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      1 
      2 /* -----------------------------------------------------------------------------------------------------------
      3 Software License for The Fraunhofer FDK AAC Codec Library for Android
      4 
      5  Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Frderung der angewandten Forschung e.V.
      6   All rights reserved.
      7 
      8  1.    INTRODUCTION
      9 The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
     10 the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
     11 This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
     12 
     13 AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
     14 audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
     15 independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
     16 of the MPEG specifications.
     17 
     18 Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
     19 may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
     20 individually for the purpose of encoding or decoding bit streams in products that are compliant with
     21 the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
     22 these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
     23 software may already be covered under those patent licenses when it is used for those licensed purposes only.
     24 
     25 Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
     26 are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
     27 applications information and documentation.
     28 
     29 2.    COPYRIGHT LICENSE
     30 
     31 Redistribution and use in source and binary forms, with or without modification, are permitted without
     32 payment of copyright license fees provided that you satisfy the following conditions:
     33 
     34 You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
     35 your modifications thereto in source code form.
     36 
     37 You must retain the complete text of this software license in the documentation and/or other materials
     38 provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
     39 You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
     40 modifications thereto to recipients of copies in binary form.
     41 
     42 The name of Fraunhofer may not be used to endorse or promote products derived from this library without
     43 prior written permission.
     44 
     45 You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
     46 software or your modifications thereto.
     47 
     48 Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
     49 and the date of any change. For modified versions of the FDK AAC Codec, the term
     50 "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
     51 "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
     52 
     53 3.    NO PATENT LICENSE
     54 
     55 NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
     56 ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
     57 respect to this software.
     58 
     59 You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
     60 by appropriate patent licenses.
     61 
     62 4.    DISCLAIMER
     63 
     64 This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
     65 "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
     66 of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
     67 CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
     68 including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
     69 or business interruption, however caused and on any theory of liability, whether in contract, strict
     70 liability, or tort (including negligence), arising in any way out of the use of this software, even if
     71 advised of the possibility of such damage.
     72 
     73 5.    CONTACT INFORMATION
     74 
     75 Fraunhofer Institute for Integrated Circuits IIS
     76 Attention: Audio and Multimedia Departments - FDK AAC LL
     77 Am Wolfsmantel 33
     78 91058 Erlangen, Germany
     79 
     80 www.iis.fraunhofer.de/amm
     81 amm-info (at) iis.fraunhofer.de
     82 ----------------------------------------------------------------------------------------------------------- */
     83 
     84 /*****************************  MPEG-4 AAC Decoder  **************************
     85 
     86    Author(s):   Manuel Jander
     87 
     88 ******************************************************************************/
     89 
     90 /**
     91  * \file   aacdecoder_lib.h
     92  * \brief  FDK AAC decoder library interface header file.
     93  *
     94 
     95 \page INTRO Introduction
     96 
     97 \section SCOPE Scope
     98 
     99 This document describes the high-level interface and usage of the ISO/MPEG-2/4 AAC Decoder
    100 library developed by the Fraunhofer Institute for Integrated Circuits (IIS).
    101 Depending on the library configuration, it implements decoding of AAC-LC (Low-Complexity),
    102 HE-AAC (High-Efficiency AAC, v1 and v2), AAC-LD (Low-Delay) and AAC-ELD (Enhanced Low-Delay).
    103 
    104 All references to SBR (Spectral Band Replication) are only applicable to HE-AAC and AAC-ELD
    105 versions of the library. All references to PS (Parametric Stereo) are only applicable to
    106 HE-AAC v2 versions of the library.
    107 
    108 \section DecoderBasics Decoder Basics
    109 
    110 This document can only give a rough overview about the ISO/MPEG-2 and ISO/MPEG-4 AAC audio
    111 coding standard. To understand all the terms in this document, you are encouraged to read
    112 the following documents.
    113 
    114 - ISO/IEC 13818-7 (MPEG-2 AAC), which defines the syntax of MPEG-2 AAC audio bitstreams.
    115 - ISO/IEC 14496-3 (MPEG-4 AAC, subpart 1 and 4), which defines the syntax of MPEG-4 AAC audio bitstreams.
    116 - Lutzky, Schuller, Gayer, Krämer, Wabnik, "A guideline to audio codec delay", 116th AES Convention, May 8, 2004
    117 
    118 MPEG Advanced Audio Coding is based on a time-to-frequency mapping of the signal. The signal
    119 is partitioned into overlapping portions and transformed into frequency domain. The spectral
    120 components are then quantized and coded.\n
    121 An MPEG2 or MPEG4 AAC audio bitstream is composed of frames. Contrary to MPEG-1/2 Layer-3 (mp3),
    122 the length of individual frames is not restricted to a fixed number of bytes, but can take on
    123 any length between 1 and 768 bytes.
