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diff --git a/libAACdec/include/aacdecoder_lib.h b/libAACdec/include/aacdecoder_lib.h new file mode 100644 index 0000000..79b4ba1 --- /dev/null +++ b/libAACdec/include/aacdecoder_lib.h @@ -0,0 +1,688 @@ + +/* ----------------------------------------------------------------------------------------------------------- +Software License for The Fraunhofer FDK AAC Codec Library for Android + +© Copyright 1995 - 2012 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 +----------------------------------------------------------------------------------------------------------- */ + +/***************************** MPEG-4 AAC Decoder ************************** + + Author(s): Manuel Jander + +******************************************************************************/ + +/** + * \file aacdecoder_lib.h + * \brief FDK AAC decoder library interface header file. + * + +\page INTRO Introduction + +\section SCOPE Scope + +This document describes the high-level interface and usage of the ISO/MPEG-2/4 AAC Decoder +library developed by the Fraunhofer Institute for Integrated Circuits (IIS). +Depending on the library configuration, it implements decoding of AAC-LC (Low-Complexity), +HE-AAC (High-Efficiency AAC, v1 and v2), AAC-LD (Low-Delay) and AAC-ELD (Enhanced Low-Delay). + +All references to SBR (Spectral Band Replication) are only applicable to HE-AAC and AAC-ELD +versions of the library. All references to PS (Parametric Stereo) are only applicable to +HE-AAC v2 versions of the library. + +\section DecoderBasics Decoder 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, subpart 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 MPEG2 or MPEG4 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 InterfaceDescritpion API Description + +All API header files are located in the folder /include of the release package. They are described in +detail in this document. All header files are provided for usage in C/C++ programs. The AAC decoder library +API functions are located at aacdecoder_lib.h. + +In binary releases the decoder core resides in statically linkable libraries called for example libAACdec.a, +(Linux) or FDK_aacDec_lib (Microsoft Visual C++). + +\section Calling_Sequence Calling Sequence + +For decoding of ISO/MPEG-2/4 AAC or HE-AAC v2 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 in +main.cpp uses file-based input/output, and in such case call mpegFileRead_Open() to open an input file and +to allocate memory for the required structures, and the corresponding mpegFileRead_Close() to close opened +files and to de-allocate associated structures. mpegFileRead_Open() tries to detect the bitstream format and +in case of MPEG-4 file format or Raw Packets file format (a Fraunhofer IIS proprietary format) reads the Audio +Specific Config data (ASC). An unsuccessful attempt to recognize the bitstream format requires the user to +provide this information manually (see \ref CommandLineUsage). For any other bitstream formats that are +usually applicable in streaming applications, the decoder itself will try to synchronize and parse the given +bitstream fragment using the FDK transport library. Hence, for streaming applications (without file access) +this step is not necessary. + +-# Call aacDecoder_Open() to open and retrieve a handle to a new AAC decoder instance. +\dontinclude main.cpp +\skipline aacDecoder_Open +-# If out-of-band config data (Audio Specific Config (ASC) or Stream Mux Config (SMC)) is available, call +aacDecoder_ConfigRaw() to pass it to the decoder and before the decoding process starts. If this data is +not available in advance, the decoder will get it from the bitstream and configure itself while decoding +with aacDecoder_DecodeFrame(). +-# Begin decoding loop. +\skipline do { +-# Read data from bitstream file or stream into a client-supplied input buffer ("inBuffer" in main.cpp). +If it is very small like just 4, aacDecoder_DecodeFrame() will +repeatedly return ::AAC_DEC_NOT_ENOUGH_BITS until enough bits were fed by aacDecoder_Fill(). Only read data +when this buffer has completely been processed and is then empty. For file-based input execute +mpegFileRead_Read() or any other implementation with similar functionality. +-# Call aacDecoder_Fill() to fill the decoder's internal bitstream input buffer with the client-supplied +external bitstream input buffer. +\skipline aacDecoder_Fill +-# Call aacDecoder_DecodeFrame() which writes decoded PCM audio data to a client-supplied buffer. It is the +client's responsibility to allocate a buffer which is large enough to hold this output data. +\skipline aacDecoder_DecodeFrame +If the bitstream's configuration (number of channels, sample rate, frame size) is not known in advance, you may +call aacDecoder_GetStreamInfo() to retrieve a structure containing this information and then initialize an audio +output device. In the example main.cpp, if the number of channels or the sample rate has changed since program +start or since the previously decoded frame, the audio output device will be re-initialized. If WAVE file output +is chosen, a new WAVE file for each new configuration will be created. +\skipline aacDecoder_GetStreamInfo +-# Repeat steps 5 to 7 until no data to decode is available anymore, or if an error occured. +\skipline } while +-# Call aacDecoder_Close() to de-allocate all AAC decoder and transport layer structures. +\skipline aacDecoder_Close + +\section BufferSystem Buffer System + +There are three main buffers in an AAC decoder application. One external input buffer to hold bitstream +data from file I/O or elsewhere, one decoder-internal input buffer, and one to hold the decoded output +PCM sample data, whereas this output buffer may overlap with the external input buffer. + +The external input buffer is set in the example framework main.cpp and its size is defined by ::IN_BUF_SIZE. +You may freely choose different sizes here. To feed the data to the decoder-internal input buffer, use the +function aacDecoder_Fill(). This function returns important information about how many bytes in the +external input buffer have not yet been copied into the internal input buffer (variable bytesValid). +Once the external buffer has been fully copied, it can be re-filled again. +In case you want to re-fill it when there are still unprocessed bytes (bytesValid is unequal 0), you +would have to additionally perform a memcpy(), so that just means unnecessary computational overhead +and therefore we recommend to re-fill the buffer only when bytesValid is 0. + +\image latex dec_buffer.png "Lifecycle of the external input buffer" width=9cm + +The size of the decoder-internal input buffer is set in tpdec_lib.h (see define ::TRANSPORTDEC_INBUF_SIZE). +You may choose a smaller size under the following considerations: + +- each input channel requires 768 bytes +- the whole buffer must be of size 2^n + +So for example a stereo decoder: + +\f[ +TRANSPORTDEC\_INBUF\_SIZE = 2 * 768 = 1536 => 2048 +\f] + +tpdec_lib.h and TRANSPORTDEC_INBUF_SIZE are not part of the decoder's library interface. Therefore +only source-code clients may change this setting. If you received a library release, please ask us and +we can change this in order to meet your memory requirements. + +\page OutputFormat Decoder audio output + +\section OutputFormatObtaining Obtaining channel mapping information + +The decoded audio output format is indicated by a set of variables of