1 /* ----------------------------------------------------------------------------- 2 Software License for The Fraunhofer FDK AAC Codec Library for Android 3 4 © Copyright 1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten 5 Forschung e.V. All rights reserved. 6 7 1. INTRODUCTION 8 The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software 9 that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding 10 scheme for digital audio. This FDK AAC Codec software is intended to be used on 11 a wide variety of Android devices. 12 13 AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient 14 general perceptual audio codecs. AAC-ELD is considered the best-performing 15 full-bandwidth communications codec by independent studies and is widely 16 deployed. AAC has been standardized by ISO and IEC as part of the MPEG 17 specifications. 18 19 Patent licenses for necessary patent claims for the FDK AAC Codec (including 20 those of Fraunhofer) may be obtained through Via Licensing 21 (www.vialicensing.com) or through the respective patent owners individually for 22 the purpose of encoding or decoding bit streams in products that are compliant 23 with the ISO/IEC MPEG audio standards. Please note that most manufacturers of 24 Android devices already license these patent claims through Via Licensing or 25 directly from the patent owners, and therefore FDK AAC Codec software may 26 already be covered under those patent licenses when it is used for those 27 licensed purposes only. 28 29 Commercially-licensed AAC software libraries, including floating-point versions 30 with enhanced sound quality, are also available from Fraunhofer. Users are 31 encouraged to check the Fraunhofer website for additional applications 32 information and documentation. 33 34 2. COPYRIGHT LICENSE 35 36 Redistribution and use in source and binary forms, with or without modification, 37 are permitted without payment of copyright license fees provided that you 38 satisfy the following conditions: 39 40 You must retain the complete text of this software license in redistributions of 41 the FDK AAC Codec or your modifications thereto in source code form. 42 43 You must retain the complete text of this software license in the documentation 44 and/or other materials provided with redistributions of the FDK AAC Codec or 45 your modifications thereto in binary form. You must make available free of 46 charge copies of the complete source code of the FDK AAC Codec and your 47 modifications thereto to recipients of copies in binary form. 48 49 The name of Fraunhofer may not be used to endorse or promote products derived 50 from this library without prior written permission. 51 52 You may not charge copyright license fees for anyone to use, copy or distribute 53 the FDK AAC Codec software or your modifications thereto. 54 55 Your modified versions of the FDK AAC Codec must carry prominent notices stating 56 that you changed the software and the date of any change. For modified versions 57 of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android" 58 must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK 59 AAC Codec Library for Android." 60 61 3. NO PATENT LICENSE 62 63 NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without 64 limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE. 65 Fraunhofer provides no warranty of patent non-infringement with respect to this 66 software. 67 68 You may use this FDK AAC Codec software or modifications thereto only for 69 purposes that are authorized by appropriate patent licenses. 70 71 4. DISCLAIMER 72 73 This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright 74 holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, 75 including but not limited to the implied warranties of merchantability and 76 fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR 77 CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, 78 or consequential damages, including but not limited to procurement of substitute 79 goods or services; loss of use, data, or profits, or business interruption, 80 however caused and on any theory of liability, whether in contract, strict 81 liability, or tort (including negligence), arising in any way out of the use of 82 this software, even if advised of the possibility of such damage. 83 84 5. CONTACT INFORMATION 85 86 Fraunhofer Institute for Integrated Circuits IIS 87 Attention: Audio and Multimedia Departments - FDK AAC LL 88 Am Wolfsmantel 33 89 91058 Erlangen, Germany 90 91 www.iis.fraunhofer.de/amm 92 amm-info@iis.fraunhofer.de 93 ----------------------------------------------------------------------------- */ 94 95 /**************************** AAC encoder library ****************************** 96 97 Author(s): M. Lohwasser 98 99 Description: 100 101 *******************************************************************************/ 102 103 /** 104 * \file aacenc_lib.h 105 * \brief FDK AAC Encoder library interface header file. 106 * 107 \mainpage Introduction 108 109 \section Scope 110 111 This document describes the high-level interface and usage of the ISO/MPEG-2/4 112 AAC Encoder library developed by the Fraunhofer Institute for Integrated 113 Circuits (IIS). 114 115 The library implements encoding on the basis of the MPEG-2 and MPEG-4 AAC 116 Low-Complexity standard, and depending on the library's configuration, MPEG-4 117 High-Efficiency AAC v2 and/or AAC-ELD standard. 118 119 All references to SBR (Spectral Band Replication) are only applicable to HE-AAC 120 or AAC-ELD versions of the library. All references to PS (Parametric Stereo) are 121 only applicable to HE-AAC v2 versions of the library. 122 123 \section encBasics Encoder Basics 124 125 This document can only give a rough overview about the ISO/MPEG-2 and ISO/MPEG-4 126 AAC audio coding standard. To understand all the terms in this document, you are 127 encouraged to read the following documents. 128 129 - ISO/IEC 13818-7 (MPEG-2 AAC), which defines the syntax of MPEG-2 AAC audio 130 bitstreams. 131 - ISO/IEC 14496-3 (MPEG-4 AAC, subparts 1 and 4), which defines the syntax of 132 MPEG-4 AAC audio bitstreams. 133 - Lutzky, Schuller, Gayer, Krämer, Wabnik, "A guideline to audio codec 134 delay", 116th AES Convention, May 8, 2004 135 136 MPEG Advanced Audio Coding is based on a time-to-frequency mapping of the 137 signal. The signal is partitioned into overlapping portions and transformed into 138 frequency domain. The spectral components are then quantized and coded. \n An 139 MPEG-2 or MPEG-4 AAC audio bitstream is composed of frames. Contrary to MPEG-1/2 140 Layer-3 (mp3), the length of individual frames is not restricted to a fixed 141 number of bytes, but can take on any length between 1 and 768 bytes. 142 143 144 \page LIBUSE Library Usage 145 146 \section InterfaceDescription API Files 147 148 All API header files are located in the folder /include of the release package. 149 All header files are provided for usage in C/C++ programs. The AAC encoder 150 library API functions are located in aacenc_lib.h. 151 152 In binary releases the encoder core resides in statically linkable libraries 153 called for example libAACenc.a/libFDK.a (LINUX) or FDK_fastaaclib.lib (MS Visual 154 C++) for the plain AAC-LC core encoder and libSBRenc.a (LINUX) or 155 FDK_sbrEncLib.lib (MS Visual C++) for the SBR (Spectral Band Replication) and PS 156 (Parametric Stereo) modules. 157 158 \section CallingSequence Calling Sequence 159 160 For encoding of ISO/MPEG-2/4 AAC bitstreams the following sequence is mandatory. 161 Input read and output write functions as well as the corresponding open and 162 close functions are left out, since they may be implemented differently 163 according to the user's specific requirements. The example implementation uses 164 file-based input/output. 165 166 -# Call aacEncOpen() to allocate encoder instance with required \ref encOpen 167 "configuration". \code HANDLE_AACENCODER hAacEncoder = NULL; if ( (ErrorStatus = 168 aacEncOpen(&hAacEncoder,0,0)) != AACENC_OK ) { \endcode 169 -# Call aacEncoder_SetParam() for each parameter to be set. AOT, samplingrate, 170 channelMode, bitrate and transport type are \ref encParams "mandatory". \code 171 ErrorStatus = aacEncoder_SetParam(hAacEncoder, parameter, value); 172 \endcode 173 -# Call aacEncEncode() with NULL parameters to \ref encReconf "initialize" 174 encoder instance with present parameter set. \code ErrorStatus = 175 aacEncEncode(hAacEncoder, NULL, NULL, NULL, NULL); \endcode 176 -# Call aacEncInfo() to retrieve a configuration data block to be transmitted 177 out of band. This is required when using RFC3640 or RFC3016 like transport. 178 \code 179 AACENC_InfoStruct encInfo; 180 aacEncInfo(hAacEncoder, &encInfo); 181 \endcode 182 -# Encode input audio data in loop. 183 \code 184 do 185 { 186 \endcode 187 Feed \ref feedInBuf "input buffer" with new audio data and provide input/output 188 \ref bufDes "arguments" to aacEncEncode(). \code ErrorStatus = 189 aacEncEncode(hAacEncoder, &inBufDesc, &outBufDesc, &inargs, &outargs); \endcode 190 Write \ref writeOutData "output data" to file or audio device. 191 \code 192 } while (ErrorStatus==AACENC_OK); 193 \endcode 194 -# Call aacEncClose() and destroy encoder instance. 195 \code 196 aacEncClose(&hAacEncoder); 197 \endcode 198 199 200 \section encOpen Encoder Instance Allocation 201 202 The assignment of the aacEncOpen() function is very flexible and can be used in 203 the following way. 204 - If the amount of memory consumption is not an issue, the encoder instance can 205 be allocated for the maximum number of possible audio channels (for example 6 or 206 8) with the full functional range supported by the library. This is the default 207 open procedure for the AAC encoder if memory consumption does not need to be 208 minimized. \code aacEncOpen(&hAacEncoder,0,0) \endcode 209 - If the required MPEG-4 AOTs do not call for the full functional range of the 210 library, encoder modules can be allocated selectively. \verbatim 211 ------------------------------------------------------ 212 AAC | SBR | PS | MD | FLAGS | value 213 -----+-----+-----+----+-----------------------+------- 214 X | - | - | - | (0x01) | 0x01 215 X | X | - | - | (0x01|0x02) | 0x03 216 X | X | X | - | (0x01|0x02|0x04) | 0x07 217 X | - | - | X | (0x01 |0x10) | 0x11 218 X | X | - | X | (0x01|0x02 |0x10) | 0x13 219 X | X | X | X | (0x01|0x02|0x04|0x10) | 0x17 220 ------------------------------------------------------ 221 - AAC: Allocate AAC Core Encoder module. 