1 2 /* ----------------------------------------------------------------------------------------------------------- 3 Software License for The Fraunhofer FDK AAC Codec Library for Android 4 5 Copyright 1995 - 2013 Fraunhofer-Gesellschaft zur Frderung der angewandten Forschung e.V. 6 All rights reserved. 7 8 1. INTRODUCTION 9 The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements 10 the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio. 11 This FDK AAC Codec software is intended to be used on a wide variety of Android devices. 12 13 AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual 14 audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by 15 independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part 16 of the MPEG specifications. 17 18 Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer) 19 may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners 20 individually for the purpose of encoding or decoding bit streams in products that are compliant with 21 the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license 22 these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec 23 software may already be covered under those patent licenses when it is used for those licensed purposes only. 24 25 Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality, 26 are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional 27 applications information and documentation. 28 29 2. COPYRIGHT LICENSE 30 31 Redistribution and use in source and binary forms, with or without modification, are permitted without 32 payment of copyright license fees provided that you satisfy the following conditions: 33 34 You must retain the complete text of this software license in redistributions of the FDK AAC Codec or 35 your modifications thereto in source code form. 36 37 You must retain the complete text of this software license in the documentation and/or other materials 38 provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form. 39 You must make available free of charge copies of the complete source code of the FDK AAC Codec and your 40 modifications thereto to recipients of copies in binary form. 41 42 The name of Fraunhofer may not be used to endorse or promote products derived from this library without 43 prior written permission. 44 45 You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec 46 software or your modifications thereto. 47 48 Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software 49 and the date of any change. For modified versions of the FDK AAC Codec, the term 50 "Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term 51 "Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android." 52 53 3. NO PATENT LICENSE 54 55 NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer, 56 ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with 57 respect to this software. 58 59 You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized 60 by appropriate patent licenses. 61 62 4. DISCLAIMER 63 64 This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors 65 "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties 66 of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR 67 CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages, 68 including but not limited to procurement of substitute goods or services; loss of use, data, or profits, 69 or business interruption, however caused and on any theory of liability, whether in contract, strict 70 liability, or tort (including negligence), arising in any way out of the use of this software, even if 71 advised of the possibility of such damage. 72 73 5. CONTACT INFORMATION 74 75 Fraunhofer Institute for Integrated Circuits IIS 76 Attention: Audio and Multimedia Departments - FDK AAC LL 77 Am Wolfsmantel 33 78 91058 Erlangen, Germany 79 80 www.iis.fraunhofer.de/amm 81 amm-info (at) iis.fraunhofer.de 82 ----------------------------------------------------------------------------------------------------------- */ 83 84 /**************************** MPEG-4 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
