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      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 HE-AAC Encoder **************************
     85 
     86   Initial author:       M. Lohwasser
     87 ******************************************************************************/
     88 
     89 /**
     90  * \file   aacenc_lib.h
     91  * \brief  FDK AAC Encoder library interface header file.
     92  *
     93 \mainpage  Introduction
     94 
     95 \section Scope
     96 
     97 This document describes the high-level interface and usage of the ISO/MPEG-2/4 AAC Encoder
     98 library developed by the Fraunhofer Institute for Integrated Circuits (IIS).
     99 
    100 The library implements encoding on the basis of the MPEG-2 and MPEG-4 AAC Low-Complexity
    101 standard, and depending on the library's configuration, MPEG-4 High-Efficiency AAC v2 and/or AAC-ELD standard.
    102 
    103 All references to SBR (Spectral Band Replication) are only applicable to HE-AAC or AAC-ELD versions
    104 of the library. All references to PS (Parametric Stereo) are only applicable to HE-AAC v2
    105 versions of the library.
    106 
    107 \section encBasics Encoder Basics
    108 
    109 This document can only give a rough overview about the ISO/MPEG-2 and ISO/MPEG-4 AAC audio coding
    110 standard. To understand all the terms in this document, you are encouraged to read the following documents.
    111 
    112 - ISO/IEC 13818-7 (MPEG-2 AAC), which defines the syntax of MPEG-2 AAC audio bitstreams.
    113 - ISO/IEC 14496-3 (MPEG-4 AAC, subparts 1 and 4), which defines the syntax of MPEG-4 AAC audio bitstreams.
    114 - Lutzky, Schuller, Gayer, Krämer, Wabnik, "A guideline to audio codec delay", 116th AES Convention, May 8, 2004
    115 
    116 MPEG Advanced Audio Coding is based on a time-to-frequency mapping of the signal. The signal is
    117 partitioned into overlapping portions and transformed into frequency domain. The spectral components
    118 are then quantized and coded. \n
    119 An MPEG-2 or MPEG-4 AAC audio bitstream is composed of frames. Contrary to MPEG-1/2 Layer-3 (mp3), the
    120 length of individual frames is not restricted to a fixed number of bytes, but can take on any length
    121 between 1 and 768 bytes.
    122 
    123 
    124 \page LIBUSE Library Usage
    125 
    126 \section InterfaceDescription API Files
    127 
    128 All API header files are located in the folder /include of the release package. All header files
    129 are provided for usage in C/C++ programs. The AAC encoder library API functions are located at
    130 aacenc_lib.h.
    131 
    132 In binary releases the encoder core resides in statically linkable libraries called for example
    133 libAACenc.a/libFDK.a (LINUX) or FDK_fastaaclib.lib (MS Visual C++) for the plain AAC-LC core encoder
    134 and libSBRenc.a (LINUX) or FDK_sbrEncLib.lib (MS Visual C++) for the SBR (Spectral Band
    135 Replication) and PS (Parametric Stereo) modules.
    136 
    137 \section CallingSequence Calling Sequence
    138 
    139 For encoding of ISO/MPEG-2/4 AAC bitstreams the following sequence is mandatory. Input read and output
    140 write functions as well as the corresponding open and close functions are left out, since they may be
    141 implemented differently according to the user's specific requirements. The example implementation in
    142 main.cpp uses file-based input/output.
    143 
    144 -# Call aacEncOpen() to allocate encoder instance with required \ref encOpen "configuration".\n
    145 \dontinclude main.cpp
    146 \skipline hAacEncoder =
    147 \skipline aacEncOpen
    148 -# Call aacEncoder_SetParam() for each parameter to be set. AOT, samplingrate, channelMode, bitrate and transport type are \ref encParams "mandatory".
    149 \code
    150     ErrorStatus = aacEncoder_SetParam(hAacEncoder, parameter, value);
    151 \endcode
    152 -# Call aacEncEncode() with NULL parameters to \ref encReconf "initialize" encoder instance with present parameter set.
    153 \skipline aacEncEncode
    154 -# Call aacEncInfo() to retrieve a configuration data block to be transmitted out of band. This is required when using RFC3640 or RFC3016 like transport.
    155 \dontinclude main.cpp
    156 \skipline encInfo
    157 \skipline aacEncInfo
    158 -# Encode input audio data in loop.
    159 \skip Encode as long as
    160 \skipline do
    161 \until {
    162 Feed \ref feedInBuf "input buffer" with new audio data and provide input/output \ref bufDes "arguments" to aacEncEncode().
    163 \skipline aacEncEncode
    164 \until ;
    165 Write \ref writeOutData "output data" to file or audio device. \skipline while
    166 -# Call aacEncClose() and destroy encoder instance.
    167 \skipline aacEncClose
    168 
    169 \section encOpen Encoder Instance Allocation
    170 
    171 The assignment of the aacEncOpen() function is very flexible and can be used in the following way.
    172 - If the amount of memory consumption is not an issue, the encoder instance can be allocated
    173 for the maximum number of possible audio channels (for example 6 or 8) with the full functional range supported by the library.
    174 This is the default open procedure for the AAC encoder if memory consumption does not need to be minimized.
    175 \code aacEncOpen(&hAacEncoder,0,0) \endcode
    176 - If the required MPEG-4 AOTs do not call for the full functional range of the library, encoder modules can be allocated selectively.
    177 \verbatim
    178 ------------------------------------------------------
    179  AAC | SBR |  PS | MD |         FLAGS         | value
    180 -----+-----+-----+----+-----------------------+-------
    181   X  |  -  |  -  |  - | (0x01)                |  0x01
    182   X  |  X  |  -  |  - | (0x01|0x02)           |  0x03
    183   X  |  X  |  X  |  - | (0x01|0x02|0x04)      |  0x07
    184   X  |  -  |  -  |  X | (0x01          |0x10) |  0x11
    185   X  |  X  |  -  |  X | (0x01|0x02     |0x10) |  0x13
    186   X  |  X  |  X  |  X | (0x01|0x02|0x04|0x10) |  0x17
    187 ------------------------------------------------------
    188  - AAC: Allocate AAC Core Encoder module.
    189  - SBR: Allocate Spectral Band Replication module.
    190  - PS: Allocate Parametric Stereo module.
    191  - MD: Allocate Meta Data module within AAC encoder.
    192 \endverbatim
    193 \code aacEncOpen(&hAacEncoder,value,0) \endcode
    194 - Specifying the maximum number of channels to be supported in the encoder instance can be done as follows.
    195  - For example allocate an encoder instance which supports 2 channels for all supported AOTs.
    196    The library itself may be capable of encoding up to 6 or 8 channels but in this example only 2 channel encoding is required and thus only buffers for 2 channels are allocated to save data memory.
    197 \code aacEncOpen(&hAacEncoder,0,2) \endcode
    198  - Additionally the maximum number of supported channels in the SBR module can be denoted separately.\n
    199    In this example the encoder instance provides a maximum of 6 channels out of which up to 2 channels support SBR.
    200    This encoder instance can produce for example 5.1 channel AAC-LC streams or stereo HE-AAC (v2) streams.
    201    HE-AAC 5.1 multi channel is not possible since only 2 out of 6 channels support SBR, which saves data memory.
    202 \code aacEncOpen(&hAacEncoder,0,6|(2<<8)) \endcode
    203 \n
    204 
    205 \section bufDes Input/Output Arguments
    206 
    207 \subsection allocIOBufs Provide Buffer Descriptors
    208 In the present encoder API, the input and output buffers are described with \ref AACENC_BufDesc "buffer descriptors". This mechanism allows a flexible handling
    209 of input and output buffers without impact to the actual encoding call. Optional buffers are necessary e.g. for ancillary data, meta data input or additional output
    210 buffers describing superframing data in DAB+ or DRM+.\n
    211 At least one input buffer for audio input data and one output buffer for bitstream data must be allocated. The input buffer size can be a user defined multiple
    212 of the number of input channels. PCM input data will be copied from the user defined PCM buffer to an internal input buffer and so input data can be less than one AAC audio frame.
    213 The output buffer size should be 6144 bits per channel excluding the LFE channel.
    214 If the output data does not fit into the provided buffer, an AACENC_ERROR will be returned by aacEncEncode().
    215 \dontinclude main.cpp
    216 \skipline inputBuffer
    217 \until outputBuffer
    218 All input and output buffer must be clustered in input and output buffer arrays.
    219 \skipline inBuffer
    220 \until outBufferElSize
    221 Allocate buffer descriptors
    222 \skipline AACENC_BufDesc
    223 \skipline AACENC_BufDesc
    224 Initialize input buffer descriptor
    225 \skipline inBufDesc
    226 \until bufElSizes
    227 Initialize output buffer descriptor
    228 \skipline outBufDesc
    229 \until bufElSizes
    230 
    231 \subsection argLists Provide Input/Output Argument Lists
    232 The input and output arguments of an aacEncEncode() call are described in argument structures.
    233 \dontinclude main.cpp
    234 \skipline AACENC_InArgs
    235 \skipline AACENC_OutArgs
    236 
    237 \section feedInBuf Feed Input Buffer
    238 The input buffer should be handled as a modulo buffer. New audio data in the form of pulse-code-
    239 modulated samples (PCM) must be read from external and be fed to the input buffer depending on its
    240 fill level. The required sample bitrate (represented by the data type INT_PCM which is 16, 24 or 32
    241 bits wide) is fixed and depends on library configuration (usually 16 bit).
