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| author | Matthias P. Braendli <matthias.braendli@mpb.li> | 2019-11-11 11:38:02 +0100 |
|---|---|---|
| committer | Matthias P. Braendli <matthias.braendli@mpb.li> | 2019-11-11 11:38:02 +0100 |
| commit | 0e5af65c467b2423a0b857ae3ad98c91acc1e190 (patch) | |
| tree | d07f69550d8886271e44fe79c4dcfb299cafbd38 /fdk-aac/libAACenc/include/aacenc_lib.h | |
| parent | efe406d9724f959c8bc2a31802559ca6d41fd897 (diff) | |
| download | ODR-AudioEnc-0e5af65c467b2423a0b857ae3ad98c91acc1e190.tar.gz ODR-AudioEnc-0e5af65c467b2423a0b857ae3ad98c91acc1e190.tar.bz2 ODR-AudioEnc-0e5af65c467b2423a0b857ae3ad98c91acc1e190.zip | |
Include patched FDK-AAC in the repository
The initial idea was to get the DAB+ patch into upstream, but since
that follows the android source releases, there is no place for a custom
DAB+ patch there.
So instead of having to maintain a patched fdk-aac that has to have the
same .so version as the distribution package on which it is installed,
we prefer having a separate fdk-aac-dab library to avoid collision.
At that point, there's no reason to keep fdk-aac in a separate
repository, as odr-audioenc is the only tool that needs DAB+ encoding
support. Including it here simplifies installation, and makes it
consistent with toolame-dab, also shipped in this repository.
DAB+ decoding support (needed by ODR-SourceCompanion, dablin, etisnoop,
welle.io and others) can be done using upstream FDK-AAC.
Diffstat (limited to 'fdk-aac/libAACenc/include/aacenc_lib.h')
| -rw-r--r-- | fdk-aac/libAACenc/include/aacenc_lib.h | 1733 |
1 files changed, 1733 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 */ |