222 - SBR: Allocate Spectral Band Replication module. 223 - PS: Allocate Parametric Stereo module. 224 - MD: Allocate Meta Data module within AAC encoder. 225 \endverbatim 226 \code aacEncOpen(&hAacEncoder,value,0) \endcode 227 - Specifying the maximum number of channels to be supported in the encoder 228 instance can be done as follows. 229 - For example allocate an encoder instance which supports 2 channels for all 230 supported AOTs. The library itself may be capable of encoding up to 6 or 8 231 channels but in this example only 2 channel encoding is required and thus only 232 buffers for 2 channels are allocated to save data memory. \code 233 aacEncOpen(&hAacEncoder,0,2) \endcode 234 - Additionally the maximum number of supported channels in the SBR module can 235 be denoted separately.\n In this example the encoder instance provides a maximum 236 of 6 channels out of which up to 2 channels support SBR. This encoder instance 237 can produce for example 5.1 channel AAC-LC streams or stereo HE-AAC (v2) 238 streams. HE-AAC 5.1 multi channel is not possible since only 2 out of 6 channels 239 support SBR, which saves data memory. \code aacEncOpen(&hAacEncoder,0,6|(2<<8)) 240 \endcode \n 241 242 \section bufDes Input/Output Arguments 243 244 \subsection allocIOBufs Provide Buffer Descriptors 245 In the present encoder API, the input and output buffers are described with \ref 246 AACENC_BufDesc "buffer descriptors". This mechanism allows a flexible handling 247 of input and output buffers without impact to the actual encoding call. Optional 248 buffers are necessary e.g. for ancillary data, meta data input or additional 249 output buffers describing superframing data in DAB+ or DRM+.\n At least one 250 input buffer for audio input data and one output buffer for bitstream data must 251 be allocated. The input buffer size can be a user defined multiple of the number 252 of input channels. PCM input data will be copied from the user defined PCM 253 buffer to an internal input buffer and so input data can be less than one AAC 254 audio frame. The output buffer size should be 6144 bits per channel excluding 255 the LFE channel. If the output data does not fit into the provided buffer, an 256 AACENC_ERROR will be returned by aacEncEncode(). \code static INT_PCM 257 inputBuffer[8*2048]; static UCHAR ancillaryBuffer[50]; static 258 AACENC_MetaData metaDataSetup; static UCHAR outputBuffer[8192]; 259 \endcode 260 261 All input and output buffer must be clustered in input and output buffer arrays. 262 \code 263 static void* inBuffer[] = { inputBuffer, ancillaryBuffer, &metaDataSetup 264 }; static INT inBufferIds[] = { IN_AUDIO_DATA, IN_ANCILLRY_DATA, 265 IN_METADATA_SETUP }; static INT inBufferSize[] = { sizeof(inputBuffer), 266 sizeof(ancillaryBuffer), sizeof(metaDataSetup) }; static INT inBufferElSize[] 267 = { sizeof(INT_PCM), sizeof(UCHAR), sizeof(AACENC_MetaData) }; 268 269 static void* outBuffer[] = { outputBuffer }; 270 static INT outBufferIds[] = { OUT_BITSTREAM_DATA }; 271 static INT outBufferSize[] = { sizeof(outputBuffer) }; 272 static INT outBufferElSize[] = { sizeof(UCHAR) }; 273 \endcode 274 275 Allocate buffer descriptors 276 \code 277 AACENC_BufDesc inBufDesc; 278 AACENC_BufDesc outBufDesc; 279 \endcode 280 281 Initialize input buffer descriptor 282 \code 283 inBufDesc.numBufs = sizeof(inBuffer)/sizeof(void*); 284 inBufDesc.bufs = (void**)&inBuffer; 285 inBufDesc.bufferIdentifiers = inBufferIds; 286 inBufDesc.bufSizes = inBufferSize; 287 inBufDesc.bufElSizes = inBufferElSize; 288 \endcode 289 290 Initialize output buffer descriptor 291 \code 292 outBufDesc.numBufs = sizeof(outBuffer)/sizeof(void*); 293 outBufDesc.bufs = (void**)&outBuffer; 294 outBufDesc.bufferIdentifiers = outBufferIds; 295 outBufDesc.bufSizes = outBufferSize; 296 outBufDesc.bufElSizes = outBufferElSize; 297 \endcode 298 299 \subsection argLists Provide Input/Output Argument Lists 300 The input and output arguments of an aacEncEncode() call are described in 301 argument structures. \code AACENC_InArgs inargs; AACENC_OutArgs outargs; 302 \endcode 303 304 \section feedInBuf Feed Input Buffer 305 The input buffer should be handled as a modulo buffer. New audio data in the 306 form of pulse-code- modulated samples (PCM) must be read from external and be 307 fed to the input buffer depending on its fill level. The required sample bitrate 308 (represented by the data type INT_PCM which is 16, 24 or 32 bits wide) is fixed 309 and depends on library configuration (usually 16 bit). \code inargs.numInSamples 310 += WAV_InputRead ( wavIn, &inputBuffer[inargs.numInSamples], 311 FDKmin(encInfo.inputChannels*encInfo.frameLength, 312 sizeof(inputBuffer) / 313 sizeof(INT_PCM)-inargs.numInSamples), 314 SAMPLE_BITS 315 ); 316 \endcode 317 318 After the encoder's internal buffer is fed with incoming audio samples, and 319 aacEncEncode() processed the new input data, update/move remaining samples in 320 input buffer, simulating a modulo buffer: \code if (outargs.numInSamples>0) { 321 FDKmemmove( inputBuffer, 322 &inputBuffer[outargs.numInSamples], 323 sizeof(INT_PCM)*(inargs.numInSamples-outargs.numInSamples) ); 324 inargs.numInSamples -= outargs.numInSamples; 325 } 326 \endcode 327 328 \section writeOutData Output Bitstream Data 329 If any AAC bitstream data is available, write it to output file or device. This 330 can be done once the following condition is true: \code if 331 (outargs.numOutBytes>0) { 332 333 } 334 \endcode 335 336 If you use file I/O then for example call mpegFileWrite_Write() from the library 337 libMpegFileWrite \code mpegFileWrite_Write(hMpegFile, outputBuffer, 338 outargs.numOutBytes, aacEncoder_GetParam(hAacEncoder, AACENC_GRANULE_LENGTH)); 339 \endcode 340 341 \section cfgMetaData Meta Data Configuration 342 343 If the present library is configured with Metadata support, it is possible to 344 insert meta data side info into the generated audio bitstream while encoding. 345 346 To work with meta data the encoder instance has to be \ref encOpen "allocated" 347 with meta data support. The meta data mode must be be configured with the 348 ::AACENC_METADATA_MODE parameter and aacEncoder_SetParam() function. \code 349 aacEncoder_SetParam(hAacEncoder, AACENC_METADATA_MODE, 0-3); \endcode 350 351 This configuration indicates how to embed meta data into bitstrem. Either no 352 insertion, MPEG or ETSI style. The meta data itself must be specified within the 353 meta data setup structure AACENC_MetaData. 354 355 Changing one of the AACENC_MetaData setup parameters can be achieved from 356 outside the library within ::IN_METADATA_SETUP input buffer. There is no need to 357 supply meta data setup structure every frame. If there is no new meta setup data 358 available, the encoder uses the previous setup or the default configuration in 359 initial state. 360 361 In general the audio compressor and limiter within the encoder library can be 362 configured with the ::AACENC_METADATA_DRC_PROFILE parameter 363 AACENC_MetaData::drc_profile and and AACENC_MetaData::comp_profile. 364 \n 365 366 \section encReconf Encoder Reconfiguration 367 368 The encoder library allows reconfiguration of the encoder instance with new 369 settings continuously between encoding frames. Each parameter to be changed must 370 be set with a single aacEncoder_SetParam() call. The internal status of each 371 parameter can be retrieved with an aacEncoder_GetParam() call.\n There is no 372 stand-alone reconfiguration function available. When parameters were modified 373 from outside the library, an internal control mechanism triggers the necessary 374 reconfiguration process which will be applied at the beginning of the following 375 aacEncEncode() call. This state can be observed from external via the 376 AACENC_INIT_STATUS and aacEncoder_GetParam() function. The reconfiguration 377 process can also be applied immediately when all parameters of an aacEncEncode() 378 call are NULL with a valid encoder handle.\n\n The internal reconfiguration 379 process can be controlled from extern with the following access. \code 380 aacEncoder_SetParam(hAacEncoder, AACENC_CONTROL_STATE, AACENC_CTRLFLAGS); 381 \endcode 382 383 384 \section encParams Encoder Parametrization 385 386 All parameteres listed in ::AACENC_PARAM can be modified within an encoder 387 instance. 388 389 \subsection encMandatory Mandatory Encoder Parameters 390 The following parameters must be specified when the encoder instance is 391 initialized. \code aacEncoder_SetParam(hAacEncoder, AACENC_AOT, value); 392 aacEncoder_SetParam(hAacEncoder, AACENC_BITRATE, value); 393 aacEncoder_SetParam(hAacEncoder, AACENC_SAMPLERATE, value); 394 aacEncoder_SetParam(hAacEncoder, AACENC_CHANNELMODE, value); 395 \endcode 396 Beyond that is an internal auto mode which preinitizializes the ::AACENC_BITRATE 397 parameter if the parameter was not set from extern. The bitrate depends on the 398 number of effective channels and sampling rate and is determined as follows. 399 \code 400 AAC-LC (AOT_AAC_LC): 1.5 bits per sample 401 HE-AAC (AOT_SBR): 0.625 bits per sample (dualrate sbr) 402 HE-AAC (AOT_SBR): 1.125 bits per sample (downsampled sbr) 403 HE-AAC v2 (AOT_PS): 0.5 bits per sample 404 \endcode 405 406 \subsection channelMode Channel Mode Configuration 407 The input audio data is described with the ::AACENC_CHANNELMODE parameter in the 408 aacEncoder_SetParam() call. It is not possible to use the encoder instance with 409 a 'number of input channels' argument. Instead, the channelMode must be set as 410 follows. \code aacEncoder_SetParam(hAacEncoder, AACENC_CHANNELMODE, value); 411 \endcode The parameter is specified in ::CHANNEL_MODE and can be mapped from the 412 number of input channels in the following way. \code CHANNEL_MODE chMode = 413 MODE_INVALID; 414 415 switch (nChannels) { 416 case 1: chMode = MODE_1; break; 417 case 2: chMode = MODE_2; break; 418 case 3: chMode = MODE_1_2; break; 419 case 4: chMode = MODE_1_2_1; break; 420 case 5: chMode = MODE_1_2_2; break; 421 case 6: chMode = MODE_1_2_2_1; break; 422 case 7: chMode = MODE_6_1; break; 423 case 8: chMode = MODE_7_1_BACK; break; 424 default: 425 chMode = MODE_INVALID; 426 } 427 return chMode; 428 \endcode 429 430 \subsection bitreservoir Bitreservoir Configuration 431 In AAC, the default bitreservoir configuration depends on the chosen bitrate per 432 frame and the number of effective channels. The size can be determined as below. 