the CStreamInfo structure. +While the members sampleRate, frameSize and numChannels might be quite self explaining, +pChannelType and pChannelIndices might require some more detailed explanation. + +These two arrays indicate what is each output channel supposed to be. Both array have +CStreamInfo::numChannels cells. Each cell of pChannelType indicates the channel type, described in +the enum ::AUDIO_CHANNEL_TYPE defined in FDK_audio.h. The cells of pChannelIndices indicate the sub index +among the channels starting with 0 among all channels of the same audio channel type. + +The indexing scheme is the same as for MPEG-2/4. Thus indices are counted upwards starting from the front +direction (thus a center channel if any, will always be index 0). Then the indices count up, starting always +with the left side, pairwise from front toward back. For detailed explanation, please refer to +ISO/IEC 13818-7:2005(E), chapter 8.5.3.2. + +In case a Program Config is included in the audio configuration, the channel mapping described within +it will be adopted. + +In case of MPEG-D Surround the channel mapping will follow the same criteria described in ISO/IEC 13818-7:2005(E), +but adding corresponding top channels to the channel types front, side and back, in order to avoid any +loss of information. + +\section OutputFormatChange Changing the audio output format + +The channel interleaving scheme and the actual channel order can be changed at runtime through the +parameters ::AAC_PCM_OUTPUT_INTERLEAVED and ::AAC_PCM_OUTPUT_CHANNEL_MAPPING. See the description of those +parameters and the decoder library function aacDecoder_SetParam() for more detail. + +\section OutputFormatExample Channel mapping examples + +The following examples illustrate the location of individual audio samples in the audio buffer that +is passed to aacDecoder_DecodeFrame() and the expected data in the CStreamInfo structure which can be obtained +by calling aacDecoder_GetStreamInfo(). + +\subsection ExamplesStereo Stereo + +In case of ::AAC_PCM_OUTPUT_INTERLEAVED set to 0 and ::AAC_PCM_OUTPUT_CHANNEL_MAPPING set to 1, +a AAC-LC bit stream which has channelConfiguration = 2 in its audio specific config would lead +to the following values in CStreamInfo: + +CStreamInfo::numChannels = 2 + +CStreamInfo::pChannelType = { ::ACT_FRONT, ::ACT_FRONT } + +CStreamInfo::pChannelIndices = { 0, 1 } + +Since ::AAC_PCM_OUTPUT_INTERLEAVED is set to 0, the audio channels will be located as contiguous blocks +in the output buffer as follows: + +\verbatim + <left sample 0> <left sample 1> <left sample 2> ... <left sample N> + <right sample 0> <right sample 1> <right sample 2> ... <right sample N> +\endverbatim + +Where N equals to CStreamInfo::frameSize . + +\subsection ExamplesSurround Surround 5.1 + +In case of ::AAC_PCM_OUTPUT_INTERLEAVED set to 1 and ::AAC_PCM_OUTPUT_CHANNEL_MAPPING set to 1, +a AAC-LC bit stream which has channelConfiguration = 6 in its audio specific config, would lead +to the following values in CStreamInfo: + +CStreamInfo::numChannels = 6 + +CStreamInfo::pChannelType = { ::ACT_FRONT, ::ACT_FRONT, ::ACT_FRONT, ::ACT_LFE, ::ACT_BACK, ::ACT_BACK } + +CStreamInfo::pChannelIndices = { 1, 2, 0, 0, 0, 1 } + +Since ::AAC_PCM_OUTPUT_CHANNEL_MAPPING is 1, WAV file channel ordering will be used. For a 5.1 channel +scheme, thus the channels would be: front left, front right, center, LFE, surround left, surround right. +Thus the third channel is the center channel, receiving the index 0. The other front channels are +front left, front right being placed as first and second channels with indices 1 and 2 correspondingly. +There is only one LFE, placed as the fourth channel and index 0. Finally both surround +channels get the