    242 
    243 \dontinclude main.cpp
    244 \skipline WAV_InputRead
    245 \until ;
    246 After the encoder's internal buffer is fed with incoming audio samples, and aacEncEncode()
    247 processed the new input data, update/move remaining samples in input buffer, simulating a modulo buffer:
    248 \skipline outargs.numInSamples>0
    249 \until }
    250 
    251 \section writeOutData Output Bitstream Data
    252 If any AAC bitstream data is available, write it to output file or device. This can be done once the
    253 following condition is true:
    254 \dontinclude main.cpp
    255 \skip Valid bitstream available
    256 \skipline outargs
    257 
    258 \skipline outBytes>0
    259 
    260 If you use file I/O then for example call mpegFileWrite_Write() from the library libMpegFileWrite
    261 
    262 \dontinclude main.cpp
    263 \skipline mpegFileWrite_Write
    264 
    265 \section cfgMetaData Meta Data Configuration
    266 
    267 If the present library is configured with Metadata support, it is possible to insert meta data side info into the generated
    268 audio bitstream while encoding.
    269 
    270 To work with meta data the encoder instance has to be \ref encOpen "allocated" with meta data support. The meta data mode must be be configured with
    271 the ::AACENC_METADATA_MODE parameter and aacEncoder_SetParam() function.
    272 \code aacEncoder_SetParam(hAacEncoder, AACENC_METADATA_MODE, 0-2); \endcode
    273 
    274 This configuration indicates how to embed meta data into bitstrem. Either no insertion, MPEG or ETSI style.
    275 The meta data itself must be specified within the meta data setup structure AACENC_MetaData.
    276 
    277 Changing one of the AACENC_MetaData setup parameters can be achieved from outside the library within ::IN_METADATA_SETUP input
    278 buffer. There is no need to supply meta data setup structure every frame. If there is no new meta setup data available, the
    279 encoder uses the previous setup or the default configuration in initial state.
    280 
    281 In general the audio compressor and limiter within the encoder library can be configured with the ::AACENC_METADATA_DRC_PROFILE parameter
    282 AACENC_MetaData::drc_profile and and AACENC_MetaData::comp_profile.
    283 \n
    284 
    285 \section encReconf Encoder Reconfiguration
    286 
    287 The encoder library allows reconfiguration of the encoder instance with new settings
    288 continuously between encoding frames. Each parameter to be changed must be set with
    289 a single aacEncoder_SetParam() call. The internal status of each parameter can be
    290 retrieved with an aacEncoder_GetParam() call.\n
    291 There is no stand-alone reconfiguration function available. When parameters were
    292 modified from outside the library, an internal control mechanism triggers the necessary
    293 reconfiguration process which will be applied at the beginning of the following
    294 aacEncEncode() call. This state can be observed from external via the AACENC_INIT_STATUS
    295 and aacEncoder_GetParam() function. The reconfiguration process can also be applied
    296 immediately when all parameters of an aacEncEncode() call are NULL with a valid encoder
    297 handle.\n\n
    298 The internal reconfiguration process can be controlled from extern with the following access.
    299 \code aacEncoder_SetParam(hAacEncoder, AACENC_CONTROL_STATE, AACENC_CTRLFLAGS); \endcode
    300 
    301 
    302 \section encParams Encoder Parametrization
    303 
    304 All parameteres listed in ::AACENC_PARAM can be modified within an encoder instance.
    305 
    306 \subsection encMandatory Mandatory Encoder Parameters
    307 The following parameters must be specified when the encoder instance is initialized.
    308 \code
    309 aacEncoder_SetParam(hAacEncoder, AACENC_AOT, value);
    310 aacEncoder_SetParam(hAacEncoder, AACENC_BITRATE, value);
    311 aacEncoder_SetParam(hAacEncoder, AACENC_SAMPLERATE, value);
    312 aacEncoder_SetParam(hAacEncoder, AACENC_CHANNELMODE, value);
    313 \endcode
    314 Beyond that is an internal auto mode which preinitizializes the ::AACENC_BITRATE parameter
    315 if the parameter was not set from extern. The bitrate depends on the number of effective
    316 channels and sampling rate and is determined as follows.
    317 \code
    318 AAC-LC (AOT_AAC_LC): 1.5 bits per sample
    319 HE-AAC (AOT_SBR): 0.625 bits per sample (dualrate sbr)
    320 HE-AAC (AOT_SBR): 1.125 bits per sample (downsampled sbr)
    321 HE-AAC v2 (AOT_PS): 0.5 bits per sample
    322 \endcode
    323 
    324 \subsection channelMode Channel Mode Configuration
    325 The input audio data is described with the ::AACENC_CHANNELMODE parameter in the
    326 aacEncoder_SetParam() call. It is not possible to use the encoder instance with a 'number of
    327 input channels' argument. Instead, the channelMode must be set as follows.
    328 \code aacEncoder_SetParam(hAacEncoder, AACENC_CHANNELMODE, value); \endcode
    329 The parameter is specified in ::CHANNEL_MODE and can be mapped from the number of input channels
    330 in the following way.
    331 \dontinclude main.cpp
    332 \skip CHANNEL_MODE chMode = MODE_INVALID;
    333 \until return
    334 
    335 \subsection encQual Audio Quality Considerations
    336 The default encoder configuration is suggested to be used. Encoder tools such as TNS and PNS
    337 are activated by default and are internally controlled (see \ref BEHAVIOUR_TOOLS).
    338 
    339 There is an additional quality parameter called ::AACENC_AFTERBURNER. In the default
    340 configuration this quality switch is deactivated because it would cause a workload
    341 increase which might be significant. If workload is not an issue in the application
    342 we recommended to activate this feature.
    343 \code aacEncoder_SetParam(hAacEncoder, AACENC_AFTERBURNER, 1); \endcode
    344 
    345 \subsection encELD ELD Auto Configuration Mode
    346 For ELD configuration a so called auto configurator is available which configures SBR and the SBR ratio by itself.
    347 The configurator is used when the encoder parameter ::AACENC_SBR_MODE and ::AACENC_SBR_RATIO are not set explicitely.
    348 
    349 Based on sampling rate and chosen bitrate per channel a reasonable SBR configuration will be used.
    350 \verbatim
    351 ------------------------------------------------------------
    352   Sampling Rate  | Channel Bitrate |  SBR |       SBR Ratio
    353 -----------------+-----------------+------+-----------------
    354  ]min, 16] kHz   |     min - 27999 |   on | downsampled SBR
    355                  |   28000 -   max |  off |             ---
    356 -----------------+-----------------+------+-----------------
    357  ]16 - 24] kHz   |     min - 39999 |   on | downsampled SBR
    358                  |   40000 -   max |  off |             ---
    359 -----------------+-----------------+------+-----------------
    360  ]24 - 32] kHz   |     min - 27999 |   on |    dualrate SBR
    361                  |   28000 - 55999 |   on | downsampled SBR
    362                  |   56000 -   max |  off |             ---
    363 -----------------+-----------------+------+-----------------
    364  ]32 - 44.1] kHz |     min - 63999 |   on |    dualrate SBR
    365                  |   64000 -   max |  off |             ---
    366 -----------------+-----------------+------+-----------------
    367  ]44.1 - 48] kHz |     min - 63999 |   on |    dualrate SBR
    368                  |   64000 - max   |  off |             ---
    369 ------------------------------------------------------------
    370 \endverbatim
    371 
    372 
    373 \section audiochCfg Audio Channel Configuration
    374 The MPEG standard refers often to the so-called Channel Configuration. This Channel Configuration is used for a fixed Channel
    375 Mapping. The configurations 1-7 are predefined in MPEG standard and used for implicit signalling within the encoded bitstream.
    376 For user defined Configurations the Channel Configuration is set to 0 and the Channel Mapping must be explecitly described with an appropriate
    377 Program Config Element. The present Encoder implementation does not allow the user to configure this Channel Configuration from
    378 extern. The Encoder implementation supports fixed Channel Modes which are mapped to Channel Configuration as follow.