433 \f[ 434 bitreservoir = nEffChannels*6144 - (bitrate*framelength/samplerate) 435 \f] 436 Due to audio quality concerns it is not recommended to change the bitreservoir 437 size to a lower value than the default setting! However, for minimizing the 438 delay for streaming applications or for achieving a constant size of the 439 bitstream packages in each frame, it may be necessaray to change the 440 bitreservoir size. This can be done with the ::AACENC_PEAK_BITRATE parameter. 441 \code 442 aacEncoder_SetParam(hAacEncoder, AACENC_PEAK_BITRATE, value); 443 \endcode 444 By setting ::AACENC_BITRATEMODE to fixed framing, the bitreservoir is disabled. 445 A disabled bitreservoir results in a constant size for each bitstream package. 446 Please note that especially at lower bitrates a disabled bitreservoir can 447 downgrade the audio quality considerably! The default bitreservoir configuration 448 can be achieved as follows. \code aacEncoder_SetParam(hAacEncoder, 449 AACENC_BITRESERVOIR, -1); \endcode 450 451 To achieve acceptable audio quality with a reduced bitreservoir size setting at 452 least 1000 bits per audio channel is recommended. For a multichannel audio file 453 with 5.1 channels the bitreservoir reduced to 5000 bits results in acceptable 454 audio quality. 455 456 457 \subsection vbrmode Variable Bitrate Mode 458 The encoder provides various Variable Bitrate Modes that differ in audio quality 459 and average overall bitrate. The given values are averages over time, different 460 encoder settings and strongly depend on the type of audio signal. The VBR 461 configurations can be adjusted via ::AACENC_BITRATEMODE encoder parameter. 462 \verbatim 463 -------------------------------------------- 464 VBR_MODE | Approx. Bitrate in kbps/channel 465 | AAC-LC | AAC-LD/AC_ELD 466 ----------+---------------+----------------- 467 VBR_1 | 32 - 48 | 32 - 56 468 VBR_2 | 40 - 56 | 40 - 64 469 VBR_3 | 48 - 64 | 48 - 72 470 VBR_4 | 64 - 80 | 64 - 88 471 VBR_5 | 96 - 120 | 112 - 144 472 -------------------------------------------- 473 \endverbatim 474 The bitrate ranges apply for individual audio channels. In case of multichannel 475 configurations the average bitrate might be estimated by multiplying with the 476 number of effective channels. This corresponds to all audio input channels 477 exclusively the low frequency channel. At configurations which are making use of 478 downmix modules the AAC core channels respectively downmix channels shall be 479 considered. For ::AACENC_AOT which are using SBR, the average bitrate can be 480 estimated by using the ratio of 0.5 for dualrate SBR and 0.75 for downsampled 481 SBR configurations. 482 483 484 \subsection encQual Audio Quality Considerations 485 The default encoder configuration is suggested to be used. Encoder tools such as 486 TNS and PNS are activated by default and are internally controlled (see \ref 487 BEHAVIOUR_TOOLS). 488 489 There is an additional quality parameter called ::AACENC_AFTERBURNER. In the 490 default configuration this quality switch is deactivated because it would cause 491 a workload increase which might be significant. If workload is not an issue in 492 the application we recommended to activate this feature. \code 493 aacEncoder_SetParam(hAacEncoder, AACENC_AFTERBURNER, 0/1); \endcode 494 495 \subsection encELD ELD Auto Configuration Mode 496 For ELD configuration a so called auto configurator is available which 497 configures SBR and the SBR ratio by itself. The configurator is used when the 498 encoder parameter ::AACENC_SBR_MODE and ::AACENC_SBR_RATIO are not set 499 explicitly. 500 501 Based on sampling rate and chosen bitrate a reasonable SBR configuration will be 502 used. \verbatim 503 ------------------------------------------------------------------ 504 Sampling Rate | Total Bitrate | No. of | SBR | SBR Ratio 505 [kHz] | [bit/s] | Chan | | 506 | | | | 507 ---------------+-----------------+--------+-----+----------------- 508 ]min, 16[ | min - max | 1 | off | --- 509 ---------------+-----------------+--------------+----------------- 510 [16] | min - 27999 | 1 | on | downsampled SBR 511 | 28000 - max | 1 | off | --- 512 ---------------+-----------------+--------------+----------------- 513 ]16 - 24] | min - 39999 | 1 | on | downsampled SBR 514 | 40000 - max | 1 | off | --- 515 ---------------+-----------------+--------------+----------------- 516 ]24 - 32] | min - 27999 | 1 | on | dualrate SBR 517 | 28000 - 55999 | 1 | on | downsampled SBR 518 | 56000 - max | 1 | off | --- 519 ---------------+-----------------+--------------+----------------- 520 ]32 - 44.1] | min - 63999 | 1 | on | dualrate SBR 521 | 64000 - max | 1 | off | --- 522 ---------------+-----------------+--------------+----------------- 523 ]44.1 - 48] | min - 63999 | 1 | on | dualrate SBR 524 | 64000 - max | 1 | off | --- 525 | | | | 526 ---------------+-----------------+--------+-----+----------------- 527 ]min, 16[ | min - max | 2 | off | --- 528 ---------------+-----------------+--------------+----------------- 529 [16] | min - 31999 | 2 | on | downsampled SBR 530 | 32000 - 63999 | 2 | on | downsampled SBR 531 | 64000 - max | 2 | off | --- 532 ---------------+-----------------+--------------+----------------- 533 ]16 - 24] | min - 47999 | 2 | on | downsampled SBR 534 | 48000 - 79999 | 2 | on | downsampled SBR 535 | 80000 - max | 2 | off | --- 536 ---------------+-----------------+--------------+----------------- 537 ]24 - 32] | min - 31999 | 2 | on | dualrate SBR 538 | 32000 - 67999 | 2 | on | dualrate SBR 539 | 68000 - 95999 | 2 | on | downsampled SBR 540 | 96000 - max | 2 | off | --- 541 ---------------+-----------------+--------------+----------------- 542 ]32 - 44.1] | min - 43999 | 2 | on | dualrate SBR 543 | 44000 - 127999 | 2 | on | dualrate SBR 544 | 128000 - max | 2 | off | --- 545 ---------------+-----------------+--------------+----------------- 546 ]44.1 - 48] | min - 43999 | 2 | on | dualrate SBR 547 | 44000 - 127999 | 2 | on | dualrate SBR 548 | 128000 - max | 2 | off | --- 549 | | | 550 ------------------------------------------------------------------ 551 \endverbatim 552 553 \subsection encDsELD Reduced Delay (Downscaled) Mode 554 The downscaled mode of AAC-ELD reduces the algorithmic delay of AAC-ELD by 555 virtually increasing the sampling rate. When using the downscaled mode, the 556 bitrate should be increased for keeping the same audio quality level. For common 557 signals, the bitrate should be increased by 25% for a downscale factor of 2. 558 559 Currently, downscaling factors 2 and 4 are supported. 560 To enable the downscaled mode in the encoder, the framelength parameter 561 AACENC_GRANULE_LENGTH must be set accordingly to 256 or 240 for a downscale 562 factor of 2 or 128 or 120 for a downscale factor of 4. The default values of 512 563 or 480 mean that no downscaling is applied. \code 564 aacEncoder_SetParam(hAacEncoder, AACENC_GRANULE_LENGTH, 256); 565 aacEncoder_SetParam(hAacEncoder, AACENC_GRANULE_LENGTH, 128); 566 \endcode 567 568 Downscaled bitstreams are fully backwards compatible. However, the legacy 569 decoder needs to support high sample rate, e.g. 96kHz. The signaled sampling 570 rate is multiplied by the downscale factor. Although not required, downscaling 571 should be applied when decoding downscaled bitstreams. It reduces CPU workload 572 and the output will have the same sampling rate as the input. In an ideal 573 configuration both encoder and decoder should run with the same downscale 574 factor. 575 576 The following table shows approximate filter bank delays in ms for common 577 sampling rates(sr) at framesize(fs), and downscale factor(dsf), based on this 578 formula: \f[ 1000 * fs / (dsf * sr) \f] 579 580 \verbatim 581 -------------------------------------- 582 | 512/2 | 512/4 | 480/2 | 480/4 583 ------+-------+-------+-------+------- 584 22050 | 17.41 | 8.71 | 16.33 | 8.16 585 32000 | 12.00 | 6.00 | 11.25 | 5.62 586 44100 | 8.71 | 4.35 | 8.16 | 4.08 587 48000 | 8.00 | 4.00 | 7.50 | 3.75 588 -------------------------------------- 589 \endverbatim 590 591 \section audiochCfg Audio Channel Configuration 592 The MPEG standard refers often to the so-called Channel Configuration. This 593 Channel Configuration is used for a fixed Channel Mapping. The configurations 594 1-7 and 11,12,14 are predefined in MPEG standard and used for implicit 595 signalling within the encoded bitstream. For user defined Configurations the 596 Channel Configuration is set to 0 and the Channel Mapping must be explecitly 597 described with an appropriate Program Config Element. The present Encoder 598 implementation does not allow the user to configure this Channel Configuration 599 from extern. The Encoder implementation supports fixed Channel Modes which are 600 mapped to Channel Configuration as follow. \verbatim 601 ---------------------------------------------------------------------------------------- 602 ChannelMode | ChCfg | Height | front_El | side_El | back_El | 603 lfe_El 604 -----------------------+-------+--------+---------------+----------+----------+--------- 605 MODE_1 | 1 | NORM | SCE | | | 606 MODE_2 | 2 | NORM | CPE | | | 607 MODE_1_2 | 3 | NORM | SCE, CPE | | | 608 MODE_1_2_1 | 4 | NORM | SCE, CPE | | SCE | 609 MODE_1_2_2 | 5 | NORM | SCE, CPE | | CPE | 610 MODE_1_2_2_1 | 6 | NORM | SCE, CPE | | CPE | 611 LFE MODE_1_2_2_2_1 | 7 | NORM | SCE, CPE, CPE | | CPE 612 | LFE MODE_6_1 | 11 | NORM | SCE, CPE | | CPE, 613 SCE | LFE MODE_7_1_BACK | 12 | NORM | SCE, CPE | | 614 CPE, CPE | LFE 615 -----------------------+-------+--------+---------------+----------+----------+--------- 616 MODE_7_1_TOP_FRONT | 14 | NORM | SCE, CPE | | CPE | 617 LFE | | TOP | CPE | | | 618 -----------------------+-------+--------+---------------+----------+----------+--------- 619 MODE_7_1_REAR_SURROUND | 0 | NORM | SCE, CPE | | CPE, CPE | 620 LFE MODE_7_1_FRONT_CENTER | 0 | NORM | SCE, CPE, CPE | | CPE 621 | LFE 622 ---------------------------------------------------------------------------------------- 623 - NORM: Normal Height Layer. - TOP: Top Height Layer. - BTM: Bottom Height 624 Layer. 