aacenc_lib.h revision 3aec97e388e29a1d03f0197b27b893bc6aaf8ac3
1 2/* ----------------------------------------------------------------------------------------------------------- 3Software License for The Fraunhofer FDK AAC Codec Library for Android 4 5� Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur F�rderung der angewandten Forschung e.V. 6 All rights reserved. 7 8 1. INTRODUCTION 9The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements 10the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. 11This FDK AAC Codec software is intended to be used on a wide variety of Android devices. 12 13AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual 14audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by 15independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part 16of the MPEG specifications. 17 18Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) 19may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners 20individually for the purpose of encoding or decoding bit streams in products that are compliant with 21the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license 22these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec 23software may already be covered under those patent licenses when it is used for those licensed purposes only. 24 25Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, 26are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional 27applications information and documentation. 28 292. COPYRIGHT LICENSE 30 31Redistribution and use in source and binary forms, with or without modification, are permitted without 32payment of copyright license fees provided that you satisfy the following conditions: 33 34You must retain the complete text of this software license in redistributions of the FDK AAC Codec or 35your modifications thereto in source code form. 36 37You must retain the complete text of this software license in the documentation and/or other materials 38provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. 39You must make available free of charge copies of the complete source code of the FDK AAC Codec and your 40modifications thereto to recipients of copies in binary form. 41 42The name of Fraunhofer may not be used to endorse or promote products derived from this library without 43prior written permission. 44 45You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec 46software or your modifications thereto. 47 48Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software 49and 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." 52 533. NO PATENT LICENSE 54 55NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, 56ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with 57respect to this software. 58 59You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized 60by appropriate patent licenses. 61 624. DISCLAIMER 63 64This 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 66of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR 67CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, 68including but not limited to procurement of substitute goods or services; loss of use, data, or profits, 69or business interruption, however caused and on any theory of liability, whether in contract, strict 70liability, or tort (including negligence), arising in any way out of the use of this software, even if 71advised of the possibility of such damage. 72 735. CONTACT INFORMATION 74 75Fraunhofer Institute for Integrated Circuits IIS 76Attention: Audio and Multimedia Departments - FDK AAC LL 77Am Wolfsmantel 33 7891058 Erlangen, Germany 79 80www.iis.fraunhofer.de/amm 81amm-info@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 97This document describes the high-level interface and usage of the ISO/MPEG-2/4 AAC Encoder 98library developed by the Fraunhofer Institute for Integrated Circuits (IIS). 99 100The library implements encoding on the basis of the MPEG-2 and MPEG-4 AAC Low-Complexity 101standard, and depending on the library's configuration, MPEG-4 High-Efficiency AAC v2 and/or AAC-ELD standard. 102 103All references to SBR (Spectral Band Replication) are only applicable to HE-AAC or AAC-ELD versions 104of the library. All references to PS (Parametric Stereo) are only applicable to HE-AAC v2 105versions of the library. 106 107\section encBasics Encoder Basics 108 109This document can only give a rough overview about the ISO/MPEG-2 and ISO/MPEG-4 AAC audio coding 110standard. 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 116MPEG Advanced Audio Coding is based on a time-to-frequency mapping of the signal. The signal is 117partitioned into overlapping portions and transformed into frequency domain. The spectral components 118are then quantized and coded. \n 119An MPEG-2 or MPEG-4 AAC audio bitstream is composed of frames. Contrary to MPEG-1/2 Layer-3 (mp3), the 120length of individual frames is not restricted to a fixed number of bytes, but can take on any length 121between 1 and 768 bytes. 122 123 124\page LIBUSE Library Usage 125 126\section InterfaceDescription API Files 127 128All API header files are located in the folder /include of the release package. All header files 129are provided for usage in C/C++ programs. The AAC encoder library API functions are located at 130aacenc_lib.h. 131 132In binary releases the encoder core resides in statically linkable libraries called for example 133libAACenc.a/libFDK.a (LINUX) or FDK_fastaaclib.lib (MS Visual C++) for the plain AAC-LC core encoder 134and libSBRenc.a (LINUX) or FDK_sbrEncLib.lib (MS Visual C++) for the SBR (Spectral Band 135Replication) and PS (Parametric Stereo) modules. 136 137\section CallingSequence Calling Sequence 138 139For encoding of ISO/MPEG-2/4 AAC bitstreams the following sequence is mandatory. Input read and output 140write functions as well as the corresponding open and close functions are left out, since they may be 141implemented differently according to the user's specific requirements. The example implementation in 142main.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 { 162Feed \ref feedInBuf "input buffer" with new audio data and provide input/output \ref bufDes "arguments" to aacEncEncode(). 163\skipline aacEncEncode 164\until ; 165Write \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 171The 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 173for the maximum number of possible audio channels (for example 6 or 8) with the full functional range supported by the library. 174This 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 208In the present encoder API, the input and output buffers are described with \ref AACENC_BufDesc "buffer descriptors". This mechanism allows a flexible handling 209of 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 210buffers describing superframing data in DAB+ or DRM+.