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1 
2 /* -----------------------------------------------------------------------------------------------------------
3 Software License for The Fraunhofer FDK AAC Codec Library for Android
4 
5 � Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur F�rderung 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@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 
902   AACENC_BITRATE                  = 0x0101,  /*!< Total encoder bitrate. This parameter is mandatory and interacts with ::AACENC_BITRATEMODE.
903                                                   - CBR: Bitrate in bits/second.
904                                                     See \ref suppBitrates for details. */
905 
906   AACENC_BITRATEMODE              = 0x0102,  /*!< Bitrate mode. Configuration can be different kind of bitrate configurations:
907                                                   - 0: Constant bitrate, use bitrate according to ::AACENC_BITRATE. (default)
908                                                        Within none LD/ELD ::AUDIO_OBJECT_TYPE, the CBR mode makes use of full allowed bitreservoir.
909                                                        In contrast, at Low-Delay ::AUDIO_OBJECT_TYPE the bitreservoir is kept very small.
910                                                   - 8: LD/ELD full bitreservoir for packet based transmission. */
911 
912   AACENC_SAMPLERATE               = 0x0103,  /*!< Audio input data sampling rate. Encoder supports following sampling rates:
913                                                   8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, 64000, 88200, 96000 */
914 
915   AACENC_SBR_MODE                 = 0x0104,  /*!< Configure SBR independently of the chosen Audio Object Type ::AUDIO_OBJECT_TYPE.
916                                                   This parameter is for ELD audio object type only.
917                                                   - -1: Use ELD SBR auto configurator (default).
918                                                   - 0: Disable Spectral Band Replication.
919                                                   - 1: Enable Spectral Band Replication. */
920 
921   AACENC_GRANULE_LENGTH           = 0x0105,  /*!< Core encoder (AAC) audio frame length in samples:
922                                                   - 1024: Default configuration.
923                                                   - 512: Default LD/ELD configuration.
924                                                   - 480: Optional length in LD/ELD configuration. */
925 
926   AACENC_CHANNELMODE              = 0x0106,  /*!< Set explicit channel mode. Channel mode must match with number of input channels.
927                                                   - 1-7 and 33,34: MPEG channel modes supported, see ::CHANNEL_MODE in FDK_audio.h. */
928 
929   AACENC_CHANNELORDER             = 0x0107,  /*!< Input audio data channel ordering scheme:
930                                                   - 0: MPEG channel ordering (e. g. 5.1: C, L, R, SL, SR, LFE). (default)
931                                                   - 1: WAVE file format channel ordering (e. g. 5.1: L, R, C, LFE, SL, SR). */
932 
933   AACENC_SBR_RATIO                = 0x0108,  /*!<  Controls activation of downsampled SBR. With downsampled SBR, the delay will be
934                                                    shorter. On the other hand, for achieving the same quality level, downsampled SBR
935                                                    needs more bits than dual-rate SBR.
936                                                    With downsampled SBR, the AAC encoder will work at the same sampling rate as the
937                                                    SBR encoder (single rate).
938                                                    Downsampled SBR is supported for AAC-ELD and HE-AACv1.
939                                                    - 1: Downsampled SBR (default for ELD).
940                                                    - 2: Dual-rate SBR   (default for HE-AAC). */
941 
942   AACENC_AFTERBURNER              = 0x0200,  /*!< This parameter controls the use of the afterburner feature.
943                                                   The afterburner is a type of analysis by synthesis algorithm which increases the
944                                                   audio quality but also the required processing power. It is recommended to always
945                                                   activate this if additional memory consumption and processing power consumption
946                                                   is not a problem. If increased MHz and memory consumption are an issue then the MHz
947                                                   and memory cost of this optional module need to be evaluated against the improvement
948                                                   in audio quality on a case by case basis.
949                                                   - 0: Disable afterburner (default).
950                                                   - 1: Enable afterburner. */
951 
952   AACENC_BANDWIDTH                = 0x0203,  /*!< Core encoder audio bandwidth:
953                                                   - 0: Determine bandwidth internally (default, see chapter \ref BEHAVIOUR_BANDWIDTH).