625 - SCE: Single Channel Element. - CPE: Channel Pair. - LFE: Low Frequency 626 Element. \endverbatim 627 628 The Table describes all fixed Channel Elements for each Channel Mode which are 629 assigned to a speaker arrangement. The arrangement includes front, side, back 630 and lfe Audio Channel Elements in the normal height layer, possibly followed by 631 front, side, and back elements in the top and bottom layer (Channel 632 Configuration 14). \n This mapping of Audio Channel Elements is defined in MPEG 633 standard for Channel Config 1-7 and 11,12,14.\n In case of Channel Config 0 or 634 writing matrix mixdown coefficients, the encoder enables the writing of Program 635 Config Element itself as described in \ref encPCE. The configuration used in 636 Program Config Element refers to the denoted Table.\n Beside the Channel Element 637 assignment the Channel Modes are resposible for audio input data channel 638 mapping. The Channel Mapping of the audio data depends on the selected 639 ::AACENC_CHANNELORDER which can be MPEG or WAV like order.\n Following table 640 describes the complete channel mapping for both Channel Order configurations. 641 \verbatim 642 --------------------------------------------------------------------------------------- 643 ChannelMode | MPEG-Channelorder | WAV-Channelorder 644 -----------------------+---+---+---+---+---+---+---+---+---+---+---+---+---+---+---+--- 645 MODE_1 | 0 | | | | | | | | 0 | | | | | | 646 | MODE_2 | 0 | 1 | | | | | | | 0 | 1 | | | | 647 | | MODE_1_2 | 0 | 1 | 2 | | | | | | 2 | 0 | 1 | | 648 | | | MODE_1_2_1 | 0 | 1 | 2 | 3 | | | | | 2 | 0 | 1 | 3 649 | | | | MODE_1_2_2 | 0 | 1 | 2 | 3 | 4 | | | | 2 | 0 | 1 650 | 3 | 4 | | | MODE_1_2_2_1 | 0 | 1 | 2 | 3 | 4 | 5 | | | 2 | 0 651 | 1 | 4 | 5 | 3 | | MODE_1_2_2_2_1 | 0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 | 2 652 | 6 | 7 | 0 | 1 | 4 | 5 | 3 MODE_6_1 | 0 | 1 | 2 | 3 | 4 | 5 | 6 | 653 | 2 | 0 | 1 | 4 | 5 | 6 | 3 | MODE_7_1_BACK | 0 | 1 | 2 | 3 | 4 | 5 | 6 654 | 7 | 2 | 0 | 1 | 6 | 7 | 4 | 5 | 3 MODE_7_1_TOP_FRONT | 0 | 1 | 2 | 3 | 4 | 655 5 | 6 | 7 | 2 | 0 | 1 | 4 | 5 | 3 | 6 | 7 656 -----------------------+---+---+---+---+---+---+---+---+---+---+---+---+---+---+---+--- 657 MODE_7_1_REAR_SURROUND | 0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 | 2 | 0 | 1 | 6 | 7 | 4 | 658 5 | 3 MODE_7_1_FRONT_CENTER | 0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 | 2 | 6 | 7 | 0 | 1 659 | 4 | 5 | 3 660 --------------------------------------------------------------------------------------- 661 \endverbatim 662 663 The denoted mapping is important for correct audio channel assignment when using 664 MPEG or WAV ordering. The incoming audio channels are distributed MPEG like 665 starting at the front channels and ending at the back channels. The distribution 666 is used as described in Table concering Channel Config and fix channel elements. 667 Please see the following example for clarification. 668 669 \verbatim 670 Example: MODE_1_2_2_1 - WAV-Channelorder 5.1 671 ------------------------------------------ 672 Input Channel | Coder Channel 673 --------------------+--------------------- 674 2 (front center) | 0 (SCE channel) 675 0 (left center) | 1 (1st of 1st CPE) 676 1 (right center) | 2 (2nd of 1st CPE) 677 4 (left surround) | 3 (1st of 2nd CPE) 678 5 (right surround) | 4 (2nd of 2nd CPE) 679 3 (LFE) | 5 (LFE) 680 ------------------------------------------ 681 \endverbatim 682 683 684 \section suppBitrates Supported Bitrates 685 686 The FDK AAC Encoder provides a wide range of supported bitrates. 687 The minimum and maximum allowed bitrate depends on the Audio Object Type. For 688 AAC-LC the minimum bitrate is the bitrate that is required to write the most 689 basic and minimal valid bitstream. It consists of the bitstream format header 690 information and other static/mandatory information within the AAC payload. The 691 maximum AAC framesize allowed by the MPEG-4 standard determines the maximum 692 allowed bitrate for AAC-LC. For HE-AAC and HE-AAC v2 a library internal look-up 693 table is used. 694 695 A good working point in terms of audio quality, sampling rate and bitrate, is at 696 1 to 1.5 bits/audio sample for AAC-LC, 0.625 bits/audio sample for dualrate 697 HE-AAC, 1.125 bits/audio sample for downsampled HE-AAC and 0.5 bits/audio sample 698 for HE-AAC v2. For example for one channel with a sampling frequency of 48 kHz, 699 the range from 48 kbit/s to 72 kbit/s achieves reasonable audio quality for 700 AAC-LC. 701 702 For HE-AAC and HE-AAC v2 the lowest possible audio input sampling frequency is 703 16 kHz because then the AAC-LC core encoder operates in dual rate mode at its 704 lowest possible sampling frequency, which is 8 kHz. HE-AAC v2 requires stereo 705 input audio data. 706 707 Please note that in HE-AAC or HE-AAC v2 mode the encoder supports much higher 708 bitrates than are appropriate for HE-AAC or HE-AAC v2. For example, at a bitrate 709 of more than 64 kbit/s for a stereo audio signal at 44.1 kHz it usually makes 710 sense to use AAC-LC, which will produce better audio quality at that bitrate 711 than HE-AAC or HE-AAC v2. 712 713 \section reommendedConfig Recommended Sampling Rate and Bitrate Combinations 714 715 The following table provides an overview of recommended encoder configuration 716 parameters which we determined by virtue of numerous listening tests. 717 718 \subsection reommendedConfigLC AAC-LC, HE-AAC, HE-AACv2 in Dualrate SBR mode. 719 \verbatim 720 ----------------------------------------------------------------------------------- 721 Audio Object Type | Bit Rate Range | Supported | Preferred | No. 722 of | [bit/s] | Sampling Rates | Sampl. | Chan. | 723 | [kHz] | Rate | | | 724 | [kHz] | 725 -------------------+------------------+-----------------------+------------+------- 726 AAC LC + SBR + PS | 8000 - 11999 | 22.05, 24.00 | 24.00 | 2 727 AAC LC + SBR + PS | 12000 - 17999 | 32.00 | 32.00 | 2 728 AAC LC + SBR + PS | 18000 - 39999 | 32.00, 44.10, 48.00 | 44.10 | 2 729 AAC LC + SBR + PS | 40000 - 64000 | 32.00, 44.10, 48.00 | 48.00 | 2 730 -------------------+------------------+-----------------------+------------+------- 731 AAC LC + SBR | 8000 - 11999 | 22.05, 24.00 | 24.00 | 1 732 AAC LC + SBR | 12000 - 17999 | 32.00 | 32.00 | 1 733 AAC LC + SBR | 18000 - 39999 | 32.00, 44.10, 48.00 | 44.10 | 1 734 AAC LC + SBR | 40000 - 64000 | 32.00, 44.10, 48.00 | 48.00 | 1 735 -------------------+------------------+-----------------------+------------+------- 736 AAC LC + SBR | 16000 - 27999 | 32.00, 44.10, 48.00 | 32.00 | 2 737 AAC LC + SBR | 28000 - 63999 | 32.00, 44.10, 48.00 | 44.10 | 2 738 AAC LC + SBR | 64000 - 128000 | 32.00, 44.10, 48.00 | 48.00 | 2 739 -------------------+------------------+-----------------------+------------+------- 740 AAC LC + SBR | 64000 - 69999 | 32.00, 44.10, 48.00 | 32.00 | 741 5, 5.1 AAC LC + SBR | 70000 - 239999 | 32.00, 44.10, 48.00 | 44.10 742 | 5, 5.1 AAC LC + SBR | 240000 - 319999 | 32.00, 44.10, 48.00 | 743 48.00 | 5, 5.1 744 -------------------+------------------+-----------------------+------------+------- 745 AAC LC | 8000 - 15999 | 11.025, 12.00, 16.00 | 12.00 | 1 746 AAC LC | 16000 - 23999 | 16.00 | 16.00 | 1 747 AAC LC | 24000 - 31999 | 16.00, 22.05, 24.00 | 24.00 | 1 748 AAC LC | 32000 - 55999 | 32.00 | 32.00 | 1 749 AAC LC | 56000 - 160000 | 32.00, 44.10, 48.00 | 44.10 | 1 750 AAC LC | 160001 - 288000 | 48.00 | 48.00 | 1 751 -------------------+------------------+-----------------------+------------+------- 752 AAC LC | 16000 - 23999 | 11.025, 12.00, 16.00 | 12.00 | 2 753 AAC LC | 24000 - 31999 | 16.00 | 16.00 | 2 754 AAC LC | 32000 - 39999 | 16.00, 22.05, 24.00 | 22.05 | 2 755 AAC LC | 40000 - 95999 | 32.00 | 32.00 | 2 756 AAC LC | 96000 - 111999 | 32.00, 44.10, 48.00 | 32.00 | 2 757 AAC LC | 112000 - 320001 | 32.00, 44.10, 48.00 | 44.10 | 2 758 AAC LC | 320002 - 576000 | 48.00 | 48.00 | 2 759 -------------------+------------------+-----------------------+------------+------- 760 AAC LC | 160000 - 239999 | 32.00 | 32.00 | 761 5, 5.1 AAC LC | 240000 - 279999 | 32.00, 44.10, 48.00 | 32.00 762 | 5, 5.1 AAC LC | 280000 - 800000 | 32.00, 44.10, 48.00 | 763 44.10 | 5, 5.1 764 ----------------------------------------------------------------------------------- 765 \endverbatim \n 766 767 \subsection reommendedConfigLD AAC-LD, AAC-ELD, AAC-ELD with SBR in Dualrate SBR 768 mode. Unlike to HE-AAC configuration the SBR is not covered by ELD audio object 769 type and needs to be enabled explicitly. Use ::AACENC_SBR_MODE to configure SBR 770 and its samplingrate ratio with ::AACENC_SBR_RATIO parameter. \verbatim 771 ----------------------------------------------------------------------------------- 772 Audio Object Type | Bit Rate Range | Supported | Preferred | No. 773 of | [bit/s] | Sampling Rates | Sampl. | Chan. | 774 | [kHz] | Rate | | | 775 | [kHz] | 776 -------------------+------------------+-----------------------+------------+------- 777 ELD + SBR | 18000 - 24999 | 32.00 - 44.10 | 32.00 | 1 778 ELD + SBR | 25000 - 31999 | 32.00 - 48.00 | 32.00 | 1 779 ELD + SBR | 32000 - 64000 | 32.00 - 48.00 | 48.00 | 1 780 -------------------+------------------+-----------------------+------------+------- 781 ELD + SBR | 32000 - 51999 | 32.00 - 48.00 | 44.10 | 2 782 ELD + SBR | 52000 - 128000 | 32.00 - 48.00 | 48.00 | 2 783 -------------------+------------------+-----------------------+------------+------- 784 ELD + SBR | 78000 - 160000 | 32.00 - 48.00 | 48.00 | 3 785 -------------------+------------------+-----------------------+------------+------- 786 ELD + SBR | 104000 - 212000 | 32.00 - 48.00 | 48.00 | 4 787 -------------------+------------------+-----------------------+------------+------- 788 ELD + SBR | 130000 - 246000 | 32.00 - 48.00 | 48.00 | 789 5, 5.1 790 -------------------+------------------+-----------------------+------------+------- 791 LD, ELD | 16000 - 19999 | 16.00 - 24.00 | 16.00 | 1 792 LD, ELD | 20000 - 39999 | 16.00 - 32.00 | 24.00 | 1 793 LD, ELD | 40000 - 49999 | 22.05 - 32.00 | 32.00 | 1 794 LD, ELD | 50000 - 61999 | 24.00 - 44.10 | 32.00 | 1 795 LD, ELD | 62000 - 84999 | 32.00 - 48.00 | 44.10 | 1 796 LD, ELD | 85000 - 192000 | 44.10 - 48.00 | 48.00 | 1 797 -------------------+------------------+-----------------------+------------+------- 798 LD, ELD | 64000 - 75999 | 24.00 - 32.00 | 32.00 | 2 799 LD, ELD | 76000 - 97999 | 24.00 - 44.10 | 32.00 | 2 800 LD, ELD | 98000 - 135999 | 32.00 - 48.00 | 44.10 | 2 801 LD, ELD | 136000 - 384000 | 44.10 - 48.00 | 48.00 | 2 802 -------------------+------------------+-----------------------+------------+------- 803 LD, ELD | 96000 - 113999 | 