    124 
    125 
    126 \page LIBUSE Library Usage
    127 
    128 \section InterfaceDescritpion API Description
    129 
    130 All API header files are located in the folder /include of the release package. They are described in
    131 detail in this document. All header files are provided for usage in C/C++ programs. The AAC decoder library
    132 API functions are located at aacdecoder_lib.h.
    133 
    134 In binary releases the decoder core resides in statically linkable libraries called for example libAACdec.a,
    135 (Linux) or FDK_aacDec_lib (Microsoft Visual C++).
    136 
    137 \section Calling_Sequence Calling Sequence
    138 
    139 For decoding of ISO/MPEG-2/4 AAC or HE-AAC v2 bitstreams the following sequence is mandatory. Input read
    140 and output write functions as well as the corresponding open and close functions are left out, since they
    141 may be implemented differently according to the user's specific requirements. The example implementation in
    142 main.cpp uses file-based input/output, and in such case call mpegFileRead_Open() to open an input file and
    143 to allocate memory for the required structures, and the corresponding mpegFileRead_Close() to close opened
    144 files and to de-allocate associated structures. mpegFileRead_Open() tries to detect the bitstream format and
    145 in case of MPEG-4 file format or Raw Packets file format (a Fraunhofer IIS proprietary format) reads the Audio
    146 Specific Config data (ASC). An unsuccessful attempt to recognize the bitstream format requires the user to
    147 provide this information manually (see \ref CommandLineUsage). For any other bitstream formats that are
    148 usually applicable in streaming applications, the decoder itself will try to synchronize and parse the given
    149 bitstream fragment using the FDK transport library. Hence, for streaming applications (without file access)
    150 this step is not necessary.
    151 
    152 -# Call aacDecoder_Open() to open and retrieve a handle to a new AAC decoder instance.
    153 \dontinclude main.cpp
    154 \skipline aacDecoder_Open
    155 -# If out-of-band config data (Audio Specific Config (ASC) or Stream Mux Config (SMC)) is available, call
    156 aacDecoder_ConfigRaw() to pass it to the decoder and before the decoding process starts. If this data is
    157 not available in advance, the decoder will get it from the bitstream  and configure itself while decoding
    158 with aacDecoder_DecodeFrame().
    159 -# Begin decoding loop.
    160 \skipline do {
    161 -# Read data from bitstream file or stream into a client-supplied input buffer ("inBuffer" in main.cpp).
    162 If it is very small like just 4, aacDecoder_DecodeFrame() will
    163 repeatedly return ::AAC_DEC_NOT_ENOUGH_BITS until enough bits were fed by aacDecoder_Fill(). Only read data
    164 when this buffer has completely been processed and is then empty. For file-based input execute
    165 mpegFileRead_Read() or any other implementation with similar functionality.
    166 -# Call aacDecoder_Fill() to fill the decoder's internal bitstream input buffer with the client-supplied
    167 external bitstream input buffer.
    168 \skipline aacDecoder_Fill
    169 -# Call aacDecoder_DecodeFrame() which writes decoded PCM audio data to a client-supplied buffer. It is the
    170 client's responsibility to allocate a buffer which is large enough to hold this output data.
    171 \skipline aacDecoder_DecodeFrame
    172 If the bitstream's configuration (number of channels, sample rate, frame size) is not known in advance, you may
    173 call aacDecoder_GetStreamInfo() to retrieve a structure containing this information and then initialize an audio
    174 output device. In the example main.cpp, if the number of channels or the sample rate has changed since program
    175 start or since the previously decoded frame, the audio output device will be re-initialized. If WAVE file output
    176 is chosen, a new WAVE file for each new configuration will be created.
    177 \skipline aacDecoder_GetStreamInfo
    178 -# Repeat steps 5 to 7 until no data to decode is available anymore, or if an error occured.
    179 \skipline } while
    180 -# Call aacDecoder_Close() to de-allocate all AAC decoder and transport layer structures.
    181 \skipline aacDecoder_Close
    182 
    183 \section BufferSystem Buffer System
    184 
    185 There are three main buffers in an AAC decoder application. One external input buffer to hold bitstream
    186 data from file I/O or elsewhere, one decoder-internal input buffer, and one to hold the decoded output
    187 PCM sample data, whereas this output buffer may overlap with the external input buffer.
    188 
    189 The external input buffer is set in the example framework main.cpp and its size is defined by ::IN_BUF_SIZE.
    190 You may freely choose different sizes here. To feed the data to the decoder-internal input buffer, use the
    191 function aacDecoder_Fill(). This function returns important information about how many bytes in the
    192 external input buffer have not yet been copied into the internal input buffer (variable bytesValid).
    193 Once the external buffer has been fully copied, it can be re-filled again.
    194 In case you want to re-fill it when there are still unprocessed bytes (bytesValid is unequal 0), you
    195 would have to additionally perform a memcpy(), so that just means unnecessary computational overhead
    196 and therefore we recommend to re-fill the buffer only when bytesValid is 0.