type definition ACT_BACK, and the indices 0 and 1. + +Since ::AAC_PCM_OUTPUT_INTERLEAVED is set to 1, the audio channels will be placed in the output buffer +as follows: + +\verbatim +<front left sample 0> <front right sample 0> +<center sample 0> <LFE sample 0> +<surround left sample 0> <surround right sample 0> + +<front left sample 1> <front right sample 1> +<center sample 1> <LFE sample 1> +<surround left sample 1> <surround right sample 1> + +... + +<front left sample N> <front right sample N> +<center sample N> <LFE sample N> +<surround left sample N> <surround right sample N> +\endverbatim + +Where N equals to CStreamInfo::frameSize . + +\subsection ExamplesArib ARIB coding mode 2/1 + +In case of ::AAC_PCM_OUTPUT_INTERLEAVED set to 1 and ::AAC_PCM_OUTPUT_CHANNEL_MAPPING set to 1, +in case of a ARIB bit stream using coding mode 2/1 as described in ARIB STD-B32 Part 2 Version 2.1-E1, page 61, +would lead to the following values in CStreamInfo: + +CStreamInfo::numChannels = 3 + +CStreamInfo::pChannelType = { ::ACT_FRONT, ::ACT_FRONT,:: ACT_BACK } + +CStreamInfo::pChannelIndices = { 0, 1, 0 } + +The audio channels will be placed as follows in the audio output buffer: + +\verbatim +<front left sample 0> <front right sample 0> <mid surround sample 0> + +<front left sample 1> <front right sample 1> <mid surround sample 1> + +... + +<front left sample N> <front right sample N> <mid surround sample N> + +Where N equals to CStreamInfo::frameSize . + +\endverbatim + +*/ + +#ifndef AACDECODER_LIB_H +#define AACDECODER_LIB_H + +#include "machine_type.h" +#include "FDK_audio.h" + +#include "genericStds.h" + +/** + * \brief AAC decoder error codes. + */ +typedef enum { + AAC_DEC_OK = 0x0000, /*!< No error occured. Output buffer is valid and error free. */ + AAC_DEC_OUT_OF_MEMORY = 0x0002, /*!< Heap returned NULL pointer. Output buffer is invalid. */ + AAC_DEC_UNKNOWN = 0x0005, /*!< Error condition is of unknown reason, or from a another module. Output buffer is invalid. */ + + /* Synchronization errors. Output buffer is invalid. */ + aac_dec_sync_error_start = 0x1000, + AAC_DEC_TRANSPORT_SYNC_ERROR = 0x1001, /*!< The transport decoder had syncronisation problems. Do not exit decoding. Just feed new + bitstream data. */ + AAC_DEC_NOT_ENOUGH_BITS = 0x1002, /*!< The input buffer ran out of bits. */ + aac_dec_sync_error_end = 0x1FFF, + + /* Initialization errors. Output buffer is invalid. */ + aac_dec_init_error_start = 0x2000, + AAC_DEC_INVALID_HANDLE = 0x2001, /*!< The handle passed to the function call was invalid (NULL). */ + AAC_DEC_UNSUPPORTED_AOT = 0x2002, /*!< The AOT found in the configuration is not supported. */ + AAC_DEC_UNSUPPORTED_FORMAT = 0x2003, /*!< The bitstream format is not supported. */ + AAC_DEC_UNSUPPORTED_ER_FORMAT = 0x2004, /*!< The error resilience tool format is not supported. */ + AAC_DEC_UNSUPPORTED_EPCONFIG = 0x2005, /*!< The error protection format is not supported. */ + AAC_DEC_UNSUPPORTED_MULTILAYER = 0x2006, /*!< More than one layer for AAC scalable is not supported. */ + AAC_DEC_UNSUPPORTED_CHANNELCONFIG = 0x2007, /*!< The channel configuration (either number or arrangement) is not supported. */ + AAC_DEC_UNSUPPORTED_SAMPLINGRATE = 0x2008, /*!< The sample rate specified in the configuration is not supported. */ + AAC_DEC_INVALID_SBR_CONFIG = 0x2009, /*!< The SBR configuration is not supported. */ + AAC_DEC_SET_PARAM_FAIL = 0x200A, /*!< The parameter could not be set. Either the value was out of range or the parameter does + not exist. */ + AAC_DEC_NEED_TO_RESTART = 0x200B, /*!< The decoder needs to be restarted, since the requiered configuration change cannot be + performed. */ + aac_dec_init_error_end = 0x2FFF, + + /* Decode errors. Output buffer is valid but concealed. */ + aac_dec_decode_error_start = 0x4000, + AAC_DEC_TRANSPORT_ERROR = 0x4001, /*!