954                                                   - 1 to fs/2: Frequency bandwidth in Hertz. (Experts only, better do not
955                                                                touch this value to avoid degraded audio quality) */
956 
957   AACENC_PEAK_BITRATE             = 0x0207,  /*!< Peak bitrate configuration parameter to adjust maximum bits per audio frame. Bitrate is in bits/second.
958                                                   The peak bitrate will internally be limited to the chosen bitrate ::AACENC_BITRATE as lower limit
959                                                   and the number_of_effective_channels*6144 bit as upper limit.
960 
961                                                   Setting the peak bitrate equal to ::AACENC_BITRATE does not necessarily mean that the audio frames
962                                                   will be of constant size. Since the peak bitate is in bits/second, the frame sizes can vary by
963                                                   one byte in one or the other direction over various frames. However, it is not recommended to reduce
964                                                   the peak pitrate to ::AACENC_BITRATE - it would disable the bitreservoir, which would affect the
965                                                   audio quality by a large amount. */
966 
967   AACENC_TRANSMUX                 = 0x0300,  /*!< Transport type to be used. See ::TRANSPORT_TYPE in FDK_audio.h. Following
968                                                   types can be configured in encoder library:
969                                                   - 0: raw access units
970                                                   - 1: ADIF bitstream format
971                                                   - 2: ADTS bitstream format
972                                                   - 6: Audio Mux Elements (LATM) with muxConfigPresent = 1
973                                                   - 7: Audio Mux Elements (LATM) with muxConfigPresent = 0, out of band StreamMuxConfig
974                                                   - 10: Audio Sync Stream (LOAS) */
975 
976   AACENC_HEADER_PERIOD            = 0x0301,  /*!< Frame count period for sending in-band configuration buffers within LATM/LOAS
977                                                   transport layer. Additionally this parameter configures the PCE repetition period
978                                                   in raw_data_block(). See \ref encPCE.
979                                                   - 0xFF: auto-mode default 10 for TT_MP4_ADTS, TT_MP4_LOAS and TT_MP4_LATM_MCP1, otherwise 0.
980                                                   - n: Frame count period. */
981 
982   AACENC_SIGNALING_MODE           = 0x0302,  /*!< Signaling mode of the extension AOT:
983                                                   - 0: Implicit backward compatible signaling (default for non-MPEG-4 based
984                                                        AOT's and for the transport formats ADIF and ADTS)
985                                                        - A stream that uses implicit signaling can be decoded by every AAC decoder, even AAC-LC-only decoders
986                                                        - An AAC-LC-only decoder will only decode the low-frequency part of the stream, resulting in a band-limited output
987                                                        - This method works with all transport formats
988                                                        - This method does not work with downsampled SBR
989                                                   - 1: Explicit backward compatible signaling
990                                                        - A stream that uses explicit backward compatible signaling can be decoded by every AAC decoder, even AAC-LC-only decoders
991                                                        - An AAC-LC-only decoder will only decode the low-frequency part of the stream, resulting in a band-limited output
992                                                        - A decoder not capable of decoding PS will only decode the AAC-LC+SBR part.
993                                                          If the stream contained PS, the result will be a a decoded mono downmix
994                                                        - This method does not work with ADIF or ADTS. For LOAS/LATM, it only works with AudioMuxVersion==1
995                                                        - This method does work with downsampled SBR
996                                                   - 2: Explicit hierarchical signaling (default for MPEG-4 based AOT's and for all transport formats excluding ADIF and ADTS)
997                                                        - A stream that uses explicit hierarchical signaling can be decoded only by HE-AAC decoders
998                                                        - An AAC-LC-only decoder will not decode a stream that uses explicit hierarchical signaling
999                                                        - A decoder not capable of decoding PS will not decode the stream at all if it contained PS
1000                                                        - This method does not work with ADIF or ADTS. It works with LOAS/LATM and the MPEG-4 File format
1001                                                        - This method does work with downsampled SBR
1002 
1003                                                    For making sure that the listener always experiences the best audio quality,
1004                                                    explicit hierarchical signaling should be used.
1005                                                    This makes sure that only a full HE-AAC-capable decoder will decode those streams.
1006                                                    The audio is played at full bandwidth.
1007                                                    For best backwards compatibility, it is recommended to encode with implicit SBR signaling.
1008                                                    A decoder capable of AAC-LC only will then only decode the AAC part, which means the decoded
1009                                                    audio will sound band-limited.