    197 
    198 \image latex dec_buffer.png "Lifecycle of the external input buffer" width=9cm
    199 
    200 The size of the decoder-internal input buffer is set in tpdec_lib.h (see define ::TRANSPORTDEC_INBUF_SIZE).
    201 You may choose a smaller size under the following considerations:
    202 
    203 - each input channel requires 768 bytes
    204 - the whole buffer must be of size 2^n
    205 
    206 So for example a stereo decoder:
    207 
    208 \f[
    209 TRANSPORTDEC\_INBUF\_SIZE = 2 * 768 = 1536 => 2048
    210 \f]
    211 
    212 tpdec_lib.h and TRANSPORTDEC_INBUF_SIZE are not part of the decoder's library interface. Therefore
    213 only source-code clients may change this setting. If you received a library release, please ask us and
    214 we can change this in order to meet your memory requirements.
    215 
    216 \page OutputFormat Decoder audio output
    217 
    218 \section OutputFormatObtaining Obtaining channel mapping information
    219 
    220 The decoded audio output format is indicated by a set of variables of the CStreamInfo structure.
    221 While the members sampleRate, frameSize and numChannels might be quite self explaining,
    222 pChannelType and pChannelIndices might require some more detailed explanation.
    223 
    224 These two arrays indicate what is each output channel supposed to be. Both array have
    225 CStreamInfo::numChannels cells. Each cell of pChannelType indicates the channel type, described in
    226 the enum ::AUDIO_CHANNEL_TYPE defined in FDK_audio.h. The cells of pChannelIndices indicate the sub index
    227 among the channels starting with 0 among all channels of the same audio channel type.
    228 
    229 The indexing scheme is the same as for MPEG-2/4. Thus indices are counted upwards starting from the front
    230 direction (thus a center channel if any, will always be index 0). Then the indices count up, starting always
    231 with the left side, pairwise from front toward back. For detailed explanation, please refer to
    232 ISO/IEC 13818-7:2005(E), chapter 8.5.3.2.
    233 
    234 In case a Program Config is included in the audio configuration, the channel mapping described within
    235 it will be adopted.
    236 
    237 In case of MPEG-D Surround the channel mapping will follow the same criteria described in ISO/IEC 13818-7:2005(E),
    238 but adding corresponding top channels to the channel types front, side and back, in order to avoid any
    239 loss of information.
    240 
    241 \section OutputFormatChange Changing the audio output format
    242 
    243 The channel interleaving scheme and the actual channel order can be changed at runtime through the
    244 parameters ::AAC_PCM_OUTPUT_INTERLEAVED and ::AAC_PCM_OUTPUT_CHANNEL_MAPPING. See the description of those
    245 parameters and the decoder library function aacDecoder_SetParam() for more detail.
    246 
    247 \section OutputFormatExample Channel mapping examples
    248 
    249 The following examples illustrate the location of individual audio samples in the audio buffer that
    250 is passed to aacDecoder_DecodeFrame() and the expected data in the CStreamInfo structure which can be obtained
    251 by calling aacDecoder_GetStreamInfo().
    252 
    253 \subsection ExamplesStereo Stereo
    254 
    255 In case of ::AAC_PCM_OUTPUT_INTERLEAVED set to 0 and ::AAC_PCM_OUTPUT_CHANNEL_MAPPING set to 1,
    256 a AAC-LC bit stream which has channelConfiguration = 2 in its audio specific config would lead
    257 to the following values in CStreamInfo:
    258 
    259 CStreamInfo::numChannels = 2
    260 
    261 CStreamInfo::pChannelType = { ::ACT_FRONT, ::ACT_FRONT }
    262 
    263 CStreamInfo::pChannelIndices = { 0, 1 }
    264 
    265 Since ::AAC_PCM_OUTPUT_INTERLEAVED is set to 0, the audio channels will be located as contiguous blocks
    266 in the output buffer as follows:
    267 
    268 \verbatim
    269   <left sample 0>  <left sample 1>  <left sample 2>  ... <left sample N>
    270   <right sample 0> <right sample 1> <right sample 2> ... <right sample N>
    271 \endverbatim
    272 
    273 Where N equals to CStreamInfo::frameSize .
    274 
    275 \subsection ExamplesSurround Surround 5.1
    276 
    277 In case of ::AAC_PCM_OUTPUT_INTERLEAVED set to 1 and ::AAC_PCM_OUTPUT_CHANNEL_MAPPING set to 1,
    278 a AAC-LC bit stream which has channelConfiguration = 6 in its audio specific config, would lead
    279 to the following values in CStreamInfo:
    280 
    281 CStreamInfo::numChannels = 6
    282 
    283 CStreamInfo::pChannelType = { ::ACT_FRONT, ::ACT_FRONT, ::ACT_FRONT, ::ACT_LFE, ::ACT_BACK, ::ACT_BACK }
    284 
    285 CStreamInfo::pChannelIndices = { 1, 2, 0, 0, 0, 1 }
    286 
    287 Since ::AAC_PCM_OUTPUT_CHANNEL_MAPPING is 1, WAV file channel ordering will be used. For a 5.1 channel
    288 scheme, thus the channels would be: front left, front right, center, LFE, surround left, surround right.
    289 Thus the third channel is the center channel, receiving the index 0. The other front channels are
    290 front left, front right being placed as first and second channels with indices 1 and 2 correspondingly.