    379 \verbatim
    380 -------------------------------------------------------------------------------
    381  ChannelMode           | ChCfg  | front_El      | side_El  | back_El  | lfe_El
    382 -----------------------+--------+---------------+----------+----------+--------
    383 MODE_1                 |      1 | SCE           |          |          |
    384 MODE_2                 |      2 | CPE           |          |          |
    385 MODE_1_2               |      3 | SCE, CPE      |          |          |
    386 MODE_1_2_1             |      4 | SCE, CPE      |          | SCE      |
    387 MODE_1_2_2             |      5 | SCE, CPE      |          | CPE      |
    388 MODE_1_2_2_1           |      6 | SCE, CPE      |          | CPE      | LFE
    389 MODE_1_2_2_2_1         |      7 | SCE, CPE, CPE |          | CPE      | LFE
    390 -----------------------+--------+---------------+----------+----------+--------
    391 MODE_7_1_REAR_SURROUND |      0 | SCE, CPE      |          | CPE, CPE | LFE
    392 MODE_7_1_FRONT_CENTER  |      0 | SCE, CPE, CPE |          | CPE      | LFE
    393 -------------------------------------------------------------------------------
    394  - SCE: Single Channel Element.
    395  - CPE: Channel Pair.
    396  - SCE: Low Frequency Element.
    397 \endverbatim
    398 
    399 Moreover, the Table describes all fixed Channel Elements for each Channel Mode which are assigned to a speaker arrangement. The
    400 arrangement includes front, side, back and lfe Audio Channel Elements.\n
    401 This mapping of Audio Channel Elements is defined in MPEG standard for Channel Config 1-7. The Channel assignment for MODE_1_1,
    402 MODE_2_2 and MODE_2_1 is used from the ARIB standard. All other configurations are defined as suggested in MPEG.\n
    403 In case of Channel Config 0 or writing matrix mixdown coefficients, the encoder enables the writing of Program Config Element
    404 itself as described in \ref encPCE. The configuration used in Program Config Element refers to the denoted Table.\n
    405 Beside the Channel Element assignment the Channel Modes are resposible for audio input data channel mapping. The Channel Mapping
    406 of the audio data depends on the selected ::AACENC_CHANNELORDER which can be MPEG or WAV like order.\n
    407 Following Table describes the complete channel mapping for both Channel Order configurations.
    408 \verbatim
    409 ---------------------------------------------------------------------------------------
    410 ChannelMode            |  MPEG-Channelorder            |  WAV-Channelorder
    411 -----------------------+---+---+---+---+---+---+---+---+---+---+---+---+---+---+---+---
    412 MODE_1                 | 0 |   |   |   |   |   |   |   | 0 |   |   |   |   |   |   |
    413 MODE_2                 | 0 | 1 |   |   |   |   |   |   | 0 | 1 |   |   |   |   |   |
    414 MODE_1_2               | 0 | 1 | 2 |   |   |   |   |   | 2 | 0 | 1 |   |   |   |   |
    415 MODE_1_2_1             | 0 | 1 | 2 | 3 |   |   |   |   | 2 | 0 | 1 | 3 |   |   |   |
    416 MODE_1_2_2             | 0 | 1 | 2 | 3 | 4 |   |   |   | 2 | 0 | 1 | 3 | 4 |   |   |
    417 MODE_1_2_2_1           | 0 | 1 | 2 | 3 | 4 | 5 |   |   | 2 | 0 | 1 | 4 | 5 | 3 |   |
    418 MODE_1_2_2_2_1         | 0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 | 2 | 6 | 7 | 0 | 1 | 4 | 5 | 3
    419 -----------------------+---+---+---+---+---+---+---+---+---+---+---+---+---+---+---+---
    420 MODE_7_1_REAR_SURROUND | 0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 | 2 | 0 | 1 | 6 | 7 | 4 | 5 | 3
    421 MODE_7_1_FRONT_CENTER  | 0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 | 2 | 6 | 7 | 0 | 1 | 4 | 5 | 3
    422 ---------------------------------------------------------------------------------------
    423 \endverbatim
    424 
    425 The denoted mapping is important for correct audio channel assignment when using MPEG or WAV ordering. The incoming audio
    426 channels are distributed MPEG like starting at the front channels and ending at the back channels. The distribution is used as
    427 described in Table concering Channel Config and fix channel elements. Please see the following example for clarification.
    428 
    429 \verbatim
    430 Example: MODE_1_2_2_1 - WAV-Channelorder 5.1
    431 ------------------------------------------
    432  Input Channel      | Coder Channel
    433 --------------------+---------------------
    434  2 (front center)   | 0 (SCE channel)
    435  0 (left center)    | 1 (1st of 1st CPE)
    436  1 (right center)   | 2 (2nd of 1st CPE)
    437  4 (left surround)  | 3 (1st of 2nd CPE)
    438  5 (right surround) | 4 (2nd of 2nd CPE)
    439  3 (LFE)            | 5 (LFE)
    440 ------------------------------------------
    441 \endverbatim
    442 
    443 
    444 \section suppBitrates Supported Bitrates
    445 
    446 The FDK AAC Encoder provides a wide range of supported bitrates.
    447 The minimum and maximum allowed bitrate depends on the Audio Object Type. For AAC-LC the minimum
    448 bitrate is the bitrate that is required to write the most basic and minimal valid bitstream.
    449 It consists of the bitstream format header information and other static/mandatory information
    450 within the AAC payload. The maximum AAC framesize allowed by the MPEG-4 standard
    451 determines the maximum allowed bitrate for AAC-LC. For HE-AAC and HE-AAC v2 a library internal
    452 look-up table is used.
    453 
    454 A good working point in terms of audio quality, sampling rate and bitrate, is at 1 to 1.5
    455 bits/audio sample for AAC-LC, 0.625 bits/audio sample for dualrate HE-AAC, 1.125 bits/audio sample
    456 for downsampled HE-AAC and 0.5 bits/audio sample for HE-AAC v2.
    457 For example for one channel with a sampling frequency of 48 kHz, the range from
    458 48 kbit/s to 72 kbit/s achieves reasonable audio quality for AAC-LC.
    459 
    460 For HE-AAC and HE-AAC v2 the lowest possible audio input sampling frequency is 16 kHz because then the
    461 AAC-LC core encoder operates in dual rate mode at its lowest possible sampling frequency, which is 8 kHz.
    462 HE-AAC v2 requires stereo input audio data.
    463 
    464 Please note that in HE-AAC or HE-AAC v2 mode the encoder supports much higher bitrates than are
    465 appropriate for HE-AAC or HE-AAC v2. For example, at a bitrate of more than 64 kbit/s for a stereo
    466 audio signal at 44.1 kHz it usually makes sense to use AAC-LC, which will produce better audio
    467 quality at that bitrate than HE-AAC or HE-AAC v2.
    468 
    469 \section reommendedConfig Recommended Sampling Rate and Bitrate Combinations
    470 
    471 The following table provides an overview of recommended encoder configuration parameters
    472 which we determined by virtue of numerous listening tests.
    473 
    474 \subsection reommendedConfigLC AAC-LC, HE-AAC, HE-AACv2 in Dualrate SBR mode.
    475 \verbatim
    476 -----------------------------------------------------------------------------------
    477 Audio Object Type  |  Bit Rate Range  |            Supported  | Preferred  | No. of
    478                    |         [bit/s]  |       Sampling Rates  |    Sampl.  |  Chan.