\n 211At 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 212of 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. 213The output buffer size should be 6144 bits per channel excluding the LFE channel. 214If 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 218All input and output buffer must be clustered in input and output buffer arrays. 219\skipline inBuffer 220\until outBufferElSize 221Allocate buffer descriptors 222\skipline AACENC_BufDesc 223\skipline AACENC_BufDesc 224Initialize input buffer descriptor 225\skipline inBufDesc 226\until bufElSizes 227Initialize output buffer descriptor 228\skipline outBufDesc 229\until bufElSizes 230 231\subsection argLists Provide Input/Output Argument Lists 232The 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 238The input buffer should be handled as a modulo buffer. New audio data in the form of pulse-code- 239modulated samples (PCM) must be read from external and be fed to the input buffer depending on its 240fill level. The required sample bitrate (represented by the data type INT_PCM which is 16, 24 or 32 241bits wide) is fixed and depends on library configuration (usually 16 bit). 242 243\dontinclude main.cpp 244\skipline WAV_InputRead 245\until ; 246After the encoder's internal buffer is fed with incoming audio samples, and aacEncEncode() 247processed 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 252If any AAC bitstream data is available, write it to output file or device. This can be done once the 253following condition is true: 254\dontinclude main.cpp 255\skip Valid bitstream available 256\skipline outargs 257 258\skipline outBytes>0 259 260If 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 267If the present library is configured with Metadata support, it is possible to insert meta data side info into the generated 268audio bitstream while encoding. 269 270To 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 271the ::AACENC_METADATA_MODE parameter and aacEncoder_SetParam() function. 272\code aacEncoder_SetParam(hAacEncoder, AACENC_METADATA_MODE, 0-2); \endcode 273 274This configuration indicates how to embed meta data into bitstrem. Either no insertion, MPEG or ETSI style. 275The meta data itself must be specified within the meta data setup structure AACENC_MetaData. 276 277Changing one of the AACENC_MetaData setup parameters can be achieved from outside the library within ::IN_METADATA_SETUP input 278buffer. There is no need to supply meta data setup structure every frame. If there is no new meta setup data available, the 279encoder uses the previous setup or the default configuration in initial state. 280 281In general the audio compressor and limiter within the encoder library can be configured with the ::AACENC_METADATA_DRC_PROFILE parameter 282AACENC_MetaData::drc_profile and and AACENC_MetaData::comp_profile. 283\n 284 285\section encReconf Encoder Reconfiguration 286 287The encoder library allows reconfiguration of the encoder instance with new settings 288continuously between encoding frames. Each parameter to be changed must be set with 289a single aacEncoder_SetParam() call. The internal status of each parameter can be 290retrieved with an aacEncoder_GetParam() call.\n 291There is no stand-alone reconfiguration function available. When parameters were 292modified from outside the library, an internal control mechanism triggers the necessary 293reconfiguration process which will be applied at the beginning of the following 294aacEncEncode() call. This state can be observed from external via the AACENC_INIT_STATUS 295and aacEncoder_GetParam() function. The reconfiguration process can also be applied 296immediately when all parameters of an aacEncEncode() call are NULL with a valid encoder 297handle.\n\n 298The 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 304All parameteres listed in ::AACENC_PARAM can be modified within an encoder instance. 305 306\subsection encMandatory Mandatory Encoder Parameters 307The following parameters must be specified when the encoder instance is initialized. 308\code 309aacEncoder_SetParam(hAacEncoder, AACENC_AOT, value); 310aacEncoder_SetParam(hAacEncoder, AACENC_BITRATE, value); 311aacEncoder_SetParam(hAacEncoder, AACENC_SAMPLERATE, value); 312aacEncoder_SetParam(hAacEncoder, AACENC_CHANNELMODE, value); 313\endcode 314Beyond that is an internal auto mode which preinitizializes the ::AACENC_BITRATE parameter 315if the parameter was not set from extern. The bitrate depends on the number of effective 316channels and sampling rate and is determined as follows. 317\code 318AAC-LC (AOT_AAC_LC): 1.5 bits per sample 319HE-AAC (AOT_SBR): 0.625 bits per sample (dualrate sbr) 320HE-AAC (AOT_SBR): 1.125 bits per sample (downsampled sbr) 321HE-AAC v2 (AOT_PS): 0.5 bits per sample 322\endcode 323 324\subsection channelMode Channel Mode Configuration 325The input audio data is described with the ::AACENC_CHANNELMODE parameter in the 326aacEncoder_SetParam() call. It is not possible to use the encoder instance with a 'number of 327input channels' argument. Instead, the channelMode must be set as follows. 328\code aacEncoder_SetParam(hAacEncoder, AACENC_CHANNELMODE, value); \endcode 329The parameter is specified in ::CHANNEL_MODE and can be mapped from the number of input channels 330in the following way. 331\dontinclude main.cpp 332\skip CHANNEL_MODE chMode = MODE_INVALID; 333\until return 334 335\subsection encQual Audio Quality Considerations 336The default encoder configuration is suggested to be used. Encoder tools such as TNS and PNS 337are activated by default and are internally controlled (see \ref BEHAVIOUR_TOOLS). 338 339There is an additional quality parameter called ::AACENC_AFTERBURNER. In the default 340configuration this quality switch is deactivated because it would cause a workload 341increase which might be significant. If workload is not an issue in the application 342we recommended to activate this feature. 343\code aacEncoder_SetParam(hAacEncoder, AACENC_AFTERBURNER, 1); \endcode 344 345 346\section audiochCfg Audio Channel Configuration 347The MPEG standard refers often to the so-called Channel Configuration. This Channel Configuration is used for a fixed Channel 348Mapping. The configurations 1-7 are predefined in MPEG standard and used for implicit signalling within the encoded bitstream. 