    291 There is only one LFE, placed as the fourth channel and index 0. Finally both surround
    292 channels get the type definition ACT_BACK, and the indices 0 and 1.
    293 
    294 Since ::AAC_PCM_OUTPUT_INTERLEAVED is set to 1, the audio channels will be placed in the output buffer
    295 as follows:
    296 
    297 \verbatim
    298 <front left sample 0> <front right sample 0>
    299 <center sample 0> <LFE sample 0>
    300 <surround left sample 0> <surround right sample 0>
    301 
    302 <front left sample 1> <front right sample 1>
    303 <center sample 1> <LFE sample 1>
    304 <surround left sample 1> <surround right sample 1>
    305 
    306 ...
    307 
    308 <front left sample N> <front right sample N>
    309 <center sample N> <LFE sample N>
    310 <surround left sample N> <surround right sample N>
    311 \endverbatim
    312 
    313 Where N equals to CStreamInfo::frameSize .
    314 
    315 \subsection ExamplesArib ARIB coding mode 2/1
    316 
    317 In case of ::AAC_PCM_OUTPUT_INTERLEAVED set to 1 and ::AAC_PCM_OUTPUT_CHANNEL_MAPPING set to 1,
    318 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,
    319 would lead to the following values in CStreamInfo:
    320 
    321 CStreamInfo::numChannels = 3
    322 
    323 CStreamInfo::pChannelType = { ::ACT_FRONT, ::ACT_FRONT,:: ACT_BACK }
    324 
    325 CStreamInfo::pChannelIndices = { 0, 1, 0 }
    326 
    327 The audio channels will be placed as follows in the audio output buffer:
    328 
    329 \verbatim
    330 <front left sample 0> <front right sample 0>  <mid surround sample 0>
    331 
    332 <front left sample 1> <front right sample 1> <mid surround sample 1>
    333 
    334 ...
    335 
    336 <front left sample N> <front right sample N> <mid surround sample N>
    337 
    338 Where N equals to CStreamInfo::frameSize .
    339 
    340 \endverbatim
    341 
    342 */
    343 
    344 #ifndef AACDECODER_LIB_H
    345 #define AACDECODER_LIB_H
    346 
    347 #include "machine_type.h"
    348 #include "FDK_audio.h"
    349 
    350 #include "genericStds.h"
    351 
    352 /**
    353  * \brief  AAC decoder error codes.
    354  */
    355 typedef enum {
    356   AAC_DEC_OK                             = 0x0000,  /*!< No error occured. Output buffer is valid and error free. */
    357   AAC_DEC_OUT_OF_MEMORY                  = 0x0002,  /*!< Heap returned NULL pointer. Output buffer is invalid. */
    358   AAC_DEC_UNKNOWN                        = 0x0005,  /*!< Error condition is of unknown reason, or from a another module. Output buffer is invalid. */
    359 
    360   /* Synchronization errors. Output buffer is invalid. */
    361   aac_dec_sync_error_start               = 0x1000,
    362   AAC_DEC_TRANSPORT_SYNC_ERROR           = 0x1001,  /*!< The transport decoder had syncronisation problems. Do not exit decoding. Just feed new
    363                                                          bitstream data. */
    364   AAC_DEC_NOT_ENOUGH_BITS                = 0x1002,  /*!< The input buffer ran out of bits. */
    365   aac_dec_sync_error_end                 = 0x1FFF,
    366 
    367   /* Initialization errors. Output buffer is invalid. */
    368   aac_dec_init_error_start               = 0x2000,
    369   AAC_DEC_INVALID_HANDLE                 = 0x2001,  /*!< The handle passed to the function call was invalid (NULL). */
    370   AAC_DEC_UNSUPPORTED_AOT                = 0x2002,  /*!< The AOT found in the configuration is not supported. */
    371   AAC_DEC_UNSUPPORTED_FORMAT             = 0x2003,  /*!< The bitstream format is not supported.  */
    372   AAC_DEC_UNSUPPORTED_ER_FORMAT          = 0x2004,  /*!< The error resilience tool format is not supported. */
    373   AAC_DEC_UNSUPPORTED_EPCONFIG           = 0x2005,  /*!< The error protection format is not supported. */
    374   AAC_DEC_UNSUPPORTED_MULTILAYER         = 0x2006,  /*!< More than one layer for AAC scalable is not supported. */
    375   AAC_DEC_UNSUPPORTED_CHANNELCONFIG      = 0x2007,  /*!< The channel configuration (either number or arrangement) is not supported. */
    376   AAC_DEC_UNSUPPORTED_SAMPLINGRATE       = 0x2008,  /*!< The sample rate specified in the configuration is not supported. */
    377   AAC_DEC_INVALID_SBR_CONFIG             = 0x2009,  /*!< The SBR configuration is not supported. */
    378   AAC_DEC_SET_PARAM_FAIL                 = 0x200A,  /*!< The parameter could not be set. Either the value was out of range or the parameter does
    379                                                          not exist. */
    380   AAC_DEC_NEED_TO_RESTART                = 0x200B,  /*!< The decoder needs to be restarted, since the requiered configuration change cannot be
    381                                                          performed. */
    382   aac_dec_init_error_end                 = 0x2FFF,
    383 
    384   /* Decode errors. Output buffer is valid but concealed. */
    385   aac_dec_decode_error_start             = 0x4000,
    386   AAC_DEC_TRANSPORT_ERROR                = 0x4001,  /*!< The transport decoder encountered an unexpected error. */
    387   AAC_DEC_PARSE_ERROR                    = 0x4002,  /*!< Error while parsing the bitstream. Most probably it is corrupted, or the system crashed. */
