channel */ +}; + +static INT isConstantBitrateMode(const QCDATA_BR_MODE bitrateMode) { + return (((bitrateMode == QCDATA_BR_MODE_CBR) || + (bitrateMode == QCDATA_BR_MODE_SFR) || + (bitrateMode == QCDATA_BR_MODE_FF)) + ? 1 + : 0); +} + +typedef enum { + FRAME_LEN_BYTES_MODULO = 1, + FRAME_LEN_BYTES_INT = 2 +} FRAME_LEN_RESULT_MODE; + +/* forward declarations */ + +static INT FDKaacEnc_calcMaxValueInSfb(INT sfbCnt, INT maxSfbPerGroup, + INT sfbPerGroup, INT* RESTRICT sfbOffset, + SHORT* RESTRICT quantSpectrum, + UINT* RESTRICT maxValue); + +static void FDKaacEnc_crashRecovery(INT nChannels, + PSY_OUT_ELEMENT* psyOutElement, + QC_OUT* qcOut, QC_OUT_ELEMENT* qcElement, + INT bitsToSave, AUDIO_OBJECT_TYPE aot, + UINT syntaxFlags, SCHAR epConfig); + +static AAC_ENCODER_ERROR FDKaacEnc_reduceBitConsumption( + int* iterations, const int maxIterations, int gainAdjustment, + int* chConstraintsFulfilled, int* calculateQuant, int nChannels, + PSY_OUT_ELEMENT* psyOutElement, QC_OUT* qcOut, QC_OUT_ELEMENT* qcOutElement, + ELEMENT_BITS* elBits, AUDIO_OBJECT_TYPE aot, UINT syntaxFlags, + SCHAR epConfig); + +void FDKaacEnc_QCClose(QC_STATE** phQCstate, QC_OUT** phQC); + +/***************************************************************************** + + functionname: FDKaacEnc_calcFrameLen + description: + returns: + input: + output: + +*****************************************************************************/ +static INT FDKaacEnc_calcFrameLen(INT bitRate, INT sampleRate, + INT granuleLength, + FRAME_LEN_RESULT_MODE mode) { + INT result; + + result = ((granuleLength) >> 3) * (bitRate); + + switch (mode) { + case FRAME_LEN_BYTES_MODULO: + result %= sampleRate; + break; + case FRAME_LEN_BYTES_INT: + result /= sampleRate; + break; + } + return (result); +} + +/***************************************************************************** + + functionname:FDKaacEnc_framePadding + description: Calculates if padding is needed for actual frame + returns: + input: + output: + +*****************************************************************************/ +static INT FDKaacEnc_framePadding(INT bitRate, INT sampleRate, + INT granuleLength, INT* paddingRest) { + INT paddingOn; + INT difference; + + paddingOn = 0; + + difference = FDKaacEnc_calcFrameLen(bitRate, sampleRate, granuleLength, + FRAME_LEN_BYTES_MODULO); + *paddingRest -= difference; + + if (*paddingRest <= 0) { + paddingOn = 1; + *paddingRest += sampleRate; + } + + return (paddingOn); +} + +/********************************************************************************* + + functionname: FDKaacEnc_QCOutNew + description: + return: + +**********************************************************************************/ +AAC_ENCODER_ERROR FDKaacEnc_QCOutNew(QC_OUT** phQC, const INT nElements, + const INT nChannels, const INT nSubFrames, + UCHAR* dynamic_RAM) { + AAC_ENCODER_ERROR ErrorStatus; + int n, i; + int elInc = 0, chInc = 0; + + for (n = 0; n < nSubFrames; n++) { + phQC[n] = GetRam_aacEnc_QCout(n); + if (phQC[n] == NULL) { + ErrorStatus = AAC_ENC_NO_MEMORY; + goto QCOutNew_bail; + } + + for (i = 0; i < nChannels; i++) { + phQC[n]->pQcOutChannels[i] = GetRam_aacEnc_QCchannel(chInc, dynamic_RAM); + if (phQC[n]->pQcOutChannels[i] == NULL) { + ErrorStatus = AAC_ENC_NO_MEMORY; + goto QCOutNew_bail; + } + + chInc++; + } /* nChannels */ + + for (i = 0; i < nElements; i++) { + phQC[n]->qcElement[i] = GetRam_aacEnc_QCelement(elInc); + if (phQC[n]->qcElement[i] == NULL) { + ErrorStatus = AAC_ENC_NO_MEMORY; + goto QCOutNew_bail; + } + elInc++; + + /* initialize pointer to dynamic buffer which are used in adjust + * thresholds */ + phQC[n]->qcElement[i]->dynMem_Ah_Flag = dynamic_RAM + (P_BUF_1); + phQC[n]->qcElement[i]->dynMem_Thr_Exp = + dynamic_RAM + (P_BUF_1) + ADJ_THR_AH_FLAG_SIZE; + phQC[n]->qcElement[i]->dynMem_SfbNActiveLinesLdData = + dynamic_RAM + (P_BUF_1) + ADJ_THR_AH_FLAG_SIZE + ADJ_THR_THR_EXP_SIZE; + + } /* nElements */ + + } /* nSubFrames */ + + return AAC_ENC_OK; + +QCOutNew_bail: + return ErrorStatus; +} + +/********************************************************************************* + + functionname: FDKaacEnc_QCOutInit + description: + return: + +**********************************************************************************/ +AAC_ENCODER_ERROR FDKaacEnc_QCOutInit(QC_OUT* phQC[(1)], const INT nSubFrames, + const CHANNEL_MAPPING* cm) { + INT n, i, ch; + + for (n = 0; n < nSubFrames; n++) { + INT chInc = 0; + for (i = 0; i < cm->nElements; i++) { + for (ch = 0; ch < cm->elInfo[i].nChannelsInEl; ch++) { + phQC[n]->qcElement[i]->qcOutChannel[ch] = + phQC[n]->pQcOutChannels[chInc]; + chInc++; + } /* chInEl */ + } /* nElements */ + } /* nSubFrames */ + + return AAC_ENC_OK; +} + +/********************************************************************************* + + functionname: FDKaacEnc_QCNew + description: + return: + +**********************************************************************************/ +AAC_ENCODER_ERROR FDKaacEnc_QCNew(QC_STATE** phQC, INT nElements, + UCHAR* dynamic_RAM) { + AAC_ENCODER_ERROR ErrorStatus; + int i; + + QC_STATE* hQC = GetRam_aacEnc_QCstate(); + *phQC = hQC; + if (hQC == NULL) { + ErrorStatus = AAC_ENC_NO_MEMORY; + goto QCNew_bail; + } + + if (FDKaacEnc_AdjThrNew(&hQC->hAdjThr, nElements)) { + ErrorStatus = AAC_ENC_NO_MEMORY; + goto QCNew_bail; + } + + if (FDKaacEnc_BCNew(&(hQC->hBitCounter), dynamic_RAM)) { + ErrorStatus = AAC_ENC_NO_MEMORY; + goto QCNew_bail; + } + + for (i = 0; i < nElements; i++) { + hQC->elementBits[i] = GetRam_aacEnc_ElementBits(i); + if (hQC->elementBits[i] == NULL) { + ErrorStatus = AAC_ENC_NO_MEMORY; + goto QCNew_bail; + } + } + + return AAC_ENC_OK; + +QCNew_bail: + FDKaacEnc_QCClose(phQC, NULL); + return ErrorStatus; +} + +/********************************************************************************* + + functionname: FDKaacEnc_QCInit + description: + return: + +**********************************************************************************/ +AAC_ENCODER_ERROR FDKaacEnc_QCInit(QC_STATE* hQC, struct QC_INIT* init, + const ULONG initFlags) { + AAC_ENCODER_ERROR err = AAC_ENC_OK; + + int i; + hQC->maxBitsPerFrame = init->maxBits; + hQC->minBitsPerFrame = init->minBits; + hQC->nElements = init->channelMapping->nElements; + if ((initFlags != 0) || ((init->bitrateMode != QCDATA_BR_MODE_FF) && + (hQC->bitResTotMax != init->bitRes))) { + hQC->bitResTot = init->bitRes; + } + hQC->bitResTotMax = init->bitRes; + hQC->maxBitFac = init->maxBitFac; + hQC->bitrateMode = init->bitrateMode; + hQC->invQuant = init->invQuant; + hQC->maxIterations = init->maxIterations; + + if (isConstantBitrateMode(hQC->bitrateMode)) { + /* 0: full bitreservoir, 1: reduced bitreservoir, 2: disabled bitreservoir + */ + hQC->bitResMode = init->bitResMode; + } else { + hQC->bitResMode = AACENC_BR_MODE_FULL; /* full bitreservoir */ + } + + hQC->padding.paddingRest = init->padding.paddingRest; + + hQC->globHdrBits = init->staticBits; /* Bit overhead due to transport */ + + err = FDKaacEnc_InitElementBits( + hQC, init->channelMapping, init->bitrate, + (init->averageBits / init->nSubFrames) - hQC->globHdrBits, + hQC->maxBitsPerFrame / init->channelMapping->nChannelsEff); + if (err != AAC_ENC_OK) goto bail; + + hQC->vbrQualFactor = FL2FXCONST_DBL(0.f); + for (i = 0; + i < (int)(sizeof(tableVbrQualFactor) / sizeof(TAB_VBR_QUAL_FACTOR)); + i++) { + if (hQC->bitrateMode == tableVbrQualFactor[i].bitrateMode) { + hQC->vbrQualFactor = (FIXP_DBL)tableVbrQualFactor[i].vbrQualFactor; + break; + } + } + + if (init->channelMapping->nChannelsEff == 1 && + (init->bitrate / init->channelMapping->nChannelsEff) < + AACENC_DZQ_BR_THR && + init->isLowDelay != + 0) /* watch out here: init->bitrate is the bitrate "minus" the + standard SBR bitrate (=2500kbps) --> for the FDK the OFFSTE + tuning should start somewhere below 32000kbps-2500kbps ... so + everything is fine here */ + { + hQC->dZoneQuantEnable = 1; + } else { + hQC->dZoneQuantEnable = 0; + } + + FDKaacEnc_AdjThrInit( + hQC->hAdjThr, init->meanPe, hQC->invQuant, init->channelMapping, + init->sampleRate, /* output sample rate */ + init->bitrate, /* total bitrate */ + init->isLowDelay, /* if set, calc bits2PE factor + depending on samplerate */ + init->bitResMode /* for a small bitreservoir, the pe + correction is calc'd differently */ + , + hQC->dZoneQuantEnable, init->bitDistributionMode, hQC->vbrQualFactor); + +bail: + return err; +} + +/********************************************************************************* + + functionname: FDKaacEnc_QCMainPrepare + description: + return: + +**********************************************************************************/ +AAC_ENCODER_ERROR FDKaacEnc_QCMainPrepare( + ELEMENT_INFO* elInfo, ATS_ELEMENT* RESTRICT adjThrStateElement, + PSY_OUT_ELEMENT* RESTRICT psyOutElement, + QC_OUT_ELEMENT* RESTRICT qcOutElement, AUDIO_OBJECT_TYPE aot, + UINT syntaxFlags, SCHAR epConfig) { + AAC_ENCODER_ERROR ErrorStatus = AAC_ENC_OK; + INT nChannels = elInfo->nChannelsInEl; + + PSY_OUT_CHANNEL** RESTRICT psyOutChannel = + psyOutElement->psyOutChannel; /* may be modified in-place */ + + FDKaacEnc_CalcFormFactor(qcOutElement->qcOutChannel, psyOutChannel, + nChannels); + + /* prepare and calculate PE without reduction */ + FDKaacEnc_peCalculation(&qcOutElement->peData, psyOutChannel, + qcOutElement->qcOutChannel, &psyOutElement->toolsInfo, + adjThrStateElement, nChannels); + + ErrorStatus = FDKaacEnc_ChannelElementWrite( + NULL, elInfo, NULL, psyOutElement, psyOutElement->psyOutChannel, + syntaxFlags, aot, epConfig, &qcOutElement->staticBitsUsed, 0); + + return ErrorStatus; +} + +/********************************************************************************* + + functionname: FDKaacEnc_AdjustBitrate + description: adjusts framelength via padding on a frame to frame +basis, to achieve a bitrate that demands a non byte aligned framelength return: +errorcode + +**********************************************************************************/ +AAC_ENCODER_ERROR FDKaacEnc_AdjustBitrate( + QC_STATE* RESTRICT hQC, CHANNEL_MAPPING* RESTRICT cm, INT* avgTotalBits, + INT bitRate, /* total bitrate */ + INT sampleRate, /* output sampling rate */ + INT granuleLength) /* frame length */ +{ + INT paddingOn; + INT frameLen; + //fprintf(stderr, "hQC->padding.paddingRest=%d bytes! (before)\n", hQC->padding.paddingRest); + + /* Do we need an extra padding byte? */ + paddingOn = FDKaacEnc_framePadding(bitRate, sampleRate, granuleLength, + &hQC->padding.paddingRest); + + frameLen = + paddingOn + FDKaacEnc_calcFrameLen(bitRate, sampleRate, granuleLength, + FRAME_LEN_BYTES_INT); + + *avgTotalBits = frameLen << 3; + + return AAC_ENC_OK; +} + +#define isAudioElement(elType) \ + ((elType == ID_SCE) || (elType == ID_CPE) || (elType == ID_LFE)) + +/********************************************************************************* + + functionname: FDKaacEnc_distributeElementDynBits + description: distributes all bits over all elements. The relative bit + distibution is described in the ELEMENT_INFO of the + appropriate element. The bit distribution table is + initialized in FDKaacEnc_InitChannelMapping(). + return: errorcode + +**********************************************************************************/ +static AAC_ENCODER_ERROR FDKaacEnc_distributeElementDynBits( + QC_STATE* hQC, QC_OUT_ELEMENT* qcElement[((8))], CHANNEL_MAPPING* cm, + INT codeBits) { + INT i; /* counter variable */ + INT totalBits = 0; /* sum of bits over all elements */ + + for (i = (cm->nElements - 1); i >= 0; i--) { + if (isAudioElement(cm->elInfo[i].elType)) { + qcElement[i]->grantedDynBits = + fMax(0, fMultI(hQC->elementBits[i]->relativeBitsEl, codeBits)); + totalBits += qcElement[i]->grantedDynBits; + } + } + + /* Due to inaccuracies with the multiplication, codeBits may differ from + totalBits. For that case, the difference must be added/substracted again + to/from one element, i.e: + Negative differences are substracted from the element with the most bits. + Positive differences are added to the element with the least bits. + */ + if (codeBits != totalBits) { + INT elMaxBits = cm->nElements - 1; /* element with the most bits */ + INT elMinBits = cm->nElements - 1; /* element with the least bits */ + + /* Search for biggest and smallest audio element */ + for (i = (cm->nElements - 1); i >= 0; i--) { + if (isAudioElement(cm->elInfo[i].elType)) { + if (qcElement[i]->grantedDynBits > + qcElement[elMaxBits]->grantedDynBits) { + elMaxBits = i; + } + if (qcElement[i]->grantedDynBits < + qcElement[elMinBits]->grantedDynBits) { + elMinBits = i; + } + } + } + /* Compensate for bit distibution difference */ + if (codeBits - totalBits > 0) { + qcElement[elMinBits]->grantedDynBits += codeBits - totalBits; + } else { + qcElement[elMaxBits]->grantedDynBits += codeBits - totalBits; + } + } + + return AAC_ENC_OK; +} + +/** + * \brief Verify whether minBitsPerFrame criterion can be satisfied. + * + * This function evaluates the bit consumption only if minBitsPerFrame parameter + * is not 0. In hyperframing mode the difference between grantedDynBits and + * usedDynBits of all sub frames results the number of fillbits to be written. + * This bits can be distrubitued in superframe to reach minBitsPerFrame bit + * consumption in single AU's. The return value denotes if enough desired fill + * bits are available to achieve minBitsPerFrame in all frames. This check can + * only be used within superframes. + * + * \param qcOut Pointer to coding data struct. + * \param minBitsPerFrame Minimal number of bits to be consumed in each frame. + * \param nSubFrames Number of frames in superframe + * + * \return + * - 1: all fine + * - 0: criterion not fulfilled + */ +static int checkMinFrameBitsDemand(QC_OUT** qcOut, const INT minBitsPerFrame, + const INT nSubFrames) { + int result = 1; /* all fine*/ + return result; +} + +//////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// + +//////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// +/********************************************************************************* + + functionname: FDKaacEnc_getMinimalStaticBitdemand + description: calculate minmal size of static bits by reduction , + to zero spectrum and deactivating tns and MS + return: number of static bits + +**********************************************************************************/ +static int FDKaacEnc_getMinimalStaticBitdemand(CHANNEL_MAPPING* cm, + PSY_OUT** psyOut) { + AUDIO_OBJECT_TYPE aot = AOT_AAC_LC; + UINT syntaxFlags = 0; + SCHAR epConfig = -1; + int i, bitcount = 0; + + for (i = 0; i < cm->nElements; i++) { + ELEMENT_INFO elInfo = cm->elInfo[i]; + + if ((elInfo.elType == ID_SCE) || (elInfo.elType == ID_CPE) || + (elInfo.elType == ID_LFE)) { + INT minElBits = 0; + + FDKaacEnc_ChannelElementWrite(NULL, &elInfo, NULL, + psyOut[0]->psyOutElement[i], + psyOut[0]->psyOutElement[i]->psyOutChannel, + syntaxFlags, aot, epConfig, &minElBits, 1); + bitcount += minElBits; + } + } + + return bitcount; +} + +//////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// + +static AAC_ENCODER_ERROR FDKaacEnc_prepareBitDistribution( + QC_STATE* hQC, PSY_OUT** psyOut, QC_OUT** qcOut, CHANNEL_MAPPING* cm, + QC_OUT_ELEMENT* qcElement[(1)][((8))], INT avgTotalBits, + INT* totalAvailableBits, INT* avgTotalDynBits) { + int i; + /* get maximal allowed dynamic bits */ + qcOut[0]->grantedDynBits = + (fixMin(hQC->maxBitsPerFrame, avgTotalBits) - hQC->globHdrBits) & ~7; + qcOut[0]->grantedDynBits -= (qcOut[0]->globalExtBits + qcOut[0]->staticBits + + qcOut[0]->elementExtBits); + qcOut[0]->maxDynBits = ((hQC->maxBitsPerFrame) & ~7) - + (qcOut[0]->globalExtBits + qcOut[0]->staticBits + + qcOut[0]->elementExtBits); + /* assure that enough bits are available */ + if ((qcOut[0]->grantedDynBits + hQC->bitResTot) < 0) { + /* crash recovery allows to reduce static bits to a minimum */ + if ((qcOut[0]->grantedDynBits + hQC->bitResTot) < + (FDKaacEnc_getMinimalStaticBitdemand(cm, psyOut) - + qcOut[0]->staticBits)) + return AAC_ENC_BITRES_TOO_LOW; + } + + /* distribute dynamic bits to each element */ + FDKaacEnc_distributeElementDynBits(hQC, qcElement[0], cm, + qcOut[0]->grantedDynBits); + + *avgTotalDynBits = 0; /*frameDynBits;*/ + + *totalAvailableBits = avgTotalBits; + + /* sum up corrected granted PE */ + qcOut[0]->totalGrantedPeCorr = 0; + + for (i = 0; i < cm->nElements; i++) { + ELEMENT_INFO elInfo = cm->elInfo[i]; + int nChannels = elInfo.nChannelsInEl; + + if ((elInfo.elType == ID_SCE) || (elInfo.elType == ID_CPE) || + (elInfo.elType == ID_LFE)) { + /* for ( all sub frames ) ... */ + FDKaacEnc_DistributeBits( + hQC->hAdjThr, hQC->hAdjThr->adjThrStateElem[i], + psyOut[0]->psyOutElement[i]->psyOutChannel, &qcElement[0][i]->peData, + &qcElement[0][i]->grantedPe, &qcElement[0][i]->grantedPeCorr, + nChannels, psyOut[0]->psyOutElement[i]->commonWindow, + qcElement[0][i]->grantedDynBits, hQC->elementBits[i]->bitResLevelEl, + hQC->elementBits[i]->maxBitResBitsEl, hQC->maxBitFac, + hQC->bitResMode); + + *totalAvailableBits += hQC->elementBits[i]->bitResLevelEl; + /* get total corrected granted PE */ + qcOut[0]->totalGrantedPeCorr += qcElement[0][i]->grantedPeCorr; + } /* -end- if(ID_SCE || ID_CPE || ID_LFE) */ + + } /* -end- element loop */ + + *totalAvailableBits = fMin(hQC->maxBitsPerFrame, (*totalAvailableBits)); + + return AAC_ENC_OK; +} + +//////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////// +static AAC_ENCODER_ERROR FDKaacEnc_updateUsedDynBits( + INT* sumDynBitsConsumed, QC_OUT_ELEMENT* qcElement[((8))], + CHANNEL_MAPPING* cm) { + INT i; + + *sumDynBitsConsumed = 0; + + for (i = 0; i < cm->nElements; i++) { + ELEMENT_INFO elInfo = cm->elInfo[i]; + + if ((elInfo.elType == ID_SCE) || (elInfo.elType == ID_CPE) || + (elInfo.elType == ID_LFE)) { + /* sum up bits consumed */ + *sumDynBitsConsumed += qcElement[i]->dynBitsUsed; + } /* -end- if(ID_SCE || ID_CPE || ID_LFE) */ + + } /* -end- element loop */ + + return AAC_ENC_OK; +} + +static INT FDKaacEnc_getTotalConsumedDynBits(QC_OUT** qcOut, INT nSubFrames) { + INT c, totalBits = 0; + + /* sum up bit consumption for all sub frames */ + for (c = 0; c < nSubFrames; c++) { + /* bit consumption not valid if dynamic bits + not available in one sub frame */ + if (qcOut[c]->usedDynBits == -1) return -1; + totalBits += qcOut[c]->usedDynBits; + } + + return totalBits; +} + +static INT FDKaacEnc_getTotalConsumedBits(QC_OUT** qcOut, + QC_OUT_ELEMENT* qcElement[(1)][((8))], + CHANNEL_MAPPING* cm, INT globHdrBits, + INT nSubFrames) { + int c, i; + int totalUsedBits = 0; + + for (c = 0; c < nSubFrames; c++) { + int dataBits = 0; + for (i = 0; i < cm->nElements; i++) { + if ((cm->elInfo[i].elType == ID_SCE) || + (cm->elInfo[i].elType == ID_CPE) || + (cm->elInfo[i].elType == ID_LFE)) { + dataBits += qcElement[c][i]->dynBitsUsed + + qcElement[c][i]->staticBitsUsed + + qcElement[c][i]->extBitsUsed; + } + } + dataBits += qcOut[c]->globalExtBits; + + totalUsedBits += (8 - (dataBits) % 8) % 8; + totalUsedBits += dataBits + globHdrBits; /* header bits for every frame */ + } + return totalUsedBits; +} + +static AAC_ENCODER_ERROR FDKaacEnc_BitResRedistribution( + QC_STATE* const hQC, const CHANNEL_MAPPING* const cm, + const INT avgTotalBits) { + /* check bitreservoir fill level */ + if (hQC->bitResTot < 0) { + return AAC_ENC_BITRES_TOO_LOW; + } else if (hQC->bitResTot > hQC->bitResTotMax) { + return AAC_ENC_BITRES_TOO_HIGH; + } else { + INT i; + INT totalBits = 0, totalBits_max = 0; + + const int totalBitreservoir = + fMin(hQC->bitResTot, (hQC->maxBitsPerFrame - avgTotalBits)); + const int totalBitreservoirMax = + fMin(hQC->bitResTotMax, (hQC->maxBitsPerFrame - avgTotalBits)); + + for (i = (cm->nElements - 1); i >= 0; i--) { + if ((cm->elInfo[i].elType == ID_SCE) || + (cm->elInfo[i].elType == ID_CPE) || + (cm->elInfo[i].elType == ID_LFE)) { + hQC->elementBits[i]->bitResLevelEl = + fMultI(hQC->elementBits[i]->relativeBitsEl, totalBitreservoir); + totalBits += hQC->elementBits[i]->bitResLevelEl; + + hQC->elementBits[i]->maxBitResBitsEl = + fMultI(hQC->elementBits[i]->relativeBitsEl, totalBitreservoirMax); + totalBits_max += hQC->elementBits[i]->maxBitResBitsEl; + } + } + for (i = 0; i < cm->nElements; i++) { + if ((cm->elInfo[i].elType == ID_SCE) || + (cm->elInfo[i].elType == ID_CPE) || + (cm->elInfo[i].elType == ID_LFE)) { + int deltaBits = fMax(totalBitreservoir - totalBits, + -hQC->elementBits[i]->bitResLevelEl); + hQC->elementBits[i]->bitResLevelEl += deltaBits; + totalBits += deltaBits; + + deltaBits = fMax(totalBitreservoirMax - totalBits_max, + -hQC->elementBits[i]->maxBitResBitsEl); + hQC->elementBits[i]->maxBitResBitsEl += deltaBits; + totalBits_max += deltaBits; + } + } + } + + return AAC_ENC_OK; +} + +AAC_ENCODER_ERROR FDKaacEnc_QCMain(QC_STATE* RESTRICT hQC, PSY_OUT** psyOut, + QC_OUT** qcOut, INT avgTotalBits, + CHANNEL_MAPPING* cm, + const AUDIO_OBJECT_TYPE aot, + UINT syntaxFlags, SCHAR epConfig) { + int i, c; + AAC_ENCODER_ERROR ErrorStatus = AAC_ENC_OK; + INT avgTotalDynBits = 0; /* maximal allowed dynamic bits for all frames */ + INT totalAvailableBits = 0; + INT nSubFrames = 1; + + /*-------------------------------------------- */ + /* redistribute total bitreservoir to elements */ + ErrorStatus = FDKaacEnc_BitResRedistribution(hQC, cm, avgTotalBits); + if (ErrorStatus != AAC_ENC_OK) { + return ErrorStatus; + } + + /*-------------------------------------------- */ + /* fastenc needs one time threshold simulation, + in case of multiple frames, one more guess has to be calculated */ + + /*-------------------------------------------- */ + /* helper pointer */ + QC_OUT_ELEMENT* qcElement[(1)][((8))]; + + /* work on a copy of qcChannel and qcElement */ + for (i = 0; i < cm->nElements; i++) { + ELEMENT_INFO elInfo = cm->elInfo[i]; + + if ((elInfo.elType == ID_SCE) || (elInfo.elType == ID_CPE) || + (elInfo.elType == ID_LFE)) { + /* for ( all sub frames ) ... */ + for (c = 0; c < nSubFrames; c++) { + { qcElement[c][i] = qcOut[c]->qcElement[i]; } + } + } + } + + /*-------------------------------------------- */ + /*-------------------------------------------- */ + if (isConstantBitrateMode(hQC->bitrateMode)) { + /* calc granted dynamic bits for sub frame and + distribute it to each element */ + ErrorStatus = FDKaacEnc_prepareBitDistribution( + hQC, psyOut, qcOut, cm, qcElement, avgTotalBits, &totalAvailableBits, + &avgTotalDynBits); + + if (ErrorStatus != AAC_ENC_OK) { + return ErrorStatus; + } + } else { + qcOut[0]->grantedDynBits = + ((hQC->maxBitsPerFrame - (hQC->globHdrBits)) & ~7) - + (qcOut[0]->globalExtBits + qcOut[0]->staticBits + + qcOut[0]->elementExtBits); + qcOut[0]->maxDynBits = qcOut[0]->grantedDynBits; + + totalAvailableBits = hQC->maxBitsPerFrame; + avgTotalDynBits = 0; + } + + /* for ( all sub frames ) ... */ + for (c = 0; c < nSubFrames; c++) { + /* for CBR and VBR mode */ + FDKaacEnc_AdjustThresholds(hQC->hAdjThr, qcElement[c], qcOut[c], + psyOut[c]->psyOutElement, + isConstantBitrateMode(hQC->bitrateMode), cm); + + } /* -end- sub frame counter */ + + /*-------------------------------------------- */ + INT iterations[(1)][((8))]; + INT chConstraintsFulfilled[(1)][((8))][(2)]; + INT calculateQuant[(1)][((8))][(2)]; + INT constraintsFulfilled[(1)][((8))]; + /*-------------------------------------------- */ + + /* for ( all sub frames ) ... */ + for (c = 0; c < nSubFrames; c++) { + for (i = 0; i < cm->nElements; i++) { + ELEMENT_INFO elInfo = cm->elInfo[i]; + INT ch, nChannels = elInfo.nChannelsInEl; + + if ((elInfo.elType == ID_SCE) || (elInfo.elType == ID_CPE) || + (elInfo.elType == ID_LFE)) { + /* Turn thresholds into scalefactors, optimize bit consumption and + * verify conformance */ + FDKaacEnc_EstimateScaleFactors( + psyOut[c]->psyOutElement[i]->psyOutChannel, + qcElement[c][i]->qcOutChannel, hQC->invQuant, hQC->dZoneQuantEnable, + cm->elInfo[i].nChannelsInEl); + + /*-------------------------------------------- */ + constraintsFulfilled[c][i] = 1; + iterations[c][i] = 0; + + for (ch = 0; ch < nChannels; ch++) { + chConstraintsFulfilled[c][i][ch] = 1; + calculateQuant[c][i][ch] = 1; + } + + /*-------------------------------------------- */ + + } /* -end- if(ID_SCE || ID_CPE || ID_LFE) */ + + } /* -end- element loop */ + + qcOut[c]->usedDynBits = -1; + + } /* -end- sub frame counter */ + + INT quantizationDone = 0; + INT sumDynBitsConsumedTotal = 0; + INT decreaseBitConsumption = -1; /* no direction yet! */ + + /*-------------------------------------------- */ + /* -start- Quantization loop ... */ + /*-------------------------------------------- */ + do /* until max allowed bits per frame and maxDynBits!=-1*/ + { + quantizationDone = 0; + + c = 0; /* get frame to process */ + + for (i = 0; i < cm->nElements; i++) { + ELEMENT_INFO elInfo = cm->elInfo[i]; + INT ch, nChannels = elInfo.nChannelsInEl; + + if ((elInfo.elType == ID_SCE) || (elInfo.elType == ID_CPE) || + (elInfo.elType == ID_LFE)) { + do /* until element bits < nChannels*MIN_BUFSIZE_PER_EFF_CHAN */ + { + do /* until spectral values < MAX_QUANT */ + { + /*-------------------------------------------- */ + if (!constraintsFulfilled[c][i]) { + if ((ErrorStatus = FDKaacEnc_reduceBitConsumption( + &iterations[c][i], hQC->maxIterations, + (decreaseBitConsumption) ? 1 : -1, + chConstraintsFulfilled[c][i], calculateQuant[c][i], + nChannels, psyOut[c]->psyOutElement[i], qcOut[c], + qcElement[c][i], hQC->elementBits[i], aot, syntaxFlags, + epConfig)) != AAC_ENC_OK) { + return ErrorStatus; + } + } + + /*-------------------------------------------- */ + /*-------------------------------------------- */ + constraintsFulfilled[c][i] = 1; + + /*-------------------------------------------- */ + /* quantize spectrum (per each channel) */ + for (ch = 0; ch < nChannels; ch++) { + /*-------------------------------------------- */ + chConstraintsFulfilled[c][i][ch] = 1; + + /*-------------------------------------------- */ + + if (calculateQuant[c][i][ch]) { + QC_OUT_CHANNEL* qcOutCh = qcElement[c][i]->qcOutChannel[ch]; + PSY_OUT_CHANNEL* psyOutCh = + psyOut[c]->psyOutElement[i]->psyOutChannel[ch]; + + calculateQuant[c][i][ch] = + 0; /* calculate quantization only if necessary */ + + /*-------------------------------------------- */ + FDKaacEnc_QuantizeSpectrum( + psyOutCh->sfbCnt, psyOutCh->maxSfbPerGroup, + psyOutCh->sfbPerGroup, psyOutCh->sfbOffsets, + qcOutCh->mdctSpectrum, qcOutCh->globalGain, qcOutCh->scf, + qcOutCh->quantSpec, hQC->dZoneQuantEnable); + + /*-------------------------------------------- */ + if (FDKaacEnc_calcMaxValueInSfb( + psyOutCh->sfbCnt, psyOutCh->maxSfbPerGroup, + psyOutCh->sfbPerGroup, psyOutCh->sfbOffsets, + qcOutCh->quantSpec, + qcOutCh->maxValueInSfb) > MAX_QUANT) { + chConstraintsFulfilled[c][i][ch] = 0; + constraintsFulfilled[c][i] = 0; + /* if quanizted value out of range; increase global gain! */ + decreaseBitConsumption = 1; + } + + /*-------------------------------------------- */ + + } /* if calculateQuant[c][i][ch] */ + + } /* channel loop */ + + /*-------------------------------------------- */ + /* quantize spectrum (per each channel) */ + + /*-------------------------------------------- */ + + } while (!constraintsFulfilled[c][i]); /* does not regard bit + consumption */ + + /*-------------------------------------------- */ + /*-------------------------------------------- */ + qcElement[c][i]->dynBitsUsed = 0; /* reset dynamic bits */ + + /* quantization valid in current channel! */ + for (ch = 0; ch < nChannels; ch++) { + QC_OUT_CHANNEL* qcOutCh = qcElement[c][i]->qcOutChannel[ch]; + PSY_OUT_CHANNEL* psyOutCh = + psyOut[c]->psyOutElement[i]->psyOutChannel[ch]; + + /* count dynamic bits */ + INT chDynBits = FDKaacEnc_dynBitCount( + hQC->hBitCounter, qcOutCh->quantSpec, qcOutCh->maxValueInSfb, + qcOutCh->scf, psyOutCh->lastWindowSequence, psyOutCh->sfbCnt, + psyOutCh->maxSfbPerGroup, psyOutCh->sfbPerGroup, + psyOutCh->sfbOffsets, &qcOutCh->sectionData, psyOutCh->noiseNrg, + psyOutCh->isBook, psyOutCh->isScale, syntaxFlags); + + /* sum up dynamic channel bits */ + qcElement[c][i]->dynBitsUsed += chDynBits; + } + + /* save dynBitsUsed for correction of bits2pe relation */ + if (hQC->hAdjThr->adjThrStateElem[i]->dynBitsLast == -1) { + hQC->hAdjThr->adjThrStateElem[i]->dynBitsLast = + qcElement[c][i]->dynBitsUsed; + } + + /* hold total bit consumption in present element below maximum allowed + */ + if (qcElement[c][i]->dynBitsUsed > + ((nChannels * MIN_BUFSIZE_PER_EFF_CHAN) - + qcElement[c][i]->staticBitsUsed - + qcElement[c][i]->extBitsUsed)) { + constraintsFulfilled[c][i] = 0; + } + + } while (!constraintsFulfilled[c][i]); + + } /* -end- if(ID_SCE || ID_CPE || ID_LFE) */ + + } /* -end- element loop */ + + /* update dynBits of current subFrame */ + FDKaacEnc_updateUsedDynBits(&qcOut[c]->usedDynBits, qcElement[c], cm); + + /* get total consumed bits, dyn bits in all sub frames have to be valid */ + sumDynBitsConsumedTotal = + FDKaacEnc_getTotalConsumedDynBits(qcOut, nSubFrames); + + if (sumDynBitsConsumedTotal == -1) { + quantizationDone = 0; /* bit consumption not valid in all sub frames */ + } else { + int sumBitsConsumedTotal = FDKaacEnc_getTotalConsumedBits( + qcOut, qcElement, cm, hQC->globHdrBits, nSubFrames); + + /* in all frames are valid dynamic bits */ + if (((sumBitsConsumedTotal < totalAvailableBits) || + sumDynBitsConsumedTotal == 0) && + (decreaseBitConsumption == 1) && + checkMinFrameBitsDemand(qcOut, hQC->minBitsPerFrame, nSubFrames) + /*()*/) { + quantizationDone = 1; /* exit bit adjustment */ + } + if (sumBitsConsumedTotal > totalAvailableBits && + (decreaseBitConsumption == 0)) { + quantizationDone = 0; /* reset! */ + } + } + + /*-------------------------------------------- */ + + int emergencyIterations = 1; + int dynBitsOvershoot = 0; + + for (c = 0; c < nSubFrames; c++) { + for (i = 0; i < cm->nElements; i++) { + ELEMENT_INFO elInfo = cm->elInfo[i]; + + if ((elInfo.elType == ID_SCE) || (elInfo.elType == ID_CPE) || + (elInfo.elType == ID_LFE)) { + /* iteration limitation */ + emergencyIterations &= + ((iterations[c][i] < hQC->maxIterations) ? 