24.00 - 32.00 | 32.00 | 3 804 LD, ELD | 114000 - 146999 | 24.00 - 44.10 | 32.00 | 3 805 LD, ELD | 147000 - 203999 | 32.00 - 48.00 | 44.10 | 3 806 LD, ELD | 204000 - 576000 | 44.10 - 48.00 | 48.00 | 3 807 -------------------+------------------+-----------------------+------------+------- 808 LD, ELD | 128000 - 151999 | 24.00 - 32.00 | 32.00 | 4 809 LD, ELD | 152000 - 195999 | 24.00 - 44.10 | 32.00 | 4 810 LD, ELD | 196000 - 271999 | 32.00 - 48.00 | 44.10 | 4 811 LD, ELD | 272000 - 768000 | 44.10 - 48.00 | 48.00 | 4 812 -------------------+------------------+-----------------------+------------+------- 813 LD, ELD | 160000 - 189999 | 24.00 - 32.00 | 32.00 | 814 5, 5.1 LD, ELD | 190000 - 244999 | 24.00 - 44.10 | 32.00 815 | 5, 5.1 LD, ELD | 245000 - 339999 | 32.00 - 48.00 | 816 44.10 | 5, 5.1 LD, ELD | 340000 - 960000 | 44.10 - 48.00 | 817 48.00 | 5, 5.1 818 ----------------------------------------------------------------------------------- 819 \endverbatim \n 820 821 \subsection reommendedConfigELD AAC-ELD with SBR in Downsampled SBR mode. 822 \verbatim 823 ----------------------------------------------------------------------------------- 824 Audio Object Type | Bit Rate Range | Supported | Preferred | No. 825 of | [bit/s] | Sampling Rates | Sampl. | Chan. | 826 | [kHz] | Rate | | | 827 | [kHz] | 828 -------------------+------------------+-----------------------+------------+------- 829 ELD + SBR | 18000 - 24999 | 16.00 - 22.05 | 22.05 | 1 830 (downsampled SBR) | 25000 - 31999 | 16.00 - 24.00 | 24.00 | 1 831 | 32000 - 47999 | 22.05 - 32.00 | 32.00 | 1 832 | 48000 - 64000 | 22.05 - 48.00 | 32.00 | 1 833 -------------------+------------------+-----------------------+------------+------- 834 ELD + SBR | 32000 - 51999 | 16.00 - 24.00 | 24.00 | 2 835 (downsampled SBR) | 52000 - 59999 | 22.05 - 24.00 | 24.00 | 2 836 | 60000 - 95999 | 22.05 - 32.00 | 32.00 | 2 837 | 96000 - 128000 | 22.05 - 48.00 | 32.00 | 2 838 -------------------+------------------+-----------------------+------------+------- 839 ELD + SBR | 78000 - 99999 | 22.05 - 24.00 | 24.00 | 3 840 (downsampled SBR) | 100000 - 143999 | 22.05 - 32.00 | 32.00 | 3 841 | 144000 - 159999 | 22.05 - 48.00 | 32.00 | 3 842 | 160000 - 192000 | 32.00 - 48.00 | 32.00 | 3 843 -------------------+------------------+-----------------------+------------+------- 844 ELD + SBR | 104000 - 149999 | 22.05 - 24.00 | 24.00 | 4 845 (downsampled SBR) | 150000 - 191999 | 22.05 - 32.00 | 32.00 | 4 846 | 192000 - 211999 | 22.05 - 48.00 | 32.00 | 4 847 | 212000 - 256000 | 32.00 - 48.00 | 32.00 | 4 848 -------------------+------------------+-----------------------+------------+------- 849 ELD + SBR | 130000 - 171999 | 22.05 - 24.00 | 24.00 | 850 5, 5.1 (downsampled SBR) | 172000 - 239999 | 22.05 - 32.00 | 32.00 851 | 5, 5.1 | 240000 - 320000 | 32.00 - 48.00 | 32.00 | 5, 5.1 852 ----------------------------------------------------------------------------------- 853 \endverbatim \n 854 855 \subsection reommendedConfigELDv2 AAC-ELD v2, AAC-ELD v2 with SBR. 856 The ELD v2 212 configuration must be configured explicitly with 857 ::AACENC_CHANNELMODE parameter according MODE_212 value. SBR can be configured 858 separately through ::AACENC_SBR_MODE and ::AACENC_SBR_RATIO parameter. Following 859 configurations shall apply to both framelengths 480 and 512. For ELD v2 860 configuration without SBR and framelength 480 the supported sampling rate is 861 restricted to the range from 16 kHz up to 24 kHz. \verbatim 862 ----------------------------------------------------------------------------------- 863 Audio Object Type | Bit Rate Range | Supported | Preferred | No. 864 of | [bit/s] | Sampling Rates | Sampl. | Chan. | 865 | [kHz] | Rate | | | 866 | [kHz] | 867 -------------------+------------------+-----------------------+------------+------- 868 ELD-212 | 16000 - 19999 | 16.00 - 24.00 | 16.00 | 2 869 (without SBR) | 20000 - 39999 | 16.00 - 32.00 | 24.00 | 2 870 | 40000 - 49999 | 22.05 - 32.00 | 32.00 | 2 871 | 50000 - 61999 | 24.00 - 44.10 | 32.00 | 2 872 | 62000 - 84999 | 32.00 - 48.00 | 44.10 | 2 873 | 85000 - 192000 | 44.10 - 48.00 | 48.00 | 2 874 -------------------+------------------+-----------------------+------------+------- 875 ELD-212 + SBR | 18000 - 20999 | 32.00 | 32.00 | 2 876 (dualrate SBR) | 21000 - 25999 | 32.00 - 44.10 | 32.00 | 2 877 | 26000 - 31999 | 32.00 - 48.00 | 44.10 | 2 878 | 32000 - 64000 | 32.00 - 48.00 | 48.00 | 2 879 -------------------+------------------+-----------------------+------------+------- 880 ELD-212 + SBR | 18000 - 19999 | 16.00 - 22.05 | 22.05 | 2 881 (downsampled SBR) | 20000 - 24999 | 16.00 - 24.00 | 22.05 | 2 882 | 25000 - 31999 | 16.00 - 24.00 | 24.00 | 2 883 | 32000 - 64000 | 24.00 - 24.00 | 24.00 | 2 884 -------------------+------------------+-----------------------+------------+------- 885 \endverbatim \n 886 887 \page ENCODERBEHAVIOUR Encoder Behaviour 888 889 \section BEHAVIOUR_BANDWIDTH Bandwidth 890 891 The FDK AAC encoder usually does not use the full frequency range of the input 892 signal, but restricts the bandwidth according to certain library-internal 893 settings. They can be changed in the table "bandWidthTable" in the file 894 bandwidth.cpp (if available). 895 896 The encoder API provides the ::AACENC_BANDWIDTH parameter to adjust the 897 bandwidth explicitly. \code aacEncoder_SetParam(hAacEncoder, AACENC_BANDWIDTH, 898 value); \endcode 899 900 However it is not recommended to change these settings, because they are based 901 on numerous listening tests and careful tweaks to ensure the best overall 902 encoding quality. Also, the maximum bandwidth that can be set manually by the 903 user is 20kHz or fs/2, whichever value is smaller. 904 905 Theoretically a signal of for example 48 kHz can contain frequencies up to 24 906 kHz, but to use this full range in an audio encoder usually does not make sense. 907 Usually the encoder has a very limited amount of bits to spend (typically 128 908 kbit/s for stereo 48 kHz content) and to allow full range bandwidth would waste 909 a lot of these bits for frequencies the human ear is hardly able to perceive 910 anyway, if at all. Hence it is wise to use the available bits for the really 911 important frequency range and just skip the rest. At lower bitrates (e. g. <= 80 912 kbit/s for stereo 48 kHz content) the encoder will choose an even smaller 913 bandwidth, because an encoded signal with smaller bandwidth and hence less 914 artifacts sounds better than a signal with higher bandwidth but then more coding 915 artefacts across all frequencies. These artefacts would occur if small bitrates 916 and high bandwidths are chosen because the available bits are just not enough to 917 encode all frequencies well. 918 919 Unfortunately some people evaluate encoding quality based on possible bandwidth 920 as well, but it is a double-edged sword considering the trade-off described 921 above. 922 923 Another aspect is workload consumption. The higher the allowed bandwidth, the 924 more frequency lines have to be processed, which in turn increases the workload. 925 926 \section FRAMESIZES_AND_BIT_RESERVOIR Frame Sizes & Bit Reservoir 927 928 For AAC there is a difference between constant bit rate and constant frame 929 length due to the so-called bit reservoir technique, which allows the encoder to 930 use less bits in an AAC frame for those audio signal sections which are easy to 931 encode, and then spend them at a later point in time for more complex audio 932 sections. The extent to which this "bit exchange" is done is limited to allow 933 for reliable and relatively low delay real time streaming. Therefore, for 934 AAC-ELD, the bitreservoir is limited. It varies between 500 and 4000 bits/frame, 935 depending on the bitrate/channel. 936 - For a bitrate of 12kbps/channel and below, the AAC-ELD bitreservoir is 500 937 bits/frame. 938 - For a bitrate of 70kbps/channel and above, the AAC-ELD bitreservoir is 4000 939 bits/frame. 940 - Between 12kbps/channel and 70kbps/channel, the AAC-ELD bitrervoir is increased 941 linearly. 942 - For AAC-LC, the bitrate is only limited by the maximum AAC frame length. It 943 is, regardless of the available bit reservoir, defined as 6144 bits per channel. 944 945 Over a longer period in time the bitrate will be constant in the AAC constant 946 bitrate mode, e.g. for ISDN transmission. This means that in AAC each bitstream 947 frame will in general have a different length in bytes but over time it 948 will reach the target bitrate. 949 950 951 One could also make an MPEG compliant 952 AAC encoder which always produces constant length packages for each AAC frame, 953 but the audio quality would be considerably worse since the bit reservoir 954 technique would have to be switched off completely. A higher bit rate would have 955 to be used to get the same audio quality as with an enabled bit reservoir. 956 957 For mp3 by the way, the same bit reservoir technique exists, but there each bit 958 stream frame has a constant length for a given bit rate (ignoring the 959 padding byte). In mp3 there is a so-called "back pointer" which tells 960 the decoder which bits belong to the current mp3 frame - and in general some or 961 many bits have been transmitted in an earlier mp3 frame. Basically this leads to 962 the same "bit exchange between mp3 frames" as in AAC but with virtually constant 963 length frames. 964 965 This variable frame length at "constant bit rate" is not something special 966 in this Fraunhofer IIS AAC encoder. AAC has been designed in that way. 967 968 \subsection BEHAVIOUR_ESTIM_AVG_FRAMESIZES Estimating Average Frame Sizes 969 970 A HE-AAC v1 or v2 audio frame contains 2048 PCM samples per channel (there is 971 also one mode with 1920 samples per channel but this is only for special 972 purposes such as DAB+ digital radio). 973 974 The number of HE-AAC frames \f$N\_FRAMES\f$ per second at 44.1 kHz is: 975 976 \f[ 977 N\_FRAMES = 44100 / 2048 = 21.5332 978 \f] 979 980 At a bit rate of 8 kbps the average number of bits per frame 981 \f$N\_BITS\_PER\_FRAME\f$ is: 982 983 \f[ 984 N\_BITS\_PER\_FRAME = 8000 / 21.5332 = 371.52 985 \f] 986 987 which is about 46.44 bytes per encoded frame. 988 989 At a bit rate of 32 kbps, which is quite high for single channel HE-AAC v1, it 990 is: 991 992 \f[ 993 N\_BITS\_PER\_FRAME = 32000 / 21.5332 = 1486 994 \f] 995 996 which is about 185.76 bytes per encoded frame. 