    388   AAC_DEC_UNSUPPORTED_EXTENSION_PAYLOAD  = 0x4003,  /*!< Error while parsing the extension payload of the bitstream. The extension payload type
    389                                                          found is not supported. */
    390   AAC_DEC_DECODE_FRAME_ERROR             = 0x4004,  /*!< The parsed bitstream value is out of range. Most probably the bitstream is corrupt, or
    391                                                          the system crashed. */
    392   AAC_DEC_CRC_ERROR                      = 0x4005,  /*!< The embedded CRC did not match. */
    393   AAC_DEC_INVALID_CODE_BOOK              = 0x4006,  /*!< An invalid codebook was signalled. Most probably the bitstream is corrupt, or the system
    394                                                          crashed. */
    395   AAC_DEC_UNSUPPORTED_PREDICTION         = 0x4007,  /*!< Predictor found, but not supported in the AAC Low Complexity profile. Most probably the
    396                                                          bitstream is corrupt, or has a wrong format. */
    397   AAC_DEC_UNSUPPORTED_CCE                = 0x4008,  /*!< A CCE element was found which is not supported. Most probably the bitstream is corrupt, or
    398                                                          has a wrong format. */
    399   AAC_DEC_UNSUPPORTED_LFE                = 0x4009,  /*!< A LFE element was found which is not supported. Most probably the bitstream is corrupt, or
    400                                                          has a wrong format. */
    401   AAC_DEC_UNSUPPORTED_GAIN_CONTROL_DATA  = 0x400A,  /*!< Gain control data found but not supported. Most probably the bitstream is corrupt, or has
    402                                                          a wrong format. */
    403   AAC_DEC_UNSUPPORTED_SBA                = 0x400B,  /*!< SBA found, but currently not supported in the BSAC profile. */
    404   AAC_DEC_TNS_READ_ERROR                 = 0x400C,  /*!< Error while reading TNS data. Most probably the bitstream is corrupt or the system
    405                                                          crashed. */
    406   AAC_DEC_RVLC_ERROR                     = 0x400D,  /*!< Error while decoding error resillient data. */
    407   aac_dec_decode_error_end               = 0x4FFF,
    408 
    409   /* Ancillary data errors. Output buffer is valid. */
    410   aac_dec_anc_data_error_start           = 0x8000,
    411   AAC_DEC_ANC_DATA_ERROR                 = 0x8001,  /*!< Non severe error concerning the ancillary data handling. */
    412   AAC_DEC_TOO_SMALL_ANC_BUFFER           = 0x8002,  /*!< The registered ancillary data buffer is too small to receive the parsed data. */
    413   AAC_DEC_TOO_MANY_ANC_ELEMENTS          = 0x8003,  /*!< More than the allowed number of ancillary data elements should be written to buffer. */
    414   aac_dec_anc_data_error_end             = 0x8FFF
    415 
    416 
    417 } AAC_DECODER_ERROR;
    418 
    419 
    420 /** Macro to identify initialization errors. */
    421 #define IS_INIT_ERROR(err)   ( (((err)>=aac_dec_init_error_start)   && ((err)<=aac_dec_init_error_end))   ? 1 : 0)
    422 /** Macro to identify decode errors. */
    423 #define IS_DECODE_ERROR(err) ( (((err)>=aac_dec_decode_error_start) && ((err)<=aac_dec_decode_error_end)) ? 1 : 0)
    424 /** Macro to identify if the audio output buffer contains valid samples after calling aacDecoder_DecodeFrame(). */
    425 #define IS_OUTPUT_VALID(err) ( ((err) == AAC_DEC_OK) || IS_DECODE_ERROR(err) )
    426 
    427 /**
    428  * \brief AAC decoder setting parameters
    429  */
    430 typedef enum
    431 {
    432   AAC_PCM_OUTPUT_INTERLEAVED              = 0x0000,  /*!< PCM output mode (1: interleaved (default); 0: not interleaved). */
    433   AAC_PCM_OUTPUT_CHANNELS                 = 0x0001,  /*!< Number of PCM output channels (if different from encoded audio channels, downmixing or
    434                                                           upmixing is applied). \n
    435                                                           -1: Disable up-/downmixing. The decoder output contains the same number of channels as the
    436                                                               encoded bitstream. \n
    437                                                            1: The decoder performs a mono matrix mix-down if the encoded audio channels are greater
    438                                                               than one. Thus it ouputs always exact one channel. \n
    439                                                            2: The decoder performs a stereo matrix mix-down if the encoded audio channels are greater
    440                                                               than two. If the encoded audio channels are smaller than two the decoder duplicates the
    441                                                               output. Thus it ouputs always exact two channels. \n */
    442   AAC_PCM_DUAL_CHANNEL_OUTPUT_MODE        = 0x0002,  /*!< Defines how the decoder processes two channel signals:
    443                                                           0: Leave both signals as they are (default).