< The transport decoder encountered an unexpected error. */ + AAC_DEC_PARSE_ERROR = 0x4002, /*!< Error while parsing the bitstream. Most probably it is corrupted, or the system crashed. */ + AAC_DEC_UNSUPPORTED_EXTENSION_PAYLOAD = 0x4003, /*!< Error while parsing the extension payload of the bitstream. The extension payload type + found is not supported. */ + AAC_DEC_DECODE_FRAME_ERROR = 0x4004, /*!< The parsed bitstream value is out of range. Most probably the bitstream is corrupt, or + the system crashed. */ + AAC_DEC_CRC_ERROR = 0x4005, /*!< The embedded CRC did not match. */ + AAC_DEC_INVALID_CODE_BOOK = 0x4006, /*!< An invalid codebook was signalled. Most probably the bitstream is corrupt, or the system + crashed. */ + AAC_DEC_UNSUPPORTED_PREDICTION = 0x4007, /*!< Predictor found, but not supported in the AAC Low Complexity profile. Most probably the + bitstream is corrupt, or has a wrong format. */ + AAC_DEC_UNSUPPORTED_CCE = 0x4008, /*!< A CCE element was found which is not supported. Most probably the bitstream is corrupt, or + has a wrong format. */ + AAC_DEC_UNSUPPORTED_LFE = 0x4009, /*!< A LFE element was found which is not supported. Most probably the bitstream is corrupt, or + has a wrong format. */ + AAC_DEC_UNSUPPORTED_GAIN_CONTROL_DATA = 0x400A, /*!< Gain control data found but not supported. Most probably the bitstream is corrupt, or has + a wrong format. */ + AAC_DEC_UNSUPPORTED_SBA = 0x400B, /*!< SBA found, but currently not supported in the BSAC profile. */ + AAC_DEC_TNS_READ_ERROR = 0x400C, /*!< Error while reading TNS data. Most probably the bitstream is corrupt or the system + crashed. */ + AAC_DEC_RVLC_ERROR = 0x400D, /*!< Error while decoding error resillient data. */ + aac_dec_decode_error_end = 0x4FFF, + + /* Ancillary data errors. Output buffer is valid. */ + aac_dec_anc_data_error_start = 0x8000, + AAC_DEC_ANC_DATA_ERROR = 0x8001, /*!< Non severe error concerning the ancillary data handling. */ + AAC_DEC_TOO_SMALL_ANC_BUFFER = 0x8002, /*!< The registered ancillary data buffer is too small to receive the parsed data. */ + AAC_DEC_TOO_MANY_ANC_ELEMENTS = 0x8003, /*!< More than the allowed number of ancillary data elements should be written to buffer. */ + aac_dec_anc_data_error_end = 0x8FFF + + +} AAC_DECODER_ERROR; + + +/** Macro to identify initialization errors. */ +#define IS_INIT_ERROR(err) ( (((err)>=aac_dec_init_error_start) && ((err)<=aac_dec_init_error_end)) ? 1 : 0) +/** Macro to identify decode errors. */ +#define IS_DECODE_ERROR(err) ( (((err)>=aac_dec_decode_error_start) && ((err)<=aac_dec_decode_error_end)) ? 1 : 0) +/** Macro to identify if the audio output buffer contains valid samples after calling aacDecoder_DecodeFrame(). */ +#define IS_OUTPUT_VALID(err) ( ((err) == AAC_DEC_OK) || IS_DECODE_ERROR(err) ) + +/** + * \brief AAC decoder setting parameters + */ +typedef enum +{ + AAC_PCM_OUTPUT_INTERLEAVED = 0x0000, /*!< PCM output mode (1: interleaved (default); 0: not interleaved). */ + AAC_PCM_OUTPUT_CHANNELS = 0x0001, /*!< Number of PCM output channels (if different from encoded audio channels, downmixing or + upmixing is applied). \n + -1: Disable up-/downmixing. The decoder output contains the same number of channels as the + encoded bitstream. \n + 1: The decoder performs a mono matrix mix-down if the encoded audio channels are greater + than one. Thus it ouputs always exact one channel. \n + 2: The decoder performs a stereo matrix mix-down if the encoded audio channels are greater + than two. If the encoded audio channels are smaller than two the decoder duplicates the + output. Thus it ouputs always exact two channels. \n */ + AAC_PCM_DUAL_CHANNEL_OUTPUT_MODE = 0x0002, /*!