349For user defined Configurations the Channel Configuration is set to 0 and the Channel Mapping must be explecitly described with an appropriate 350Program Config Element. The present Encoder implementation does not allow the user to configure this Channel Configuration from 351extern. The Encoder implementation supports fixed Channel Modes which are mapped to Channel Configuration as follow. 352\verbatim 353-------------------------------------------------------------------- 354 ChannelMode | ChCfg | front_El | side_El | back_El | lfe_El 355-----------------+--------+----------+----------+----------+-------- 356MODE_1 | 1 | SCE | | | 357MODE_2 | 2 | CPE | | | 358MODE_1_2 | 3 | SCE, CPE | | | 359MODE_1_2_1 | 4 | SCE, CPE | | SCE | 360MODE_1_2_2 | 5 | SCE, CPE | | CPE | 361MODE_1_2_2_1 | 6 | SCE, CPE | | CPE | LFE 362-------------------------------------------------------------------- 363 - SCE: Single Channel Element. 364 - CPE: Channel Pair. 365 - SCE: Low Frequency Element. 366\endverbatim 367 368Moreover, the Table describes all fixed Channel Elements for each Channel Mode which are assigned to a speaker arrangement. The 369arrangement includes front, side, back and lfe Audio Channel Elements.\n 370This mapping of Audio Channel Elements is defined in MPEG standard for Channel Config 1-7. The Channel assignment for MODE_1_1, 371MODE_2_2 and MODE_2_1 is used from the ARIB standard. All other configurations are defined as suggested in MPEG.\n 372In case of Channel Config 0 or writing matrix mixdown coefficients, the encoder enables the writing of Program Config Element 373itself as described in \ref encPCE. The configuration used in Program Config Element refers to the denoted Table.\n 374Beside the Channel Element assignment the Channel Modes are resposible for audio input data channel mapping. The Channel Mapping 375of the audio data depends on the selected ::AACENC_CHANNELORDER which can be MPEG or WAV like order.\n 376Following Table describes the complete channel mapping for both Channel Order configurations. 377\verbatim 378--------------------------------------------------------------------------------- 379ChannelMode | MPEG-Channelorder | WAV-Channelorder 380-----------------+---+---+---+---+---+---+---+---+---+---+---+---+---+---+---+--- 381MODE_1 | 0 | | | | | | | | 0 | | | | | | | 382MODE_2 | 0 | 1 | | | | | | | 0 | 1 | | | | | | 383MODE_1_2 | 0 | 1 | 2 | | | | | | 2 | 0 | 1 | | | | | 384MODE_1_2_1 | 0 | 1 | 2 | 3 | | | | | 2 | 0 | 1 | 3 | | | | 385MODE_1_2_2 | 0 | 1 | 2 | 3 | 4 | | | | 2 | 0 | 1 | 3 | 4 | | | 386MODE_1_2_2_1 | 0 | 1 | 2 | 3 | 4 | 5 | | | 2 | 0 | 1 | 4 | 5 | 3 | | 387--------------------------------------------------------------------------------- 388\endverbatim 389 390The denoted mapping is important for correct audio channel assignment when using MPEG or WAV ordering. The incoming audio 391channels are distributed MPEG like starting at the front channels and ending at the back channels. The distribution is used as 392described in Table concering Channel Config and fix channel elements. Please see the following example for clarification. 393 394\verbatim 395Example: MODE_1_2_2_1 - WAV-Channelorder 5.1 396------------------------------------------ 397 Input Channel | Coder Channel 398--------------------+--------------------- 399 2 (front center) | 0 (SCE channel) 400 0 (left center) | 1 (1st of 1st CPE) 401 1 (right center) | 2 (2nd of 1st CPE) 402 4 (left surround) | 3 (1st of 2nd CPE) 403 5 (right surround) | 4 (2nd of 2nd CPE) 404 3 (LFE) | 5 (LFE) 405------------------------------------------ 406\endverbatim 407 408 409\section suppBitrates Supported Bitrates 410 411The FDK AAC Encoder provides a wide range of supported bitrates. 412The minimum and maximum allowed bitrate depends on the Audio Object Type. For AAC-LC the minimum 413bitrate is the bitrate that is required to write the most basic and minimal valid bitstream. 414It consists of the bitstream format header information and other static/mandatory information 415within the AAC payload. The maximum AAC framesize allowed by the MPEG-4 standard 416determines the maximum allowed bitrate for AAC-LC. For HE-AAC and HE-AAC v2 a library internal 417look-up table is used. 418 419A good working point in terms of audio quality, sampling rate and bitrate, is at 1 to 1.5 420bits/audio sample for AAC-LC, 0.625 bits/audio sample for dualrate HE-AAC, 1.125 bits/audio sample 421for downsampled HE-AAC and 0.5 bits/audio sample for HE-AAC v2. 422For example for one channel with a sampling frequency of 48 kHz, the range from 42348 kbit/s to 72 kbit/s achieves reasonable audio quality for AAC-LC. 424 425For HE-AAC and HE-AAC v2 the lowest possible audio input sampling frequency is 16 kHz because then the 426AAC-LC core encoder operates in dual rate mode at its lowest possible sampling frequency, which is 8 kHz. 427HE-AAC v2 requires stereo input audio data. 428 429Please note that in HE-AAC or HE-AAC v2 mode the encoder supports much higher bitrates than are 430appropriate for HE-AAC or HE-AAC v2. For example, at a bitrate of more than 64 kbit/s for a stereo 431audio signal at 44.1 kHz it usually makes sense to use AAC-LC, which will produce better audio 432quality at that bitrate than HE-AAC or HE-AAC v2. 433 434\section reommendedConfig Recommended Sampling Rate and Bitrate Combinations 435 436The following table provides an overview of recommended encoder configuration parameters 437which we determined by virtue of numerous listening tests. 438 439\subsection reommendedConfigLC AAC-LC, HE-AAC, HE-AACv2 in Dualrate SBR mode. 440\verbatim 441----------------------------------------------------------------------------------- 442Audio Object Type | Bit Rate Range | Supported | Preferred | No. of 443 | [bit/s] | Sampling Rates | Sampl. | Chan. 444 | | [kHz] | Rate | 445 | | | [kHz] | 446-------------------+------------------+-----------------------+------------+------- 447AAC LC + SBR + PS | 8000 - 11999 | 22.05, 24.00 | 24.00 | 2 448AAC LC + SBR + PS | 12000 - 17999 | 32.00 | 32.00 | 2 449AAC LC + SBR + PS | 18000 - 39999 | 32.00, 44.10, 48.00 | 44.10 | 2 450AAC LC + SBR + PS | 40000 - 56000 | 32.00, 44.10, 48.00 | 48.00 | 2 451-------------------+------------------+-----------------------+------------+------- 452AAC LC + SBR | 8000 - 11999 | 22.05, 24.00 | 24.00 | 1 453AAC LC + SBR | 12000 - 17999 | 32.00 | 32.00 | 1 454AAC LC + SBR | 18000 - 39999 | 32.00, 44.10, 48.00 | 44.10 | 1 455AAC LC + SBR | 40000 - 56000 | 32.00, 44.10, 48.00 | 48.00 | 1 456AAC LC + SBR | 16000 - 27999 | 32.00, 44.10, 48.00 | 32.00 | 2 457AAC LC + SBR | 28000 - 63999 | 32.00, 44.10, 48.00 | 44.10 | 2 458AAC LC + SBR | 64000 - 128000 | 32.00, 44.10, 48.00 | 48.00 | 2 459-------------------+------------------+-----------------------+------------+------- 