    444                                                           1: Create a dual mono output signal from channel 1.
    445                                                           2: Create a dual mono output signal from channel 2.
    446                                                           3: Create a dual mono output signal by mixing both channels (L' = R' = 0.5*Ch1 + 0.5*Ch2). */
    447   AAC_PCM_OUTPUT_CHANNEL_MAPPING          = 0x0003,  /*!< Output buffer channel ordering. 0: MPEG PCE style order, 1: WAV file channel order (default). */
    448 
    449   AAC_CONCEAL_METHOD                      = 0x0100,  /*!< Error concealment: Processing method. \n
    450                                                           0: Spectral muting. \n
    451                                                           1: Noise substitution (see ::CONCEAL_NOISE). \n
    452                                                           2: Energy interpolation (adds additional signal delay of one frame, see ::CONCEAL_INTER). \n */
    453 
    454   AAC_DRC_BOOST_FACTOR                    = 0x0200,  /*!< Dynamic Range Control: Scaling factor for boosting gain values.
    455                                                           Defines how the boosting DRC factors (conveyed in the bitstream) will be applied to the
    456                                                           decoded signal. The valid values range from 0 (don't apply boost factors) to 127 (fully
    457                                                           apply all boosting factors). */
    458   AAC_DRC_ATTENUATION_FACTOR              = 0x0201,  /*!< Dynamic Range Control: Scaling factor for attenuating gain values. Same as
    459                                                           AAC_DRC_BOOST_FACTOR but for attenuating DRC factors. */
    460   AAC_DRC_REFERENCE_LEVEL                 = 0x0202,  /*!< Dynamic Range Control: Target reference level. Defines the level below full-scale
    461                                                           (quantized in steps of 0.25dB) to which the output audio signal will be normalized to by
    462                                                           the DRC module. The valid values range from 0 (full-scale) to 127 (31.75 dB below
    463                                                           full-scale). The value smaller than 0 switches off normalization. */
    464   AAC_DRC_HEAVY_COMPRESSION               = 0x0203,  /*!< Dynamic Range Control: En-/Disable DVB specific heavy compression (aka RF mode).
    465                                                           If set to 1, the decoder will apply the compression values from the DVB specific ancillary
    466                                                           data field. At the same time the MPEG-4 Dynamic Range Control tool will be disabled. By
    467                                                           default heavy compression is disabled. */
    468 
    469   AAC_QMF_LOWPOWER                        = 0x0300,  /*!< Quadrature Mirror Filter (QMF) Bank processing mode. \n
    470                                                           -1: Use internal default. Implies MPEG Surround partially complex accordingly. \n
    471                                                            0: Use complex QMF data mode. \n
    472                                                            1: Use real (low power) QMF data mode. \n */
    473 
    474   AAC_MPEGS_ENABLE                        = 0x0500,  /*!< MPEG Surround: Allow/Disable decoding of MPS content. Available only for decoders with MPEG
    475                                                           Surround support. */
    476 
    477   AAC_TPDEC_CLEAR_BUFFER                  = 0x0603   /*!< Clear internal bit stream buffer of transport layers. The decoder will start decoding
    478                                                           at new data passed after this event and any previous data is discarded. */
    479 
    480 } AACDEC_PARAM;
    481 
    482 /**
    483  * \brief This structure gives information about the currently decoded audio data.
    484  *        All fields are read-only.
    485  */
    486 typedef struct
    487 {
    488   /* These three members are the only really relevant ones for the user.                                                           */
    489   INT               sampleRate;          /*!< The samplerate in Hz of the fully decoded PCM audio signal (after SBR processing).   */
    490   INT               frameSize;           /*!< The frame size of the decoded PCM audio signal. \n
    491                                               1024 or 960 for AAC-LC \n
    492                                               2048 or 1920 for HE-AAC (v2) \n
    493                                               512 or 480 for AAC-LD and AAC-ELD                                                    */
    494   INT               numChannels;         /*!< The number of output audio channels in the decoded and interleaved PCM audio signal. */
    495   AUDIO_CHANNEL_TYPE *pChannelType;       /*!< Audio channel type of each output audio channel.           */
    496   UCHAR             *pChannelIndices;     /*!< Audio channel index for each output audio channel.
    497                                                See ISO/IEC 13818-7:2005(E), 8.5.3.2 Explicit channel mapping using a program_config_element() */
    498   /* Decoder internal members. */
    499   INT               aacSampleRate;       /*!< sampling rate in Hz without SBR (from configuration info).                           */
    500   INT               profile;             /*!< MPEG-2 profile (from file header) (-1: not applicable (e. g. MPEG-4)).               */
    501   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). */
    502   INT               channelConfig;       /*!< Channel configuration (0: PCE defined, 1: mono, 2: stereo, ...                       */
    503   INT               bitRate;             /*!< Instantaneous bit rate.                   */
    504   INT               aacSamplesPerFrame;  /*!< Samples per frame for the AAC core (from ASC). \n
    505                                               1024 or 960 for AAC-LC \n
    506                                               512 or 480 for AAC-LD and AAC-ELD         */
    507   INT               aacNumChannels;      /*!< The number of audio channels after AAC core processing (before PS or MPS processing).