< Defines how the decoder processes two channel signals: + 0: Leave both signals as they are (default). + 1: Create a dual mono output signal from channel 1. + 2: Create a dual mono output signal from channel 2. + 3: Create a dual mono output signal by mixing both channels (L' = R' = 0.5*Ch1 + 0.5*Ch2). */ + AAC_PCM_OUTPUT_CHANNEL_MAPPING = 0x0003, /*!< Output buffer channel ordering. 0: MPEG PCE style order, 1: WAV file channel order (default). */ + + AAC_CONCEAL_METHOD = 0x0100, /*!< Error concealment: Processing method. \n + 0: Spectral muting. \n + 1: Noise substitution (see ::CONCEAL_NOISE). \n + 2: Energy interpolation (adds additional signal delay of one frame, see ::CONCEAL_INTER). \n */ + + AAC_DRC_BOOST_FACTOR = 0x0200, /*!< Dynamic Range Control: Scaling factor for boosting gain values. + Defines how the boosting DRC factors (conveyed in the bitstream) will be applied to the + decoded signal. The valid values range from 0 (don't apply boost factors) to 127 (fully + apply all boosting factors). */ + AAC_DRC_ATTENUATION_FACTOR = 0x0201, /*!< Dynamic Range Control: Scaling factor for attenuating gain values. Same as + AAC_DRC_BOOST_FACTOR but for attenuating DRC factors. */ + AAC_DRC_REFERENCE_LEVEL = 0x0202, /*!< Dynamic Range Control: Target reference level. Defines the level below full-scale + (quantized in steps of 0.25dB) to which the output audio signal will be normalized to by + the DRC module. The valid values range from 0 (full-scale) to 127 (31.75 dB below + full-scale). The value smaller than 0 switches off normalization. */ + AAC_DRC_HEAVY_COMPRESSION = 0x0203, /*!< Dynamic Range Control: En-/Disable DVB specific heavy compression (aka RF mode). + If set to 1, the decoder will apply the compression values from the DVB specific ancillary + data field. At the same time the MPEG-4 Dynamic Range Control tool will be disabled. By + default heavy compression is disabled. */ + + AAC_QMF_LOWPOWER = 0x0300, /*!< Quadrature Mirror Filter (QMF) Bank processing mode. \n + -1: Use internal default. Implies MPEG Surround partially complex accordingly. \n + 0: Use complex QMF data mode. \n + 1: Use real (low power) QMF data mode. \n */ + + AAC_MPEGS_ENABLE = 0x0500, /*!< MPEG Surround: Allow/Disable decoding of MPS content. Available only for decoders with MPEG + Surround support. */ + + AAC_TPDEC_CLEAR_BUFFER = 0x0603 /*!< Clear internal bit stream buffer of transport layers. The decoder will start decoding + at new data passed after this event and any previous data is discarded. */ + +} AACDEC_PARAM; + +/** + * \brief This structure gives information about the currently decoded audio data. + * All fields are read-only. + */ +typedef struct +{ + /* These three members are the only really relevant ones for the user. */ + INT sampleRate; /*!< The samplerate in Hz of the fully decoded PCM audio signal (after SBR processing). */ + INT frameSize; /*!< The frame size of the decoded PCM audio signal. \n + 1024 or 960 for AAC-LC \n + 2048 or 1920 for HE-AAC (v2) \n + 512 or 480 for AAC-LD and AAC-ELD */ + INT numChannels; /*!< The number of output audio channels in the decoded and interleaved PCM audio signal. */ + AUDIO_CHANNEL_TYPE *pChannelType; /*!< Audio channel type of each output audio channel. */ + UCHAR *pChannelIndices; /*!< Audio channel index for each output audio channel. + See ISO/IEC 13818-7:2005(E), 8.5.3.2 Explicit channel mapping using a program_config_element() */ + /* Decoder internal members. */ + INT aacSampleRate; /*!< sampling rate in Hz without SBR (from configuration info). */ + INT profile; /*!< MPEG-2 profile (from file header) (-1: not applicable (e. g. MPEG-4)). */ + AUDIO_OBJECT_TYPE aot; /*!