    479                    |                  |                [kHz]  |      Rate  |
    480                    |                  |                       |     [kHz]  |
    481 -------------------+------------------+-----------------------+------------+-------
    482 AAC LC + SBR + PS  |   8000 -  11999  |         22.05, 24.00  |     24.00  |      2
    483 AAC LC + SBR + PS  |  12000 -  17999  |                32.00  |     32.00  |      2
    484 AAC LC + SBR + PS  |  18000 -  39999  |  32.00, 44.10, 48.00  |     44.10  |      2
    485 AAC LC + SBR + PS  |  40000 -  56000  |  32.00, 44.10, 48.00  |     48.00  |      2
    486 -------------------+------------------+-----------------------+------------+-------
    487 AAC LC + SBR       |   8000 -  11999  |         22.05, 24.00  |     24.00  |      1
    488 AAC LC + SBR       |  12000 -  17999  |                32.00  |     32.00  |      1
    489 AAC LC + SBR       |  18000 -  39999  |  32.00, 44.10, 48.00  |     44.10  |      1
    490 AAC LC + SBR       |  40000 -  56000  |  32.00, 44.10, 48.00  |     48.00  |      1
    491 AAC LC + SBR       |  16000 -  27999  |  32.00, 44.10, 48.00  |     32.00  |      2
    492 AAC LC + SBR       |  28000 -  63999  |  32.00, 44.10, 48.00  |     44.10  |      2
    493 AAC LC + SBR       |  64000 - 128000  |  32.00, 44.10, 48.00  |     48.00  |      2
    494 -------------------+------------------+-----------------------+------------+-------
    495 AAC LC + SBR       |  64000 -  69999  |  32.00, 44.10, 48.00  |     32.00  | 5, 5.1
    496 AAC LC + SBR       |  70000 - 159999  |  32.00, 44.10, 48.00  |     44.10  | 5, 5.1
    497 AAC LC + SBR       | 160000 - 245999  |  32.00, 44.10, 48.00  |     48.00  |      5
    498 AAC LC + SBR       | 160000 - 265999  |  32.00, 44.10, 48.00  |     48.00  |    5.1
    499 -------------------+------------------+-----------------------+------------+-------
    500 AAC LC             |   8000 -  15999  | 11.025, 12.00, 16.00  |     12.00  |      1
    501 AAC LC             |  16000 -  23999  |                16.00  |     16.00  |      1
    502 AAC LC             |  24000 -  31999  |  16.00, 22.05, 24.00  |     24.00  |      1
    503 AAC LC             |  32000 -  55999  |                32.00  |     32.00  |      1
    504 AAC LC             |  56000 - 160000  |  32.00, 44.10, 48.00  |     44.10  |      1
    505 AAC LC             | 160001 - 288000  |                48.00  |     48.00  |      1
    506 -------------------+------------------+-----------------------+------------+-------
    507 AAC LC             |  16000 -  23999  | 11.025, 12.00, 16.00  |     12.00  |      2
    508 AAC LC             |  24000 -  31999  |                16.00  |     16.00  |      2
    509 AAC LC             |  32000 -  39999  |  16.00, 22.05, 24.00  |     22.05  |      2
    510 AAC LC             |  40000 -  95999  |                32.00  |     32.00  |      2
    511 AAC LC             |  96000 - 111999  |  32.00, 44.10, 48.00  |     32.00  |      2
    512 AAC LC             | 112000 - 320001  |  32.00, 44.10, 48.00  |     44.10  |      2
    513 AAC LC             | 320002 - 576000  |                48.00  |     48.00  |      2
    514 -------------------+------------------+-----------------------+------------+-------
    515 AAC LC             | 160000 - 239999  |                32.00  |     32.00  | 5, 5.1
    516 AAC LC             | 240000 - 279999  |  32.00, 44.10, 48.00  |     32.00  | 5, 5.1
    517 AAC LC             | 280000 - 800000  |  32.00, 44.10, 48.00  |     44.10  | 5, 5.1
    518 -----------------------------------------------------------------------------------
    519 \endverbatim \n
    520 
    521 \subsection reommendedConfigLD AAC-LD, AAC-ELD, AAC-ELD with SBR in Dualrate SBR mode.
    522 \verbatim
    523 -----------------------------------------------------------------------------------
    524 Audio Object Type  |  Bit Rate Range  |            Supported  | Preferred  | No. of
    525                    |         [bit/s]  |       Sampling Rates  |    Sampl.  |  Chan.
    526                    |                  |                [kHz]  |      Rate  |
    527                    |                  |                       |     [kHz]  |
    528 -------------------+------------------+-----------------------+------------+-------
    529 ELD + SBR          |  18000 -  24999  |        32.00 - 44.10  |     32.00  |      1
    530 ELD + SBR          |  25000 -  31999  |        32.00 - 48.00  |     32.00  |      1
    531 ELD + SBR          |  32000 -  64000  |        32.00 - 48.00  |     48.00  |      1
    532 -------------------+------------------+-----------------------+------------+-------
    533 ELD + SBR          |  32000 -  51999  |        32.00 - 48.00  |     44.10  |      2
    534 ELD + SBR          |  52000 - 128000  |        32.00 - 48.00  |     48.00  |      2
    535 -------------------+------------------+-----------------------+------------+-------
    536 ELD + SBR          |  72000 - 160000  |        44.10 - 48.00  |     48.00  |      3
    537 -------------------+------------------+-----------------------+------------+-------
    538 ELD + SBR          |  96000 - 212000  |        44.10 - 48.00  |     48.00  |      4
    539 -------------------+------------------+-----------------------+------------+-------
    540 ELD + SBR          | 120000 - 246000  |        44.10 - 48.00  |     48.00  |      5
    541 -------------------+------------------+-----------------------+------------+-------
    542 ELD + SBR          | 120000 - 266000  |        44.10 - 48.00  |     48.00  |    5.1
    543 -------------------+------------------+-----------------------+------------+-------
    544 LD, ELD            |  16000 -  19999  |        16.00 - 24.00  |     16.00  |      1
    545 LD, ELD            |  20000 -  39999  |        16.00 - 32.00  |     24.00  |      1
    546 LD, ELD            |  40000 -  49999  |        22.05 - 32.00  |     32.00  |      1
    547 LD, ELD            |  50000 -  61999  |        24.00 - 44.10  |     32.00  |      1
    548 LD, ELD            |  62000 -  84999  |        32.00 - 48.00  |     44.10  |      1
    549 LD, ELD            |  85000 - 192000  |        44.10 - 48.00  |     48.00  |      1
    550 -------------------+------------------+-----------------------+------------+-------
    551 LD, ELD            |  64000 -  75999  |        24.00 - 32.00  |     32.00  |      2
    552 LD, ELD            |  76000 -  97999  |        24.00 - 44.10  |     32.00  |      2
    553 LD, ELD            |  98000 - 135999  |        32.00 - 48.00  |     44.10  |      2
    554 LD, ELD            | 136000 - 384000  |        44.10 - 48.00  |     48.00  |      2
    555 -------------------+------------------+-----------------------+------------+-------
    556 LD, ELD            |  96000 - 113999  |        24.00 - 32.00  |     32.00  |      3
    557 LD, ELD            | 114000 - 146999  |        24.00 - 44.10  |     32.00  |      3
    558 LD, ELD            | 147000 - 203999  |        32.00 - 48.00  |     44.10  |      3
    559 LD, ELD            | 204000 - 576000  |        44.10 - 48.00  |     48.00  |      3
    560 -------------------+------------------+-----------------------+------------+-------
    561 LD, ELD            | 128000 - 151999  |        24.00 - 32.00  |     32.00  |      4
    562 LD, ELD            | 152000 - 195999  |        24.00 - 44.10  |     32.00  |      4
    563 LD, ELD            | 196000 - 271999  |        32.00 - 48.00  |     44.10  |      4
    564 LD, ELD            | 272000 - 768000  |        44.10 - 48.00  |     48.00  |      4
    565 -------------------+------------------+-----------------------+------------+-------
    566 LD, ELD            | 160000 - 189999  |        24.00 - 32.00  |     32.00  |      5
    567 LD, ELD            | 190000 - 244999  |        24.00 - 44.10  |     32.00  |      5
    568 LD, ELD            | 245000 - 339999  |        32.00 - 48.00  |     44.10  |      5
    569 LD, ELD            | 340000 - 960000  |        44.10 - 48.00  |     48.00  |      5
    570 -----------------------------------------------------------------------------------
    571 \endverbatim \n
    572 
    573 \subsection reommendedConfigELD AAC-ELD with SBR in Downsampled SBR mode.
    574 \verbatim
    575 -----------------------------------------------------------------------------------
    576 Audio Object Type  |  Bit Rate Range  |            Supported  | Preferred  | No. of
    577                    |         [bit/s]  |       Sampling Rates  |    Sampl.  |  Chan.
    578                    |                  |                [kHz]  |      Rate  |
    579                    |                  |                       |     [kHz]  |
    580 -------------------+------------------+-----------------------+------------+-------
    581 ELD + SBR          |  18000 -  24999  |        16.00 - 22.05  |     22.05  |      1
    582 (downsampled SBR)  |  25000 -  35999  |        22.05 - 32.00  |     24.00  |      1
    583                    |  36000 -  64000  |        32.00 - 48.00  |     32.00  |      1
    584 -----------------------------------------------------------------------------------
    585 \endverbatim \n
    586 
    587 
    588 \page ENCODERBEHAVIOUR Encoder Behaviour
    589 
    590 \section BEHAVIOUR_BANDWIDTH Bandwidth
    591 
    592 The FDK AAC encoder usually does not use the full frequency range of the input signal, but restricts the bandwidth
    593 according to certain library-internal settings. They can be changed in the table "bandWidthTable" in the
    594 file bandwidth.cpp (if available).
    595 
    596 The encoder API provides the ::AACENC_BANDWIDTH parameter to adjust the bandwidth explicitly.
    597 \code
    598 aacEncoder_SetParam(hAacEncoder, AACENC_BANDWIDTH, value);
    599 \endcode
    600 
    601 However it is not recommended to change these settings, because they are based on numerious listening
    602 tests and careful tweaks to ensure the best overall encoding quality.
    603 
    604 Theoretically a signal of for example 48 kHz can contain frequencies up to 24 kHz, but to use this full range
    605 in an audio encoder usually does not make sense. Usually the encoder has a very limited amount of
    606 bits to spend (typically 128 kbit/s for stereo 48 kHz content) and to allow full range bandwidth would
    607 waste a lot of these bits for frequencies the human ear is hardly able to perceive anyway, if at all. Hence it
    608 is wise to use the available bits for the really important frequency range and just skip the rest.