    508                                               CAUTION: This are not the final number of output channels! */
    509   AUDIO_OBJECT_TYPE extAot;              /*!< Extension Audio Object Type (from ASC)   */
    510   INT               extSamplingRate;     /*!< Extension sampling rate in Hz (from ASC) */
    511 
    512   UINT              flags;               /*!< Copy if internal flags. Only to be written by the decoder, and only to be read externally. */
    513 
    514   SCHAR             epConfig;            /*!< epConfig level (from ASC): only level 0 supported, -1 means no ER (e. g. AOT=2, MPEG-2 AAC, etc.)  */
    515 
    516   /* Statistics */
    517   INT               numLostAccessUnits;  /*!< This integer will reflect the estimated amount of lost access units in case aacDecoder_DecodeFrame()
    518                                               returns AAC_DEC_TRANSPORT_SYNC_ERROR. It will be < 0 if the estimation failed. */
    519 
    520   UINT              numTotalBytes;       /*!< This is the number of total bytes that have passed through the decoder. */
    521   UINT              numBadBytes;         /*!< This is the number of total bytes that were considered with errors from numTotalBytes. */
    522   UINT              numTotalAccessUnits; /*!< This is the number of total access units that have passed through the decoder. */
    523   UINT              numBadAccessUnits;   /*!< This is the number of total access units that were considered with errors from numTotalBytes. */
    524 
    525 } CStreamInfo;
    526 
    527 
    528 typedef struct AAC_DECODER_INSTANCE *HANDLE_AACDECODER;
    529 
    530 #ifdef __cplusplus
    531 extern "C"
    532 {
    533 #endif
    534 
    535 /**
    536  * \brief Initialize ancillary data buffer.
    537  *
    538  * \param self    AAC decoder handle.
    539  * \param buffer  Pointer to (external) ancillary data buffer.
    540  * \param size    Size of the buffer pointed to by buffer.
    541  * \return        Error code.
    542  */
    543 LINKSPEC_H AAC_DECODER_ERROR
    544 aacDecoder_AncDataInit ( HANDLE_AACDECODER self,
    545                          UCHAR            *buffer,
    546                          int               size );
    547 
    548 /**
    549  * \brief Get one ancillary data element.
    550  *
    551  * \param self   AAC decoder handle.
    552  * \param index  Index of the ancillary data element to get.
    553  * \param ptr    Pointer to a buffer receiving a pointer to the requested ancillary data element.
    554  * \param size   Pointer to a buffer receiving the length of the requested ancillary data element.
    555  * \return       Error code.
    556  */
    557 LINKSPEC_H AAC_DECODER_ERROR
    558 aacDecoder_AncDataGet ( HANDLE_AACDECODER self,
    559                         int               index,
    560                         UCHAR           **ptr,
    561                         int              *size );
    562 
    563 /**
    564  * \brief Set one single decoder parameter.
    565  *
    566  * \param self   AAC decoder handle.
    567  * \param param  Parameter to be set.
    568  * \param value  Parameter value.
    569  * \return       Error code.
    570  */
    571 LINKSPEC_H AAC_DECODER_ERROR
    572 aacDecoder_SetParam ( const HANDLE_AACDECODER  self,
    573                       const AACDEC_PARAM       param,
    574                       const INT                value );
    575 
    576 
    577 /**
    578  * \brief              Get free bytes inside decoder internal buffer
    579  * \param self    Handle of AAC decoder instance
    580  * \param pFreeBytes Pointer to variable receving amount of free bytes inside decoder internal buffer
    581  * \return             Error code
    582  */
    583 LINKSPEC_H AAC_DECODER_ERROR
    584 aacDecoder_GetFreeBytes ( const HANDLE_AACDECODER  self,
    585                                             UINT *pFreeBytes);
    586 
    587 /**
    588  * \brief               Open an AAC decoder instance
    589  * \param transportFmt  The transport type to be used
    590  * \return              AAC decoder handle
    591  */
    592 LINKSPEC_H HANDLE_AACDECODER
    593 aacDecoder_Open ( TRANSPORT_TYPE transportFmt, UINT nrOfLayers );
    594 
    595 /**
    596  * \brief Explicitly configure the decoder by passing a raw AudioSpecificConfig (ASC) or a StreamMuxConfig (SMC),
    597  *  contained in a binary buffer. This is required for MPEG-4 and Raw Packets file format bitstreams
    598  *  as well as for LATM bitstreams with no in-band SMC. If the transport format is LATM with or without
    599  *  LOAS, configuration is assumed to be an SMC, for all other file formats an ASC.