0 : 1); + } + } + /* detection if used dyn bits exceeds the maximal allowed criterion */ + dynBitsOvershoot |= + ((qcOut[c]->usedDynBits > qcOut[c]->maxDynBits) ? 1 : 0); + } + + if (quantizationDone == 0 || dynBitsOvershoot) { + int sumBitsConsumedTotal = FDKaacEnc_getTotalConsumedBits( + qcOut, qcElement, cm, hQC->globHdrBits, nSubFrames); + + if ((sumDynBitsConsumedTotal >= avgTotalDynBits) || + (sumDynBitsConsumedTotal == 0)) { + quantizationDone = 1; + } + if (emergencyIterations && (sumBitsConsumedTotal < totalAvailableBits)) { + quantizationDone = 1; + } + if ((sumBitsConsumedTotal > totalAvailableBits) || + !checkMinFrameBitsDemand(qcOut, hQC->minBitsPerFrame, nSubFrames)) { + quantizationDone = 0; + } + if ((sumBitsConsumedTotal < totalAvailableBits) && + checkMinFrameBitsDemand(qcOut, hQC->minBitsPerFrame, nSubFrames)) { + decreaseBitConsumption = 0; + } else { + decreaseBitConsumption = 1; + } + + if (dynBitsOvershoot) { + quantizationDone = 0; + decreaseBitConsumption = 1; + } + + /* reset constraints fullfilled flags */ + FDKmemclear(constraintsFulfilled, sizeof(constraintsFulfilled)); + FDKmemclear(chConstraintsFulfilled, sizeof(chConstraintsFulfilled)); + + } /* quantizationDone */ + + } while (!quantizationDone); + + /*-------------------------------------------- */ + /* ... -end- Quantization loop */ + /*-------------------------------------------- */ + + /*-------------------------------------------- */ + /*-------------------------------------------- */ + + return AAC_ENC_OK; +} + +static AAC_ENCODER_ERROR FDKaacEnc_reduceBitConsumption( + int* iterations, const int maxIterations, int gainAdjustment, + int* chConstraintsFulfilled, int* calculateQuant, int nChannels, + PSY_OUT_ELEMENT* psyOutElement, QC_OUT* qcOut, QC_OUT_ELEMENT* qcOutElement, + ELEMENT_BITS* elBits, AUDIO_OBJECT_TYPE aot, UINT syntaxFlags, + SCHAR epConfig) { + int ch; + + /** SOLVING PROBLEM **/ + if ((*iterations) < maxIterations) { + /* increase gain (+ next iteration) */ + for (ch = 0; ch < nChannels; ch++) { + if (!chConstraintsFulfilled[ch]) { + qcOutElement->qcOutChannel[ch]->globalGain += gainAdjustment; + calculateQuant[ch] = 1; /* global gain has changed, recalculate + quantization in next iteration! */ + } + } + } else if ((*iterations) == maxIterations) { + if (qcOutElement->dynBitsUsed == 0) { + return AAC_ENC_QUANT_ERROR; + } else { + /* crash recovery */ + INT bitsToSave = 0; + if ((bitsToSave = fixMax( + (qcOutElement->dynBitsUsed + 8) - + (elBits->bitResLevelEl + qcOutElement->grantedDynBits), + (qcOutElement->dynBitsUsed + qcOutElement->staticBitsUsed + 8) - + (elBits->maxBitsEl))) > 0) { + FDKaacEnc_crashRecovery(nChannels, psyOutElement, qcOut, qcOutElement, + bitsToSave, aot, syntaxFlags, epConfig); + } else { + for (ch = 0; ch < nChannels; ch++) { + qcOutElement->qcOutChannel[ch]->globalGain += 1; + } + } + for (ch = 0; ch < nChannels; ch++) { + calculateQuant[ch] = 1; + } + } + } else { + /* (*iterations) > maxIterations */ + return AAC_ENC_QUANT_ERROR; + } + (*iterations)++; + + return AAC_ENC_OK; +} + +AAC_ENCODER_ERROR FDKaacEnc_updateFillBits(CHANNEL_MAPPING* cm, + QC_STATE* qcKernel, + ELEMENT_BITS* RESTRICT elBits[((8))], + QC_OUT** qcOut) { + switch (qcKernel->bitrateMode) { + case QCDATA_BR_MODE_SFR: + break; + + case QCDATA_BR_MODE_FF: + break; + case QCDATA_BR_MODE_VBR_1: + case QCDATA_BR_MODE_VBR_2: + case QCDATA_BR_MODE_VBR_3: + case QCDATA_BR_MODE_VBR_4: + case QCDATA_BR_MODE_VBR_5: + qcOut[0]->totFillBits = + (qcOut[0]->grantedDynBits - qcOut[0]->usedDynBits) & + 7; /* precalculate alignment bits */ + qcOut[0]->totalBits = qcOut[0]->staticBits + qcOut[0]->usedDynBits + + qcOut[0]->totFillBits + qcOut[0]->elementExtBits + + qcOut[0]->globalExtBits; + qcOut[0]->totFillBits += + (fixMax(0, qcKernel->minBitsPerFrame - qcOut[0]->totalBits) + 7) & ~7; + break; + case QCDATA_BR_MODE_CBR: + case QCDATA_BR_MODE_INVALID: + default: + INT bitResSpace = qcKernel->bitResTotMax - qcKernel->bitResTot; + /* processing fill-bits */ + INT deltaBitRes = qcOut[0]->grantedDynBits - qcOut[0]->usedDynBits; + qcOut[0]->totFillBits = fixMax( + (deltaBitRes & 7), (deltaBitRes - (fixMax(0, bitResSpace - 7) & ~7))); + qcOut[0]->totalBits = qcOut[0]->staticBits + qcOut[0]->usedDynBits + + qcOut[0]->totFillBits + qcOut[0]->elementExtBits + + qcOut[0]->globalExtBits; + qcOut[0]->totFillBits += + (fixMax(0, qcKernel->minBitsPerFrame - qcOut[0]->totalBits) + 7) & ~7; + break; + } /* switch (qcKernel->bitrateMode) */ + + return AAC_ENC_OK; +} + +/********************************************************************************* + + functionname: FDKaacEnc_calcMaxValueInSfb + description: + return: + +**********************************************************************************/ + +static INT FDKaacEnc_calcMaxValueInSfb(INT sfbCnt, INT maxSfbPerGroup, + INT sfbPerGroup, INT* RESTRICT sfbOffset, + SHORT* RESTRICT quantSpectrum, + UINT* RESTRICT maxValue) { + INT sfbOffs, sfb; + INT maxValueAll = 0; + + for (sfbOffs = 0; sfbOffs < sfbCnt; sfbOffs += sfbPerGroup) + for (sfb = 0; sfb < maxSfbPerGroup; sfb++) { + INT line; + INT maxThisSfb = 0; + for (line = sfbOffset[sfbOffs + sfb]; line < sfbOffset[sfbOffs + sfb + 1]; + line++) { + INT tmp = fixp_abs(quantSpectrum[line]); + maxThisSfb = fixMax(tmp, maxThisSfb); + } + + maxValue[sfbOffs + sfb] = maxThisSfb; + maxValueAll = fixMax(maxThisSfb, maxValueAll); + } + return maxValueAll; +} + +/********************************************************************************* + + functionname: FDKaacEnc_updateBitres + description: + return: + +**********************************************************************************/ +void FDKaacEnc_updateBitres(CHANNEL_MAPPING* cm, QC_STATE* qcKernel, + QC_OUT** qcOut) { + switch (qcKernel->bitrateMode) { + case QCDATA_BR_MODE_VBR_1: + case QCDATA_BR_MODE_VBR_2: + case QCDATA_BR_MODE_VBR_3: + case QCDATA_BR_MODE_VBR_4: + case QCDATA_BR_MODE_VBR_5: + /* variable bitrate */ + qcKernel->bitResTot = + fMin(qcKernel->maxBitsPerFrame, qcKernel->bitResTotMax); + break; + case QCDATA_BR_MODE_CBR: + case QCDATA_BR_MODE_SFR: + case QCDATA_BR_MODE_INVALID: + default: + int c = 0; + /* constant bitrate */ + { + qcKernel->bitResTot += qcOut[c]->grantedDynBits - + (qcOut[c]->usedDynBits + qcOut[c]->totFillBits + + qcOut[c]->alignBits); + } + break; + } +} + +/********************************************************************************* + + functionname: FDKaacEnc_FinalizeBitConsumption + description: + return: + +**********************************************************************************/ +AAC_ENCODER_ERROR FDKaacEnc_FinalizeBitConsumption( + CHANNEL_MAPPING* cm, QC_STATE* qcKernel, QC_OUT* qcOut, + QC_OUT_ELEMENT** qcElement, HANDLE_TRANSPORTENC hTpEnc, + AUDIO_OBJECT_TYPE aot, UINT syntaxFlags, SCHAR epConfig) { + QC_OUT_EXTENSION fillExtPayload; + INT totFillBits, alignBits; + + /* Get total consumed bits in AU */ + qcOut->totalBits = qcOut->staticBits + qcOut->usedDynBits + + qcOut->totFillBits + qcOut->elementExtBits + + qcOut->globalExtBits; + + if (qcKernel->bitrateMode == QCDATA_BR_MODE_CBR) { + /* Now we can get the exact transport bit amount, and hopefully it is equal + * to the estimated value */ + INT exactTpBits = transportEnc_GetStaticBits(hTpEnc, qcOut->totalBits); + + if (exactTpBits != qcKernel->globHdrBits) { + INT diffFillBits = 0; + + /* How many bits can be take by bitreservoir */ + const INT bitresSpace = + qcKernel->bitResTotMax - + (qcKernel->bitResTot + + (qcOut->grantedDynBits - (qcOut->usedDynBits + qcOut->totFillBits))); + + /* Number of bits which can be moved to bitreservoir. */ + const INT bitsToBitres = qcKernel->globHdrBits - exactTpBits; + FDK_ASSERT(bitsToBitres >= 0); /* is always positive */ + + /* If bitreservoir can not take all bits, move ramaining bits to fillbits + */ + diffFillBits = fMax(0, bitsToBitres - bitresSpace); + + /* Assure previous alignment */ + diffFillBits = (diffFillBits + 7) & ~7; + + /* Move as many bits as possible to bitreservoir */ + qcKernel->bitResTot += (bitsToBitres - diffFillBits); + + /* Write remaing bits as fill bits */ + qcOut->totFillBits += diffFillBits; + qcOut->totalBits += diffFillBits; + qcOut->grantedDynBits += diffFillBits; + + /* Get new header bits */ + qcKernel->globHdrBits = + transportEnc_GetStaticBits(hTpEnc, qcOut->totalBits); + + if (qcKernel->globHdrBits != exactTpBits) { + /* In previous step, fill bits and corresponding total bits were changed + when bitreservoir was completely filled. Now we can take the too much + taken bits caused by header overhead from bitreservoir. + */ + qcKernel->bitResTot -= (qcKernel->globHdrBits - exactTpBits); + } + } + + } /* MODE_CBR */ + + /* Update exact number of consumed header bits. */ + qcKernel->globHdrBits = transportEnc_GetStaticBits(hTpEnc, qcOut->totalBits); + + /* Save total fill bits and distribut to alignment and fill bits */ + totFillBits = qcOut->totFillBits; + + /* fake a fill extension payload */ + FDKmemclear(&fillExtPayload, sizeof(QC_OUT_EXTENSION)); + + fillExtPayload.type = EXT_FILL_DATA; + fillExtPayload.nPayloadBits = totFillBits; + + /* ask bitstream encoder how many of that bits can be written in a fill + * extension data entity */ + qcOut->totFillBits = FDKaacEnc_writeExtensionData(NULL, &fillExtPayload, 0, 0, + syntaxFlags, aot, epConfig); + + //fprintf(stderr, "FinalizeBitConsumption(): totFillBits=%d, qcOut->totFillBits=%d \n", totFillBits, qcOut->totFillBits); + + /* now distribute extra fillbits and alignbits */ + alignBits = + 7 - (qcOut->staticBits + qcOut->usedDynBits + qcOut->elementExtBits + + qcOut->totFillBits + qcOut->globalExtBits - 1) % + 8; + + /* Maybe we could remove this */ + if (((alignBits + qcOut->totFillBits - totFillBits) == 8) && + (qcOut->totFillBits > 8)) + qcOut->totFillBits -= 8; + + qcOut->totalBits = qcOut->staticBits + qcOut->usedDynBits + + qcOut->totFillBits + alignBits + qcOut->elementExtBits + + qcOut->globalExtBits; + + if ((qcOut->totalBits > qcKernel->maxBitsPerFrame) || + (qcOut->totalBits < qcKernel->minBitsPerFrame)) { + return AAC_ENC_QUANT_ERROR; + } + + qcOut->alignBits = alignBits; + + return AAC_ENC_OK; +} + +/********************************************************************************* + + functionname: FDKaacEnc_crashRecovery + description: fulfills constraints by means of brute force... + => bits are saved by cancelling out spectral lines!! + (beginning at the highest frequencies) + return: errorcode + +**********************************************************************************/ + +static void FDKaacEnc_crashRecovery(INT nChannels, + PSY_OUT_ELEMENT* psyOutElement, + QC_OUT* qcOut, QC_OUT_ELEMENT* qcElement, + INT bitsToSave, AUDIO_OBJECT_TYPE aot, + UINT syntaxFlags, SCHAR epConfig) { + INT ch; + INT savedBits = 0; + INT sfb, sfbGrp; + INT bitsPerScf[(2)][MAX_GROUPED_SFB]; + INT sectionToScf[(2)][MAX_GROUPED_SFB]; + INT* sfbOffset; + INT sect, statBitsNew; + QC_OUT_CHANNEL** qcChannel = qcElement->qcOutChannel; + PSY_OUT_CHANNEL** psyChannel = psyOutElement->psyOutChannel; + + /* create a table which converts frq-bins to bit-demand... [bitsPerScf] */ + /* ...and another one which holds the corresponding sections [sectionToScf] */ + for (ch = 0; ch < nChannels; ch++) { + sfbOffset = psyChannel[ch]->sfbOffsets; + + for (sect = 0; sect < qcChannel[ch]->sectionData.noOfSections; sect++) { + INT codeBook = qcChannel[ch]->sectionData.huffsection[sect].codeBook; + + for (sfb = qcChannel[ch]->sectionData.huffsection[sect].sfbStart; + sfb < qcChannel[ch]->sectionData.huffsection[sect].sfbStart + + qcChannel[ch]->sectionData.huffsection[sect].sfbCnt; + sfb++) { + bitsPerScf[ch][sfb] = 0; + if ((codeBook != CODE_BOOK_PNS_NO) /*&& + (sfb < (qcChannel[ch]->sectionData.noOfGroups*qcChannel[ch]->sectionData.maxSfbPerGroup))*/) { + INT sfbStartLine = sfbOffset[sfb]; + INT noOfLines = sfbOffset[sfb + 1] - sfbStartLine; + bitsPerScf[ch][sfb] = FDKaacEnc_countValues( + &(qcChannel[ch]->quantSpec[sfbStartLine]), noOfLines, codeBook); + } + sectionToScf[ch][sfb] = sect; + } + } + } + + /* LOWER [maxSfb] IN BOTH CHANNELS!! */ + /* Attention: in case of stereo: maxSfbL == maxSfbR, GroupingL == GroupingR ; + */ + + for (sfb = qcChannel[0]->sectionData.maxSfbPerGroup - 1; sfb >= 0; sfb--) { + for (sfbGrp = 0; sfbGrp < psyChannel[0]->sfbCnt; + sfbGrp += psyChannel[0]->sfbPerGroup) { + for (ch = 0; ch < nChannels; ch++) { + sect = sectionToScf[ch][sfbGrp + sfb]; + qcChannel[ch]->sectionData.huffsection[sect].sfbCnt--; + savedBits += bitsPerScf[ch][sfbGrp + sfb]; + + if (qcChannel[ch]->sectionData.huffsection[sect].sfbCnt == 0) { + savedBits += (psyChannel[ch]->lastWindowSequence != SHORT_WINDOW) + ? FDKaacEnc_sideInfoTabLong[0] + : FDKaacEnc_sideInfoTabShort[0]; + } + } + } + + /* ...have enough bits been saved? */ + if (savedBits >= bitsToSave) break; + + } /* sfb loop */ + + /* if not enough bits saved, + clean whole spectrum and remove side info overhead */ + if (sfb == -1) { + sfb = 0; + } + + for (ch = 0; ch < nChannels; ch++) { + qcChannel[ch]->sectionData.maxSfbPerGroup = sfb; + psyChannel[ch]->maxSfbPerGroup = sfb; + /* when no spectrum is coded save tools info in bitstream */ + if (sfb == 0) { + FDKmemclear(&psyChannel[ch]->tnsInfo, sizeof(TNS_INFO)); + FDKmemclear(&psyOutElement->toolsInfo, sizeof(TOOLSINFO)); + } + } + /* dynamic bits will be updated in iteration loop */ + + { /* if stop sfb has changed save bits in side info, e.g. MS or TNS coding */ + ELEMENT_INFO elInfo; + + FDKmemclear(&elInfo, sizeof(ELEMENT_INFO)); + elInfo.nChannelsInEl = nChannels; + elInfo.elType = (nChannels == 2) ? ID_CPE : ID_SCE; + + FDKaacEnc_ChannelElementWrite(NULL, &elInfo, NULL, psyOutElement, + psyChannel, syntaxFlags, aot, epConfig, + &statBitsNew, 0); + } + + savedBits = qcElement->staticBitsUsed - statBitsNew; + + /* update static and dynamic bits */ + qcElement->staticBitsUsed -= savedBits; + qcElement->grantedDynBits += savedBits; + + qcOut->staticBits -= savedBits; + qcOut->grantedDynBits += savedBits; + qcOut->maxDynBits += savedBits; +} + +void FDKaacEnc_QCClose(QC_STATE** phQCstate, QC_OUT** phQC) { + int n, i; + + if (phQC != NULL) { + for (n = 0; n < (1); n++) { + if (phQC[n] != NULL) { + QC_OUT* hQC = phQC[n]; + for (i = 0; i < (8); i++) { + } + + for (i = 0; i < ((8)); i++) { + if (hQC->qcElement[i]) FreeRam_aacEnc_QCelement(&hQC->qcElement[i]); + } + + FreeRam_aacEnc_QCout(&phQC[n]); + } + } + } + + if (phQCstate != NULL) { + if (*phQCstate != NULL) { + QC_STATE* hQCstate = *phQCstate; + + if (hQCstate->hAdjThr != NULL) FDKaacEnc_AdjThrClose(&hQCstate->hAdjThr); + + if (hQCstate->hBitCounter != NULL) + FDKaacEnc_BCClose(&hQCstate->hBitCounter); + + for (i = 0; i < ((8)); i++) { + if (hQCstate->elementBits[i] != NULL) { + FreeRam_aacEnc_ElementBits(&hQCstate->elementBits[i]); + } + } + FreeRam_aacEnc_QCstate(phQCstate); + } + } +} diff --git a/fdk-aac/libAACenc/src/qc_main.h b/fdk-aac/libAACenc/src/qc_main.h new file mode 100644 index 0000000..b9e8e2d --- /dev/null +++ b/fdk-aac/libAACenc/src/qc_main.h @@ -0,0 +1,158 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M. Werner + + Description: Quantizing & coding + +*******************************************************************************/ + +#ifndef QC_MAIN_H +#define QC_MAIN_H + +#include "aacenc.h" +#include "qc_data.h" +#include "interface.h" +#include "psy_main.h" +#include "tpenc_lib.h" + +/* Quantizing & coding stage */ + +AAC_ENCODER_ERROR FDKaacEnc_QCOutNew(QC_OUT **phQC, const INT nElements, + const INT nChannels, const INT nSubFrames, + UCHAR *dynamic_RAM); + +AAC_ENCODER_ERROR FDKaacEnc_QCOutInit(QC_OUT *phQC[(1)], const INT nSubFrames, + const CHANNEL_MAPPING *cm); + +AAC_ENCODER_ERROR FDKaacEnc_QCNew(QC_STATE **phQC, INT nElements, + UCHAR *dynamic_RAM); + +AAC_ENCODER_ERROR FDKaacEnc_QCInit(QC_STATE *hQC, struct QC_INIT *init, + const ULONG initFlags); + +AAC_ENCODER_ERROR FDKaacEnc_QCMainPrepare( + ELEMENT_INFO *elInfo, ATS_ELEMENT *RESTRICT adjThrStateElement, + PSY_OUT_ELEMENT *RESTRICT psyOutElement, + QC_OUT_ELEMENT *RESTRICT qcOutElement, /* returns error code */ + AUDIO_OBJECT_TYPE aot, UINT syntaxFlags, SCHAR epConfig); + +AAC_ENCODER_ERROR FDKaacEnc_QCMain(QC_STATE *RESTRICT hQC, PSY_OUT **psyOut, + QC_OUT **qcOut, INT avgTotalBits, + CHANNEL_MAPPING *cm, AUDIO_OBJECT_TYPE aot, + UINT syntaxFlags, SCHAR epConfig); + +AAC_ENCODER_ERROR FDKaacEnc_updateFillBits(CHANNEL_MAPPING *cm, + QC_STATE *qcKernel, + ELEMENT_BITS *RESTRICT elBits[((8))], + QC_OUT **qcOut); + +void FDKaacEnc_updateBitres(CHANNEL_MAPPING *cm, QC_STATE *qcKernel, + QC_OUT **qcOut); + +AAC_ENCODER_ERROR FDKaacEnc_FinalizeBitConsumption( + CHANNEL_MAPPING *cm, QC_STATE *hQC, QC_OUT *qcOut, + QC_OUT_ELEMENT **qcElement, HANDLE_TRANSPORTENC hTpEnc, + AUDIO_OBJECT_TYPE aot, UINT syntaxFlags, SCHAR epConfig); + +AAC_ENCODER_ERROR FDKaacEnc_AdjustBitrate(QC_STATE *RESTRICT hQC, + CHANNEL_MAPPING *RESTRICT cm, + INT *avgTotalBits, INT bitRate, + INT sampleRate, INT granuleLength); + +void FDKaacEnc_QCClose(QC_STATE **phQCstate, QC_OUT **phQC); + +#endif /* QC_MAIN_H */ diff --git a/fdk-aac/libAACenc/src/quantize.cpp b/fdk-aac/libAACenc/src/quantize.cpp new file mode 100644 index 0000000..4d25263 --- /dev/null +++ b/fdk-aac/libAACenc/src/quantize.cpp @@ -0,0 +1,401 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M.Werner + + Description: Quantization + +*******************************************************************************/ + +#include "quantize.h" + +#include "aacEnc_rom.h" + +/***************************************************************************** + + functionname: FDKaacEnc_quantizeLines + description: quantizes spectrum lines + returns: + input: global gain, number of lines to process, spectral data + output: quantized spectrum + +*****************************************************************************/ +static void FDKaacEnc_quantizeLines(INT gain, INT noOfLines, + const FIXP_DBL *mdctSpectrum, + SHORT *quaSpectrum, INT dZoneQuantEnable) { + int line; + FIXP_DBL k = FL2FXCONST_DBL(0.0f); + FIXP_QTD quantizer = FDKaacEnc_quantTableQ[(-gain) & 3]; + INT quantizershift = ((-gain) >> 2) + 1; + const INT kShift = 16; + + if (dZoneQuantEnable) + k = FL2FXCONST_DBL(0.23f) >> kShift; + else + k = FL2FXCONST_DBL(-0.0946f + 0.5f) >> kShift; + + for (line = 0; line < noOfLines; line++) { + FIXP_DBL accu = fMultDiv2(mdctSpectrum[line], quantizer); + + if (accu < FL2FXCONST_DBL(0.0f)) { + accu = -accu; + /* normalize */ + INT accuShift = CntLeadingZeros(accu) - 1; /* CountLeadingBits() is not + necessary here since test + value is always > 0 */ + accu <<= accuShift; + INT tabIndex = + (INT)(accu >> (DFRACT_BITS - 2 - MANT_DIGITS)) & (~MANT_SIZE); + INT totalShift = quantizershift - accuShift + 1; + accu = fMultDiv2(FDKaacEnc_mTab_3_4[tabIndex], + FDKaacEnc_quantTableE[totalShift & 3]); + totalShift = (16 - 4) - (3 * (totalShift >> 2)); + FDK_ASSERT(totalShift >= 0); /* MAX_QUANT_VIOLATION */ + accu >>= fixMin(totalShift, DFRACT_BITS - 1); + quaSpectrum[line] = + (SHORT)(-((LONG)(k + accu) >> (DFRACT_BITS - 1 - 16))); + } else if (accu > FL2FXCONST_DBL(0.0f)) { + /* normalize */ + INT accuShift = CntLeadingZeros(accu) - 1; /* CountLeadingBits() is not + necessary here since test + value is always > 0 */ + accu <<= accuShift; + INT tabIndex = + (INT)(accu >> (DFRACT_BITS - 2 - MANT_DIGITS)) & (~MANT_SIZE); + INT totalShift = quantizershift - accuShift + 1; + accu = fMultDiv2(FDKaacEnc_mTab_3_4[tabIndex], + FDKaacEnc_quantTableE[totalShift & 3]); + totalShift = (16 - 4) - (3 * (totalShift >> 2)); + FDK_ASSERT(totalShift >= 0); /* MAX_QUANT_VIOLATION */ + accu >>= fixMin(totalShift, DFRACT_BITS - 1); + quaSpectrum[line] = (SHORT)((LONG)(k + accu) >> (DFRACT_BITS - 1 - 16)); + } else { + quaSpectrum[line] = 0; + } + } +} + +/***************************************************************************** + + functionname:iFDKaacEnc_quantizeLines + description: iquantizes spectrum lines + mdctSpectrum = iquaSpectrum^4/3 *2^(0.25*gain) + input: global gain, number of lines to process,quantized spectrum + output: spectral data + +*****************************************************************************/ +static void FDKaacEnc_invQuantizeLines(INT gain, INT noOfLines, + SHORT *quantSpectrum, + FIXP_DBL *mdctSpectrum) + +{ + INT iquantizermod; + INT iquantizershift; + INT line; + + iquantizermod = gain & 3; + iquantizershift = gain >> 2; + + for (line = 0; line < noOfLines; line++) { + if (quantSpectrum[line] < 0) { + FIXP_DBL accu; + INT ex, specExp, tabIndex; + FIXP_DBL s, t; + + accu = (FIXP_DBL)-quantSpectrum[line]; + + ex = CountLeadingBits(accu); + accu <<= ex; + specExp = (DFRACT_BITS - 1) - ex; + + FDK_ASSERT(specExp < 14); /* this fails if abs(value) > 8191 */ + + tabIndex = (INT)(accu >> (DFRACT_BITS - 2 - MANT_DIGITS)) & (~MANT_SIZE); + + /* calculate "mantissa" ^4/3 */ + s = FDKaacEnc_mTab_4_3Elc[tabIndex]; + + /* get approperiate exponent multiplier for specExp^3/4 combined with + * scfMod */ + t = FDKaacEnc_specExpMantTableCombElc[iquantizermod][specExp]; + + /* multiply "mantissa" ^4/3 with exponent multiplier */ + accu = fMult(s, t); + + /* get approperiate exponent shifter */ + specExp = FDKaacEnc_specExpTableComb[iquantizermod][specExp] - + 1; /* -1 to avoid overflows in accu */ + + if ((-iquantizershift - specExp) < 0) + accu <<= -(-iquantizershift - specExp); + else + accu >>= -iquantizershift - specExp; + + mdctSpectrum[line] = -accu; + } else if (quantSpectrum[line] > 0) { + FIXP_DBL accu; + INT ex, specExp, tabIndex; + FIXP_DBL s, t; + + accu = (FIXP_DBL)(INT)quantSpectrum[line]; + + ex = CountLeadingBits(accu); + accu <<= ex; + specExp = (DFRACT_BITS - 1) - ex; + + FDK_ASSERT(specExp < 14); /* this fails if abs(value) > 8191 */ + + tabIndex = (INT)(accu >> (DFRACT_BITS - 2 - MANT_DIGITS)) & (~MANT_SIZE); + + /* calculate "mantissa" ^4/3 */ + s = FDKaacEnc_mTab_4_3Elc[tabIndex]; + + /* get approperiate exponent multiplier for specExp^3/4 combined with + * scfMod */ + t = FDKaacEnc_specExpMantTableCombElc[iquantizermod][specExp]; + + /* multiply "mantissa" ^4/3 with exponent multiplier */ + accu = fMult(s, t); + + /* get approperiate exponent shifter */ + specExp = FDKaacEnc_specExpTableComb[iquantizermod][specExp] - + 1; /* -1 to avoid overflows in accu */ + + if ((-iquantizershift - specExp) < 0) + accu <<= -(-iquantizershift - specExp); + else + accu >>= -iquantizershift - specExp; + + mdctSpectrum[line] = accu; + } else { + mdctSpectrum[line] = FL2FXCONST_DBL(0.0f); + } + } +} + +/***************************************************************************** + + functionname: FDKaacEnc_QuantizeSpectrum + description: quantizes the entire spectrum + returns: + input: number of scalefactor bands to be quantized, ... + output: quantized spectrum + +*****************************************************************************/ +void FDKaacEnc_QuantizeSpectrum(INT sfbCnt, INT maxSfbPerGroup, INT sfbPerGroup, + const INT *sfbOffset, + const FIXP_DBL *mdctSpectrum, INT globalGain, + const INT *scalefactors, + SHORT *quantizedSpectrum, + INT dZoneQuantEnable) { + INT sfbOffs, sfb; + + /* in FDKaacEnc_quantizeLines quaSpectrum is calculated with: + spec^(3/4) * 2^(-3/16*QSS) * 2^(3/4*scale) + k + simplify scaling calculation and reduce QSS before: + spec^(3/4) * 2^(-3/16*(QSS - 4*scale)) */ + + for (sfbOffs = 0; sfbOffs < sfbCnt; sfbOffs += sfbPerGroup) + for (sfb = 0; sfb < maxSfbPerGroup; sfb++) { + INT scalefactor = scalefactors[sfbOffs + sfb]; + + FDKaacEnc_quantizeLines( + globalGain - scalefactor, /* QSS */ + sfbOffset[sfbOffs + sfb + 1] - sfbOffset[sfbOffs + sfb], + mdctSpectrum + sfbOffset[sfbOffs + sfb], + quantizedSpectrum + sfbOffset[sfbOffs + sfb], dZoneQuantEnable); + } +} + +/***************************************************************************** + + functionname: FDKaacEnc_calcSfbDist + description: calculates distortion of quantized values + returns: distortion + input: gain, number of lines to process, spectral data + output: + +*****************************************************************************/ +FIXP_DBL FDKaacEnc_calcSfbDist(const FIXP_DBL *mdctSpectrum, + SHORT *quantSpectrum, INT noOfLines, INT gain, + INT dZoneQuantEnable) { + INT i, scale; + FIXP_DBL xfsf; + FIXP_DBL diff; + FIXP_DBL invQuantSpec; + + xfsf = FL2FXCONST_DBL(0.0f); + + for (i = 0; i < noOfLines; i++) { + /* quantization */ + FDKaacEnc_quantizeLines(gain, 1, &mdctSpectrum[i], &quantSpectrum[i], + dZoneQuantEnable); + + if (fAbs(quantSpectrum[i]) > MAX_QUANT) { + return FL2FXCONST_DBL(0.0f); + } + /* inverse quantization */ + FDKaacEnc_invQuantizeLines(gain, 1, &quantSpectrum[i], &invQuantSpec); + + /* dist */ + diff = fixp_abs(fixp_abs(invQuantSpec) - fixp_abs(mdctSpectrum[i] >> 1)); + + scale = CountLeadingBits(diff); + diff = scaleValue(diff, scale); + diff = fPow2(diff); + scale = fixMin(2 * (scale - 1), DFRACT_BITS - 1); + + diff = scaleValue(diff, -scale); + + xfsf = xfsf + diff; + } + + xfsf = CalcLdData(xfsf); + + return xfsf; +} + +/***************************************************************************** + + functionname: FDKaacEnc_calcSfbQuantEnergyAndDist + description: calculates energy and distortion of quantized values + returns: + input: gain, number of lines to process, quantized spectral data, + spectral data + output: energy, distortion + +*****************************************************************************/ +void FDKaacEnc_calcSfbQuantEnergyAndDist(FIXP_DBL *mdctSpectrum, + SHORT *quantSpectrum, INT noOfLines, + INT gain, FIXP_DBL *en, + FIXP_DBL *dist) { + INT i, scale; + FIXP_DBL invQuantSpec; + FIXP_DBL diff; + + FIXP_DBL energy = FL2FXCONST_DBL(0.0f); + FIXP_DBL distortion = FL2FXCONST_DBL(0.0f); + + for (i = 0; i < noOfLines; i++) { + if (fAbs(quantSpectrum[i]) > MAX_QUANT) { + *en = FL2FXCONST_DBL(0.0f); + *dist = FL2FXCONST_DBL(0.0f); + return; + } + + /* inverse quantization */ + FDKaacEnc_invQuantizeLines(gain, 1, &quantSpectrum[i], &invQuantSpec); + + /* energy */ + energy += fPow2(invQuantSpec); + + /* dist */ + diff = fixp_abs(fixp_abs(invQuantSpec) - fixp_abs(mdctSpectrum[i] >> 1)); + + scale = CountLeadingBits(diff); + diff = scaleValue(diff, scale); + diff = fPow2(diff); + + scale = fixMin(2 * (scale - 1), DFRACT_BITS - 1); + + diff = scaleValue(diff, -scale); + + distortion += diff; + } + + *en = CalcLdData(energy) + FL2FXCONST_DBL(0.03125f); + *dist = CalcLdData(distortion); +} diff --git a/fdk-aac/libAACenc/src/quantize.h b/fdk-aac/libAACenc/src/quantize.h new file mode 100644 index 0000000..dfc2206 --- /dev/null +++ b/fdk-aac/libAACenc/src/quantize.h @@ -0,0 +1,127 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M.Werner + + Description: Quantization + +*******************************************************************************/ + +#ifndef QUANTIZE_H +#define QUANTIZE_H + +#include "common_fix.h" + +/* quantizing */ + +#define MAX_QUANT 8191 + +void FDKaacEnc_QuantizeSpectrum(INT sfbCnt, INT maxSfbPerGroup, INT sfbPerGroup, + const INT *sfbOffset, + const FIXP_DBL *mdctSpectrum, INT globalGain, + const INT *scalefactors, + SHORT *quantizedSpectrum, INT dZoneQuantEnable); + +FIXP_DBL FDKaacEnc_calcSfbDist(const FIXP_DBL *mdctSpectrum, + SHORT *quantSpectrum, INT noOfLines, INT gain, + INT dZoneQuantEnable); + +void FDKaacEnc_calcSfbQuantEnergyAndDist(FIXP_DBL *mdctSpectrum, + SHORT *quantSpectrum, INT noOfLines, + INT gain, FIXP_DBL *en, + FIXP_DBL *dist); + +#endif /* QUANTIZE_H */ diff --git a/fdk-aac/libAACenc/src/sf_estim.cpp b/fdk-aac/libAACenc/src/sf_estim.cpp new file mode 100644 index 0000000..17a8ae2 --- /dev/null +++ b/fdk-aac/libAACenc/src/sf_estim.cpp @@ -0,0 +1,1292 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M. Werner + + Description: Scale factor estimation + +*******************************************************************************/ + +#include "sf_estim.h" +#include "aacEnc_rom.h" +#include "quantize.h" +#include "bit_cnt.h" + +#ifdef __arm__ +#endif + +#define UPCOUNT_LIMIT 1 +#define AS_PE_FAC_SHIFT 7 +#define DIST_FAC_SHIFT 3 +#define AS_PE_FAC_FLOAT (float)(1 << AS_PE_FAC_SHIFT) +static const INT MAX_SCF_DELTA = 60; + +static const FIXP_DBL PE_C1 = FL2FXCONST_DBL( + 3.0f / AS_PE_FAC_FLOAT); /* (log(8.0)/log(2)) >> AS_PE_FAC_SHIFT */ +static const FIXP_DBL PE_C2 = FL2FXCONST_DBL( + 1.3219281f / AS_PE_FAC_FLOAT); /* (log(2.5)/log(2)) >> AS_PE_FAC_SHIFT */ +static const FIXP_DBL PE_C3 = FL2FXCONST_DBL(0.5593573f); /* 1-C2/C1 */ + +/* + Function; FDKaacEnc_FDKaacEnc_CalcFormFactorChannel + + Description: Calculates the formfactor + + sf: scale factor of the mdct spectrum + sfbFormFactorLdData is scaled with the factor 1/(((2^sf)^0.5) * + (2^FORM_FAC_SHIFT)) +*/ +static void FDKaacEnc_FDKaacEnc_CalcFormFactorChannel( + FIXP_DBL *RESTRICT sfbFormFactorLdData, + PSY_OUT_CHANNEL *RESTRICT psyOutChan) { + INT j, sfb, sfbGrp; + FIXP_DBL formFactor; + + int tmp0 = psyOutChan->sfbCnt; + int tmp1 = psyOutChan->maxSfbPerGroup; + int step = psyOutChan->sfbPerGroup; + for (sfbGrp = 0; sfbGrp < tmp0; sfbGrp += step) { + for (sfb = 0; sfb < tmp1; sfb++) { + formFactor = FL2FXCONST_DBL(0.0f); + /* calc sum of sqrt(spec) */ + for (j = psyOutChan->sfbOffsets[sfbGrp + sfb]; + j < psyOutChan->sfbOffsets[sfbGrp + sfb + 1]; j++) { + formFactor += + sqrtFixp(fixp_abs(psyOutChan->mdctSpectrum[j])) >> FORM_FAC_SHIFT; + } + sfbFormFactorLdData[sfbGrp + sfb] = CalcLdData(formFactor); + } + /* set sfbFormFactor for sfbs with zero spec to zero. Just for debugging. */ + for (; sfb < psyOutChan->sfbPerGroup; sfb++) { + sfbFormFactorLdData[sfbGrp + sfb] = FL2FXCONST_DBL(-1.0f); + } + } +} + +/* + Function: FDKaacEnc_CalcFormFactor + + Description: Calls FDKaacEnc_FDKaacEnc_CalcFormFactorChannel() for each + channel +*/ + +void FDKaacEnc_CalcFormFactor(QC_OUT_CHANNEL *qcOutChannel[(2)], + PSY_OUT_CHANNEL *psyOutChannel[(2)], + const INT nChannels) { + INT j; + for (j = 0; j < nChannels; j++) { + FDKaacEnc_FDKaacEnc_CalcFormFactorChannel( + qcOutChannel[j]->sfbFormFactorLdData, psyOutChannel[j]); + } +} + +/* + Function: FDKaacEnc_calcSfbRelevantLines + + Description: Calculates sfbNRelevantLines + + sfbNRelevantLines is scaled with the factor 1/((2^FORM_FAC_SHIFT) * 2.0) +*/ +static void FDKaacEnc_calcSfbRelevantLines( + const FIXP_DBL *const sfbFormFactorLdData, + const FIXP_DBL *const sfbEnergyLdData, + const FIXP_DBL *const sfbThresholdLdData, const INT *const sfbOffsets, + const INT sfbCnt, const INT sfbPerGroup, const INT maxSfbPerGroup, + FIXP_DBL *sfbNRelevantLines) { + INT sfbOffs, sfb; + FIXP_DBL sfbWidthLdData; + FIXP_DBL asPeFacLdData = + FL2FXCONST_DBL(0.109375); /* AS_PE_FAC_SHIFT*ld64(2) */ + FIXP_DBL accu; + + /* sfbNRelevantLines[i] = 2^( (sfbFormFactorLdData[i] - 0.25 * + * (sfbEnergyLdData[i] - ld64(sfbWidth[i]/(2^7)) - AS_PE_FAC_SHIFT*ld64(2)) * + * 64); */ + + FDKmemclear(sfbNRelevantLines, sfbCnt * sizeof(FIXP_DBL)); + + for (sfbOffs = 0; sfbOffs < sfbCnt; sfbOffs += sfbPerGroup) { + for (sfb = 0; sfb < maxSfbPerGroup; sfb++) { + /* calc sum of sqrt(spec) */ + if ((FIXP_DBL)sfbEnergyLdData[sfbOffs + sfb] > + (FIXP_DBL)sfbThresholdLdData[sfbOffs + sfb]) { + INT sfbWidth = + sfbOffsets[sfbOffs + sfb + 1] - sfbOffsets[sfbOffs + sfb]; + + /* avgFormFactorLdData = + * sqrtFixp(sqrtFixp(sfbEnergyLdData[sfbOffs+sfb]/sfbWidth)); */ + /* sfbNRelevantLines[sfbOffs+sfb] = sfbFormFactor[sfbOffs+sfb] / + * avgFormFactorLdData; */ + sfbWidthLdData = + (FIXP_DBL)(sfbWidth << (DFRACT_BITS - 1 - AS_PE_FAC_SHIFT)); + sfbWidthLdData = CalcLdData(sfbWidthLdData); + + accu = sfbEnergyLdData[sfbOffs + sfb] - sfbWidthLdData - asPeFacLdData; + accu = sfbFormFactorLdData[sfbOffs + sfb] - (accu >> 2); + + sfbNRelevantLines[sfbOffs + sfb] = CalcInvLdData(accu) >> 1; + } + } + } +} + +/* + Function: FDKaacEnc_countSingleScfBits + + Description: + + scfBitsFract is scaled by 1/(2^(2*AS_PE_FAC_SHIFT)) +*/ +static FIXP_DBL FDKaacEnc_countSingleScfBits(INT scf, INT scfLeft, + INT scfRight) { + FIXP_DBL scfBitsFract; + + scfBitsFract = (FIXP_DBL)(FDKaacEnc_bitCountScalefactorDelta(scfLeft - scf) + + FDKaacEnc_bitCountScalefactorDelta(scf - scfRight)); + + scfBitsFract = scfBitsFract << (DFRACT_BITS - 1 - (2 * AS_PE_FAC_SHIFT)); + + return scfBitsFract; /* output scaled by 1/(2^(2*AS_PE_FAC)) */ +} + +/* + Function: FDKaacEnc_calcSingleSpecPe + + specPe is scaled by 1/(2^(2*AS_PE_FAC_SHIFT)) +*/ +static FIXP_DBL FDKaacEnc_calcSingleSpecPe(INT scf, FIXP_DBL sfbConstPePart, + FIXP_DBL nLines) { + FIXP_DBL specPe = FL2FXCONST_DBL(0.0f); + FIXP_DBL ldRatio; + FIXP_DBL scfFract; + + scfFract = (FIXP_DBL)(scf << (DFRACT_BITS - 1 - AS_PE_FAC_SHIFT)); + + ldRatio = sfbConstPePart - fMult(FL2FXCONST_DBL(0.375f), scfFract); + + if (ldRatio >= PE_C1) { + specPe = fMult(FL2FXCONST_DBL(0.7f), fMult(nLines, ldRatio)); + } else { + specPe = fMult(FL2FXCONST_DBL(0.7f), + fMult(nLines, (PE_C2 + fMult(PE_C3, ldRatio)))); + } + + return specPe; /* output scaled by 1/(2^(2*AS_PE_FAC)) */ +} + +/* + Function: FDKaacEnc_countScfBitsDiff + + scfBitsDiff is scaled by 1/(2^(2*AS_PE_FAC_SHIFT)) +*/ +static FIXP_DBL FDKaacEnc_countScfBitsDiff(INT *scfOld, INT *scfNew, INT sfbCnt, + INT startSfb, INT stopSfb) { + FIXP_DBL scfBitsFract; + INT scfBitsDiff = 0; + INT sfb = 0, sfbLast; + INT sfbPrev, sfbNext; + + /* search for first relevant sfb */ + sfbLast = startSfb; + while ((sfbLast < stopSfb) && (scfOld[sfbLast] == FDK_INT_MIN)) sfbLast++; + /* search for previous relevant sfb and count diff */ + sfbPrev = startSfb - 1; + while ((sfbPrev >= 0) && (scfOld[sfbPrev] == FDK_INT_MIN)) sfbPrev--; + if (sfbPrev >= 0) + scfBitsDiff += + FDKaacEnc_bitCountScalefactorDelta(scfNew[sfbPrev] - scfNew[sfbLast]) - + FDKaacEnc_bitCountScalefactorDelta(scfOld[sfbPrev] - scfOld[sfbLast]); + /* now loop through all sfbs and count diffs of relevant sfbs */ + for (sfb = sfbLast + 1; sfb < stopSfb; sfb++) { + if (scfOld[sfb] != FDK_INT_MIN) { + scfBitsDiff += + FDKaacEnc_bitCountScalefactorDelta(scfNew[sfbLast] - scfNew[sfb]) - + FDKaacEnc_bitCountScalefactorDelta(scfOld[sfbLast] - scfOld[sfb]); + sfbLast = sfb; + } + } + /* search for next relevant sfb and count diff */ + sfbNext = stopSfb; + while ((sfbNext < sfbCnt) && (scfOld[sfbNext] == FDK_INT_MIN)) sfbNext++; + if (sfbNext < sfbCnt) + scfBitsDiff += + FDKaacEnc_bitCountScalefactorDelta(scfNew[sfbLast] - scfNew[sfbNext]) - + FDKaacEnc_bitCountScalefactorDelta(scfOld[sfbLast] - scfOld[sfbNext]); + + scfBitsFract = + (FIXP_DBL)(scfBitsDiff << (DFRACT_BITS - 1 - (2 * AS_PE_FAC_SHIFT))); + + return scfBitsFract; +} + +/* + Function: FDKaacEnc_calcSpecPeDiff + + specPeDiff is scaled by 1/(2^(2*AS_PE_FAC_SHIFT)) +*/ +static FIXP_DBL FDKaacEnc_calcSpecPeDiff( + PSY_OUT_CHANNEL *psyOutChan, QC_OUT_CHANNEL *qcOutChannel, INT *scfOld, + INT *scfNew, FIXP_DBL *sfbConstPePart, FIXP_DBL *sfbFormFactorLdData, + FIXP_DBL *sfbNRelevantLines, INT startSfb, INT stopSfb) { + FIXP_DBL specPeDiff = FL2FXCONST_DBL(0.0f); + FIXP_DBL scfFract = FL2FXCONST_DBL(0.0f); + INT sfb; + + /* loop through all sfbs and count pe difference */ + for (sfb = startSfb; sfb < stopSfb; sfb++) { + if (scfOld[sfb] != FDK_INT_MIN) { + FIXP_DBL ldRatioOld, ldRatioNew, pOld, pNew; + + /* sfbConstPePart[sfb] = (float)log(psyOutChan->sfbEnergy[sfb] * 6.75f / + * sfbFormFactor[sfb]) * LOG2_1; */ + /* 0.02152255861f = log(6.75)/log(2)/AS_PE_FAC_FLOAT; LOG2_1 is 1.0 for + * log2 */ + /* 0.09375f = log(64.0)/log(2.0)/64.0 = scale of sfbFormFactorLdData */ + if (sfbConstPePart[sfb] == (FIXP_DBL)FDK_INT_MIN) + sfbConstPePart[sfb] = + ((psyOutChan->sfbEnergyLdData[sfb] - sfbFormFactorLdData[sfb] - + FL2FXCONST_DBL(0.09375f)) >> + 1) + + FL2FXCONST_DBL(0.02152255861f); + + scfFract = (FIXP_DBL)(scfOld[sfb] << (DFRACT_BITS - 1 - AS_PE_FAC_SHIFT)); + ldRatioOld = + sfbConstPePart[sfb] - fMult(FL2FXCONST_DBL(0.375f), scfFract); + + scfFract = (FIXP_DBL)(scfNew[sfb] << (DFRACT_BITS - 1 - AS_PE_FAC_SHIFT)); + ldRatioNew = + sfbConstPePart[sfb] - fMult(FL2FXCONST_DBL(0.375f), scfFract); + + if (ldRatioOld >= PE_C1) + pOld = ldRatioOld; + else + pOld = PE_C2 + fMult(PE_C3, ldRatioOld); + + if (ldRatioNew >= PE_C1) + pNew = ldRatioNew; + else + pNew = PE_C2 + fMult(PE_C3, ldRatioNew); + + specPeDiff += fMult(FL2FXCONST_DBL(0.7f), + fMult(sfbNRelevantLines[sfb], (pNew - pOld))); + } + } + + return specPeDiff; +} + +/* + Function: FDKaacEnc_improveScf + + Description: Calculate the distortion by quantization and inverse quantization + of the spectrum with various scalefactors. The scalefactor which provides the + best results will be used. +*/ +static INT FDKaacEnc_improveScf(const FIXP_DBL *spec, SHORT *quantSpec, + SHORT *quantSpecTmp, INT sfbWidth, + FIXP_DBL threshLdData, INT scf, INT minScf, + FIXP_DBL *distLdData, INT *minScfCalculated, + INT dZoneQuantEnable) { + FIXP_DBL sfbDistLdData; + INT scfBest = scf; + INT k; + FIXP_DBL distFactorLdData = FL2FXCONST_DBL(-0.0050301265); /* ld64(1/1.25) */ + + /* calc real distortion */ + sfbDistLdData = + FDKaacEnc_calcSfbDist(spec, quantSpec, sfbWidth, scf, dZoneQuantEnable); + *minScfCalculated = scf; + /* nmr > 1.25 -> try to improve nmr */ + if (sfbDistLdData > (threshLdData - distFactorLdData)) { + INT scfEstimated = scf; + FIXP_DBL sfbDistBestLdData = sfbDistLdData; + INT cnt; + /* improve by bigger scf ? */ + cnt = 0; + + while ((sfbDistLdData > (threshLdData - distFactorLdData)) && + (cnt++ < UPCOUNT_LIMIT)) { + scf++; + sfbDistLdData = FDKaacEnc_calcSfbDist(spec, quantSpecTmp, sfbWidth, scf, + dZoneQuantEnable); + + if (sfbDistLdData < sfbDistBestLdData) { + scfBest = scf; + sfbDistBestLdData = sfbDistLdData; + for (k = 0; k < sfbWidth; k++) quantSpec[k] = quantSpecTmp[k]; + } + } + /* improve by smaller scf ? */ + cnt = 0; + scf = scfEstimated; + sfbDistLdData = sfbDistBestLdData; + while ((sfbDistLdData > (threshLdData - distFactorLdData)) && (cnt++ < 1) && + (scf > minScf)) { + scf--; + sfbDistLdData = FDKaacEnc_calcSfbDist(spec, quantSpecTmp, sfbWidth, scf, + dZoneQuantEnable); + + if (sfbDistLdData < sfbDistBestLdData) { + scfBest = scf; + sfbDistBestLdData = sfbDistLdData; + for (k = 0; k < sfbWidth; k++) quantSpec[k] = quantSpecTmp[k]; + } + *minScfCalculated = scf; + } + *distLdData = sfbDistBestLdData; + } else { /* nmr <= 1.25 -> try to find bigger scf to use less bits */ + FIXP_DBL sfbDistBestLdData = sfbDistLdData; + FIXP_DBL sfbDistAllowedLdData = + fixMin(sfbDistLdData - distFactorLdData, threshLdData); + int cnt; + for (cnt = 0; cnt < UPCOUNT_LIMIT; cnt++) { + scf++; + sfbDistLdData = FDKaacEnc_calcSfbDist(spec, quantSpecTmp, sfbWidth, scf, + dZoneQuantEnable); + + if (sfbDistLdData < sfbDistAllowedLdData) { + *minScfCalculated = scfBest + 1; + scfBest = scf; + sfbDistBestLdData = sfbDistLdData; + for (k = 0; k < sfbWidth; k++) quantSpec[k] = quantSpecTmp[k]; + } + } + *distLdData = sfbDistBestLdData; + } + + /* return best scalefactor */ + return scfBest; +} + +/* + Function: FDKaacEnc_assimilateSingleScf + +*/ +static void FDKaacEnc_assimilateSingleScf( + const PSY_OUT_CHANNEL *psyOutChan, const QC_OUT_CHANNEL *qcOutChannel, + SHORT *quantSpec, SHORT *quantSpecTmp, INT dZoneQuantEnable, INT *scf, + const INT *minScf, FIXP_DBL *sfbDist, FIXP_DBL *sfbConstPePart, + const FIXP_DBL *sfbFormFactorLdData, const FIXP_DBL *sfbNRelevantLines, + INT *minScfCalculated, INT restartOnSuccess) { + INT sfbLast, sfbAct, sfbNext; + INT scfAct, *scfLast, *scfNext, scfMin, scfMax; + INT sfbWidth, sfbOffs; + FIXP_DBL enLdData; + FIXP_DBL sfbPeOld, sfbPeNew; + FIXP_DBL sfbDistNew; + INT i, k; + INT success = 0; + FIXP_DBL deltaPe = FL2FXCONST_DBL(0.0f); + FIXP_DBL deltaPeNew, deltaPeTmp; + INT prevScfLast[MAX_GROUPED_SFB], prevScfNext[MAX_GROUPED_SFB]; + FIXP_DBL deltaPeLast[MAX_GROUPED_SFB]; + INT updateMinScfCalculated; + + for (i = 0; i < psyOutChan->sfbCnt; i++) { + prevScfLast[i] = FDK_INT_MAX; + prevScfNext[i] = FDK_INT_MAX; + deltaPeLast[i] = (FIXP_DBL)FDK_INT_MAX; + } + + sfbLast = -1; + sfbAct = -1; + sfbNext = -1; + scfLast = 0; + scfNext = 0; + scfMin = FDK_INT_MAX; + scfMax = FDK_INT_MAX; + do { + /* search for new relevant sfb */ + sfbNext++; + while ((sfbNext < psyOutChan->sfbCnt) && (scf[sfbNext] == FDK_INT_MIN)) + sfbNext++; + if ((sfbLast >= 0) && (sfbAct >= 0) && (sfbNext < psyOutChan->sfbCnt)) { + /* relevant scfs to the left and to the right */ + scfAct = scf[sfbAct]; + scfLast = scf + sfbLast; + scfNext = scf + sfbNext; + scfMin = fixMin(*scfLast, *scfNext); + scfMax = fixMax(*scfLast, *scfNext); + } else if ((sfbLast == -1) && (sfbAct >= 0) && + (sfbNext < psyOutChan->sfbCnt)) { + /* first relevant scf */ + scfAct = scf[sfbAct]; + scfLast = &scfAct; + scfNext = scf + sfbNext; + scfMin = *scfNext; + scfMax = *scfNext; + } else if ((sfbLast >= 0) && (sfbAct >= 0) && + (sfbNext == psyOutChan->sfbCnt)) { + /* last relevant scf */ + scfAct = scf[sfbAct]; + scfLast = scf + sfbLast; + scfNext = &scfAct; + scfMin = *scfLast; + scfMax = *scfLast; + } + if (sfbAct >= 0) scfMin = fixMax(scfMin, minScf[sfbAct]); + + if ((sfbAct >= 0) && (sfbLast >= 0 || sfbNext < psyOutChan->sfbCnt) && + (scfAct > scfMin) && (scfAct <= scfMin + MAX_SCF_DELTA) && + (scfAct >= scfMax - MAX_SCF_DELTA) && + (scfAct <= + fixMin(scfMin, fixMin(*scfLast, *scfNext)) + MAX_SCF_DELTA) && + (*scfLast != prevScfLast[sfbAct] || *scfNext != prevScfNext[sfbAct] || + deltaPe < deltaPeLast[sfbAct])) { + /* bigger than neighbouring scf found, try to use smaller scf */ + success = 0; + + sfbWidth = + psyOutChan->sfbOffsets[sfbAct + 1] - psyOutChan->sfbOffsets[sfbAct]; + sfbOffs = psyOutChan->sfbOffsets[sfbAct]; + + /* estimate required bits for actual scf */ + enLdData = qcOutChannel->sfbEnergyLdData[sfbAct]; + + /* sfbConstPePart[sfbAct] = (float)log(6.75f*en/sfbFormFactor[sfbAct]) * + * LOG2_1; */ + /* 0.02152255861f = log(6.75)/log(2)/AS_PE_FAC_FLOAT; LOG2_1 is 1.0 for + * log2 */ + /* 0.09375f = log(64.0)/log(2.0)/64.0 = scale of sfbFormFactorLdData */ + if (sfbConstPePart[sfbAct] == (FIXP_DBL)FDK_INT_MIN) { + sfbConstPePart[sfbAct] = ((enLdData - sfbFormFactorLdData[sfbAct] - + FL2FXCONST_DBL(0.09375f)) >> + 1) + + FL2FXCONST_DBL(0.02152255861f); + } + + sfbPeOld = FDKaacEnc_calcSingleSpecPe(scfAct, sfbConstPePart[sfbAct], + sfbNRelevantLines[sfbAct]) + + FDKaacEnc_countSingleScfBits(scfAct, *scfLast, *scfNext); + + deltaPeNew = deltaPe; + updateMinScfCalculated = 1; + + do { + /* estimate required bits for smaller scf */ + scfAct--; + /* check only if the same check was not done before */ + if (scfAct < minScfCalculated[sfbAct] && + scfAct >= scfMax - MAX_SCF_DELTA) { + /* estimate required bits for new scf */ + sfbPeNew = FDKaacEnc_calcSingleSpecPe(scfAct, sfbConstPePart[sfbAct], + sfbNRelevantLines[sfbAct]) + + FDKaacEnc_countSingleScfBits(scfAct, *scfLast, *scfNext); + + /* use new scf if no increase in pe and + quantization error is smaller */ + deltaPeTmp = deltaPe + sfbPeNew - sfbPeOld; + /* 0.0006103515625f = 10.0f/(2^(2*AS_PE_FAC_SHIFT)) */ + if (deltaPeTmp < FL2FXCONST_DBL(0.0006103515625f)) { + /* distortion of new scf */ + sfbDistNew = FDKaacEnc_calcSfbDist( + qcOutChannel->mdctSpectrum + sfbOffs, quantSpecTmp + sfbOffs, + sfbWidth, scfAct, dZoneQuantEnable); + + if (sfbDistNew < sfbDist[sfbAct]) { + /* success, replace scf by new one */ + scf[sfbAct] = scfAct; + sfbDist[sfbAct] = sfbDistNew; + + for (k = 0; k < sfbWidth; k++) + quantSpec[sfbOffs + k] = quantSpecTmp[sfbOffs + k]; + + deltaPeNew = deltaPeTmp; + success = 1; + } + /* mark as already checked */ + if (updateMinScfCalculated) minScfCalculated[sfbAct] = scfAct; + } else { + /* from this scf value on not all new values have been checked */ + updateMinScfCalculated = 0; + } + } + } while (scfAct > scfMin); + + deltaPe = deltaPeNew; + + /* save parameters to avoid multiple computations of the same sfb */ + prevScfLast[sfbAct] = *scfLast; + prevScfNext[sfbAct] = *scfNext; + deltaPeLast[sfbAct] = deltaPe; + } + + if (success && restartOnSuccess) { + /* start again at first sfb */ + sfbLast = -1; + sfbAct = -1; + sfbNext = -1; + scfLast = 0; + scfNext = 0; + scfMin = FDK_INT_MAX; + scfMax = FDK_INT_MAX; + success = 0; + } else { + /* shift sfbs for next band */ + sfbLast = sfbAct; + sfbAct = sfbNext; + } + } while (sfbNext < psyOutChan->sfbCnt); +} + +/* + Function: FDKaacEnc_assimilateMultipleScf + +*/ +static void FDKaacEnc_assimilateMultipleScf( + PSY_OUT_CHANNEL *psyOutChan, QC_OUT_CHANNEL *qcOutChannel, SHORT *quantSpec, + SHORT *quantSpecTmp, INT dZoneQuantEnable, INT *scf, const INT *minScf, + FIXP_DBL *sfbDist, FIXP_DBL *sfbConstPePart, FIXP_DBL *sfbFormFactorLdData, + FIXP_DBL *sfbNRelevantLines) { + INT sfb, startSfb, stopSfb; + INT scfTmp[MAX_GROUPED_SFB], scfMin, scfMax, scfAct; + INT possibleRegionFound; + INT sfbWidth, sfbOffs, i, k; + FIXP_DBL sfbDistNew[MAX_GROUPED_SFB], distOldSum, distNewSum; + INT deltaScfBits; + FIXP_DBL deltaSpecPe; + FIXP_DBL deltaPe = FL2FXCONST_DBL(0.0f); + FIXP_DBL deltaPeNew; + INT sfbCnt = psyOutChan->sfbCnt; + + /* calc min and max scalfactors */ + scfMin = FDK_INT_MAX; + scfMax = FDK_INT_MIN; + for (sfb = 0; sfb < sfbCnt; sfb++) { + if (scf[sfb] != FDK_INT_MIN) { + scfMin = fixMin(scfMin, scf[sfb]); + scfMax = fixMax(scfMax, scf[sfb]); + } + } + + if (scfMax != FDK_INT_MIN && scfMax <= scfMin + MAX_SCF_DELTA) { + scfAct = scfMax; + + do { + /* try smaller scf */ + scfAct--; + for (i = 0; i < MAX_GROUPED_SFB; i++) scfTmp[i] = scf[i]; + stopSfb = 0; + do { + /* search for region where all scfs are bigger than scfAct */ + sfb = stopSfb; + while (sfb < sfbCnt && (scf[sfb] == FDK_INT_MIN || scf[sfb] <= scfAct)) + sfb++; + startSfb = sfb; + sfb++; + while (sfb < sfbCnt && (scf[sfb] == FDK_INT_MIN || scf[sfb] > scfAct)) + sfb++; + stopSfb = sfb; + + /* check if in all sfb of a valid region scfAct >= minScf[sfb] */ + possibleRegionFound = 0; + if (startSfb < sfbCnt) { + possibleRegionFound = 1; + for (sfb = startSfb; sfb < stopSfb; sfb++) { + if (scf[sfb] != FDK_INT_MIN) + if (scfAct < minScf[sfb]) { + possibleRegionFound = 0; + break; + } + } + } + + if (possibleRegionFound) { /* region found */ + + /* replace scfs in region by scfAct */ + for (sfb = startSfb; sfb < stopSfb; sfb++) { + if (scfTmp[sfb] != FDK_INT_MIN) scfTmp[sfb] = scfAct; + } + + /* estimate change in bit demand for new scfs */ + deltaScfBits = FDKaacEnc_countScfBitsDiff(scf, scfTmp, sfbCnt, + startSfb, stopSfb); + + deltaSpecPe = FDKaacEnc_calcSpecPeDiff( + psyOutChan, qcOutChannel, scf, scfTmp, sfbConstPePart, + sfbFormFactorLdData, sfbNRelevantLines, startSfb, stopSfb); + + deltaPeNew = deltaPe + (FIXP_DBL)deltaScfBits + deltaSpecPe; + + /* new bit demand small enough ? */ + /* 0.0006103515625f = 10.0f/(2^(2*AS_PE_FAC_SHIFT)) */ + if (deltaPeNew < FL2FXCONST_DBL(0.0006103515625f)) { + /* quantize and calc sum of new distortion */ + distOldSum = distNewSum = FL2FXCONST_DBL(0.0f); + for (sfb = startSfb; sfb < stopSfb; sfb++) { + if (scfTmp[sfb] != FDK_INT_MIN) { + distOldSum += CalcInvLdData(sfbDist[sfb]) >> DIST_FAC_SHIFT; + + sfbWidth = psyOutChan->sfbOffsets[sfb + 1] - + psyOutChan->sfbOffsets[sfb]; + sfbOffs = psyOutChan->sfbOffsets[sfb]; + + sfbDistNew[sfb] = FDKaacEnc_calcSfbDist( + qcOutChannel->mdctSpectrum + sfbOffs, + quantSpecTmp + sfbOffs, sfbWidth, scfAct, dZoneQuantEnable); + + if (sfbDistNew[sfb] > qcOutChannel->sfbThresholdLdData[sfb]) { + /* no improvement, skip further dist. calculations */ + distNewSum = distOldSum << 1; + break; + } + distNewSum += CalcInvLdData(sfbDistNew[sfb]) >> DIST_FAC_SHIFT; + } + } + /* distortion smaller ? -> use new scalefactors */ + if (distNewSum < distOldSum) { + deltaPe = deltaPeNew; + for (sfb = startSfb; sfb < stopSfb; sfb++) { + if (scf[sfb] != FDK_INT_MIN) { + sfbWidth = psyOutChan->sfbOffsets[sfb + 1] - + psyOutChan->sfbOffsets[sfb]; + sfbOffs = psyOutChan->sfbOffsets[sfb]; + scf[sfb] = scfAct; + sfbDist[sfb] = sfbDistNew[sfb]; + + for (k = 0; k < sfbWidth; k++) + quantSpec[sfbOffs + k] = quantSpecTmp[sfbOffs + k]; + } + } + } + } + } + + } while (stopSfb <= sfbCnt); + + } while (scfAct > scfMin); + } +} + +/* + Function: FDKaacEnc_FDKaacEnc_assimilateMultipleScf2 + +*/ +static void FDKaacEnc_FDKaacEnc_assimilateMultipleScf2( + PSY_OUT_CHANNEL *psyOutChan, QC_OUT_CHANNEL *qcOutChannel, SHORT *quantSpec, + SHORT *quantSpecTmp, INT dZoneQuantEnable, INT *scf, const INT *minScf, + FIXP_DBL *sfbDist, FIXP_DBL *sfbConstPePart, FIXP_DBL *sfbFormFactorLdData, + FIXP_DBL *sfbNRelevantLines) { + INT sfb, startSfb, stopSfb; + INT scfTmp[MAX_GROUPED_SFB], scfAct, scfNew; + INT scfPrev, scfNext, scfPrevNextMin, scfPrevNextMax, scfLo, scfHi; + INT scfMin, scfMax; + INT *sfbOffs = psyOutChan->sfbOffsets; + FIXP_DBL sfbDistNew[MAX_GROUPED_SFB], sfbDistMax[MAX_GROUPED_SFB]; + FIXP_DBL distOldSum, distNewSum; + INT deltaScfBits; + FIXP_DBL deltaSpecPe; + FIXP_DBL deltaPe = FL2FXCONST_DBL(0.0f); + FIXP_DBL deltaPeNew = FL2FXCONST_DBL(0.0f); + INT sfbCnt = psyOutChan->sfbCnt; + INT bSuccess, bCheckScf; + INT i, k; + + /* calc min and max scalfactors */ + scfMin = FDK_INT_MAX; + scfMax = FDK_INT_MIN; + for (sfb = 0; sfb < sfbCnt; sfb++) { + if (scf[sfb] != FDK_INT_MIN) { + scfMin = fixMin(scfMin, scf[sfb]); + scfMax = fixMax(scfMax, scf[sfb]); + } + } + + stopSfb = 0; + scfAct = FDK_INT_MIN; + do { + /* search for region with same scf values scfAct */ + scfPrev = scfAct; + + sfb = stopSfb; + while (sfb < sfbCnt && (scf[sfb] == FDK_INT_MIN)) sfb++; + startSfb = sfb; + scfAct = scf[startSfb]; + sfb++; + while (sfb < sfbCnt && + ((scf[sfb] == FDK_INT_MIN) || (scf[sfb] == scf[startSfb]))) + sfb++; + stopSfb = sfb; + + if (stopSfb < sfbCnt) + scfNext = scf[stopSfb]; + else + scfNext = scfAct; + + if (scfPrev == FDK_INT_MIN) scfPrev = scfAct; + + scfPrevNextMax = fixMax(scfPrev, scfNext); + scfPrevNextMin = fixMin(scfPrev, scfNext); + + /* try to reduce bits by checking scf values in the range + scf[startSfb]...scfHi */ + scfHi = fixMax(scfPrevNextMax, scfAct); + /* try to find a better solution by reducing the scf difference to + the nearest possible lower scf */ + if (scfPrevNextMax >= scfAct) + scfLo = fixMin(scfAct, scfPrevNextMin); + else + scfLo = scfPrevNextMax; + + if (startSfb < sfbCnt && + scfHi - scfLo <= MAX_SCF_DELTA) { /* region found */ + /* 1. try to save bits by coarser quantization */ + if (scfHi > scf[startSfb]) { + /* calculate the allowed distortion */ + for (sfb = startSfb; sfb < stopSfb; sfb++) { + if (scf[sfb] != FDK_INT_MIN) { + /* sfbDistMax[sfb] = + * (float)pow(qcOutChannel->sfbThreshold[sfb]*sfbDist[sfb]*sfbDist[sfb],1.0f/3.0f); + */ + /* sfbDistMax[sfb] = + * fixMax(sfbDistMax[sfb],qcOutChannel->sfbEnergy[sfb]*FL2FXCONST_DBL(1.e-3f)); + */ + /* -0.15571537944 = ld64(1.e-3f)*/ + sfbDistMax[sfb] = fMult(FL2FXCONST_DBL(1.0f / 3.0f), + qcOutChannel->sfbThresholdLdData[sfb]) + + fMult(FL2FXCONST_DBL(1.0f / 3.0f), sfbDist[sfb]) + + fMult(FL2FXCONST_DBL(1.0f / 3.0f), sfbDist[sfb]); + sfbDistMax[sfb] = + fixMax(sfbDistMax[sfb], qcOutChannel->sfbEnergyLdData[sfb] - + FL2FXCONST_DBL(0.15571537944)); + sfbDistMax[sfb] = + fixMin(sfbDistMax[sfb], qcOutChannel->sfbThresholdLdData[sfb]); + } + } + + /* loop over all possible scf values for this region */ + bCheckScf = 1; + for (scfNew = scf[startSfb] + 1; scfNew <= scfHi; scfNew++) { + for (k = 0; k < MAX_GROUPED_SFB; k++) scfTmp[k] = scf[k]; + + /* replace scfs in region by scfNew */ + for (sfb = startSfb; sfb < stopSfb; sfb++) { + if (scfTmp[sfb] != FDK_INT_MIN) scfTmp[sfb] = scfNew; + } + + /* estimate change in bit demand for new scfs */ + deltaScfBits = FDKaacEnc_countScfBitsDiff(scf, scfTmp, sfbCnt, + startSfb, stopSfb); + + deltaSpecPe = FDKaacEnc_calcSpecPeDiff( + psyOutChan, qcOutChannel, scf, scfTmp, sfbConstPePart, + sfbFormFactorLdData, sfbNRelevantLines, startSfb, stopSfb); + + deltaPeNew = deltaPe + (FIXP_DBL)deltaScfBits + deltaSpecPe; + + /* new bit demand small enough ? */ + if (deltaPeNew < FL2FXCONST_DBL(0.0f)) { + bSuccess = 1; + + /* quantize and calc sum of new distortion */ + for (sfb = startSfb; sfb < stopSfb; sfb++) { + if (scfTmp[sfb] != FDK_INT_MIN) { + sfbDistNew[sfb] = FDKaacEnc_calcSfbDist( + qcOutChannel->mdctSpectrum + sfbOffs[sfb], + quantSpecTmp + sfbOffs[sfb], + sfbOffs[sfb + 1] - sfbOffs[sfb], scfNew, dZoneQuantEnable); + + if (sfbDistNew[sfb] > sfbDistMax[sfb]) { + /* no improvement, skip further dist. calculations */ + bSuccess = 0; + if (sfbDistNew[sfb] == qcOutChannel->sfbEnergyLdData[sfb]) { + /* if whole sfb is already quantized to 0, further + checks with even coarser quant. are useless*/ + bCheckScf = 0; + } + break; + } + } + } + if (bCheckScf == 0) /* further calculations useless ? */ + break; + /* distortion small enough ? -> use new scalefactors */ + if (bSuccess) { + deltaPe = deltaPeNew; + for (sfb = startSfb; sfb < stopSfb; sfb++) { + if (scf[sfb] != FDK_INT_MIN) { + scf[sfb] = scfNew; + sfbDist[sfb] = sfbDistNew[sfb]; + + for (k = 0; k < sfbOffs[sfb + 1] - sfbOffs[sfb]; k++) + quantSpec[sfbOffs[sfb] + k] = + quantSpecTmp[sfbOffs[sfb] + k]; + } + } + } + } + } + } + + /* 2. only if coarser quantization was not successful, try to find + a better solution by finer quantization and reducing bits for + scalefactor coding */ + if (scfAct == scf[startSfb] && scfLo < scfAct && + scfMax - scfMin <= MAX_SCF_DELTA) { + int bminScfViolation = 0; + + for (k = 0; k < MAX_GROUPED_SFB; k++) scfTmp[k] = scf[k]; + + scfNew = scfLo; + + /* replace scfs in region by scfNew and + check if in all sfb scfNew >= minScf[sfb] */ + for (sfb = startSfb; sfb < stopSfb; sfb++) { + if (scfTmp[sfb] != FDK_INT_MIN) { + scfTmp[sfb] = scfNew; + if (scfNew < minScf[sfb]) bminScfViolation = 1; + } + } + + if (!bminScfViolation) { + /* estimate change in bit demand for new scfs */ + deltaScfBits = FDKaacEnc_countScfBitsDiff(scf, scfTmp, sfbCnt, + startSfb, stopSfb); + + deltaSpecPe = FDKaacEnc_calcSpecPeDiff( + psyOutChan, qcOutChannel, scf, scfTmp, sfbConstPePart, + sfbFormFactorLdData, sfbNRelevantLines, startSfb, stopSfb); + + deltaPeNew = deltaPe + (FIXP_DBL)deltaScfBits + deltaSpecPe; + } + + /* new bit demand small enough ? */ + if (!bminScfViolation && deltaPeNew < FL2FXCONST_DBL(0.0f)) { + /* quantize and calc sum of new distortion */ + distOldSum = distNewSum = FL2FXCONST_DBL(0.0f); + for (sfb = startSfb; sfb < stopSfb; sfb++) { + if (scfTmp[sfb] != FDK_INT_MIN) { + distOldSum += CalcInvLdData(sfbDist[sfb]) >> DIST_FAC_SHIFT; + + sfbDistNew[sfb] = FDKaacEnc_calcSfbDist( + qcOutChannel->mdctSpectrum + sfbOffs[sfb], + quantSpecTmp + sfbOffs[sfb], sfbOffs[sfb + 1] - sfbOffs[sfb], + scfNew, dZoneQuantEnable); + + if (sfbDistNew[sfb] > qcOutChannel->sfbThresholdLdData[sfb]) { + /* no improvement, skip further dist. calculations */ + distNewSum = distOldSum << 1; + break; + } + distNewSum += CalcInvLdData(sfbDistNew[sfb]) >> DIST_FAC_SHIFT; + } + } + /* distortion smaller ? -> use new scalefactors */ + if (distNewSum < fMult(FL2FXCONST_DBL(0.8f), distOldSum)) { + deltaPe = deltaPeNew; + for (sfb = startSfb; sfb < stopSfb; sfb++) { + if (scf[sfb] != FDK_INT_MIN) { + scf[sfb] = scfNew; + sfbDist[sfb] = sfbDistNew[sfb]; + + for (k = 0; k < sfbOffs[sfb + 1] - sfbOffs[sfb]; k++) + quantSpec[sfbOffs[sfb] + k] = quantSpecTmp[sfbOffs[sfb] + k]; + } + } + } + } + } + + /* 3. try to find a better solution (save bits) by only reducing the + scalefactor without new quantization */ + if (scfMax - scfMin <= + MAX_SCF_DELTA - 3) { /* 3 bec. scf is reduced 3 times, + see for loop below */ + + for (k = 0; k < sfbCnt; k++) scfTmp[k] = scf[k]; + + for (i = 0; i < 3; i++) { + scfNew = scfTmp[startSfb] - 1; + /* replace scfs in region by scfNew */ + for (sfb = startSfb; sfb < stopSfb; sfb++) { + if (scfTmp[sfb] != FDK_INT_MIN) scfTmp[sfb] = scfNew; + } + /* estimate change in bit demand for new scfs */ + deltaScfBits = FDKaacEnc_countScfBitsDiff(scf, scfTmp, sfbCnt, + startSfb, stopSfb); + deltaPeNew = deltaPe + (FIXP_DBL)deltaScfBits; + /* new bit demand small enough ? */ + if (deltaPeNew <= FL2FXCONST_DBL(0.0f)) { + bSuccess = 1; + distOldSum = distNewSum = FL2FXCONST_DBL(0.0f); + for (sfb = startSfb; sfb < stopSfb; sfb++) { + if (scfTmp[sfb] != FDK_INT_MIN) { + FIXP_DBL sfbEnQ; + /* calc the energy and distortion of the quantized spectrum for + a smaller scf */ + FDKaacEnc_calcSfbQuantEnergyAndDist( + qcOutChannel->mdctSpectrum + sfbOffs[sfb], + quantSpec + sfbOffs[sfb], sfbOffs[sfb + 1] - sfbOffs[sfb], + scfNew, &sfbEnQ, &sfbDistNew[sfb]); + + distOldSum += CalcInvLdData(sfbDist[sfb]) >> DIST_FAC_SHIFT; + distNewSum += CalcInvLdData(sfbDistNew[sfb]) >> DIST_FAC_SHIFT; + + /* 0.00259488556167 = ld64(1.122f) */ + /* -0.00778722686652 = ld64(0.7079f) */ + if ((sfbDistNew[sfb] > + (sfbDist[sfb] + FL2FXCONST_DBL(0.00259488556167f))) || + (sfbEnQ < (qcOutChannel->sfbEnergyLdData[sfb] - + FL2FXCONST_DBL(0.00778722686652f)))) { + bSuccess = 0; + break; + } + } + } + /* distortion smaller ? -> use new scalefactors */ + if (distNewSum < distOldSum && bSuccess) { + deltaPe = deltaPeNew; + for (sfb = startSfb; sfb < stopSfb; sfb++) { + if (scf[sfb] != FDK_INT_MIN) { + scf[sfb] = scfNew; + sfbDist[sfb] = sfbDistNew[sfb]; + } + } + } + } + } + } + } + } while (stopSfb <= sfbCnt); +} + +static void FDKaacEnc_EstimateScaleFactorsChannel( + QC_OUT_CHANNEL *qcOutChannel, PSY_OUT_CHANNEL *psyOutChannel, + INT *RESTRICT scf, INT *RESTRICT globalGain, + FIXP_DBL *RESTRICT sfbFormFactorLdData, const INT invQuant, + SHORT *RESTRICT quantSpec, const INT dZoneQuantEnable) { + INT i, j, sfb, sfbOffs; + INT scfInt; + INT maxSf; + INT minSf; + FIXP_DBL threshLdData; + FIXP_DBL energyLdData; + FIXP_DBL energyPartLdData; + FIXP_DBL thresholdPartLdData; + FIXP_DBL scfFract; + FIXP_DBL maxSpec; + INT minScfCalculated[MAX_GROUPED_SFB]; + FIXP_DBL