460AAC LC + SBR | 64000 - 69999 | 32.00, 44.10, 48.00 | 32.00 | 5, 5.1 461AAC LC + SBR | 70000 - 159999 | 32.00, 44.10, 48.00 | 44.10 | 5, 5.1 462AAC LC + SBR | 160000 - 245999 | 32.00, 44.10, 48.00 | 48.00 | 5 463AAC LC + SBR | 160000 - 265999 | 32.00, 44.10, 48.00 | 48.00 | 5.1 464-------------------+------------------+-----------------------+------------+------- 465AAC LC | 8000 - 15999 | 11.025, 12.00, 16.00 | 12.00 | 1 466AAC LC | 16000 - 23999 | 16.00 | 16.00 | 1 467AAC LC | 24000 - 31999 | 16.00, 22.05, 24.00 | 24.00 | 1 468AAC LC | 32000 - 55999 | 32.00 | 32.00 | 1 469AAC LC | 56000 - 160000 | 32.00, 44.10, 48.00 | 44.10 | 1 470AAC LC | 160001 - 288000 | 48.00 | 48.00 | 1 471-------------------+------------------+-----------------------+------------+------- 472AAC LC | 16000 - 23999 | 11.025, 12.00, 16.00 | 12.00 | 2 473AAC LC | 24000 - 31999 | 16.00 | 16.00 | 2 474AAC LC | 32000 - 39999 | 16.00, 22.05, 24.00 | 22.05 | 2 475AAC LC | 40000 - 95999 | 32.00 | 32.00 | 2 476AAC LC | 96000 - 111999 | 32.00, 44.10, 48.00 | 32.00 | 2 477AAC LC | 112000 - 320001 | 32.00, 44.10, 48.00 | 44.10 | 2 478AAC LC | 320002 - 576000 | 48.00 | 48.00 | 2 479-------------------+------------------+-----------------------+------------+------- 480AAC LC | 160000 - 239999 | 32.00 | 32.00 | 5, 5.1 481AAC LC | 240000 - 279999 | 32.00, 44.10, 48.00 | 32.00 | 5, 5.1 482AAC LC | 280000 - 800000 | 32.00, 44.10, 48.00 | 44.10 | 5, 5.1 483----------------------------------------------------------------------------------- 484\endverbatim \n 485 486\subsection reommendedConfigLD AAC-LD, AAC-ELD, AAC-ELD with SBR in Dualrate SBR mode. 487\verbatim 488----------------------------------------------------------------------------------- 489Audio Object Type | Bit Rate Range | Supported | Preferred | No. of 490 | [bit/s] | Sampling Rates | Sampl. | Chan. 491 | | [kHz] | Rate | 492 | | | [kHz] | 493-------------------+------------------+-----------------------+------------+------- 494ELD + SBR | 18000 - 24999 | 32.00 - 44.10 | 32.00 | 1 495ELD + SBR | 25000 - 31999 | 32.00 - 48.00 | 32.00 | 1 496ELD + SBR | 32000 - 64000 | 32.00 - 48.00 | 48.00 | 1 497-------------------+------------------+-----------------------+------------+------- 498ELD + SBR | 32000 - 51999 | 32.00 - 48.00 | 44.10 | 2 499ELD + SBR | 52000 - 128000 | 32.00 - 48.00 | 48.00 | 2 500-------------------+------------------+-----------------------+------------+------- 501ELD + SBR | 72000 - 160000 | 44.10 - 48.00 | 48.00 | 3 502-------------------+------------------+-----------------------+------------+------- 503ELD + SBR | 96000 - 212000 | 44.10 - 48.00 | 48.00 | 4 504-------------------+------------------+-----------------------+------------+------- 505ELD + SBR | 120000 - 246000 | 44.10 - 48.00 | 48.00 | 5 506-------------------+------------------+-----------------------+------------+------- 507ELD + SBR | 120000 - 266000 | 44.10 - 48.00 | 48.00 | 5.1 508-------------------+------------------+-----------------------+------------+------- 509LD, ELD | 16000 - 19999 | 16.00 - 24.00 | 16.00 | 1 510LD, ELD | 20000 - 39999 | 16.00 - 32.00 | 24.00 | 1 511LD, ELD | 40000 - 49999 | 22.05 - 32.00 | 32.00 | 1 512LD, ELD | 50000 - 61999 | 24.00 - 44.10 | 32.00 | 1 513LD, ELD | 62000 - 84999 | 32.00 - 48.00 | 44.10 | 1 514LD, ELD | 85000 - 192000 | 44.10 - 48.00 | 48.00 | 1 515-------------------+------------------+-----------------------+------------+------- 516LD, ELD | 64000 - 75999 | 24.00 - 32.00 | 32.00 | 2 517LD, ELD | 76000 - 97999 | 24.00 - 44.10 | 32.00 | 2 518LD, ELD | 98000 - 135999 | 32.00 - 48.00 | 44.10 | 2 519LD, ELD | 136000 - 384000 | 44.10 - 48.00 | 48.00 | 2 520-------------------+------------------+-----------------------+------------+------- 521LD, ELD | 96000 - 113999 | 24.00 - 32.00 | 32.00 | 3 522LD, ELD | 114000 - 146999 | 24.00 - 44.10 | 32.00 | 3 523LD, ELD | 147000 - 203999 | 32.00 - 48.00 | 44.10 | 3 524LD, ELD | 204000 - 576000 | 44.10 - 48.00 | 48.00 | 3 525-------------------+------------------+-----------------------+------------+------- 526LD, ELD | 128000 - 151999 | 24.00 - 32.00 | 32.00 | 4 527LD, ELD | 152000 - 195999 | 24.00 - 44.10 | 32.00 | 4 528LD, ELD | 196000 - 271999 | 32.00 - 48.00 | 44.10 | 4 529LD, ELD | 272000 - 768000 | 44.10 - 48.00 | 48.00 | 4 530-------------------+------------------+-----------------------+------------+------- 531LD, ELD | 160000 - 189999 | 24.00 - 32.00 | 32.00 | 5 532LD, ELD | 190000 - 244999 | 24.00 - 44.10 | 32.00 | 5 533LD, ELD | 245000 - 339999 | 32.00 - 48.00 | 44.10 | 5 534LD, ELD | 340000 - 960000 | 44.10 - 48.00 | 48.00 | 5 535----------------------------------------------------------------------------------- 536\endverbatim \n 537 538\subsection reommendedConfigELD AAC-ELD with SBR in Downsampled SBR mode. 539\verbatim 540----------------------------------------------------------------------------------- 541Audio Object Type | Bit Rate Range | Supported | Preferred | No. of 542 | [bit/s] | Sampling Rates | Sampl. | Chan. 543 | | [kHz] | Rate | 544 | | | [kHz] | 545-------------------+------------------+-----------------------+------------+------- 546ELD + SBR | 18000 - 24999 | 16.00 - 22.05 | 22.05 | 1 547(downsampled SBR) | 25000 - 35999 | 22.05 - 32.00 | 24.00 | 1 548 | 36000 - 64000 | 32.00 - 48.00 | 32.00 | 1 549----------------------------------------------------------------------------------- 550\endverbatim \n 551 552 553\page ENCODERBEHAVIOUR Encoder Behaviour 554 555\section BEHAVIOUR_BANDWIDTH Bandwidth 556 557The FDK AAC encoder usually does not use the full frequency range of the input signal, but restricts the bandwidth 558according to certain library-internal settings. They can be changed in the table "bandWidthTable" in the 559file bandwidth.cpp (if available). 560 561The encoder API provides the ::AACENC_BANDWIDTH parameter to adjust the bandwidth explicitly. 562\code 563aacEncoder_SetParam(hAacEncoder, AACENC_BANDWIDTH, value); 564\endcode 565 566However it is not recommended to change these settings, because they are based on numerious listening 567tests and careful tweaks to ensure the best overall encoding quality. 