    609 At lower bitrates (e. g. <= 80 kbit/s for stereo 48 kHz content) the encoder will choose an even smaller
    610 bandwidth, because an encoded signal with smaller bandwidth and hence less artifacts sounds better than a signal
    611 with higher bandwidth but then more coding artefacts across all frequencies. These artefacts would occur if
    612 small bitrates and high bandwidths are chosen because the available bits are just not enough to encode all
    613 frequencies well.
    614 
    615 Unfortunately some people evaluate encoding quality based on possible bandwidth as well, but it is a two-sided
    616 sword considering the trade-off described above.
    617 
    618 Another aspect is workload consumption. The higher the allowed bandwidth, the more frequency lines have to be
    619 processed, which in turn increases the workload.
    620 
    621 \section FRAMESIZES_AND_BIT_RESERVOIR Frame Sizes & Bit Reservoir
    622 
    623 For AAC there is a difference between constant bit rate and constant frame
    624 length due to the so-called bit reservoir technique, which allows the encoder to use less
    625 bits in an AAC frame for those audio signal sections which are easy to encode,
    626 and then spend them at a later point in
    627 time for more complex audio sections. The extent to which this "bit exchange"
    628 is done is limited to allow for reliable and relatively low delay real time
    629 streaming.
    630 Over a longer period in time the bitrate will be constant in the AAC constant
    631 bitrate mode, e.g. for ISDN transmission. This means that in AAC each bitstream
    632 frame will in general have a different length in bytes but over time it
    633 will reach the target bitrate. One could also make an MPEG compliant
    634 AAC encoder which always produces constant length packages for each AAC frame,
    635 but the audio quality would be considerably worse since the bit reservoir
    636 technique would have to be switched off completely. A higher bit rate would have
    637 to be used to get the same audio quality as with an enabled bit reservoir.
    638 
    639 The maximum AAC frame length, regardless of the available bit reservoir, is defined
    640 as 6144 bits per channel.
    641 
    642 For mp3 by the way, the same bit reservoir technique exists, but there each bit
    643 stream frame has a constant length for a given bit rate (ignoring the
    644 padding byte). In mp3 there is a so-called "back pointer" which tells
    645 the decoder which bits belong to the current mp3 frame - and in general some or
    646 many bits have been transmitted in an earlier mp3 frame. Basically this leads to
    647 the same "bit exchange between mp3 frames" as in AAC but with virtually constant
    648 length frames.
    649 
    650 This variable frame length at "constant bit rate" is not something special
    651 in this Fraunhofer IIS AAC encoder. AAC has been designed in that way.
    652 
    653 \subsection BEHAVIOUR_ESTIM_AVG_FRAMESIZES Estimating Average Frame Sizes
    654 
    655 A HE-AAC v1 or v2 audio frame contains 2048 PCM samples per channel (there is
    656 also one mode with 1920 samples per channel but this is only for special purposes
    657 such as DAB+ digital radio).
    658 
    659 The number of HE-AAC frames \f$N\_FRAMES\f$ per second at 44.1 kHz is:
    660 
    661 \f[
    662 N\_FRAMES = 44100 / 2048 = 21.5332
    663 \f]
    664 
    665 At a bit rate of 8 kbps the average number of bits per frame \f$N\_BITS\_PER\_FRAME\f$ is:
    666 
    667 \f[
    668 N\_BITS\_PER\_FRAME = 8000 / 21.5332 = 371.52
    669 \f]
    670 
    671 which is about 46.44 bytes per encoded frame.
    672 
    673 At a bit rate of 32 kbps, which is quite high for single channel HE-AAC v1, it is:
    674 
    675 \f[
    676 N\_BITS\_PER\_FRAME = 32000 / 21.5332 = 1486
    677 \f]
    678 
    679 which is about 185.76 bytes per encoded frame.
    680 
    681 These bits/frame figures are average figures where each AAC frame generally has a different
    682 size in bytes. To calculate the same for AAC-LC just use 1024 instead of 2048 PCM samples per
    683 frame and channel.
    684 For AAC-LD/ELD it is either 480 or 512 PCM samples per frame and channel.
    685 
    686 
    687 \section BEHAVIOUR_TOOLS Encoder Tools
    688 
    689 The AAC encoder supports TNS, PNS, MS, Intensity and activates these tools depending on the audio signal and
    690 the encoder configuration (i.e. bitrate or AOT). It is not required to configure these tools manually.
    691 
    692 PNS improves encoding quality only for certain bitrates. Therefore it makes sense to activate PNS only for
    693 these bitrates and save the processing power required for PNS (about 10 % of the encoder) when using other
    694 bitrates. This is done automatically inside the encoder library. PNS is disabled inside the encoder library if
    695 an MPEG-2 AOT is choosen since PNS is an MPEG-4 AAC feature.
    696 
    697 If SBR is activated, the encoder automatically deactivates PNS internally. If TNS is disabled but PNS is allowed,
    698 the encoder deactivates PNS calculation internally.
    699 
    700 */
    701 
    702 #ifndef _AAC_ENC_LIB_H_
    703 #define _AAC_ENC_LIB_H_
    704 
    705 #include "machine_type.h"
    706 #include "FDK_audio.h"
    707 
    708 
    709 /**
    710  *  AAC encoder error codes.
    711  */
    712 typedef enum {
    713     AACENC_OK                     = 0x0000,  /*!< No error happened. All fine. */
    714 
    715     AACENC_INVALID_HANDLE         = 0x0020,  /*!< Handle passed to function call was invalid. */
    716     AACENC_MEMORY_ERROR           = 0x0021,  /*!< Memory allocation failed. */
    717     AACENC_UNSUPPORTED_PARAMETER  = 0x0022,  /*!< Parameter not available. */
    718     AACENC_INVALID_CONFIG         = 0x0023,  /*!< Configuration not provided. */
    719 
    720     AACENC_INIT_ERROR             = 0x0040,  /*!< General initialization error. */
    721     AACENC_INIT_AAC_ERROR         = 0x0041,  /*!< AAC library initialization error. */
    722     AACENC_INIT_SBR_ERROR         = 0x0042,  /*!< SBR library initialization error. */
    723     AACENC_INIT_TP_ERROR          = 0x0043,  /*!< Transport library initialization error. */
    724     AACENC_INIT_META_ERROR        = 0x0044,  /*!< Meta data library initialization error. */
    725 
    726     AACENC_ENCODE_ERROR           = 0x0060,  /*!< The encoding process was interrupted by an unexpected error. */
    727 
    728     AACENC_ENCODE_EOF             = 0x0080   /*!< End of file reached. */
    729 
    730 } AACENC_ERROR;
    731 
    732 
    733 /**
    734  *  AAC encoder buffer descriptors identifier.
    735  *  This identifier are used within buffer descriptors AACENC_BufDesc::bufferIdentifiers.
    736  */
    737 typedef enum {
    738     /* Input buffer identifier. */
    739     IN_AUDIO_DATA      = 0,                  /*!< Audio input buffer, interleaved INT_PCM samples. */
    740     IN_ANCILLRY_DATA   = 1,                  /*!< Ancillary data to be embedded into bitstream. */
    741     IN_METADATA_SETUP  = 2,                  /*!< Setup structure for embedding meta data. */
    742 
    743     /* Output buffer identifier. */
    744     OUT_BITSTREAM_DATA = 3,                  /*!< Buffer holds bitstream output data. */
    745     OUT_AU_SIZES       = 4                   /*!< Buffer contains sizes of each access unit. This information
    746                                                   is necessary for superframing. */
    747 
    748 } AACENC_BufferIdentifier;
    749 
    750 
    751 /**
    752  *  AAC encoder handle.
    753  */
    754 typedef struct AACENCODER *HANDLE_AACENCODER;
    755 
    756 
    757 /**
    758  *  Provides some info about the encoder configuration.
    759  */
    760 typedef struct {
    761 
    762     UINT                maxOutBufBytes;      /*!< Maximum number of encoder bitstream bytes within one frame.
    763                                                   Size depends on maximum number of supported channels in encoder instance.
    764                                                   For superframing (as used for example in DAB+), size has to be a multiple accordingly. */
    765 
    766     UINT                maxAncBytes;         /*!< Maximum number of ancillary data bytes which can be inserted into
    767                                                   bitstream within one frame. */
    768 
    769     UINT                inBufFillLevel;      /*!< Internal input buffer fill level in samples per channel. This parameter
    770                                                   will automatically be cleared if samplingrate or channel(Mode/Order) changes. */
    771 
    772     UINT                inputChannels;       /*!< Number of input channels expected in encoding process. */
    773 
    774     UINT                frameLength;         /*!< Amount of input audio samples consumed each frame per channel, depending
    775                                                   on audio object type configuration. */
    776 
    777     UINT                encoderDelay;        /*!< Codec delay in PCM samples/channel. Depends on framelength and AOT. Does not
    778                                                   include framing delay for filling up encoder PCM input buffer. */
    779 
    780     UCHAR               confBuf[64];         /*!< Configuration buffer in binary format as an AudioSpecificConfig
    781                                                   or StreamMuxConfig according to the selected transport type. */
    782 
    783     UINT                confSize;            /*!< Number of valid bytes in confBuf. */
    784 
    785 } AACENC_InfoStruct;
    786 
    787 
    788 /**
    789  *  Describes the input and output buffers for an aacEncEncode() call.