< Audio Object Type (from ASC): is set to the appropriate value for MPEG-2 bitstreams (e. g. 2 for AAC-LC). */ + INT channelConfig; /*!< Channel configuration (0: PCE defined, 1: mono, 2: stereo, ... */ + INT bitRate; /*!< Instantaneous bit rate. */ + INT aacSamplesPerFrame; /*!< Samples per frame for the AAC core (from ASC). \n + 1024 or 960 for AAC-LC \n + 512 or 480 for AAC-LD and AAC-ELD */ + + AUDIO_OBJECT_TYPE extAot; /*!< Extension Audio Object Type (from ASC) */ + INT extSamplingRate; /*!< Extension sampling rate in Hz (from ASC) */ + + UINT flags; /*!< Copy if internal flags. Only to be written by the decoder, and only to be read externally. */ + + SCHAR epConfig; /*!< epConfig level (from ASC): only level 0 supported, -1 means no ER (e. g. AOT=2, MPEG-2 AAC, etc.) */ + + /* Statistics */ + INT numLostAccessUnits; /*!< This integer will reflect the estimated amount of lost access units in case aacDecoder_DecodeFrame() + returns AAC_DEC_TRANSPORT_SYNC_ERROR. It will be < 0 if the estimation failed. */ + + UINT numTotalBytes; /*!< This is the number of total bytes that have passed through the decoder. */ + UINT numBadBytes; /*!< This is the number of total bytes that were considered with errors from numTotalBytes. */ + UINT numTotalAccessUnits; /*!< This is the number of total access units that have passed through the decoder. */ + UINT numBadAccessUnits; /*!< This is the number of total access units that were considered with errors from numTotalBytes. */ + +} CStreamInfo; + + +typedef struct AAC_DECODER_INSTANCE *HANDLE_AACDECODER; + +#ifdef __cplusplus +extern "C" +{ +#endif + +/** + * \brief Initialize ancillary data buffer. + * + * \param self AAC decoder handle. + * \param buffer Pointer to (external) ancillary data buffer. + * \param size Size of the buffer pointed to by buffer. + * \return Error code. + */ +LINKSPEC_H AAC_DECODER_ERROR +aacDecoder_AncDataInit ( HANDLE_AACDECODER self, + UCHAR *buffer, + int size ); + +/** + * \brief Get one ancillary data element. + * + * \param self AAC decoder handle. + * \param index Index of the ancillary data element to get. + * \param ptr Pointer to a buffer receiving a pointer to the requested ancillary data element. + * \param size Pointer to a buffer receiving the length of the requested ancillary data element. + * \return Error code. + */ +LINKSPEC_H AAC_DECODER_ERROR +aacDecoder_AncDataGet ( HANDLE_AACDECODER self, + int index, + UCHAR **ptr, + int *size ); + +/** + * \brief Set one single decoder parameter. + * + * \param self AAC decoder handle. + * \param param Parameter to be set. + * \param value Parameter value. + * \return Error code. + */ +LINKSPEC_H AAC_DECODER_ERROR +aacDecoder_SetParam ( const HANDLE_AACDECODER self, + const AACDEC_PARAM param, + const INT value ); + + +/** + * \brief Get free bytes inside decoder internal buffer + * \param self Handle of AAC decoder instance + * \param pFreeBytes Pointer to variable receving amount of free bytes inside decoder internal buffer + * \return Error code + */ +LINKSPEC_H AAC_DECODER_ERROR +aacDecoder_GetFreeBytes ( const HANDLE_AACDECODER self, + UINT *pFreeBytes); + +/** + * \brief Open an AAC decoder instance + * \param transportFmt The transport type to be used + * \return AAC decoder handle + */ +LINKSPEC_H HANDLE_AACDECODER +aacDecoder_Open ( TRANSPORT_TYPE transportFmt, UINT nrOfLayers ); + +/** + * \brief Explicitly configure the decoder by passing a raw AudioSpecificConfig (ASC) or a StreamMuxConfig (SMC), + * contained in a binary buffer. This is required for MPEG-4 and Raw Packets file format bitstreams + * as well as for LATM bitstreams with no in-band SMC. If the transport format is LATM with or without + * LOAS, configuration is assumed to be an SMC, for