568 569Theoretically a signal of for example 48 kHz can contain frequencies up to 24 kHz, but to use this full range 570in an audio encoder usually does not make sense. Usually the encoder has a very limited amount of 571bits to spend (typically 128 kbit/s for stereo 48 kHz content) and to allow full range bandwidth would 572waste a lot of these bits for frequencies the human ear is hardly able to perceive anyway, if at all. Hence it 573is wise to use the available bits for the really important frequency range and just skip the rest. 574At lower bitrates (e. g. <= 80 kbit/s for stereo 48 kHz content) the encoder will choose an even smaller 575bandwidth, because an encoded signal with smaller bandwidth and hence less artifacts sounds better than a signal 576with higher bandwidth but then more coding artefacts across all frequencies. These artefacts would occur if 577small bitrates and high bandwidths are chosen because the available bits are just not enough to encode all 578frequencies well. 579 580Unfortunately some people evaluate encoding quality based on possible bandwidth as well, but it is a two-sided 581sword considering the trade-off described above. 582 583Another aspect is workload consumption. The higher the allowed bandwidth, the more frequency lines have to be 584processed, which in turn increases the workload. 585 586\section FRAMESIZES_AND_BIT_RESERVOIR Frame Sizes & Bit Reservoir 587 588For AAC there is a difference between constant bit rate and constant frame 589length due to the so-called bit reservoir technique, which allows the encoder to use less 590bits in an AAC frame for those audio signal sections which are easy to encode, 591and then spend them at a later point in 592time for more complex audio sections. The extent to which this "bit exchange" 593is done is limited to allow for reliable and relatively low delay real time 594streaming. 595Over a longer period in time the bitrate will be constant in the AAC constant 596bitrate mode, e.g. for ISDN transmission. This means that in AAC each bitstream 597frame will in general have a different length in bytes but over time it 598will reach the target bitrate. One could also make an MPEG compliant 599AAC encoder which always produces constant length packages for each AAC frame, 600but the audio quality would be considerably worse since the bit reservoir 601technique would have to be switched off completely. A higher bit rate would have 602to be used to get the same audio quality as with an enabled bit reservoir. 603 604The maximum AAC frame length, regardless of the available bit reservoir, is defined 605as 6144 bits per channel. 606 607For mp3 by the way, the same bit reservoir technique exists, but there each bit 608stream frame has a constant length for a given bit rate (ignoring the 609padding byte). In mp3 there is a so-called "back pointer" which tells 610the decoder which bits belong to the current mp3 frame - and in general some or 611many bits have been transmitted in an earlier mp3 frame. Basically this leads to 612the same "bit exchange between mp3 frames" as in AAC but with virtually constant 613length frames. 614 615This variable frame length at "constant bit rate" is not something special 616in this Fraunhofer IIS AAC encoder. AAC has been designed in that way. 617 618\subsection BEHAVIOUR_ESTIM_AVG_FRAMESIZES Estimating Average Frame Sizes 619 620A HE-AAC v1 or v2 audio frame contains 2048 PCM samples per channel (there is 621also one mode with 1920 samples per channel but this is only for special purposes 622such as DAB+ digital radio). 623 624The number of HE-AAC frames \f$N\_FRAMES\f$ per second at 44.1 kHz is: 625 626\f[ 627N\_FRAMES = 44100 / 2048 = 21.5332 628\f] 629 630At a bit rate of 8 kbps the average number of bits per frame \f$N\_BITS\_PER\_FRAME\f$ is: 631 632\f[ 633N\_BITS\_PER\_FRAME = 8000 / 21.5332 = 371.52 634\f] 635 636which is about 46.44 bytes per encoded frame. 637 638At a bit rate of 32 kbps, which is quite high for single channel HE-AAC v1, it is: 639 640\f[ 641N\_BITS\_PER\_FRAME = 32000 / 21.5332 = 1486 642\f] 643 644which is about 185.76 bytes per encoded frame. 645 646These bits/frame figures are average figures where each AAC frame generally has a different 647size in bytes. To calculate the same for AAC-LC just use 1024 instead of 2048 PCM samples per 648frame and channel. 649For AAC-LD/ELD it is either 480 or 512 PCM samples per frame and channel. 650 651 652\section BEHAVIOUR_TOOLS Encoder Tools 653 654The AAC encoder supports TNS, PNS, MS, Intensity and activates these tools depending on the audio signal and 655the encoder configuration (i.e. bitrate or AOT). It is not required to configure these tools manually. 656 657PNS improves encoding quality only for certain bitrates. Therefore it makes sense to activate PNS only for 658these bitrates and save the processing power required for PNS (about 10 % of the encoder) when using other 659bitrates. This is done automatically inside the encoder library. PNS is disabled inside the encoder library if 660an MPEG-2 AOT is choosen since PNS is an MPEG-4 AAC feature. 661 662If SBR is activated, the encoder automatically deactivates PNS internally. If TNS is disabled but PNS is allowed, 663the encoder deactivates PNS calculation internally. 664 665*/ 666 667#ifndef _AAC_ENC_LIB_H_ 668#define _AAC_ENC_LIB_H_ 669 670#include "machine_type.h" 671#include "FDK_audio.h" 672 673 674/** 675 * AAC encoder error codes. 676 */ 677typedef enum { 678 AACENC_OK = 0x0000, /*!< No error happened. All fine. */ 679 680 AACENC_INVALID_HANDLE = 0x0020, /*!< Handle passed to function call was invalid. */ 681 AACENC_MEMORY_ERROR = 0x0021, /*!< Memory allocation failed. */ 682 AACENC_UNSUPPORTED_PARAMETER = 0x0022, /*!< Parameter not available. */ 683 AACENC_INVALID_CONFIG = 0x0023, /*!< Configuration not provided. */ 684 685 AACENC_INIT_ERROR = 0x0040, /*!< General initialization error. */ 686 AACENC_INIT_AAC_ERROR = 0x0041, /*!< AAC library initialization error. */ 687 AACENC_INIT_SBR_ERROR = 0x0042, /*!< SBR library initialization error. */ 688 AACENC_INIT_TP_ERROR = 0x0043, /*!< Transport library initialization error. */ 689 AACENC_INIT_META_ERROR = 0x0044, /*!< Meta data library initialization error. */ 690 691 AACENC_ENCODE_ERROR = 0x0060, /*!< The encoding process was interrupted by an unexpected error. */ 692 693 AACENC_ENCODE_EOF = 0x0080 /*!< End of file reached. */ 694 695} AACENC_ERROR; 696 697 698/** 699 * AAC encoder buffer descriptors identifier. 700 * This identifier are used within buffer descriptors AACENC_BufDesc::bufferIdentifiers. 701 */ 702typedef enum { 703 /* Input buffer identifier. */ 704 IN_AUDIO_DATA = 0, /*!< Audio input buffer, interleaved INT_PCM samples. */ 705 IN_ANCILLRY_DATA = 1, /*!< Ancillary data to be embedded into bitstream. */ 706 IN_METADATA_SETUP = 2, /*!< Setup structure for embedding meta data. */ 707 708 /* Output buffer identifier. */ 709 OUT_BITSTREAM_DATA = 3, /*!< Buffer holds bitstream output data. */ 710 OUT_AU_SIZES = 4 /*!< Buffer contains sizes of each access unit. This information 711 is necessary for superframing. */ 712 713} AACENC_BufferIdentifier; 714 715 716/** 717 * AAC encoder handle. 