    600  *
    601  * \param self    AAC decoder handle.
    602  * \param conf    Pointer to an unsigned char buffer containing the binary configuration buffer (either ASC or SMC).
    603  * \param length  Length of the configuration buffer in bytes.
    604  * \return        Error code.
    605  */
    606 LINKSPEC_H AAC_DECODER_ERROR
    607 aacDecoder_ConfigRaw ( HANDLE_AACDECODER self,
    608                        UCHAR            *conf[],
    609                        const UINT        length[] );
    610 
    611 
    612 /**
    613  * \brief Fill AAC decoder's internal input buffer with bitstream data from the external input buffer.
    614  *  The function only copies such data as long as the decoder-internal input buffer is not full.
    615  *  So it grabs whatever it can from pBuffer and returns information (bytesValid) so that at a
    616  *  subsequent call of %aacDecoder_Fill(), the right position in pBuffer can be determined to
    617  *  grab the next data.
    618  *
    619  * \param self        AAC decoder handle.
    620  * \param pBuffer     Pointer to external input buffer.
    621  * \param bufferSize  Size of external input buffer. This argument is required because decoder-internally
    622  *                    we need the information to calculate the offset to pBuffer, where the next
    623  *                    available data is, which is then fed into the decoder-internal buffer (as much
    624  *                    as possible). Our example framework implementation fills the buffer at pBuffer
    625  *                    again, once it contains no available valid bytes anymore (meaning bytesValid equal 0).
    626  * \param bytesValid  Number of bitstream bytes in the external bitstream buffer that have not yet been
    627  *                    copied into the decoder's internal bitstream buffer by calling this function.
    628  *                    The value is updated according to the amount of newly copied bytes.
    629  * \return            Error code.
    630  */
    631 LINKSPEC_H AAC_DECODER_ERROR
    632 aacDecoder_Fill ( HANDLE_AACDECODER  self,
    633                   UCHAR             *pBuffer[],
    634                   const UINT         bufferSize[],
    635                   UINT              *bytesValid );
    636 
    637 #define AACDEC_CONCEAL  1 /*!< Flag for aacDecoder_DecodeFrame(): do not consider new input data. Do concealment. */
    638 #define AACDEC_FLUSH    2 /*!< Flag for aacDecoder_DecodeFrame(): Do not consider new input data. Flush filterbanks (output delayed audio). */
    639 #define AACDEC_INTR     4 /*!< Flag for aacDecoder_DecodeFrame(): Signal an input bit stream data discontinuity. Resync any internals as necessary. */
    640 #define AACDEC_CLRHIST  8 /*!< Flag for aacDecoder_DecodeFrame(): Clear all signal delay lines and history buffers.
    641                                Caution: This can cause discontinuities in the output signal. */
    642 
    643 /**
    644  * \brief            Decode one audio frame
    645  *
    646  * \param self       AAC decoder handle.
    647  * \param pTimeData  Pointer to external output buffer where the decoded PCM samples will be stored into.
    648  * \param flags      Bit field with flags for the decoder: \n
    649  *                   (flags & AACDEC_CONCEAL) == 1: Do concealment. \n
    650  *                   (flags & AACDEC_FLUSH) == 2: Discard input data. Flush filter banks (output delayed audio). \n
    651  *                   (flags & AACDEC_INTR) == 4: Input data is discontinuous. Resynchronize any internals as necessary.
    652  * \return           Error code.
    653  */
    654 LINKSPEC_H AAC_DECODER_ERROR
    655 aacDecoder_DecodeFrame ( HANDLE_AACDECODER  self,
    656                          INT_PCM           *pTimeData,
    657                          const INT          timeDataSize,
    658                          const UINT         flags );
    659 
    660 /**
    661  * \brief       De-allocate all resources of an AAC decoder instance.
    662  *
    663  * \param self  AAC decoder handle.
    664  * \return      void
    665  */
    666 LINKSPEC_H void aacDecoder_Close ( HANDLE_AACDECODER self );
    667 
    668 /**
    669  * \brief       Get CStreamInfo handle from decoder.
    670  *
    671  * \param self  AAC decoder handle.
    672  * \return      Reference to requested CStreamInfo.
    673  */
    674 LINKSPEC_H CStreamInfo* aacDecoder_GetStreamInfo( HANDLE_AACDECODER self );
    675 
    676 /**
    677  * \brief       Get decoder library info.
    678  *
    679  * \param info  Pointer to an allocated LIB_INFO structure.
    680  * \return      0 on success
    681  */
    682 LINKSPEC_H INT aacDecoder_GetLibInfo( LIB_INFO *info );
    683 
    684 
    685 #ifdef __cplusplus
    686 }
    687 #endif
    688 
    689 #endif /* AACDECODER_LIB_H */
    690