997 998 These bits/frame figures are average figures where each AAC frame generally has 999 a different size in bytes. To calculate the same for AAC-LC just use 1024 1000 instead of 2048 PCM samples per frame and channel. For AAC-LD/ELD it is either 1001 480 or 512 PCM samples per frame and channel. 1002 1003 1004 \section BEHAVIOUR_TOOLS Encoder Tools 1005 1006 The AAC encoder supports TNS, PNS, MS, Intensity and activates these tools 1007 depending on the audio signal and the encoder configuration (i.e. bitrate or 1008 AOT). It is not required to configure these tools manually. 1009 1010 PNS improves encoding quality only for certain bitrates. Therefore it makes 1011 sense to activate PNS only for these bitrates and save the processing power 1012 required for PNS (about 10 % of the encoder) when using other bitrates. This is 1013 done automatically inside the encoder library. PNS is disabled inside the 1014 encoder library if an MPEG-2 AOT is choosen since PNS is an MPEG-4 AAC feature. 1015 1016 If SBR is activated, the encoder automatically deactivates PNS internally. If 1017 TNS is disabled but PNS is allowed, the encoder deactivates PNS calculation 1018 internally. 1019 1020 */ 1021 1022 #ifndef AACENC_LIB_H 1023 #define AACENC_LIB_H 1024 1025 #include "machine_type.h" 1026 #include "FDK_audio.h" 1027 1028 /** 1029 * AAC encoder error codes. 1030 */ 1031 typedef enum { 1032 AACENC_OK = 0x0000, /*!< No error happened. All fine. */ 1033 1034 AACENC_INVALID_HANDLE = 1035 0x0020, /*!< Handle passed to function call was invalid. */ 1036 AACENC_MEMORY_ERROR = 0x0021, /*!< Memory allocation failed. */ 1037 AACENC_UNSUPPORTED_PARAMETER = 0x0022, /*!< Parameter not available. */ 1038 AACENC_INVALID_CONFIG = 0x0023, /*!< Configuration not provided. */ 1039 1040 AACENC_INIT_ERROR = 0x0040, /*!< General initialization error. */ 1041 AACENC_INIT_AAC_ERROR = 0x0041, /*!< AAC library initialization error. */ 1042 AACENC_INIT_SBR_ERROR = 0x0042, /*!< SBR library initialization error. */ 1043 AACENC_INIT_TP_ERROR = 0x0043, /*!< Transport library initialization error. */ 1044 AACENC_INIT_META_ERROR = 1045 0x0044, /*!< Meta data library initialization error. */ 1046 AACENC_INIT_MPS_ERROR = 0x0045, /*!< MPS library initialization error. */ 1047 1048 AACENC_ENCODE_ERROR = 0x0060, /*!< The encoding process was interrupted by an 1049 unexpected error. */ 1050 1051 AACENC_ENCODE_EOF = 0x0080 /*!< End of file reached. */ 1052 1053 } AACENC_ERROR; 1054 1055 /** 1056 * AAC encoder buffer descriptors identifier. 1057 * This identifier are used within buffer descriptors 1058 * AACENC_BufDesc::bufferIdentifiers. 1059 */ 1060 typedef enum { 1061 /* Input buffer identifier. */ 1062 IN_AUDIO_DATA = 0, /*!< Audio input buffer, interleaved INT_PCM samples. */ 1063 IN_ANCILLRY_DATA = 1, /*!< Ancillary data to be embedded into bitstream. */ 1064 IN_METADATA_SETUP = 2, /*!< Setup structure for embedding meta data. */ 1065 1066 /* Output buffer identifier. */ 1067 OUT_BITSTREAM_DATA = 3, /*!< Buffer holds bitstream output data. */ 1068 OUT_AU_SIZES = 1069 4 /*!< Buffer contains sizes of each access unit. This information 1070 is necessary for superframing. */ 1071 1072 } AACENC_BufferIdentifier; 1073 1074 /** 1075 * AAC encoder handle. 1076 */ 1077 typedef struct AACENCODER *HANDLE_AACENCODER; 1078 1079 /** 1080 * Provides some info about the encoder configuration. 1081 */ 1082 typedef struct { 1083 UINT maxOutBufBytes; /*!< Maximum number of encoder bitstream bytes within one 1084 frame. Size depends on maximum number of supported 1085 channels in encoder instance. For superframing (as 1086 used for example in DAB+), size has to be a multiple 1087 accordingly. */ 1088 1089 UINT maxAncBytes; /*!< Maximum number of ancillary data bytes which can be 1090 inserted into bitstream within one frame. */ 1091 1092 UINT inBufFillLevel; /*!< Internal input buffer fill level in samples per 1093 channel. This parameter will automatically be cleared 1094 if samplingrate or channel(Mode/Order) changes. */ 1095 1096 UINT inputChannels; /*!< Number of input channels expected in encoding 1097 process. */ 1098 1099 UINT frameLength; /*!< Amount of input audio samples consumed each frame per 1100 channel, depending on audio object type configuration. */ 1101 1102 UINT nDelay; /*!< Codec delay in PCM samples/channel. Depends on framelength 1103 and AOT. Does not include framing delay for filling up encoder 1104 PCM input buffer. */ 1105 1106 UINT nDelayCore; /*!< Codec delay in PCM samples/channel, w/o delay caused by 1107 the decoder SBR module. This delay is needed to correctly 1108 write edit lists for gapless playback. The decoder may not 1109 know how much delay is introdcued by SBR, since it may not 1110 know if SBR is active at all (implicit signaling), 1111 therefore the deocder must take into account any delay 1112 caused by the SBR module. */ 1113 1114 UCHAR confBuf[64]; /*!< Configuration buffer in binary format as an 1115 AudioSpecificConfig or StreamMuxConfig according to the 1116 selected transport type. */ 1117 1118 UINT confSize; /*!< Number of valid bytes in confBuf. */ 1119 1120 } AACENC_InfoStruct; 1121 1122 /** 1123 * Describes the input and output buffers for an aacEncEncode() call. 1124 */ 1125 typedef struct { 1126 INT numBufs; /*!< Number of buffers. */ 1127 void **bufs; /*!< Pointer to vector containing buffer addresses. */ 1128 INT *bufferIdentifiers; /*!< Identifier of each buffer element. See 1129 ::AACENC_BufferIdentifier. */ 1130 INT *bufSizes; /*!< Size of each buffer in 8-bit bytes. */ 1131 INT *bufElSizes; /*!< Size of each buffer element in bytes. */ 1132 1133 } AACENC_BufDesc; 1134 1135 /** 1136 * Defines the input arguments for an aacEncEncode() call. 1137 */ 1138 typedef struct { 1139 INT numInSamples; /*!< Number of valid input audio samples (multiple of input 1140 channels). */ 1141 INT numAncBytes; /*!< Number of ancillary data bytes to be encoded. */ 1142 1143 } AACENC_InArgs; 1144 1145 /** 1146 * Defines the output arguments for an aacEncEncode() call. 1147 */ 1148 typedef struct { 1149 INT numOutBytes; /*!< Number of valid bitstream bytes generated during 1150 aacEncEncode(). */ 1151 INT numInSamples; /*!< Number of input audio samples consumed by the encoder. 1152 */ 1153 INT numAncBytes; /*!< Number of ancillary data bytes consumed by the encoder. 1154 */ 1155 INT bitResState; /*!< State of the bit reservoir in bits. */ 1156 1157 } AACENC_OutArgs; 1158 1159 /** 1160 * Meta Data Compression Profiles. 1161 */ 1162 typedef enum { 1163 AACENC_METADATA_DRC_NONE = 0, /*!< None. */ 1164 AACENC_METADATA_DRC_FILMSTANDARD = 1, /*!< Film standard. */ 1165 AACENC_METADATA_DRC_FILMLIGHT = 2, /*!< Film light. */ 1166 AACENC_METADATA_DRC_MUSICSTANDARD = 3, /*!< Music standard. */ 1167 AACENC_METADATA_DRC_MUSICLIGHT = 4, /*!< Music light. */ 1168 AACENC_METADATA_DRC_SPEECH = 5, /*!< Speech. */ 1169 AACENC_METADATA_DRC_NOT_PRESENT = 1170 256 /*!< Disable writing gain factor (used for comp_profile only). */ 1171 1172 } AACENC_METADATA_DRC_PROFILE; 1173 1174 /** 1175 * Meta Data setup structure. 1176 */ 1177 typedef struct { 1178 AACENC_METADATA_DRC_PROFILE 1179 drc_profile; /*!< MPEG DRC compression profile. See 1180 ::AACENC_METADATA_DRC_PROFILE. */ 1181 AACENC_METADATA_DRC_PROFILE 1182 comp_profile; /*!< ETSI heavy compression profile. See 1183 ::AACENC_METADATA_DRC_PROFILE. */ 1184 1185 INT drc_TargetRefLevel; /*!< Used to define expected level to: 1186 Scaled with 16 bit. x*2^16. */ 1187 INT comp_TargetRefLevel; /*!< Adjust limiter to avoid overload. 1188 Scaled with 16 bit. x*2^16. */ 1189 1190 INT prog_ref_level_present; /*!< Flag, if prog_ref_level is present */ 1191 INT prog_ref_level; /*!< Programme Reference Level = Dialogue Level: 1192 -31.75dB .. 0 dB ; stepsize: 0.25dB 1193 Scaled with 16 bit. x*2^16.*/ 1194 1195 UCHAR PCE_mixdown_idx_present; /*!< Flag, if dmx-idx should be written in 1196 programme config element */ 1197 UCHAR ETSI_DmxLvl_present; /*!< Flag, if dmx-lvl should be written in 1198 ETSI-ancData */ 1199 1200 SCHAR centerMixLevel; /*!< Center downmix level (0...7, according to table) */ 1201 SCHAR surroundMixLevel; /*!< Surround downmix level (0...7, according to 1202 table) */ 1203 1204 UCHAR 1205 dolbySurroundMode; /*!< Indication for Dolby Surround Encoding Mode. 1206 - 0: Dolby Surround mode not indicated 1207 - 1: 2-ch audio part is not Dolby surround encoded 1208 - 2: 2-ch audio part is Dolby surround encoded */ 1209 1210 UCHAR drcPresentationMode; /*!< Indicatin for DRC Presentation Mode. 1211 - 0: Presentation mode not inticated 1212 - 1: Presentation mode 1 1213 - 2: Presentation mode 2 */ 1214 1215 struct { 1216 /* extended ancillary data */ 1217 UCHAR extAncDataEnable; /*< Indicates if MPEG4_ext_ancillary_data() exists. 1218 - 0: No MPEG4_ext_ancillary_data(). 1219 - 1: Insert MPEG4_ext_ancillary_data(). */ 1220 1221 UCHAR 1222 extDownmixLevelEnable; /*< Indicates if ext_downmixing_levels() exists. 1223 - 0: No ext_downmixing_levels(). 1224 - 1: Insert ext_downmixing_levels(). */ 1225 UCHAR extDownmixLevel_A; /*< Downmix level index A (0...7, according to 1226 table) */ 1227 UCHAR extDownmixLevel_B; /*< Downmix level index B (0...7, according to 1228 table) */ 1229 1230 UCHAR dmxGainEnable; /*< Indicates if ext_downmixing_global_gains() exists. 1231 - 0: No ext_downmixing_global_gains(). 1232 - 1: Insert ext_downmixing_global_gains(). */ 1233 INT dmxGain5; /*< Gain factor for downmix to 5 channels. 1234 -15.75dB .. -15.75dB; stepsize: 0.25dB 1235 Scaled with 16 bit. x*2^16.*/ 1236 INT dmxGain2; /*< Gain factor for downmix to 2 channels. 1237 -15.75dB .. -15.75dB; stepsize: 0.25dB 1238 Scaled with 16 bit. x*2^16.*/ 1239 1240 UCHAR lfeDmxEnable; /*< Indicates if ext_downmixing_lfe_level() exists. 1241 - 0: No ext_downmixing_lfe_level(). 1242 - 1: Insert ext_downmixing_lfe_level(). */ 1243 UCHAR lfeDmxLevel; /*< Downmix level index for LFE (0..15, according to 1244 table) */ 1245 1246 } ExtMetaData; 1247 1248 } AACENC_MetaData; 1249 1250 /** 1251 * AAC encoder control flags. 1252 * 1253 * In interaction with the ::AACENC_CONTROL_STATE parameter it is possible to 1254 * get information about the internal initialization process. It is also 1255 * possible to overwrite the internal state from extern when necessary. 1256 */ 1257 typedef enum { 1258 AACENC_INIT_NONE = 0x0000, /*!< Do not trigger initialization. */ 1259 AACENC_INIT_CONFIG = 1260 0x0001, /*!