    790  */
    791 typedef struct {
    792     INT                 numBufs;             /*!< Number of buffers. */
    793     void              **bufs;                /*!< Pointer to vector containing buffer addresses. */
    794     INT                *bufferIdentifiers;   /*!< Identifier of each buffer element. See ::AACENC_BufferIdentifier. */
    795     INT                *bufSizes;            /*!< Size of each buffer in 8-bit bytes. */
    796     INT                *bufElSizes;          /*!< Size of each buffer element in bytes. */
    797 
    798 } AACENC_BufDesc;
    799 
    800 
    801 /**
    802  *  Defines the input arguments for an aacEncEncode() call.
    803  */
    804 typedef struct {
    805     INT                 numInSamples;        /*!< Number of valid input audio samples (multiple of input channels). */
    806     INT                 numAncBytes;         /*!< Number of ancillary data bytes to be encoded. */
    807 
    808 } AACENC_InArgs;
    809 
    810 
    811 /**
    812  *  Defines the output arguments for an aacEncEncode() call.
    813  */
    814 typedef struct {
    815     INT                 numOutBytes;         /*!< Number of valid bitstream bytes generated during aacEncEncode(). */
    816     INT                 numInSamples;        /*!< Number of input audio samples consumed by the encoder. */
    817     INT                 numAncBytes;         /*!< Number of ancillary data bytes consumed by the encoder. */
    818 
    819 } AACENC_OutArgs;
    820 
    821 
    822 /**
    823  *  Meta Data Compression Profiles.
    824  */
    825 typedef enum {
    826     AACENC_METADATA_DRC_NONE          = 0,   /*!< None. */
    827     AACENC_METADATA_DRC_FILMSTANDARD  = 1,   /*!< Film standard. */
    828     AACENC_METADATA_DRC_FILMLIGHT     = 2,   /*!< Film light. */
    829     AACENC_METADATA_DRC_MUSICSTANDARD = 3,   /*!< Music standard. */
    830     AACENC_METADATA_DRC_MUSICLIGHT    = 4,   /*!< Music light. */
    831     AACENC_METADATA_DRC_SPEECH        = 5    /*!< Speech. */
    832 
    833 } AACENC_METADATA_DRC_PROFILE;
    834 
    835 
    836 /**
    837  *  Meta Data setup structure.
    838  */
    839 typedef struct {
    840 
    841   AACENC_METADATA_DRC_PROFILE drc_profile;             /*!< MPEG DRC compression profile. See ::AACENC_METADATA_DRC_PROFILE. */
    842   AACENC_METADATA_DRC_PROFILE comp_profile;            /*!< ETSI heavy compression profile. See ::AACENC_METADATA_DRC_PROFILE. */
    843 
    844   INT                         drc_TargetRefLevel;      /*!< Used to define expected level to:
    845                                                             Scaled with 16 bit. x*2^16. */
    846   INT                         comp_TargetRefLevel;     /*!< Adjust limiter to avoid overload.
    847                                                             Scaled with 16 bit. x*2^16. */
    848 
    849   INT                         prog_ref_level_present;  /*!< Flag, if prog_ref_level is present */
    850   INT                         prog_ref_level;          /*!< Programme Reference Level = Dialogue Level:
    851                                                             -31.75dB .. 0 dB ; stepsize: 0.25dB
    852                                                             Scaled with 16 bit. x*2^16.*/
    853 
    854   UCHAR                       PCE_mixdown_idx_present; /*!< Flag, if dmx-idx should be written in programme config element */
    855   UCHAR                       ETSI_DmxLvl_present;     /*!< Flag, if dmx-lvl should be written in ETSI-ancData */
    856 
    857   SCHAR                       centerMixLevel;          /*!< Center downmix level (0...7, according to table) */
    858   SCHAR                       surroundMixLevel;        /*!< Surround downmix level (0...7, according to table) */
    859 
    860   UCHAR                       dolbySurroundMode;       /*!< Indication for Dolby Surround Encoding Mode.
    861                                                             - 0: Dolby Surround mode not indicated
    862                                                             - 1: 2-ch audio part is not Dolby surround encoded
    863                                                             - 2: 2-ch audio part is Dolby surround encoded */
    864 } AACENC_MetaData;
    865 
    866 
    867 /**
    868  * AAC encoder control flags.
    869  *
    870  * In interaction with the ::AACENC_CONTROL_STATE parameter it is possible to get information about the internal
    871  * initialization process. It is also possible to overwrite the internal state from extern when necessary.
    872  */
    873 typedef enum
    874 {
    875     AACENC_INIT_NONE              = 0x0000,  /*!< Do not trigger initialization. */
    876     AACENC_INIT_CONFIG            = 0x0001,  /*!< Initialize all encoder modules configuration. */
    877     AACENC_INIT_STATES            = 0x0002,  /*!< Reset all encoder modules history buffer. */
    878     AACENC_INIT_TRANSPORT         = 0x1000,  /*!< Initialize transport lib with new parameters. */
    879     AACENC_RESET_INBUFFER         = 0x2000,  /*!< Reset fill level of internal input buffer. */
    880     AACENC_INIT_ALL               = 0xFFFF   /*!< Initialize all. */
    881 }
    882 AACENC_CTRLFLAGS;
    883 
    884 
    885 /**
    886  * \brief  AAC encoder setting parameters.
    887  *
    888  * Use aacEncoder_SetParam() function to configure, or use aacEncoder_GetParam() function to read
    889  * the internal status of the following parameters.
    890  */
    891 typedef enum
    892 {
    893   AACENC_AOT                      = 0x0100,  /*!< Audio object type. See ::AUDIO_OBJECT_TYPE in FDK_audio.h.
    894                                                   - 2: MPEG-4 AAC Low Complexity.
    895                                                   - 5: MPEG-4 AAC Low Complexity with Spectral Band Replication (HE-AAC).
    896                                                   - 29: MPEG-4 AAC Low Complexity with Spectral Band Replication and Parametric Stereo (HE-AAC v2).
    897                                                         This configuration can be used only with stereo input audio data.
    898                                                   - 23: MPEG-4 AAC Low-Delay.
    899                                                   - 39: MPEG-4 AAC Enhanced Low-Delay. Since there is no ::AUDIO_OBJECT_TYPE for ELD in
    900                                                         combination with SBR defined, enable SBR explicitely by ::AACENC_SBR_MODE parameter.
    901                                                   - 129: MPEG-2 AAC Low Complexity.
    902                                                   - 132: MPEG-2 AAC Low Complexity with Spectral Band Replication (HE-AAC).
    903                                                   - 156: MPEG-2 AAC Low Complexity with Spectral Band Replication and Parametric Stereo (HE-AAC v2).
    904                                                          This configuration can be used only with stereo input audio data. */
    905 
    906   AACENC_BITRATE                  = 0x0101,  /*!< Total encoder bitrate. This parameter is mandatory and interacts with ::AACENC_BITRATEMODE.
    907                                                   - CBR: Bitrate in bits/second.
    908                                                     See \ref suppBitrates for details. */
    909 
    910   AACENC_BITRATEMODE              = 0x0102,  /*!< Bitrate mode. Configuration can be different kind of bitrate configurations:
    911                                                   - 0: Constant bitrate, use bitrate according to ::AACENC_BITRATE. (default)
    912                                                        Within none LD/ELD ::AUDIO_OBJECT_TYPE, the CBR mode makes use of full allowed bitreservoir.
    913                                                        In contrast, at Low-Delay ::AUDIO_OBJECT_TYPE the bitreservoir is kept very small.
    914                                                   - 8: LD/ELD full bitreservoir for packet based transmission. */
    915 
    916   AACENC_SAMPLERATE               = 0x0103,  /*!< Audio input data sampling rate. Encoder supports following sampling rates:
    917                                                   8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, 64000, 88200, 96000 */
    918 
    919   AACENC_SBR_MODE                 = 0x0104,  /*!< Configure SBR independently of the chosen Audio Object Type ::AUDIO_OBJECT_TYPE.
    920                                                   This parameter is for ELD audio object type only.
    921                                                   - -1: Use ELD SBR auto configurator (default).
    922                                                   - 0: Disable Spectral Band Replication.
    923                                                   - 1: Enable Spectral Band Replication. */
    924 
    925   AACENC_GRANULE_LENGTH           = 0x0105,  /*!< Core encoder (AAC) audio frame length in samples:
    926                                                   - 1024: Default configuration.
    927                                                   - 512: Default LD/ELD configuration.
    928                                                   - 480: Optional length in LD/ELD configuration. */
    929 
    930   AACENC_CHANNELMODE              = 0x0106,  /*!< Set explicit channel mode. Channel mode must match with number of input channels.