718 */ 719typedef struct AACENCODER *HANDLE_AACENCODER; 720 721 722/** 723 * Provides some info about the encoder configuration. 724 */ 725typedef struct { 726 727 UINT maxOutBufBytes; /*!< Maximum number of encoder bitstream bytes within one frame. 728 Size depends on maximum number of supported channels in encoder instance. 729 For superframing (as used for example in DAB+), size has to be a multiple accordingly. */ 730 731 UINT maxAncBytes; /*!< Maximum number of ancillary data bytes which can be inserted into 732 bitstream within one frame. */ 733 734 UINT inBufFillLevel; /*!< Internal input buffer fill level in samples per channel. This parameter 735 will automatically be cleared if samplingrate or channel(Mode/Order) changes. */ 736 737 UINT inputChannels; /*!< Number of input channels expected in encoding process. */ 738 739 UINT frameLength; /*!< Amount of input audio samples consumed each frame per channel, depending 740 on audio object type configuration. */ 741 742 UINT encoderDelay; /*!< Codec delay in PCM samples/channel. Depends on framelength and AOT. Does not 743 include framing delay for filling up encoder PCM input buffer. */ 744 745 UCHAR confBuf[64]; /*!< Configuration buffer in binary format as an AudioSpecificConfig 746 or StreamMuxConfig according to the selected transport type. */ 747 748 UINT confSize; /*!< Number of valid bytes in confBuf. */ 749 750} AACENC_InfoStruct; 751 752 753/** 754 * Describes the input and output buffers for an aacEncEncode() call. 755 */ 756typedef struct { 757 INT numBufs; /*!< Number of buffers. */ 758 void **bufs; /*!< Pointer to vector containing buffer addresses. */ 759 INT *bufferIdentifiers; /*!< Identifier of each buffer element. See ::AACENC_BufferIdentifier. */ 760 INT *bufSizes; /*!< Size of each buffer in 8-bit bytes. */ 761 INT *bufElSizes; /*!< Size of each buffer element in bytes. */ 762 763} AACENC_BufDesc; 764 765 766/** 767 * Defines the input arguments for an aacEncEncode() call. 768 */ 769typedef struct { 770 INT numInSamples; /*!< Number of valid input audio samples (multiple of input channels). */ 771 INT numAncBytes; /*!< Number of ancillary data bytes to be encoded. */ 772 773} AACENC_InArgs; 774 775 776/** 777 * Defines the output arguments for an aacEncEncode() call. 778 */ 779typedef struct { 780 INT numOutBytes; /*!< Number of valid bitstream bytes generated during aacEncEncode(). */ 781 INT numInSamples; /*!< Number of input audio samples consumed by the encoder. */ 782 INT numAncBytes; /*!< Number of ancillary data bytes consumed by the encoder. */ 783 784} AACENC_OutArgs; 785 786 787/** 788 * Meta Data Compression Profiles. 789 */ 790typedef enum { 791 AACENC_METADATA_DRC_NONE = 0, /*!< None. */ 792 AACENC_METADATA_DRC_FILMSTANDARD = 1, /*!< Film standard. */ 793 AACENC_METADATA_DRC_FILMLIGHT = 2, /*!< Film light. */ 794 AACENC_METADATA_DRC_MUSICSTANDARD = 3, /*!< Music standard. */ 795 AACENC_METADATA_DRC_MUSICLIGHT = 4, /*!< Music light. */ 796 AACENC_METADATA_DRC_SPEECH = 5 /*!< Speech. */ 797 798} AACENC_METADATA_DRC_PROFILE; 799 800 801/** 802 * Meta Data setup structure. 803 */ 804typedef struct { 805 806 AACENC_METADATA_DRC_PROFILE drc_profile; /*!< MPEG DRC compression profile. See ::AACENC_METADATA_DRC_PROFILE. */ 807 AACENC_METADATA_DRC_PROFILE comp_profile; /*!< ETSI heavy compression profile. See ::AACENC_METADATA_DRC_PROFILE. */ 808 809 INT drc_TargetRefLevel; /*!< Used to define expected level to: 810 Scaled with 16 bit. x*2^16. */ 811 INT comp_TargetRefLevel; /*!< Adjust limiter to avoid overload. 812 Scaled with 16 bit. x*2^16. */ 813 814 INT prog_ref_level_present; /*!< Flag, if prog_ref_level is present */ 815 INT prog_ref_level; /*!< Programme Reference Level = Dialogue Level: 816 -31.75dB .. 0 dB ; stepsize: 0.25dB 817 Scaled with 16 bit. x*2^16.*/ 818 819 UCHAR PCE_mixdown_idx_present; /*!< Flag, if dmx-idx should be written in programme config element */ 820 UCHAR ETSI_DmxLvl_present; /*!< Flag, if dmx-lvl should be written in ETSI-ancData */ 821 822 SCHAR centerMixLevel; /*!< Center downmix level (0...7, according to table) */ 823 SCHAR surroundMixLevel; /*!< Surround downmix level (0...7, according to table) */ 824 825 UCHAR dolbySurroundMode; /*!< Indication for Dolby Surround Encoding Mode. 826 - 0: Dolby Surround mode not indicated 827 - 1: 2-ch audio part is not Dolby surround encoded 828 - 2: 2-ch audio part is Dolby surround encoded */ 829} AACENC_MetaData; 830 831 832/** 833 * AAC encoder control flags. 834 * 835 * In interaction with the ::AACENC_CONTROL_STATE parameter it is possible to get information about the internal 836 * initialization process. It is also possible to overwrite the internal state from extern when necessary. 837 */ 838typedef enum 839{ 840 AACENC_INIT_NONE = 0x0000, /*!< Do not trigger initialization. */ 841 AACENC_INIT_CONFIG = 0x0001, /*!< Initialize all encoder modules configuration. */ 842 AACENC_INIT_STATES = 0x0002, /*!< Reset all encoder modules history buffer. */ 843 AACENC_INIT_TRANSPORT = 0x1000, /*!< Initialize transport lib with new parameters. */ 844 AACENC_RESET_INBUFFER = 0x2000, /*!< Reset fill level of internal input buffer. */ 845 AACENC_INIT_ALL = 0xFFFF /*!< Initialize all. */ 846} 847AACENC_CTRLFLAGS; 848 849 850/** 851 * \brief AAC encoder setting parameters. 852 * 853 * Use aacEncoder_SetParam() function to configure, or use aacEncoder_GetParam() function to read 854 * the internal status of the following parameters. 855 */ 856typedef enum 857{ 858 AACENC_AOT = 0x0100, /*!< Audio object type. See ::AUDIO_OBJECT_TYPE in FDK_audio.h. 859 - 2: MPEG-4 AAC Low Complexity. 860 - 5: MPEG-4 AAC Low Complexity with Spectral Band Replication (HE-AAC). 861 - 29: MPEG-4 AAC Low Complexity with Spectral Band Replication and Parametric Stereo (HE-AAC v2). 862 This configuration can be used only with stereo input audio data. 863 - 23: MPEG-4 AAC Low-Delay. 864 - 39: MPEG-4 AAC Enhanced Low-Delay. Since there is no ::AUDIO_OBJECT_TYPE for ELD in 865 combination with SBR defined, enable SBR explicitely by ::AACENC_SBR_MODE parameter. 866 - 129: MPEG-2 AAC Low Complexity. 867 - 132: MPEG-2 AAC Low Complexity with Spectral Band Replication (HE-AAC). 868 - 156: MPEG-2 AAC Low Complexity with Spectral Band Replication and Parametric Stereo (HE-AAC v2). 869 This configuration can be used only with stereo input audio data. */ 870 871 AACENC_BITRATE = 0x0101, /*!< Total encoder bitrate. This parameter is mandatory and interacts with ::AACENC_BITRATEMODE. 872 - CBR: Bitrate in bits/second. 873 See \ref suppBitrates for details. */ 874 875 AACENC_BITRATEMODE = 0x0102, /*!