< Initialize all encoder modules configuration. */ 1261 AACENC_INIT_STATES = 0x0002, /*!< Reset all encoder modules history buffer. */ 1262 AACENC_INIT_TRANSPORT = 1263 0x1000, /*!< Initialize transport lib with new parameters. */ 1264 AACENC_RESET_INBUFFER = 1265 0x2000, /*!< Reset fill level of internal input buffer. */ 1266 AACENC_INIT_ALL = 0xFFFF /*!< Initialize all. */ 1267 } AACENC_CTRLFLAGS; 1268 1269 /** 1270 * \brief AAC encoder setting parameters. 1271 * 1272 * Use aacEncoder_SetParam() function to configure, or use aacEncoder_GetParam() 1273 * function to read the internal status of the following parameters. 1274 */ 1275 typedef enum { 1276 AACENC_AOT = 1277 0x0100, /*!< Audio object type. See ::AUDIO_OBJECT_TYPE in FDK_audio.h. 1278 - 2: MPEG-4 AAC Low Complexity. 1279 - 5: MPEG-4 AAC Low Complexity with Spectral Band Replication 1280 (HE-AAC). 1281 - 29: MPEG-4 AAC Low Complexity with Spectral Band 1282 Replication and Parametric Stereo (HE-AAC v2). This 1283 configuration can be used only with stereo input audio data. 1284 - 23: MPEG-4 AAC Low-Delay. 1285 - 39: MPEG-4 AAC Enhanced Low-Delay. Since there is no 1286 ::AUDIO_OBJECT_TYPE for ELD in combination with SBR defined, 1287 enable SBR explicitely by ::AACENC_SBR_MODE parameter. The ELD 1288 v2 212 configuration can be configured by ::AACENC_CHANNELMODE 1289 parameter. 1290 - 129: MPEG-2 AAC Low Complexity. 1291 - 132: MPEG-2 AAC Low Complexity with Spectral Band 1292 Replication (HE-AAC). 1293 1294 Please note that the virtual MPEG-2 AOT's basically disables 1295 non-existing Perceptual Noise Substitution tool in AAC encoder 1296 and controls the MPEG_ID flag in adts header. The virtual 1297 MPEG-2 AOT doesn't prohibit specific transport formats. */ 1298 1299 AACENC_BITRATE = 0x0101, /*!< Total encoder bitrate. This parameter is 1300 mandatory and interacts with ::AACENC_BITRATEMODE. 1301 - CBR: Bitrate in bits/second. 1302 - VBR: Variable bitrate. Bitrate argument will 1303 be ignored. See \ref suppBitrates for details. */ 1304 1305 AACENC_BITRATEMODE = 0x0102, /*!< Bitrate mode. Configuration can be different 1306 kind of bitrate configurations: 1307 - 0: Constant bitrate, use bitrate according 1308 to ::AACENC_BITRATE. (default) Within none 1309 LD/ELD ::AUDIO_OBJECT_TYPE, the CBR mode makes 1310 use of full allowed bitreservoir. In contrast, 1311 at Low-Delay ::AUDIO_OBJECT_TYPE the 1312 bitreservoir is kept very small. 1313 - 1: Variable bitrate mode, \ref vbrmode 1314 "very low bitrate". 1315 - 2: Variable bitrate mode, \ref vbrmode 1316 "low bitrate". 1317 - 3: Variable bitrate mode, \ref vbrmode 1318 "medium bitrate". 1319 - 4: Variable bitrate mode, \ref vbrmode 1320 "high bitrate". 1321 - 5: Variable bitrate mode, \ref vbrmode 1322 "very high bitrate". */ 1323 1324 AACENC_SAMPLERATE = 0x0103, /*!< Audio input data sampling rate. Encoder 1325 supports following sampling rates: 8000, 11025, 1326 12000, 16000, 22050, 24000, 32000, 44100, 1327 48000, 64000, 88200, 96000 */ 1328 1329 AACENC_SBR_MODE = 0x0104, /*!< Configure SBR independently of the chosen Audio 1330 Object Type ::AUDIO_OBJECT_TYPE. This parameter 1331 is for ELD audio object type only. 1332 - -1: Use ELD SBR auto configurator (default). 1333 - 0: Disable Spectral Band Replication. 1334 - 1: Enable Spectral Band Replication. */ 1335 1336 AACENC_GRANULE_LENGTH = 1337 0x0105, /*!< Core encoder (AAC) audio frame length in samples: 1338 - 1024: Default configuration. 1339 - 512: Default length in LD/ELD configuration. 1340 - 480: Length in LD/ELD configuration. 1341 - 256: Length for ELD reduced delay mode (x2). 1342 - 240: Length for ELD reduced delay mode (x2). 1343 - 128: Length for ELD reduced delay mode (x4). 1344 - 120: Length for ELD reduced delay mode (x4). */ 1345 1346 AACENC_CHANNELMODE = 0x0106, /*!< Set explicit channel mode. Channel mode must 1347 match with number of input channels. 1348 - 1-7, 11,12,14 and 33,34: MPEG channel 1349 modes supported, see ::CHANNEL_MODE in 1350 FDK_audio.h. */ 1351 1352 AACENC_CHANNELORDER = 1353 0x0107, /*!< Input audio data channel ordering scheme: 1354 - 0: MPEG channel ordering (e. g. 5.1: C, L, R, SL, SR, LFE). 1355 (default) 1356 - 1: WAVE file format channel ordering (e. g. 5.1: L, R, C, 1357 LFE, SL, SR). */ 1358 1359 AACENC_SBR_RATIO = 1360 0x0108, /*!< Controls activation of downsampled SBR. With downsampled 1361 SBR, the delay will be shorter. On the other hand, for 1362 achieving the same quality level, downsampled SBR needs more 1363 bits than dual-rate SBR. With downsampled SBR, the AAC encoder 1364 will work at the same sampling rate as the SBR encoder (single 1365 rate). Downsampled SBR is supported for AAC-ELD and HE-AACv1. 1366 - 1: Downsampled SBR (default for ELD). 1367 - 2: Dual-rate SBR (default for HE-AAC). */ 1368 1369 AACENC_AFTERBURNER = 1370 0x0200, /*!< This parameter controls the use of the afterburner feature. 1371 The afterburner is a type of analysis by synthesis algorithm 1372 which increases the audio quality but also the required 1373 processing power. It is recommended to always activate this if 1374 additional memory consumption and processing power consumption 1375 is not a problem. If increased MHz and memory consumption are 1376 an issue then the MHz and memory cost of this optional module 1377 need to be evaluated against the improvement in audio quality 1378 on a case by case basis. 1379 - 0: Disable afterburner (default). 1380 - 1: Enable afterburner. */ 1381 1382 AACENC_BANDWIDTH = 0x0203, /*!< Core encoder audio bandwidth: 1383 - 0: Determine audio bandwidth internally 1384 (default, see chapter \ref BEHAVIOUR_BANDWIDTH). 1385 - 1 to fs/2: Audio bandwidth in Hertz. Limited 1386 to 20kHz max. Not usable if SBR is active. This 1387 setting is for experts only, better do not touch 1388 this value to avoid degraded audio quality. */ 1389 1390 AACENC_PEAK_BITRATE = 1391 0x0207, /*!< Peak bitrate configuration parameter to adjust maximum bits 1392 per audio frame. Bitrate is in bits/second. The peak bitrate 1393 will internally be limited to the chosen bitrate 1394 ::AACENC_BITRATE as lower limit and the 1395 number_of_effective_channels*6144 bit as upper limit. 1396 1397 Setting the peak bitrate equal to ::AACENC_BITRATE does not 1398 necessarily mean that the audio frames will be of constant 1399 size. Since the peak bitate is in bits/second, the frame sizes 1400 can vary by one byte in one or the other direction over various 1401 frames. However, it is not recommended to reduce the peak 1402 pitrate to ::AACENC_BITRATE - it would disable the 1403 bitreservoir, which would affect the audio quality by a large 1404 amount. */ 1405 1406 AACENC_TRANSMUX = 0x0300, /*!< Transport type to be used. See ::TRANSPORT_TYPE 1407 in FDK_audio.h. Following types can be configured 1408 in encoder library: 1409 - 0: raw access units 1410 - 1: ADIF bitstream format 1411 - 2: ADTS bitstream format 1412 - 6: Audio Mux Elements (LATM) with 1413 muxConfigPresent = 1 1414 - 7: Audio Mux Elements (LATM) with 1415 muxConfigPresent = 0, out of band StreamMuxConfig 1416 - 10: Audio Sync Stream (LOAS) */ 1417 1418 AACENC_HEADER_PERIOD = 1419 0x0301, /*!< Frame count period for sending in-band configuration buffers 1420 within LATM/LOAS transport layer. Additionally this parameter 1421 configures the PCE repetition period in raw_data_block(). See 1422 \ref encPCE. 1423 - 0xFF: auto-mode default 10 for TT_MP4_ADTS, TT_MP4_LOAS and 1424 TT_MP4_LATM_MCP1, otherwise 0. 1425 - n: Frame count period. */ 1426 1427 AACENC_SIGNALING_MODE = 1428 0x0302, /*!< Signaling mode of the extension AOT: 1429 - 0: Implicit backward compatible signaling (default for 1430 non-MPEG-4 based AOT's and for the transport formats ADIF and 1431 ADTS) 1432 - A stream that uses implicit signaling can be decoded 1433 by every AAC decoder, even AAC-LC-only decoders 1434 - An AAC-LC-only decoder will only decode the 1435 low-frequency part of the stream, resulting in a band-limited 1436 output 1437 - This method works with all transport formats 1438 - This method does not work with downsampled SBR 1439 - 1: Explicit backward compatible signaling 1440 - A stream that uses explicit backward compatible 1441 signaling can be decoded by every AAC decoder, even AAC-LC-only 1442 decoders 1443 - An AAC-LC-only decoder will only decode the 1444 low-frequency part of the stream, resulting in a band-limited 1445 output 1446 - A decoder not capable of decoding PS will only decode 1447 the AAC-LC+SBR part. If the stream contained PS, the result 1448 will be a a decoded mono downmix 1449 - This method does not work with ADIF or ADTS. For 1450 LOAS/LATM, it only works with AudioMuxVersion==1 1451 - This method does work with downsampled SBR 1452 - 2: Explicit hierarchical signaling (default for MPEG-4 1453 based AOT's and for all transport formats excluding ADIF and 1454 ADTS) 1455 - A stream that uses explicit hierarchical signaling can 1456 be decoded only by HE-AAC decoders 1457 - An AAC-LC-only decoder will not decode a stream that 1458 uses explicit hierarchical signaling 1459 - A decoder not capable of decoding PS will not decode 1460 the stream at all if it contained PS 1461 - This method does not work with ADIF or ADTS. It works 1462 with LOAS/LATM and the MPEG-4 File format 1463 - This method does work with downsampled SBR 1464 1465 For making sure that the listener always experiences the 1466 best audio quality, explicit hierarchical signaling should be 1467 used. This makes sure that only a full HE-AAC-capable decoder 1468 will decode those streams. The audio is played at full 1469 bandwidth. For best backwards compatibility, it is recommended 1470 to encode with implicit SBR signaling. A decoder capable of 1471 AAC-LC only will then only decode the AAC part, which means the 1472 decoded audio will sound band-limited. 1473 1474 For MPEG-2 transport types (ADTS,ADIF), only implicit 1475 signaling is possible. 1476 1477 For LOAS and LATM, explicit backwards compatible signaling 1478 only works together with AudioMuxVersion==1. The reason is 1479 that, for explicit backwards compatible signaling, additional 1480 information will be appended to the ASC. A decoder that is only 1481 capable of decoding AAC-LC will skip this part. Nevertheless, 1482 for jumping to the end of the ASC, it needs to know the ASC 1483 length. Transmitting the length of the ASC is a feature of 1484 AudioMuxVersion==1, it is not possible to transmit the length 1485 of the ASC with AudioMuxVersion==0, therefore an AAC-LC-only 1486 decoder will not be able to parse a LOAS/LATM stream that was 1487 being encoded with AudioMuxVersion==0. 