    931                                                   - 1-7 and 33,34: MPEG channel modes supported, see ::CHANNEL_MODE in FDK_audio.h. */
    932 
    933   AACENC_CHANNELORDER             = 0x0107,  /*!< Input audio data channel ordering scheme:
    934                                                   - 0: MPEG channel ordering (e. g. 5.1: C, L, R, SL, SR, LFE). (default)
    935                                                   - 1: WAVE file format channel ordering (e. g. 5.1: L, R, C, LFE, SL, SR). */
    936 
    937   AACENC_SBR_RATIO                = 0x0108,  /*!<  Controls activation of downsampled SBR. With downsampled SBR, the delay will be
    938                                                    shorter. On the other hand, for achieving the same quality level, downsampled SBR
    939                                                    needs more bits than dual-rate SBR.
    940                                                    With downsampled SBR, the AAC encoder will work at the same sampling rate as the
    941                                                    SBR encoder (single rate).
    942                                                    Downsampled SBR is supported for AAC-ELD and HE-AACv1.
    943                                                    - 1: Downsampled SBR (default for ELD).
    944                                                    - 2: Dual-rate SBR   (default for HE-AAC). */
    945 
    946   AACENC_AFTERBURNER              = 0x0200,  /*!< This parameter controls the use of the afterburner feature.
    947                                                   The afterburner is a type of analysis by synthesis algorithm which increases the
    948                                                   audio quality but also the required processing power. It is recommended to always
    949                                                   activate this if additional memory consumption and processing power consumption
    950                                                   is not a problem. If increased MHz and memory consumption are an issue then the MHz
    951                                                   and memory cost of this optional module need to be evaluated against the improvement
    952                                                   in audio quality on a case by case basis.
    953                                                   - 0: Disable afterburner (default).
    954                                                   - 1: Enable afterburner. */
    955 
    956   AACENC_BANDWIDTH                = 0x0203,  /*!< Core encoder audio bandwidth:
    957                                                   - 0: Determine bandwidth internally (default, see chapter \ref BEHAVIOUR_BANDWIDTH).
    958                                                   - 1 to fs/2: Frequency bandwidth in Hertz. (Experts only, better do not
    959                                                                touch this value to avoid degraded audio quality) */
    960 
    961   AACENC_TRANSMUX                 = 0x0300,  /*!< Transport type to be used. See ::TRANSPORT_TYPE in FDK_audio.h. Following
    962                                                   types can be configured in encoder library:
    963                                                   - 0: raw access units
    964                                                   - 1: ADIF bitstream format
    965                                                   - 2: ADTS bitstream format
    966                                                   - 6: Audio Mux Elements (LATM) with muxConfigPresent = 1
    967                                                   - 7: Audio Mux Elements (LATM) with muxConfigPresent = 0, out of band StreamMuxConfig
    968                                                   - 10: Audio Sync Stream (LOAS) */
    969 
    970   AACENC_HEADER_PERIOD            = 0x0301,  /*!< Frame count period for sending in-band configuration buffers within LATM/LOAS
    971                                                   transport layer. Additionally this parameter configures the PCE repetition period
    972                                                   in raw_data_block(). See \ref encPCE.
    973                                                   - 0xFF: auto-mode default 10 for TT_MP4_ADTS, TT_MP4_LOAS and TT_MP4_LATM_MCP1, otherwise 0.
    974                                                   - n: Frame count period. */
    975 
    976   AACENC_SIGNALING_MODE           = 0x0302,  /*!< Signaling mode of the extension AOT:
    977                                                   - 0: Implicit backward compatible signaling (default for non-MPEG-4 based
    978                                                        AOT's and for the transport formats ADIF and ADTS)
    979                                                        - A stream that uses implicit signaling can be decoded by every AAC decoder, even AAC-LC-only decoders
    980                                                        - An AAC-LC-only decoder will only decode the low-frequency part of the stream, resulting in a band-limited output
    981                                                        - This method works with all transport formats
    982                                                        - This method does not work with downsampled SBR
    983                                                   - 1: Explicit backward compatible signaling
    984                                                        - A stream that uses explicit backward compatible signaling can be decoded by every AAC decoder, even AAC-LC-only decoders
    985                                                        - An AAC-LC-only decoder will only decode the low-frequency part of the stream, resulting in a band-limited output
    986                                                        - A decoder not capable of decoding PS will only decode the AAC-LC+SBR part.
    987                                                          If the stream contained PS, the result will be a a decoded mono downmix
    988                                                        - This method does not work with ADIF or ADTS. For LOAS/LATM, it only works with AudioMuxVersion==1
    989                                                        - This method does work with downsampled SBR
    990                                                   - 2: Explicit hierarchical signaling (default for MPEG-4 based AOT's and for all transport formats excluding ADIF and ADTS)
    991                                                        - A stream that uses explicit hierarchical signaling can be decoded only by HE-AAC decoders
    992                                                        - An AAC-LC-only decoder will not decode a stream that uses explicit hierarchical signaling
    993                                                        - A decoder not capable of decoding PS will not decode the stream at all if it contained PS
    994                                                        - This method does not work with ADIF or ADTS. It works with LOAS/LATM and the MPEG-4 File format
    995                                                        - This method does work with downsampled SBR
    996 
    997                                                    For making sure that the listener always experiences the best audio quality,
    998                                                    explicit hierarchical signaling should be used.
    999                                                    This makes sure that only a full HE-AAC-capable decoder will decode those streams.
   1000                                                    The audio is played at full bandwidth.
   1001                                                    For best backwards compatibility, it is recommended to encode with implicit SBR signaling.
   1002                                                    A decoder capable of AAC-LC only will then only decode the AAC part, which means the decoded
   1003                                                    audio will sound band-limited.
   1004 
   1005                                                    For MPEG-2 transport types (ADTS,ADIF), only implicit signaling is possible.
   1006 
   1007                                                    For LOAS and LATM, explicit backwards compatible signaling only works together with AudioMuxVersion==1.
   1008                                                    The reason is that, for explicit backwards compatible signaling, additional information will be appended to the ASC.
   1009                                                    A decoder that is only capable of decoding AAC-LC will skip this part.
   1010                                                    Nevertheless, for jumping to the end of the ASC, it needs to know the ASC length.
   1011                                                    Transmitting the length of the ASC is a feature of AudioMuxVersion==1, it is not possible to transmit the
   1012                                                    length of the ASC with AudioMuxVersion==0, therefore an AAC-LC-only decoder will not be able to parse a
   1013                                                    LOAS/LATM stream that was being encoded with AudioMuxVersion==0.
   1014 
   1015                                                    For downsampled SBR, explicit signaling is mandatory. The reason for this is that the
   1016                                                    extension sampling frequency (which is in case of SBR the sampling frequqncy of the SBR part)
   1017                                                    can only be signaled in explicit mode.
   1018 
   1019                                                    For AAC-ELD, the SBR information is transmitted in the ELDSpecific Config, which is part of the
   1020                                                    AudioSpecificConfig. Therefore, the settings here will have no effect on AAC-ELD.*/
   1021 
   1022   AACENC_TPSUBFRAMES              = 0x0303,  /*!< Number of sub frames in a transport frame for LOAS/LATM or ADTS (default 1).
   1023                                                   - ADTS: Maximum number of sub frames restricted to 4.
   1024                                                   - LOAS/LATM: Maximum number of sub frames restricted to 2.*/
   1025 
   1026   AACENC_PROTECTION               = 0x0306,  /*!< Configure protection in tranpsort layer:
   1027                                                   - 0: No protection. (default)
   1028                                                   - 1: CRC active for ADTS bitstream format. */
   1029 
   1030   AACENC_ANCILLARY_BITRATE        = 0x0500,  /*!< Constant ancillary data bitrate in bits/second.
   1031                                                   - 0: Either no ancillary data or insert exact number of bytes, denoted via
   1032                                                        input parameter, numAncBytes in AACENC_InArgs.
   1033                                                   - else: Insert ancillary data with specified bitrate. */
   1034 
   1035   AACENC_METADATA_MODE            = 0x0600,  /*!< Configure Meta Data. See ::AACENC_MetaData for further details:
   1036                                                   - 0: Do not embed any metadata.
   1037                                                   - 1: Embed MPEG defined metadata only.
   1038                                                   - 2: Embed all metadata. */
   1039 
   1040   AACENC_CONTROL_STATE            = 0xFF00,  /*!< There is an automatic process which internally reconfigures the encoder instance
   1041                                                   when a configuration parameter changed or an error occured. This paramerter allows
   1042                                                   overwriting or getting the control status of this process. See ::AACENC_CTRLFLAGS. */
   1043 
   1044   AACENC_NONE                     = 0xFFFF   /*!< ------ */
   1045 
   1046 } AACENC_PARAM;
   1047 
   1048 
   1049 #ifdef __cplusplus
   1050 extern "C" {
   1051 #endif
   1052 
   1053 /**
   1054  * \brief  Open an instance of the encoder.
   1055  *
   1056  * Allocate memory for an encoder instance with a functional range denoted by the function parameters.