< Bitrate mode. Configuration can be different kind of bitrate configurations: 876 - 0: Constant bitrate, use bitrate according to ::AACENC_BITRATE. (default) 877 Within none LD/ELD ::AUDIO_OBJECT_TYPE, the CBR mode makes use of full allowed bitreservoir. 878 In contrast, at Low-Delay ::AUDIO_OBJECT_TYPE the bitreservoir is kept very small. 879 - 8: LD/ELD full bitreservoir for packet based transmission. */ 880 881 AACENC_SAMPLERATE = 0x0103, /*!< Audio input data sampling rate. Encoder supports following sampling rates: 882 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, 64000, 88200, 96000 */ 883 884 AACENC_SBR_MODE = 0x0104, /*!< Configure SBR independently of the chosen Audio Object Type ::AUDIO_OBJECT_TYPE:. 885 This parameter is only available for ELD. 886 - 0: Disable Spectral Band Replication. 887 - 1: Enable Spectral Band Replication. */ 888 889 AACENC_GRANULE_LENGTH = 0x0105, /*!< Core encoder (AAC) audio frame length in samples: 890 - 1024: Default configuration. 891 - 512: Default LD/ELD configuration. 892 - 480: Optional length in LD/ELD configuration. */ 893 894 AACENC_CHANNELMODE = 0x0106, /*!< Set explicit channel mode. Channel mode must match with number of input channels. 895 - 1-6: MPEG channel modes supported, see ::CHANNEL_MODE in FDK_audio.h. */ 896 897 AACENC_CHANNELORDER = 0x0107, /*!< Input audio data channel ordering scheme: 898 - 0: MPEG channel ordering (e. g. 5.1: C, L, R, SL, SR, LFE). (default) 899 - 1: WAVE file format channel ordering (e. g. 5.1: L, R, C, LFE, SL, SR). */ 900 901 AACENC_SBR_RATIO = 0x0108, /*!< Controls activation of downsampled SBR. With downsampled SBR, the delay will be 902 shorter. On the other hand, for achieving the same quality level, downsampled SBR 903 needs more bits than dual-rate SBR. 904 With downsampled SBR, the AAC encoder will work at the same sampling rate as the 905 SBR encoder (single rate). 906 Downsampled SBR is supported for AAC-ELD and HE-AACv1. 907 - 1: Downsampled SBR (default for ELD). 908 - 2: Dual-rate SBR (default for HE-AAC). */ 909 910 AACENC_AFTERBURNER = 0x0200, /*!< This parameter controls the use of the afterburner feature. 911 The afterburner is a type of analysis by synthesis algorithm which increases the 912 audio quality but also the required processing power. It is recommended to always 913 activate this if additional memory consumption and processing power consumption 914 is not a problem. If increased MHz and memory consumption are an issue then the MHz 915 and memory cost of this optional module need to be evaluated against the improvement 916 in audio quality on a case by case basis. 917 - 0: Disable afterburner (default). 918 - 1: Enable afterburner. */ 919 920 AACENC_BANDWIDTH = 0x0203, /*!< Core encoder audio bandwidth: 921 - 0: Determine bandwidth internally (default, see chapter \ref BEHAVIOUR_BANDWIDTH). 922 - 1 to fs/2: Frequency bandwidth in Hertz. (Experts only, better do not 923 touch this value to avoid degraded audio quality) */ 924 925 AACENC_TRANSMUX = 0x0300, /*!< Transport type to be used. See ::TRANSPORT_TYPE in FDK_audio.h. Following 926 types can be configured in encoder library: 927 - 0: raw access units 928 - 1: ADIF bitstream format 929 - 2: ADTS bitstream format 930 - 6: Audio Mux Elements (LATM) with muxConfigPresent = 1 931 - 7: Audio Mux Elements (LATM) with muxConfigPresent = 0, out of band StreamMuxConfig 932 - 10: Audio Sync Stream (LOAS) */ 933 934 AACENC_HEADER_PERIOD = 0x0301, /*!< Frame count period for sending in-band configuration buffers within LATM/LOAS 935 transport layer. Additionally this parameter configures the PCE repetition period 936 in raw_data_block(). See \ref encPCE. 937 - 0xFF: auto-mode default 10 for TT_MP4_ADTS, TT_MP4_LOAS and TT_MP4_LATM_MCP1, otherwise 0. 938 - n: Frame count period. */ 939 940 AACENC_SIGNALING_MODE = 0x0302, /*!< Signaling mode of the extension AOT: 941 - 0: Implicit backward compatible signaling. (default) 942 - 1: Explicit SBR and implicit PS signaling. 943 - 2: Explicit hierarchical signaling. 944 945 The use of backward-compatible implicit signaling is recommended if the user specically 946 aims at preserving compatibility with decoders only capable of decoding AAC-LC. Otherwise 947 use non-backward-compatible explicit signaling. 948 Bitstream formats ADTS and ADIF can only do implicit signaling. */ 949 950 AACENC_TPSUBFRAMES = 0x0303, /*!< Number of sub frames in a transport frame for LOAS/LATM or ADTS (default 1). 951 - ADTS: Maximum number of sub frames restricted to 4. 952 - LOAS/LATM: Maximum number of sub frames restricted to 2.*/ 953 954 AACENC_PROTECTION = 0x0306, /*!< Configure protection in tranpsort layer: 955 - 0: No protection. (default) 956 - 1: CRC active for ADTS bitstream format. */ 957 958 AACENC_ANCILLARY_BITRATE = 0x0500, /*!< Constant ancillary data bitrate in bits/second. 959 - 0: Either no ancillary data or insert exact number of bytes, denoted via 960 input parameter, numAncBytes in AACENC_InArgs. 961 - else: Insert ancillary data with specified bitrate. */ 962 963 AACENC_METADATA_MODE = 0x0600, /*!< Configure Meta Data. See ::AACENC_MetaData for further details: 964 - 0: Do not embed any metadata. 965 - 1: Embed MPEG defined metadata only. 966 - 2: Embed all metadata. */ 967 968 AACENC_CONTROL_STATE = 0xFF00, /*!< There is an automatic process which internally reconfigures the encoder instance 969 when a configuration parameter changed or an error occured. This paramerter allows 970 overwriting or getting the control status of this process. See ::AACENC_CTRLFLAGS. */ 971 972 AACENC_NONE = 0xFFFF /*!< ------ */ 973 974} AACENC_PARAM; 975 976 977#ifdef __cplusplus 978extern "C" { 979#endif 980 981/** 982 * \brief Open an instance of the encoder. 983 * 984 * Allocate memory for an encoder instance with a functional range denoted by the function parameters. 985 * Preinitialize encoder instance with default configuration. 986 * 987 * \param phAacEncoder A pointer to an encoder handle. Initialized on return. 988 * \param encModules Specify encoder modules to be supported in this encoder instance: 989 * - 0x0: Allocate memory for all available encoder modules. 990 * - else: Select memory allocation regarding encoder modules. Following flags are possible and can be combined. 991 * - 0x01: AAC module. 992 * - 0x02: SBR module. 993 * - 0x04: PS module. 994 * - 0x10: Metadata module. 995 * - example: (0x01|0x02|0x04|0x10) allocates all modules and is equivalent to default configuration denotet by 0x0. 996 * \param maxChannels Number of channels to be allocated. This parameter can be used in different ways: 997 * - 0: Allocate maximum number of AAC and SBR channels as supported by the library. 