1488 1489 For downsampled SBR, explicit signaling is mandatory. The 1490 reason for this is that the extension sampling frequency (which 1491 is in case of SBR the sampling frequqncy of the SBR part) can 1492 only be signaled in explicit mode. 1493 1494 For AAC-ELD, the SBR information is transmitted in the 1495 ELDSpecific Config, which is part of the AudioSpecificConfig. 1496 Therefore, the settings here will have no effect on AAC-ELD.*/ 1497 1498 AACENC_TPSUBFRAMES = 1499 0x0303, /*!< Number of sub frames in a transport frame for LOAS/LATM or 1500 ADTS (default 1). 1501 - ADTS: Maximum number of sub frames restricted to 4. 1502 - LOAS/LATM: Maximum number of sub frames restricted to 2.*/ 1503 1504 AACENC_AUDIOMUXVER = 1505 0x0304, /*!< AudioMuxVersion to be used for LATM. (AudioMuxVersionA, 1506 currently not implemented): 1507 - 0: Default, no transmission of tara Buffer fullness, no ASC 1508 length and including actual latm Buffer fullnes. 1509 - 1: Transmission of tara Buffer fullness, ASC length and 1510 actual latm Buffer fullness. 1511 - 2: Transmission of tara Buffer fullness, ASC length and 1512 maximum level of latm Buffer fullness. */ 1513 1514 AACENC_PROTECTION = 0x0306, /*!< Configure protection in transport layer: 1515 - 0: No protection. (default) 1516 - 1: CRC active for ADTS transport format. */ 1517 1518 AACENC_ANCILLARY_BITRATE = 1519 0x0500, /*!< Constant ancillary data bitrate in bits/second. 1520 - 0: Either no ancillary data or insert exact number of 1521 bytes, denoted via input parameter, numAncBytes in 1522 AACENC_InArgs. 1523 - else: Insert ancillary data with specified bitrate. */ 1524 1525 AACENC_METADATA_MODE = 0x0600, /*!< Configure Meta Data. See ::AACENC_MetaData 1526 for further details: 1527 - 0: Do not embed any metadata. 1528 - 1: Embed dynamic_range_info metadata. 1529 - 2: Embed dynamic_range_info and 1530 ancillary_data metadata. 1531 - 3: Embed ancillary_data metadata. */ 1532 1533 AACENC_CONTROL_STATE = 1534 0xFF00, /*!< There is an automatic process which internally reconfigures 1535 the encoder instance when a configuration parameter changed or 1536 an error occured. This paramerter allows overwriting or getting 1537 the control status of this process. See ::AACENC_CTRLFLAGS. */ 1538 1539 AACENC_NONE = 0xFFFF /*!< ------ */ 1540 1541 } AACENC_PARAM; 1542 1543 #ifdef __cplusplus 1544 extern "C" { 1545 #endif 1546 1547 /** 1548 * \brief Open an instance of the encoder. 1549 * 1550 * Allocate memory for an encoder instance with a functional range denoted by 1551 * the function parameters. Preinitialize encoder instance with default 1552 * configuration. 1553 * 1554 * \param phAacEncoder A pointer to an encoder handle. Initialized on return. 1555 * \param encModules Specify encoder modules to be supported in this encoder 1556 * instance: 1557 * - 0x0: Allocate memory for all available encoder 1558 * modules. 1559 * - else: Select memory allocation regarding encoder 1560 * modules. Following flags are possible and can be combined. 1561 * - 0x01: AAC module. 1562 * - 0x02: SBR module. 1563 * - 0x04: PS module. 1564 * - 0x08: MPS module. 1565 * - 0x10: Metadata module. 1566 * - example: (0x01|0x02|0x04|0x08|0x10) allocates 1567 * all modules and is equivalent to default configuration denotet by 0x0. 1568 * \param maxChannels Number of channels to be allocated. This parameter can 1569 * be used in different ways: 1570 * - 0: Allocate maximum number of AAC and SBR channels as 1571 * supported by the library. 1572 * - nChannels: Use same maximum number of channels for 1573 * allocating memory in AAC and SBR module. 1574 * - nChannels | (nSbrCh<<8): Number of SBR channels can be 1575 * different to AAC channels to save data memory. 1576 * 1577 * \return 1578 * - AACENC_OK, on succes. 1579 * - AACENC_INVALID_HANDLE, AACENC_MEMORY_ERROR, AACENC_INVALID_CONFIG, 1580 * on failure. 1581 */ 1582 AACENC_ERROR aacEncOpen(HANDLE_AACENCODER *phAacEncoder, const UINT encModules, 1583 const UINT maxChannels); 1584 1585 /** 1586 * \brief Close the encoder instance. 1587 * 1588 * Deallocate encoder instance and free whole memory. 1589 * 1590 * \param phAacEncoder Pointer to the encoder handle to be deallocated. 1591 * 1592 * \return 1593 * - AACENC_OK, on success. 1594 * - AACENC_INVALID_HANDLE, on failure. 1595 */ 1596 AACENC_ERROR aacEncClose(HANDLE_AACENCODER *phAacEncoder); 1597 1598 /** 1599 * \brief Encode audio data. 1600 * 1601 * This function is mainly for encoding audio data. In addition the function can 1602 * be used for an encoder (re)configuration process. 1603 * - PCM input data will be retrieved from external input buffer until the fill 1604 * level allows encoding a single frame. This functionality allows an external 1605 * buffer with reduced size in comparison to the AAC or HE-AAC audio frame 1606 * length. 1607 * - If the value of the input samples argument is zero, just internal 1608 * reinitialization will be applied if it is requested. 1609 * - At the end of a file the flushing process can be triggerd via setting the 1610 * value of the input samples argument to -1. The encoder delay lines are fully 1611 * flushed when the encoder returns no valid bitstream data 1612 * AACENC_OutArgs::numOutBytes. Furthermore the end of file is signaled by the 1613 * return value AACENC_ENCODE_EOF. 1614 * - If an error occured in the previous frame or any of the encoder parameters 1615 * changed, an internal reinitialization process will be applied before encoding 1616 * the incoming audio samples. 1617 * - The function can also be used for an independent reconfiguration process 1618 * without encoding. The first parameter has to be a valid encoder handle and 1619 * all other parameters can be set to NULL. 1620 * - If the size of the external bitbuffer in outBufDesc is not sufficient for 1621 * writing the whole bitstream, an internal error will be the return value and a 1622 * reconfiguration will be triggered. 1623 * 1624 * \param hAacEncoder A valid AAC encoder handle. 1625 * \param inBufDesc Input buffer descriptor, see AACENC_BufDesc: 1626 * - At least one input buffer with audio data is 1627 * expected. 1628 * - Optionally a second input buffer with 1629 * ancillary data can be fed. 1630 * \param outBufDesc Output buffer descriptor, see AACENC_BufDesc: 1631 * - Provide one output buffer for the encoded 1632 * bitstream. 1633 * \param inargs Input arguments, see AACENC_InArgs. 1634 * \param outargs Output arguments, AACENC_OutArgs. 1635 * 1636 * \return 1637 * - AACENC_OK, on success. 1638 * - AACENC_INVALID_HANDLE, AACENC_ENCODE_ERROR, on failure in encoding 1639 * process. 1640 * - AACENC_INVALID_CONFIG, AACENC_INIT_ERROR, AACENC_INIT_AAC_ERROR, 1641 * AACENC_INIT_SBR_ERROR, AACENC_INIT_TP_ERROR, AACENC_INIT_META_ERROR, 1642 * AACENC_INIT_MPS_ERROR, on failure in encoder initialization. 1643 * - AACENC_UNSUPPORTED_PARAMETER, on incorrect input or output buffer 1644 * descriptor initialization. 1645 * - AACENC_ENCODE_EOF, when flushing fully concluded. 1646 */ 1647 AACENC_ERROR aacEncEncode(const HANDLE_AACENCODER hAacEncoder, 1648 const AACENC_BufDesc *inBufDesc, 1649 const AACENC_BufDesc *outBufDesc, 1650 const AACENC_InArgs *inargs, AACENC_OutArgs *outargs); 1651 1652 /** 1653 * \brief Acquire info about present encoder instance. 1654 * 1655 * This function retrieves information of the encoder configuration. In addition 1656 * to informative internal states, a configuration data block of the current 1657 * encoder settings will be returned. The format is either Audio Specific Config 1658 * in case of Raw Packets transport format or StreamMuxConfig in case of 1659 * LOAS/LATM transport format. The configuration data block is binary coded as 1660 * specified in ISO/IEC 14496-3 (MPEG-4 audio), to be used directly for MPEG-4 1661 * File Format or RFC3016 or RFC3640 applications. 1662 * 1663 * \param hAacEncoder A valid AAC encoder handle. 1664 * \param pInfo Pointer to AACENC_InfoStruct. Filled on return. 1665 * 1666 * \return 1667 * - AACENC_OK, on succes. 1668 * - AACENC_INIT_ERROR, on failure. 1669 */ 1670 AACENC_ERROR aacEncInfo(const HANDLE_AACENCODER hAacEncoder, 1671 AACENC_InfoStruct *pInfo); 1672 1673 /** 1674 * \brief Set one single AAC encoder parameter. 1675 * 1676 * This function allows configuration of all encoder parameters specified in 1677 * ::AACENC_PARAM. Each parameter must be set with a separate function call. An 1678 * internal validation of the configuration value range will be done and an 1679 * internal reconfiguration will be signaled. The actual configuration adoption 1680 * is part of the subsequent aacEncEncode() call. 1681 * 1682 * \param hAacEncoder A valid AAC encoder handle. 1683 * \param param Parameter to be set. See ::AACENC_PARAM. 1684 * \param value Parameter value. See parameter description in 1685 * ::AACENC_PARAM. 1686 * 1687 * \return 1688 * - AACENC_OK, on success. 1689 * - AACENC_INVALID_HANDLE, AACENC_UNSUPPORTED_PARAMETER, 1690 * AACENC_INVALID_CONFIG, on failure. 1691 */ 1692 AACENC_ERROR aacEncoder_SetParam(const HANDLE_AACENCODER hAacEncoder, 1693 const AACENC_PARAM param, const UINT value); 1694 1695 /** 1696 * \brief Get one single AAC encoder parameter. 1697 * 1698 * This function is the complement to aacEncoder_SetParam(). After encoder 1699 * reinitialization with user defined settings, the internal status can be 1700 * obtained of each parameter, specified with ::AACENC_PARAM. 1701 * 1702 * \param hAacEncoder A valid AAC encoder handle. 1703 * \param param Parameter to be returned. See ::AACENC_PARAM. 1704 * 1705 * \return Internal configuration value of specifed parameter ::AACENC_PARAM. 1706 */ 1707 UINT aacEncoder_GetParam(const HANDLE_AACENCODER hAacEncoder, 1708 const AACENC_PARAM param); 1709 1710 /** 1711 * \brief Get information about encoder library build. 1712 * 1713 * Fill a given LIB_INFO structure with library version information. 1714 * 1715 * \param info Pointer to an allocated LIB_INFO struct. 1716 * 1717 * \return 1718 * - AACENC_OK, on success. 1719 * - AACENC_INVALID_HANDLE, AACENC_INIT_ERROR, on failure. 1720 */ 1721 AACENC_ERROR aacEncGetLibInfo(LIB_INFO *info); 1722 1723 #ifdef __cplusplus 1724 } 1725 #endif 1726 1727 #endif /* AACENC_LIB_H */ 1728