sfbDistLdData[MAX_GROUPED_SFB]; + C_ALLOC_SCRATCH_START(quantSpecTmp, SHORT, (1024)) + INT minSfMaxQuant[MAX_GROUPED_SFB]; + + FIXP_DBL threshConstLdData = + FL2FXCONST_DBL(0.04304511722f); /* log10(6.75)/log10(2.0)/64.0 */ + FIXP_DBL convConst = FL2FXCONST_DBL(0.30102999566f); /* log10(2.0) */ + FIXP_DBL c1Const = + FL2FXCONST_DBL(-0.27083183594f); /* C1 = -69.33295 => C1/2^8 */ + + if (invQuant > 0) { + FDKmemclear(quantSpec, (1024) * sizeof(SHORT)); + } + + /* scfs without energy or with thresh>energy are marked with FDK_INT_MIN */ + for (i = 0; i < psyOutChannel->sfbCnt; i++) { + scf[i] = FDK_INT_MIN; + } + + for (i = 0; i < MAX_GROUPED_SFB; i++) { + minSfMaxQuant[i] = FDK_INT_MIN; + } + + for (sfbOffs = 0; sfbOffs < psyOutChannel->sfbCnt; + sfbOffs += psyOutChannel->sfbPerGroup) { + for (sfb = 0; sfb < psyOutChannel->maxSfbPerGroup; sfb++) { + threshLdData = qcOutChannel->sfbThresholdLdData[sfbOffs + sfb]; + energyLdData = qcOutChannel->sfbEnergyLdData[sfbOffs + sfb]; + + sfbDistLdData[sfbOffs + sfb] = energyLdData; + + if (energyLdData > threshLdData) { + FIXP_DBL tmp; + + /* energyPart = (float)log10(sfbFormFactor[sfbOffs+sfb]); */ + /* 0.09375f = log(64.0)/log(2.0)/64.0 = scale of sfbFormFactorLdData */ + energyPartLdData = + sfbFormFactorLdData[sfbOffs + sfb] + FL2FXCONST_DBL(0.09375f); + + /* influence of allowed distortion */ + /* thresholdPart = (float)log10(6.75*thresh+FLT_MIN); */ + thresholdPartLdData = threshConstLdData + threshLdData; + + /* scf calc */ + /* scfFloat = 8.8585f * (thresholdPart - energyPart); */ + scfFract = thresholdPartLdData - energyPartLdData; + /* conversion from log2 to log10 */ + scfFract = fMult(convConst, scfFract); + /* (8.8585f * scfFract)/8 = 8/8 * scfFract + 0.8585 * scfFract/8 */ + scfFract = scfFract + fMult(FL2FXCONST_DBL(0.8585f), scfFract >> 3); + + /* integer scalefactor */ + /* scfInt = (int)floor(scfFloat); */ + scfInt = + (INT)(scfFract >> + ((DFRACT_BITS - 1) - 3 - + LD_DATA_SHIFT)); /* 3 bits => scfFract/8.0; 6 bits => ld64 */ + + /* maximum of spectrum */ + maxSpec = FL2FXCONST_DBL(0.0f); + + /* Unroll by 4, allow dual memory access */ + DWORD_ALIGNED(qcOutChannel->mdctSpectrum); + for (j = psyOutChannel->sfbOffsets[sfbOffs + sfb]; + j < psyOutChannel->sfbOffsets[sfbOffs + sfb + 1]; j += 4) { + maxSpec = fMax(maxSpec, + fMax(fMax(fAbs(qcOutChannel->mdctSpectrum[j + 0]), + fAbs(qcOutChannel->mdctSpectrum[j + 1])), + fMax(fAbs(qcOutChannel->mdctSpectrum[j + 2]), + fAbs(qcOutChannel->mdctSpectrum[j + 3])))); + } + /* lower scf limit to avoid quantized values bigger than MAX_QUANT */ + /* C1 = -69.33295f, C2 = 5.77078f = 4/log(2) */ + /* minSfMaxQuant[sfbOffs+sfb] = (int)ceil(C1 + C2*log(maxSpec)); */ + /* C1/2^8 + 4/log(2.0)*log(maxSpec)/2^8 => C1/2^8 + + * log(maxSpec)/log(2.0)*4/2^8 => C1/2^8 + log(maxSpec)/log(2.0)/64.0 */ + + // minSfMaxQuant[sfbOffs+sfb] = ((INT) ((c1Const + CalcLdData(maxSpec)) + // >> ((DFRACT_BITS-1)-8))) + 1; + tmp = CalcLdData(maxSpec); + if (c1Const > FL2FXCONST_DBL(-1.f) - tmp) { + minSfMaxQuant[sfbOffs + sfb] = + ((INT)((c1Const + tmp) >> ((DFRACT_BITS - 1) - 8))) + 1; + } else { + minSfMaxQuant[sfbOffs + sfb] = + ((INT)(FL2FXCONST_DBL(-1.f) >> ((DFRACT_BITS - 1) - 8))) + 1; + } + + scfInt = fixMax(scfInt, minSfMaxQuant[sfbOffs + sfb]); + + /* find better scalefactor with analysis by synthesis */ + if (invQuant > 0) { + scfInt = FDKaacEnc_improveScf( + qcOutChannel->mdctSpectrum + + psyOutChannel->sfbOffsets[sfbOffs + sfb], + quantSpec + psyOutChannel->sfbOffsets[sfbOffs + sfb], + quantSpecTmp + psyOutChannel->sfbOffsets[sfbOffs + sfb], + psyOutChannel->sfbOffsets[sfbOffs + sfb + 1] - + psyOutChannel->sfbOffsets[sfbOffs + sfb], + threshLdData, scfInt, minSfMaxQuant[sfbOffs + sfb], + &sfbDistLdData[sfbOffs + sfb], &minScfCalculated[sfbOffs + sfb], + dZoneQuantEnable); + } + scf[sfbOffs + sfb] = scfInt; + } + } + } + + if (invQuant > 0) { + /* try to decrease scf differences */ + FIXP_DBL sfbConstPePart[MAX_GROUPED_SFB]; + FIXP_DBL sfbNRelevantLines[MAX_GROUPED_SFB]; + + for (i = 0; i < psyOutChannel->sfbCnt; i++) + sfbConstPePart[i] = (FIXP_DBL)FDK_INT_MIN; + + FDKaacEnc_calcSfbRelevantLines( + sfbFormFactorLdData, qcOutChannel->sfbEnergyLdData, + qcOutChannel->sfbThresholdLdData, psyOutChannel->sfbOffsets, + psyOutChannel->sfbCnt, psyOutChannel->sfbPerGroup, + psyOutChannel->maxSfbPerGroup, sfbNRelevantLines); + + FDKaacEnc_assimilateSingleScf( + psyOutChannel, qcOutChannel, quantSpec, quantSpecTmp, dZoneQuantEnable, + scf, minSfMaxQuant, sfbDistLdData, sfbConstPePart, sfbFormFactorLdData, + sfbNRelevantLines, minScfCalculated, 1); + + if (invQuant > 1) { + FDKaacEnc_assimilateMultipleScf( + psyOutChannel, qcOutChannel, quantSpec, quantSpecTmp, + dZoneQuantEnable, scf, minSfMaxQuant, sfbDistLdData, sfbConstPePart, + sfbFormFactorLdData, sfbNRelevantLines); + + FDKaacEnc_FDKaacEnc_assimilateMultipleScf2( + psyOutChannel, qcOutChannel, quantSpec, quantSpecTmp, + dZoneQuantEnable, scf, minSfMaxQuant, sfbDistLdData, sfbConstPePart, + sfbFormFactorLdData, sfbNRelevantLines); + } + } + + /* get min scalefac */ + minSf = FDK_INT_MAX; + for (sfbOffs = 0; sfbOffs < psyOutChannel->sfbCnt; + sfbOffs += psyOutChannel->sfbPerGroup) { + for (sfb = 0; sfb < psyOutChannel->maxSfbPerGroup; sfb++) { + if (scf[sfbOffs + sfb] != FDK_INT_MIN) + minSf = fixMin(minSf, scf[sfbOffs + sfb]); + } + } + + /* limit scf delta */ + for (sfbOffs = 0; sfbOffs < psyOutChannel->sfbCnt; + sfbOffs += psyOutChannel->sfbPerGroup) { + for (sfb = 0; sfb < psyOutChannel->maxSfbPerGroup; sfb++) { + if ((scf[sfbOffs + sfb] != FDK_INT_MIN) && + (minSf + MAX_SCF_DELTA) < scf[sfbOffs + sfb]) { + scf[sfbOffs + sfb] = minSf + MAX_SCF_DELTA; + if (invQuant > 0) { /* changed bands need to be quantized again */ + sfbDistLdData[sfbOffs + sfb] = FDKaacEnc_calcSfbDist( + qcOutChannel->mdctSpectrum + + psyOutChannel->sfbOffsets[sfbOffs + sfb], + quantSpec + psyOutChannel->sfbOffsets[sfbOffs + sfb], + psyOutChannel->sfbOffsets[sfbOffs + sfb + 1] - + psyOutChannel->sfbOffsets[sfbOffs + sfb], + scf[sfbOffs + sfb], dZoneQuantEnable); + } + } + } + } + + /* get max scalefac for global gain */ + maxSf = FDK_INT_MIN; + for (sfbOffs = 0; sfbOffs < psyOutChannel->sfbCnt; + sfbOffs += psyOutChannel->sfbPerGroup) { + for (sfb = 0; sfb < psyOutChannel->maxSfbPerGroup; sfb++) { + maxSf = fixMax(maxSf, scf[sfbOffs + sfb]); + } + } + + /* calc loop scalefactors, if spec is not all zero (i.e. maxSf == -99) */ + if (maxSf > FDK_INT_MIN) { + *globalGain = maxSf; + for (sfbOffs = 0; sfbOffs < psyOutChannel->sfbCnt; + sfbOffs += psyOutChannel->sfbPerGroup) { + for (sfb = 0; sfb < psyOutChannel->maxSfbPerGroup; sfb++) { + if (scf[sfbOffs + sfb] == FDK_INT_MIN) { + scf[sfbOffs + sfb] = 0; + /* set band explicitely to zero */ + for (j = psyOutChannel->sfbOffsets[sfbOffs + sfb]; + j < psyOutChannel->sfbOffsets[sfbOffs + sfb + 1]; j++) { + qcOutChannel->mdctSpectrum[j] = FL2FXCONST_DBL(0.0f); + } + } else { + scf[sfbOffs + sfb] = maxSf - scf[sfbOffs + sfb]; + } + } + } + } else { + *globalGain = 0; + /* set spectrum explicitely to zero */ + for (sfbOffs = 0; sfbOffs < psyOutChannel->sfbCnt; + sfbOffs += psyOutChannel->sfbPerGroup) { + for (sfb = 0; sfb < psyOutChannel->maxSfbPerGroup; sfb++) { + scf[sfbOffs + sfb] = 0; + /* set band explicitely to zero */ + for (j = psyOutChannel->sfbOffsets[sfbOffs + sfb]; + j < psyOutChannel->sfbOffsets[sfbOffs + sfb + 1]; j++) { + qcOutChannel->mdctSpectrum[j] = FL2FXCONST_DBL(0.0f); + } + } + } + } + + /* free quantSpecTmp from scratch */ + C_ALLOC_SCRATCH_END(quantSpecTmp, SHORT, (1024)) +} + +void FDKaacEnc_EstimateScaleFactors(PSY_OUT_CHANNEL *psyOutChannel[], + QC_OUT_CHANNEL *qcOutChannel[], + const INT invQuant, + const INT dZoneQuantEnable, + const INT nChannels) { + int ch; + + for (ch = 0; ch < nChannels; ch++) { + FDKaacEnc_EstimateScaleFactorsChannel( + qcOutChannel[ch], psyOutChannel[ch], qcOutChannel[ch]->scf, + &qcOutChannel[ch]->globalGain, qcOutChannel[ch]->sfbFormFactorLdData, + invQuant, qcOutChannel[ch]->quantSpec, dZoneQuantEnable); + } +} diff --git a/fdk-aac/libAACenc/src/sf_estim.h b/fdk-aac/libAACenc/src/sf_estim.h new file mode 100644 index 0000000..ab2d3c2 --- /dev/null +++ b/fdk-aac/libAACenc/src/sf_estim.h @@ -0,0 +1,124 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M. Werner + + Description: Scale factor estimation + +*******************************************************************************/ + +#ifndef SF_ESTIM_H +#define SF_ESTIM_H + +#include "common_fix.h" + +#include "psy_const.h" +#include "qc_data.h" +#include "interface.h" + +#define FORM_FAC_SHIFT 6 + +void FDKaacEnc_CalcFormFactor(QC_OUT_CHANNEL *qcOutChannel[(2)], + PSY_OUT_CHANNEL *psyOutChannel[(2)], + const INT nChannels); + +void FDKaacEnc_EstimateScaleFactors(PSY_OUT_CHANNEL *psyOutChannel[], + QC_OUT_CHANNEL *qcOutChannel[], + const INT invQuant, + const INT dZoneQuantEnable, + const INT nChannels); + +#endif diff --git a/fdk-aac/libAACenc/src/spreading.cpp b/fdk-aac/libAACenc/src/spreading.cpp new file mode 100644 index 0000000..0fb43bb --- /dev/null +++ b/fdk-aac/libAACenc/src/spreading.cpp @@ -0,0 +1,125 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M.Werner + + Description: Spreading of energy + +*******************************************************************************/ + +#include "spreading.h" + +void FDKaacEnc_SpreadingMax(const INT pbCnt, + const FIXP_DBL *RESTRICT maskLowFactor, + const FIXP_DBL *RESTRICT maskHighFactor, + FIXP_DBL *RESTRICT pbSpreadEnergy) { + int i; + FIXP_DBL delay; + + /* slope to higher frequencies */ + delay = pbSpreadEnergy[0]; + for (i = 1; i < pbCnt; i++) { + delay = fixMax(pbSpreadEnergy[i], fMult(maskHighFactor[i], delay)); + pbSpreadEnergy[i] = delay; + } + + /* slope to lower frequencies */ + delay = pbSpreadEnergy[pbCnt - 1]; + for (i = pbCnt - 2; i >= 0; i--) { + delay = fixMax(pbSpreadEnergy[i], fMult(maskLowFactor[i], delay)); + pbSpreadEnergy[i] = delay; + } +} diff --git a/fdk-aac/libAACenc/src/spreading.h b/fdk-aac/libAACenc/src/spreading.h new file mode 100644 index 0000000..e693031 --- /dev/null +++ b/fdk-aac/libAACenc/src/spreading.h @@ -0,0 +1,113 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M.Werner + + Description: Spreading of energy and weighted tonality + +*******************************************************************************/ + +#ifndef SPREADING_H +#define SPREADING_H + +#include "common_fix.h" + +void FDKaacEnc_SpreadingMax(const INT pbCnt, + const FIXP_DBL *RESTRICT maskLowFactor, + const FIXP_DBL *RESTRICT maskHighFactor, + FIXP_DBL *RESTRICT pbSpreadEnergy); + +#endif /* #ifndef SPREADING_H */ diff --git a/fdk-aac/libAACenc/src/tns_func.h b/fdk-aac/libAACenc/src/tns_func.h new file mode 100644 index 0000000..6099bc7 --- /dev/null +++ b/fdk-aac/libAACenc/src/tns_func.h @@ -0,0 +1,129 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): Alex Goeschel + + Description: Temporal noise shaping + +*******************************************************************************/ + +#ifndef TNS_FUNC_H +#define TNS_FUNC_H + +#include "common_fix.h" + +#include "psy_configuration.h" + +AAC_ENCODER_ERROR FDKaacEnc_InitTnsConfiguration( + INT bitrate, INT samplerate, INT channels, INT blocktype, INT granuleLength, + INT isLowDelay, INT ldSbrPresent, TNS_CONFIG *tnsConfig, + PSY_CONFIGURATION *psyConfig, INT active, INT useTnsPeak); + +INT FDKaacEnc_TnsDetect(TNS_DATA *tnsData, const TNS_CONFIG *tC, + TNS_INFO *tnsInfo, INT sfbCnt, const FIXP_DBL *spectrum, + INT subBlockNumber, INT blockType); + +void FDKaacEnc_TnsSync(TNS_DATA *tnsDataDest, const TNS_DATA *tnsDataSrc, + TNS_INFO *tnsInfoDest, TNS_INFO *tnsInfoSrc, + const INT blockTypeDest, const INT blockTypeSrc, + const TNS_CONFIG *tC); + +INT FDKaacEnc_TnsEncode(TNS_INFO *tnsInfo, TNS_DATA *tnsData, + const INT numOfSfb, const TNS_CONFIG *tC, + const INT lowPassLine, FIXP_DBL *spectrum, + const INT subBlockNumber, const INT blockType); + +#endif /* TNS_FUNC_H */ diff --git a/fdk-aac/libAACenc/src/tonality.cpp b/fdk-aac/libAACenc/src/tonality.cpp new file mode 100644 index 0000000..334e0f1 --- /dev/null +++ b/fdk-aac/libAACenc/src/tonality.cpp @@ -0,0 +1,219 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M. Werner + + Description: Convert chaos measure to the tonality index + +*******************************************************************************/ + +#include "tonality.h" + +#include "chaosmeasure.h" + +#if defined(__arm__) +#endif + +static const FIXP_DBL normlog = + (FIXP_DBL)0xd977d949; /*FL2FXCONST_DBL(-0.4342944819f * + FDKlog(2.0)/FDKlog(2.7182818)); */ + +static void FDKaacEnc_CalcSfbTonality(FIXP_DBL *RESTRICT spectrum, + INT *RESTRICT sfbMaxScaleSpec, + FIXP_DBL *RESTRICT chaosMeasure, + FIXP_SGL *RESTRICT sfbTonality, + INT sfbCnt, const INT *RESTRICT sfbOffset, + FIXP_DBL *RESTRICT sfbEnergyLD64); + +void FDKaacEnc_CalculateFullTonality(FIXP_DBL *RESTRICT spectrum, + INT *RESTRICT sfbMaxScaleSpec, + FIXP_DBL *RESTRICT sfbEnergyLD64, + FIXP_SGL *RESTRICT sfbTonality, INT sfbCnt, + const INT *sfbOffset, INT usePns) { + INT j; + INT numberOfLines = sfbOffset[sfbCnt]; + + if (usePns) { + C_ALLOC_SCRATCH_START(chaosMeasurePerLine, FIXP_DBL, (1024)) + + /* calculate chaos measure */ + FDKaacEnc_CalculateChaosMeasure(spectrum, numberOfLines, + chaosMeasurePerLine); + + /* smooth ChaosMeasure */ + FIXP_DBL left = chaosMeasurePerLine[0]; + FIXP_DBL right; + for (j = 1; j < (numberOfLines - 1); j += 2) { + right = chaosMeasurePerLine[j]; + right = right - (right >> 2); + left = right + (left >> 2); + chaosMeasurePerLine[j] = left; /* 0.25 left + 0.75 right */ + + right = chaosMeasurePerLine[j + 1]; + right = right - (right >> 2); + left = right + (left >> 2); + chaosMeasurePerLine[j + 1] = left; + } + if (j == (numberOfLines - 1)) { + right = chaosMeasurePerLine[j]; + right = right - (right >> 2); + left = right + (left >> 2); + chaosMeasurePerLine[j] = left; + } + + FDKaacEnc_CalcSfbTonality(spectrum, sfbMaxScaleSpec, chaosMeasurePerLine, + sfbTonality, sfbCnt, sfbOffset, sfbEnergyLD64); + + C_ALLOC_SCRATCH_END(chaosMeasurePerLine, FIXP_DBL, (1024)) + } +} + +/***************************************************************************** + + functionname: CalculateTonalityIndex + description: computes tonality values out of unpredictability values + limits range and computes log() + returns: + input: ptr to energies, ptr to chaos measure values, + number of sfb + output: sfb wise tonality values + +*****************************************************************************/ +static void FDKaacEnc_CalcSfbTonality(FIXP_DBL *RESTRICT spectrum, + INT *RESTRICT sfbMaxScaleSpec, + FIXP_DBL *RESTRICT chaosMeasure, + FIXP_SGL *RESTRICT sfbTonality, + INT sfbCnt, const INT *RESTRICT sfbOffset, + FIXP_DBL *RESTRICT sfbEnergyLD64) { + INT i; + + for (i = 0; i < sfbCnt; i++) { + FIXP_DBL chaosMeasureSfbLD64; + INT shiftBits = + fixMax(0, sfbMaxScaleSpec[i] - + 4); /* max sfbWidth = 96 ; 2^7=128 => 7/2 = 4 (spc*spc) */ + + INT j; + FIXP_DBL chaosMeasureSfb = FL2FXCONST_DBL(0.0); + + /* calc chaosMeasurePerSfb */ + for (j = (sfbOffset[i + 1] - sfbOffset[i]) - 1; j >= 0; j--) { + FIXP_DBL tmp = (*spectrum++) << shiftBits; + FIXP_DBL lineNrg = fMultDiv2(tmp, tmp); + chaosMeasureSfb = fMultAddDiv2(chaosMeasureSfb, lineNrg, *chaosMeasure++); + } + + /* calc tonalityPerSfb */ + if (chaosMeasureSfb != FL2FXCONST_DBL(0.0)) { + /* add ld(convtone)/64 and 2/64 bec.fMultDiv2 */ + chaosMeasureSfbLD64 = CalcLdData((chaosMeasureSfb)) - sfbEnergyLD64[i]; + chaosMeasureSfbLD64 += FL2FXCONST_DBL(3.0f / 64) - + ((FIXP_DBL)(shiftBits) << (DFRACT_BITS - 6)); + + if (chaosMeasureSfbLD64 > + FL2FXCONST_DBL(-0.0519051)) /* > ld(0.05)+ld(2) */ + { + if (chaosMeasureSfbLD64 <= FL2FXCONST_DBL(0.0)) + sfbTonality[i] = + FX_DBL2FX_SGL(fMultDiv2(chaosMeasureSfbLD64, normlog) << 7); + else + sfbTonality[i] = FL2FXCONST_SGL(0.0); + } else + sfbTonality[i] = (FIXP_SGL)MAXVAL_SGL; + } else + sfbTonality[i] = (FIXP_SGL)MAXVAL_SGL; + } +} diff --git a/fdk-aac/libAACenc/src/tonality.h b/fdk-aac/libAACenc/src/tonality.h new file mode 100644 index 0000000..c5cf4c5 --- /dev/null +++ b/fdk-aac/libAACenc/src/tonality.h @@ -0,0 +1,115 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M. Lohwasser + + Description: Calculate tonality index + +*******************************************************************************/ + +#ifndef TONALITY_H +#define TONALITY_H + +#include "common_fix.h" +#include "chaosmeasure.h" + +void FDKaacEnc_CalculateFullTonality(FIXP_DBL *RESTRICT spectrum, + INT *RESTRICT sfbMaxScaleSpec, + FIXP_DBL *RESTRICT sfbEnergyLD64, + FIXP_SGL *RESTRICT sfbTonality, INT sfbCnt, + const INT *sfbOffset, INT usePns); + +#endif /* TONALITY_H */ diff --git a/fdk-aac/libAACenc/src/transform.cpp b/fdk-aac/libAACenc/src/transform.cpp new file mode 100644 index 0000000..08b1c2f --- /dev/null +++ b/fdk-aac/libAACenc/src/transform.cpp @@ -0,0 +1,294 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): Tobias Chalupka + + Description: FDKaacLdEnc_MdctTransform480: + The module FDKaacLdEnc_MdctTransform will perform the MDCT. + The MDCT supports the sine window and + the zero padded window. The algorithm of the MDCT + can be divided in Windowing, PreModulation, Fft and + PostModulation. + +*******************************************************************************/ + +#include "transform.h" +#include "dct.h" +#include "psy_const.h" +#include "aacEnc_rom.h" +#include "FDK_tools_rom.h" + +#if defined(__arm__) +#endif + +INT FDKaacEnc_Transform_Real(const INT_PCM *pTimeData, + FIXP_DBL *RESTRICT mdctData, const INT blockType, + const INT windowShape, INT *prevWindowShape, + H_MDCT mdctPers, const INT frameLength, + INT *pMdctData_e, INT filterType) { + const INT_PCM *RESTRICT timeData; + + UINT numSpec; + UINT numMdctLines; + UINT offset; + int fr; /* fr: right window slope length */ + SHORT mdctData_e[8]; + + timeData = pTimeData; + + if (blockType == SHORT_WINDOW) { + numSpec = 8; + numMdctLines = frameLength >> 3; + } else { + numSpec = 1; + numMdctLines = frameLength; + } + + offset = (windowShape == LOL_WINDOW) ? ((frameLength * 3) >> 2) : 0; + switch (blockType) { + case LONG_WINDOW: + case STOP_WINDOW: + fr = frameLength - offset; + break; + case START_WINDOW: /* or StopStartSequence */ + case SHORT_WINDOW: + fr = frameLength >> 3; + break; + default: + FDK_ASSERT(0); + return -1; + } + + mdct_block(mdctPers, timeData, frameLength, mdctData, numSpec, numMdctLines, + FDKgetWindowSlope(fr, windowShape), fr, mdctData_e); + + if (blockType == SHORT_WINDOW) { + if (!(mdctData_e[0] == mdctData_e[1] && mdctData_e[1] == mdctData_e[2] && + mdctData_e[2] == mdctData_e[3] && mdctData_e[3] == mdctData_e[4] && + mdctData_e[4] == mdctData_e[5] && mdctData_e[5] == mdctData_e[6] && + mdctData_e[6] == mdctData_e[7])) { + return -1; + } + } + *prevWindowShape = windowShape; + *pMdctData_e = mdctData_e[0]; + + return 0; +} + +INT FDKaacEnc_Transform_Real_Eld(const INT_PCM *pTimeData, + FIXP_DBL *RESTRICT mdctData, + const INT blockType, const INT windowShape, + INT *prevWindowShape, const INT frameLength, + INT *mdctData_e, INT filterType, + FIXP_DBL *RESTRICT overlapAddBuffer) { + const INT_PCM *RESTRICT timeData; + + INT i; + + /* tl: transform length + fl: left window slope length + nl: left window slope offset + fr: right window slope length + nr: right window slope offset */ + const FIXP_WTB *pWindowELD = NULL; + int N = frameLength; + int L = frameLength; + + timeData = pTimeData; + + if (blockType != LONG_WINDOW) { + return -1; + } + + /* + * MDCT scale: + * + 1: fMultDiv2() in windowing. + * + 1: Because of factor 1/2 in Princen-Bradley compliant windowed TDAC. + */ + *mdctData_e = 1 + 1; + + switch (frameLength) { + case 512: + pWindowELD = ELDAnalysis512; + break; + case 480: + pWindowELD = ELDAnalysis480; + break; + case 256: + pWindowELD = ELDAnalysis256; + *mdctData_e += 1; + break; + case 240: + pWindowELD = ELDAnalysis240; + *mdctData_e += 1; + break; + case 128: + pWindowELD = ELDAnalysis128; + *mdctData_e += 2; + break; + case 120: + pWindowELD = ELDAnalysis120; + *mdctData_e += 2; + break; + default: + FDK_ASSERT(0); + return -1; + } + + for (i = 0; i < N / 4; i++) { + FIXP_DBL z0, outval; + + z0 = (fMult((FIXP_PCM)timeData[L + N * 3 / 4 - 1 - i], + pWindowELD[N / 2 - 1 - i]) + << (WTS0 - 1)) + + (fMult((FIXP_PCM)timeData[L + N * 3 / 4 + i], pWindowELD[N / 2 + i]) + << (WTS0 - 1)); + + outval = (fMultDiv2((FIXP_PCM)timeData[L + N * 3 / 4 - 1 - i], + pWindowELD[N + N / 2 - 1 - i]) >> + (-WTS1)); + outval += (fMultDiv2((FIXP_PCM)timeData[L + N * 3 / 4 + i], + pWindowELD[N + N / 2 + i]) >> + (-WTS1)); + outval += (fMultDiv2(overlapAddBuffer[N / 2 + i], pWindowELD[2 * N + i]) >> + (-WTS2 - 1)); + + overlapAddBuffer[N / 2 + i] = overlapAddBuffer[i]; + + overlapAddBuffer[i] = z0; + mdctData[i] = overlapAddBuffer[N / 2 + i] + + (fMultDiv2(overlapAddBuffer[N + N / 2 - 1 - i], + pWindowELD[2 * N + N / 2 + i]) >> + (-WTS2 - 1)); + + mdctData[N - 1 - i] = outval; + overlapAddBuffer[N + N / 2 - 1 - i] = outval; + } + + for (i = N / 4; i < N / 2; i++) { + FIXP_DBL z0, outval; + + z0 = fMult((FIXP_PCM)timeData[L + N * 3 / 4 - 1 - i], + pWindowELD[N / 2 - 1 - i]) + << (WTS0 - 1); + + outval = (fMultDiv2((FIXP_PCM)timeData[L + N * 3 / 4 - 1 - i], + pWindowELD[N + N / 2 - 1 - i]) >> + (-WTS1)); + outval += (fMultDiv2(overlapAddBuffer[N / 2 + i], pWindowELD[2 * N + i]) >> + (-WTS2 - 1)); + + overlapAddBuffer[N / 2 + i] = + overlapAddBuffer[i] + + (fMult((FIXP_PCM)timeData[L - N / 4 + i], pWindowELD[N / 2 + i]) + << (WTS0 - 1)); + + overlapAddBuffer[i] = z0; + mdctData[i] = overlapAddBuffer[N / 2 + i] + + (fMultDiv2(overlapAddBuffer[N + N / 2 - 1 - i], + pWindowELD[2 * N + N / 2 + i]) >> + (-WTS2 - 1)); + + mdctData[N - 1 - i] = outval; + overlapAddBuffer[N + N / 2 - 1 - i] = outval; + } + dct_IV(mdctData, frameLength, mdctData_e); + + *prevWindowShape = windowShape; + + return 0; +} diff --git a/fdk-aac/libAACenc/src/transform.h b/fdk-aac/libAACenc/src/transform.h new file mode 100644 index 0000000..8f5ff46 --- /dev/null +++ b/fdk-aac/libAACenc/src/transform.h @@ -0,0 +1,163 @@ +/* ----------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten +Forschung e.V. All rights reserved. + + 1. INTRODUCTION +The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software +that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding +scheme for digital audio. This FDK AAC Codec software is intended to be used on +a wide variety of Android devices. + +AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient +general perceptual audio codecs. AAC-ELD is considered the best-performing +full-bandwidth communications codec by independent studies and is widely +deployed. AAC has been standardized by ISO and IEC as part of the MPEG +specifications. + +Patent licenses for necessary patent claims for the FDK AAC Codec (including +those of Fraunhofer) may be obtained through Via Licensing +(www.vialicensing.com) or through the respective patent owners individually for +the purpose of encoding or decoding bit streams in products that are compliant +with the ISO/IEC MPEG audio standards. Please note that most manufacturers of +Android devices already license these patent claims through Via Licensing or +directly from the patent owners, and therefore FDK AAC Codec software may +already be covered under those patent licenses when it is used for those +licensed purposes only. + +Commercially-licensed AAC software libraries, including floating-point versions +with enhanced sound quality, are also available from Fraunhofer. Users are +encouraged to check the Fraunhofer website for additional applications +information and documentation. + +2. COPYRIGHT LICENSE + +Redistribution and use in source and binary forms, with or without modification, +are permitted without payment of copyright license fees provided that you +satisfy the following conditions: + +You must retain the complete text of this software license in redistributions of +the FDK AAC Codec or your modifications thereto in source code form. + +You must retain the complete text of this software license in the documentation +and/or other materials provided with redistributions of the FDK AAC Codec or +your modifications thereto in binary form. You must make available free of +charge copies of the complete source code of the FDK AAC Codec and your +modifications thereto to recipients of copies in binary form. + +The name of Fraunhofer may not be used to endorse or promote products derived +from this library without prior written permission. + +You may not charge copyright license fees for anyone to use, copy or distribute +the FDK AAC Codec software or your modifications thereto. + +Your modified versions of the FDK AAC Codec must carry prominent notices stating +that you changed the software and the date of any change. For modified versions +of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" +must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK +AAC Codec Library for Android." + +3. NO PATENT LICENSE + +NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without +limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. +Fraunhofer provides no warranty of patent non-infringement with respect to this +software. + +You may use this FDK AAC Codec software or modifications thereto only for +purposes that are authorized by appropriate patent licenses. + +4. DISCLAIMER + +This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright +holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, +including but not limited to the implied warranties of merchantability and +fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR +CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, +or consequential damages, including but not limited to procurement of substitute +goods or services; loss of use, data, or profits, or business interruption, +however caused and on any theory of liability, whether in contract, strict +liability, or tort (including negligence), arising in any way out of the use of +this software, even if advised of the possibility of such damage. + +5. CONTACT INFORMATION + +Fraunhofer Institute for Integrated Circuits IIS +Attention: Audio and Multimedia Departments - FDK AAC LL +Am Wolfsmantel 33 +91058 Erlangen, Germany + +www.iis.fraunhofer.de/amm +amm-info@iis.fraunhofer.de +----------------------------------------------------------------------------- */ + +/**************************** AAC encoder library ****************************** + + Author(s): M. Werner + + Description: MDCT Transform + +*******************************************************************************/ + +#ifndef TRANSFORM_H +#define TRANSFORM_H + +#include "mdct.h" +#include "common_fix.h" + +#define WTS0 1 +#define WTS1 0 +#define WTS2 -2 + +/** + * \brief: Performe MDCT transform of time domain data. + * \param timeData pointer to time domain input signal. + * \param mdctData pointer to store frequency domain output data. + * \param blockType index indicating the type of block. Either + * LONG_WINDOW, START_WINDOW, SHORT_WINDOW or STOP_WINDOW. + * \param windowShape index indicating the window slope type to be used. + * Values allowed are either SINE_WINDOW or KBD_WINDOW. + * \param previndowShape index indicating the window slope type used + * in the last frame. + * Values allowed are either SINE_WINDOW or KBD_WINDOW. + * \param frameLength length of the block. Either 1024 or 960. + * \param mdctData_e pointer to an INT where the exponent of the frequency + * domain output data is stored into. + * \param filterType xxx + * \return 0 in case of success, non-zero in case of error (inconsistent + * parameters). + */ +INT FDKaacEnc_Transform_Real(const INT_PCM* pTimeData, + FIXP_DBL* RESTRICT mdctData, const INT blockType, + const INT windowShape, INT* prevWindowShape, + H_MDCT mdctPers, const INT frameLength, + INT* pMdctData_e, INT filterType); + +/** + * \brief: Performe ELD filterbnank transform of time domain data. + * \param timeData pointer to time domain input signal. + * \param mdctData pointer to store frequency domain output data. + * \param blockType index indicating the type of block. Either + * LONG_WINDOW, START_WINDOW, SHORT_WINDOW or STOP_WINDOW. + * \param windowShape index indicating the window slope type to be used. + * Values allowed are either SINE_WINDOW or KBD_WINDOW. + * \param previndowShape index indicating the window slope type used + * in the last frame. + * Values allowed are either SINE_WINDOW or KBD_WINDOW. + * \param frameLength length of the block. Either 1024 or 960. + * \param mdctData_e pointer to an INT where the exponent of the frequency + * domain output data is stored into. + * \param filterType xxx + * \param overlapAddBuffer overlap add buffer for overlap of ELD filterbank + * \return 0 in case of success, non-zero in case of error (inconsistent + * parameters). + */ +INT FDKaacEnc_Transform_Real_Eld(const INT_PCM* pTimeData, + FIXP_DBL* RESTRICT mdctData, + const INT blockType, const INT windowShape, + INT* prevWindowShape, const INT frameLength, + INT* mdctData_e, INT filterType, + FIXP_DBL* RESTRICT overlapAddBuffer); + +#endif /* #!defined (TRANSFORM_H) */ |