998 * - nChannels: Use same maximum number of channels for allocating memory in AAC and SBR module. 999 * - nChannels | (nSbrCh<<8): Number of SBR channels can be different to AAC channels to save data memory. 1000 * 1001 * \return 1002 * - AACENC_OK, on succes. 1003 * - AACENC_INVALID_HANDLE, AACENC_MEMORY_ERROR, AACENC_INVALID_CONFIG, on failure. 1004 */ 1005AACENC_ERROR aacEncOpen( 1006 HANDLE_AACENCODER *phAacEncoder, 1007 const UINT encModules, 1008 const UINT maxChannels 1009 ); 1010 1011 1012/** 1013 * \brief Close the encoder instance. 1014 * 1015 * Deallocate encoder instance and free whole memory. 1016 * 1017 * \param phAacEncoder Pointer to the encoder handle to be deallocated. 1018 * 1019 * \return 1020 * - AACENC_OK, on success. 1021 * - AACENC_INVALID_HANDLE, on failure. 1022 */ 1023AACENC_ERROR aacEncClose( 1024 HANDLE_AACENCODER *phAacEncoder 1025 ); 1026 1027 1028/** 1029 * \brief Encode audio data. 1030 * 1031 * This function is mainly for encoding audio data. In addition the function can be used for an encoder (re)configuration 1032 * process. 1033 * - PCM input data will be retrieved from external input buffer until the fill level allows encoding a single frame. 1034 * This functionality allows an external buffer with reduced size in comparison to the AAC or HE-AAC audio frame length. 1035 * - If the value of the input samples argument is zero, just internal reinitialization will be applied if it is 1036 * requested. 1037 * - At the end of a file the flushing process can be triggerd via setting the value of the input samples argument to -1. 1038 * The encoder delay lines are fully flushed when the encoder returns no valid bitstream data AACENC_OutArgs::numOutBytes. 1039 * Furthermore the end of file is signaled by the return value AACENC_ENCODE_EOF. 1040 * - If an error occured in the previous frame or any of the encoder parameters changed, an internal reinitialization 1041 * process will be applied before encoding the incoming audio samples. 1042 * - The function can also be used for an independent reconfiguration process without encoding. The first parameter has to be a 1043 * valid encoder handle and all other parameters can be set to NULL. 1044 * - If the size of the external bitbuffer in outBufDesc is not sufficient for writing the whole bitstream, an internal 1045 * error will be the return value and a reconfiguration will be triggered. 1046 * 1047 * \param hAacEncoder A valid AAC encoder handle. 1048 * \param inBufDesc Input buffer descriptor, see AACENC_BufDesc: 1049 * - At least one input buffer with audio data is expected. 1050 * - Optionally a second input buffer with ancillary data can be fed. 1051 * \param outBufDesc Output buffer descriptor, see AACENC_BufDesc: 1052 * - Provide one output buffer for the encoded bitstream. 1053 * \param inargs Input arguments, see AACENC_InArgs. 1054 * \param outargs Output arguments, AACENC_OutArgs. 1055 * 1056 * \return 1057 * - AACENC_OK, on success. 1058 * - AACENC_INVALID_HANDLE, AACENC_ENCODE_ERROR, on failure in encoding process. 1059 * - AACENC_INVALID_CONFIG, AACENC_INIT_ERROR, AACENC_INIT_AAC_ERROR, AACENC_INIT_SBR_ERROR, AACENC_INIT_TP_ERROR, 1060 * AACENC_INIT_META_ERROR, on failure in encoder initialization. 1061 * - AACENC_ENCODE_EOF, when flushing fully concluded. 1062 */ 1063AACENC_ERROR aacEncEncode( 1064 const HANDLE_AACENCODER hAacEncoder, 1065 const AACENC_BufDesc *inBufDesc, 1066 const AACENC_BufDesc *outBufDesc, 1067 const AACENC_InArgs *inargs, 1068 AACENC_OutArgs *outargs 1069 ); 1070 1071 1072/** 1073 * \brief Acquire info about present encoder instance. 1074 * 1075 * This function retrieves information of the encoder configuration. In addition to informative internal states, 1076 * a configuration data block of the current encoder settings will be returned. The format is either Audio Specific Config 1077 * in case of Raw Packets transport format or StreamMuxConfig in case of LOAS/LATM transport format. The configuration 1078 * data block is binary coded as specified in ISO/IEC 14496-3 (MPEG-4 audio), to be used directly for MPEG-4 File Format 1079 * or RFC3016 or RFC3640 applications. 1080 * 1081 * \param hAacEncoder A valid AAC encoder handle. 1082 * \param pInfo Pointer to AACENC_InfoStruct. Filled on return. 1083 * 1084 * \return 1085 * - AACENC_OK, on succes. 1086 * - AACENC_INIT_ERROR, on failure. 1087 */ 1088AACENC_ERROR aacEncInfo( 1089 const HANDLE_AACENCODER hAacEncoder, 1090 AACENC_InfoStruct *pInfo 1091 ); 1092 1093 1094/** 1095 * \brief Set one single AAC encoder parameter. 1096 * 1097 * This function allows configuration of all encoder parameters specified in ::AACENC_PARAM. Each parameter must be 1098 * set with a separate function call. An internal validation of the configuration value range will be done and an 1099 * internal reconfiguration will be signaled. The actual configuration adoption is part of the subsequent aacEncEncode() call. 1100 * 1101 * \param hAacEncoder A valid AAC encoder handle. 1102 * \param param Parameter to be set. See ::AACENC_PARAM. 1103 * \param value Parameter value. See parameter description in ::AACENC_PARAM. 1104 * 1105 * \return 1106 * - AACENC_OK, on success. 1107 * - AACENC_INVALID_HANDLE, AACENC_UNSUPPORTED_PARAMETER, AACENC_INVALID_CONFIG, on failure. 1108 */ 1109AACENC_ERROR aacEncoder_SetParam( 1110 const HANDLE_AACENCODER hAacEncoder, 1111 const AACENC_PARAM param, 1112 const UINT value 1113 ); 1114 1115 1116/** 1117 * \brief Get one single AAC encoder parameter. 1118 * 1119 * This function is the complement to aacEncoder_SetParam(). After encoder reinitialization with user defined settings, 1120 * the internal status can be obtained of each parameter, specified with ::AACENC_PARAM. 1121 * 1122 * \param hAacEncoder A valid AAC encoder handle. 1123 * \param param Parameter to be returned. See ::AACENC_PARAM. 1124 * 1125 * \return Internal configuration value of specifed parameter ::AACENC_PARAM. 1126 */ 1127UINT aacEncoder_GetParam( 1128 const HANDLE_AACENCODER hAacEncoder, 1129 const AACENC_PARAM param 1130 ); 1131 1132 1133/** 1134 * \brief Get information about encoder library build. 1135 * 1136 * Fill a given LIB_INFO structure with library version information. 1137 * 1138 * \param info Pointer to an allocated LIB_INFO struct. 1139 * 1140 * \return 1141 * - AACENC_OK, on success. 1142 * - AACENC_INVALID_HANDLE, AACENC_INIT_ERROR, on failure. 1143 */ 1144AACENC_ERROR aacEncGetLibInfo( 1145 LIB_INFO *info 1146 ); 1147 1148 1149#ifdef __cplusplus 1150} 1151#endif 1152 1153#endif /* _AAC_ENC_LIB_H_ */ 1154