all other file formats an ASC. + * + * \param self AAC decoder handle. + * \param conf Pointer to an unsigned char buffer containing the binary configuration buffer (either ASC or SMC). + * \param length Length of the configuration buffer in bytes. + * \return Error code. + */ +LINKSPEC_H AAC_DECODER_ERROR +aacDecoder_ConfigRaw ( HANDLE_AACDECODER self, + UCHAR *conf[], + const UINT length[] ); + + +/** + * \brief Fill AAC decoder's internal input buffer with bitstream data from the external input buffer. + * The function only copies such data as long as the decoder-internal input buffer is not full. + * So it grabs whatever it can from pBuffer and returns information (bytesValid) so that at a + * subsequent call of %aacDecoder_Fill(), the right position in pBuffer can be determined to + * grab the next data. + * + * \param self AAC decoder handle. + * \param pBuffer Pointer to external input buffer. + * \param bufferSize Size of external input buffer. This argument is required because decoder-internally + * we need the information to calculate the offset to pBuffer, where the next + * available data is, which is then fed into the decoder-internal buffer (as much + * as possible). Our example framework implementation fills the buffer at pBuffer + * again, once it contains no available valid bytes anymore (meaning bytesValid equal 0). + * \param bytesValid Number of bitstream bytes in the external bitstream buffer that have not yet been + * copied into the decoder's internal bitstream buffer by calling this function. + * The value is updated according to the amount of newly copied bytes. + * \return Error code. + */ +LINKSPEC_H AAC_DECODER_ERROR +aacDecoder_Fill ( HANDLE_AACDECODER self, + UCHAR *pBuffer[], + const UINT bufferSize[], + UINT *bytesValid ); + +#define AACDEC_CONCEAL 1 /*!< Flag for aacDecoder_DecodeFrame(): do not consider new input data. Do concealment. */ +#define AACDEC_FLUSH 2 /*!< Flag for aacDecoder_DecodeFrame(): Do not consider new input data. Flush filterbanks (output delayed audio). */ +#define AACDEC_INTR 4 /*!< Flag for aacDecoder_DecodeFrame(): Signal an input bit stream data discontinuity. Resync any internals as necessary. */ +#define AACDEC_CLRHIST 8 /*!< Flag for aacDecoder_DecodeFrame(): Clear all signal delay lines and history buffers. + Caution: This can cause discontinuities in the output signal. */ + +/** + * \brief Decode one audio frame + * + * \param self AAC decoder handle. + * \param pTimeData Pointer to external output buffer where the decoded PCM samples will be stored into. + * \param flags Bit field with flags for the decoder: \n + * (flags & AACDEC_CONCEAL) == 1: Do concealment. \n + * (flags & AACDEC_FLUSH) == 2: Discard input data. Flush filter banks (output delayed audio). \n + * (flags & AACDEC_INTR) == 4: Input data is discontinuous. Resynchronize any internals as necessary. + * \return Error code. + */ +LINKSPEC_H AAC_DECODER_ERROR +aacDecoder_DecodeFrame ( HANDLE_AACDECODER self, + INT_PCM *pTimeData, + const INT timeDataSize, + const UINT flags ); + +/** + * \brief De-allocate all resources of an AAC decoder instance. + * + * \param self AAC decoder handle. + * \return void + */ +LINKSPEC_H void aacDecoder_Close ( HANDLE_AACDECODER self ); + +/** + * \brief Get CStreamInfo handle from decoder. + * + * \param self AAC decoder handle. + * \return Reference to requested CStreamInfo. + */ +LINKSPEC_H CStreamInfo* aacDecoder_GetStreamInfo( HANDLE_AACDECODER self ); + +/** + * \brief Get decoder library info. + * + * \param info Pointer to an allocated LIB_INFO structure. + * \return 0 on success + */ +LINKSPEC_H INT aacDecoder_GetLibInfo( LIB_INFO *info ); + + +#ifdef __cplusplus +} +#endif + +#endif /* AACDECODER_LIB_H */ |