   1057  * Preinitialize encoder instance with default configuration.
   1058  *
   1059  * \param phAacEncoder  A pointer to an encoder handle. Initialized on return.
   1060  * \param encModules    Specify encoder modules to be supported in this encoder instance:
   1061  *                      - 0x0: Allocate memory for all available encoder modules.
   1062  *                      - else: Select memory allocation regarding encoder modules. Following flags are possible and can be combined.
   1063  *                              - 0x01: AAC module.
   1064  *                              - 0x02: SBR module.
   1065  *                              - 0x04: PS module.
   1066  *                              - 0x10: Metadata module.
   1067  *                              - example: (0x01|0x02|0x04|0x10) allocates all modules and is equivalent to default configuration denotet by 0x0.
   1068  * \param maxChannels   Number of channels to be allocated. This parameter can be used in different ways:
   1069  *                      - 0: Allocate maximum number of AAC and SBR channels as supported by the library.
   1070  *                      - nChannels: Use same maximum number of channels for allocating memory in AAC and SBR module.
   1071  *                      - nChannels | (nSbrCh<<8): Number of SBR channels can be different to AAC channels to save data memory.
   1072  *
   1073  * \return
   1074  *          - AACENC_OK, on succes.
   1075  *          - AACENC_INVALID_HANDLE, AACENC_MEMORY_ERROR, AACENC_INVALID_CONFIG, on failure.
   1076  */
   1077 AACENC_ERROR aacEncOpen(
   1078         HANDLE_AACENCODER        *phAacEncoder,
   1079         const UINT                encModules,
   1080         const UINT                maxChannels
   1081         );
   1082 
   1083 
   1084 /**
   1085  * \brief  Close the encoder instance.
   1086  *
   1087  * Deallocate encoder instance and free whole memory.
   1088  *
   1089  * \param phAacEncoder  Pointer to the encoder handle to be deallocated.
   1090  *
   1091  * \return
   1092  *          - AACENC_OK, on success.
   1093  *          - AACENC_INVALID_HANDLE, on failure.
   1094  */
   1095 AACENC_ERROR aacEncClose(
   1096         HANDLE_AACENCODER        *phAacEncoder
   1097         );
   1098 
   1099 
   1100 /**
   1101  * \brief Encode audio data.
   1102  *
   1103  * This function is mainly for encoding audio data. In addition the function can be used for an encoder (re)configuration
   1104  * process.
   1105  * - PCM input data will be retrieved from external input buffer until the fill level allows encoding a single frame.
   1106  *   This functionality allows an external buffer with reduced size in comparison to the AAC or HE-AAC audio frame length.
   1107  * - If the value of the input samples argument is zero, just internal reinitialization will be applied if it is
   1108  *   requested.
   1109  * - At the end of a file the flushing process can be triggerd via setting the value of the input samples argument to -1.
   1110  *   The encoder delay lines are fully flushed when the encoder returns no valid bitstream data AACENC_OutArgs::numOutBytes.
   1111  *   Furthermore the end of file is signaled by the return value AACENC_ENCODE_EOF.
   1112  * - If an error occured in the previous frame or any of the encoder parameters changed, an internal reinitialization
   1113  *   process will be applied before encoding the incoming audio samples.
   1114  * - The function can also be used for an independent reconfiguration process without encoding. The first parameter has to be a
   1115  *   valid encoder handle and all other parameters can be set to NULL.
   1116  * - If the size of the external bitbuffer in outBufDesc is not sufficient for writing the whole bitstream, an internal
   1117  *   error will be the return value and a reconfiguration will be triggered.
   1118  *
   1119  * \param hAacEncoder           A valid AAC encoder handle.
   1120  * \param inBufDesc             Input buffer descriptor, see AACENC_BufDesc:
   1121  *                              - At least one input buffer with audio data is expected.
   1122  *                              - Optionally a second input buffer with ancillary data can be fed.
   1123  * \param outBufDesc            Output buffer descriptor, see AACENC_BufDesc:
   1124  *                              - Provide one output buffer for the encoded bitstream.
   1125  * \param inargs                Input arguments, see AACENC_InArgs.
   1126  * \param outargs               Output arguments, AACENC_OutArgs.
   1127  *
   1128  * \return
   1129  *          - AACENC_OK, on success.
   1130  *          - AACENC_INVALID_HANDLE, AACENC_ENCODE_ERROR, on failure in encoding process.
   1131  *          - AACENC_INVALID_CONFIG, AACENC_INIT_ERROR, AACENC_INIT_AAC_ERROR, AACENC_INIT_SBR_ERROR, AACENC_INIT_TP_ERROR,
   1132  *            AACENC_INIT_META_ERROR, on failure in encoder initialization.
   1133  *          - AACENC_ENCODE_EOF, when flushing fully concluded.
   1134  */
   1135 AACENC_ERROR aacEncEncode(
   1136         const HANDLE_AACENCODER   hAacEncoder,
   1137         const AACENC_BufDesc     *inBufDesc,
   1138         const AACENC_BufDesc     *outBufDesc,
   1139         const AACENC_InArgs      *inargs,
   1140         AACENC_OutArgs           *outargs
   1141         );
   1142 
   1143 
   1144 /**
   1145  * \brief  Acquire info about present encoder instance.
   1146  *
   1147  * This function retrieves information of the encoder configuration. In addition to informative internal states,
   1148  * a configuration data block of the current encoder settings will be returned. The format is either Audio Specific Config
   1149  * in case of Raw Packets transport format or StreamMuxConfig in case of LOAS/LATM transport format. The configuration
   1150  * data block is binary coded as specified in ISO/IEC 14496-3 (MPEG-4 audio), to be used directly for MPEG-4 File Format
   1151  * or RFC3016 or RFC3640 applications.
   1152  *
   1153  * \param hAacEncoder           A valid AAC encoder handle.
   1154  * \param pInfo                 Pointer to AACENC_InfoStruct. Filled on return.
   1155  *
   1156  * \return
   1157  *          - AACENC_OK, on succes.
   1158  *          - AACENC_INIT_ERROR, on failure.
   1159  */
   1160 AACENC_ERROR aacEncInfo(
   1161         const HANDLE_AACENCODER   hAacEncoder,
   1162         AACENC_InfoStruct        *pInfo
   1163         );
   1164 
   1165 
   1166 /**
   1167  * \brief  Set one single AAC encoder parameter.
   1168  *
   1169  * This function allows configuration of all encoder parameters specified in ::AACENC_PARAM. Each parameter must be
   1170  * set with a separate function call. An internal validation of the configuration value range will be done and an
   1171  * internal reconfiguration will be signaled. The actual configuration adoption is part of the subsequent aacEncEncode() call.
   1172  *
   1173  * \param hAacEncoder           A valid AAC encoder handle.
   1174  * \param param                 Parameter to be set. See ::AACENC_PARAM.
   1175  * \param value                 Parameter value. See parameter description in ::AACENC_PARAM.
   1176  *
   1177  * \return
   1178  *          - AACENC_OK, on success.
   1179  *          - AACENC_INVALID_HANDLE, AACENC_UNSUPPORTED_PARAMETER, AACENC_INVALID_CONFIG, on failure.
   1180  */
   1181 AACENC_ERROR aacEncoder_SetParam(
   1182         const HANDLE_AACENCODER   hAacEncoder,
   1183         const AACENC_PARAM        param,
   1184         const UINT                value
   1185         );
   1186 
   1187 
   1188 /**
   1189  * \brief  Get one single AAC encoder parameter.
   1190  *
   1191  * This function is the complement to aacEncoder_SetParam(). After encoder reinitialization with user defined settings,
   1192  * the internal status can be obtained of each parameter, specified with ::AACENC_PARAM.
   1193  *
   1194  * \param hAacEncoder           A valid AAC encoder handle.
   1195  * \param param                 Parameter to be returned. See ::AACENC_PARAM.
   1196  *
   1197  * \return  Internal configuration value of specifed parameter ::AACENC_PARAM.
   1198  */
   1199 UINT aacEncoder_GetParam(
   1200         const HANDLE_AACENCODER   hAacEncoder,
   1201         const AACENC_PARAM        param
   1202         );
   1203 
   1204 
   1205 /**
   1206  * \brief  Get information about encoder library build.
   1207  *
   1208  * Fill a given LIB_INFO structure with library version information.
   1209  *
   1210  * \param info  Pointer to an allocated LIB_INFO struct.
   1211  *
   1212  * \return
   1213  *          - AACENC_OK, on success.
   1214  *          - AACENC_INVALID_HANDLE, AACENC_INIT_ERROR, on failure.
   1215  */
   1216 AACENC_ERROR aacEncGetLibInfo(
   1217         LIB_INFO                 *info
   1218         );
   1219 
   1220 
   1221 #ifdef __cplusplus
   1222 }
   1223 #endif
   1224 
   1225 #endif   /* _AAC_ENC_LIB_H_ */
   1226