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authorDave Burke <daveburke@google.com>2012-04-17 09:51:45 -0700
committerDave Burke <daveburke@google.com>2012-04-17 23:04:43 -0700
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Fraunhofer AAC codec.
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+/***************************** MPEG-4 AAC Decoder **************************
+
+ (C) Copyright Fraunhofer IIS 1999-2008
+ All Rights Reserved
+
+ Please be advised that this software and/or program delivery is
+ Confidential Information of Fraunhofer and subject to and covered by the
+
+ Fraunhofer IIS Software Evaluation Agreement
+ between Google Inc. and Fraunhofer
+ effective and in full force since March 1, 2012.
+
+ You may use this software and/or program only under the terms and
+ conditions described in the above mentioned Fraunhofer IIS Software
+ Evaluation Agreement. Any other and/or further use requires a separate agreement.
+
+
+ $Id$
+ Author(s): Manuel Jander
+
+ This software and/or program is protected by copyright law and international
+ treaties. Any reproduction or distribution of this software and/or program,
+ or any portion of it, may result in severe civil and criminal penalties, and
+ will be prosecuted to the maximum extent possible under law.
+
+******************************************************************************/
+
+/**
+ * \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, Kr&auml;mer, 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 */