1010 
1011                                                    For MPEG-2 transport types (ADTS,ADIF), only implicit signaling is possible.
1012 
1013                                                    For LOAS and LATM, explicit backwards compatible signaling only works together with AudioMuxVersion==1.
1014                                                    The reason is that, for explicit backwards compatible signaling, additional information will be appended to the ASC.
1015                                                    A decoder that is only capable of decoding AAC-LC will skip this part.
1016                                                    Nevertheless, for jumping to the end of the ASC, it needs to know the ASC length.
1017                                                    Transmitting the length of the ASC is a feature of AudioMuxVersion==1, it is not possible to transmit the
1018                                                    length of the ASC with AudioMuxVersion==0, therefore an AAC-LC-only decoder will not be able to parse a
1019                                                    LOAS/LATM stream that was being encoded with AudioMuxVersion==0.
1020 
1021                                                    For downsampled SBR, explicit signaling is mandatory. The reason for this is that the
1022                                                    extension sampling frequency (which is in case of SBR the sampling frequqncy of the SBR part)
1023                                                    can only be signaled in explicit mode.
1024 
1025                                                    For AAC-ELD, the SBR information is transmitted in the ELDSpecific Config, which is part of the
1026                                                    AudioSpecificConfig. Therefore, the settings here will have no effect on AAC-ELD.*/
1027 
1028   AACENC_TPSUBFRAMES              = 0x0303,  /*!< Number of sub frames in a transport frame for LOAS/LATM or ADTS (default 1).
1029                                                   - ADTS: Maximum number of sub frames restricted to 4.
1030                                                   - LOAS/LATM: Maximum number of sub frames restricted to 2.*/
1031 
1032   AACENC_AUDIOMUXVER              = 0x0304,  /*!< AudioMuxVersion to be used for LATM. (AudioMuxVersionA, currently not implemented):
1033                                                   - 0: Default, no transmission of tara Buffer fullness, no ASC length and including actual latm Buffer fullnes.
1034                                                   - 1: Transmission of tara Buffer fullness, ASC length and actual latm Buffer fullness.
1035                                                   - 2: Transmission of tara Buffer fullness, ASC length and maximum level of latm Buffer fullness. */
1036 
1037   AACENC_PROTECTION               = 0x0306,  /*!< Configure protection in tranpsort layer:
1038                                                   - 0: No protection. (default)
1039                                                   - 1: CRC active for ADTS bitstream format. */
1040 
1041   AACENC_ANCILLARY_BITRATE        = 0x0500,  /*!< Constant ancillary data bitrate in bits/second.
1042                                                   - 0: Either no ancillary data or insert exact number of bytes, denoted via
1043                                                        input parameter, numAncBytes in AACENC_InArgs.
1044                                                   - else: Insert ancillary data with specified bitrate. */
1045 
1046   AACENC_METADATA_MODE            = 0x0600,  /*!< Configure Meta Data. See ::AACENC_MetaData for further details:
1047                                                   - 0: Do not embed any metadata.
1048                                                   - 1: Embed MPEG defined metadata only.
1049                                                   - 2: Embed all metadata. */
1050 
1051   AACENC_CONTROL_STATE            = 0xFF00,  /*!< There is an automatic process which internally reconfigures the encoder instance
1052                                                   when a configuration parameter changed or an error occured. This paramerter allows
1053                                                   overwriting or getting the control status of this process. See ::AACENC_CTRLFLAGS. */
1054 
1055   AACENC_NONE                     = 0xFFFF   /*!< ------ */
1056 
1057 } AACENC_PARAM;
1058 
1059 
1060 #ifdef __cplusplus
1061 extern "C" {
1062 #endif
1063 
1064 /**
1065  * \brief  Open an instance of the encoder.
1066  *
1067  * Allocate memory for an encoder instance with a functional range denoted by the function parameters.
1068  * Preinitialize encoder instance with default configuration.
1069  *
1070  * \param phAacEncoder  A pointer to an encoder handle. Initialized on return.
1071  * \param encModules    Specify encoder modules to be supported in this encoder instance:
1072  *                      - 0x0: Allocate memory for all available encoder modules.
1073  *                      - else: Select memory allocation regarding encoder modules. Following flags are possible and can be combined.
1074  *                              - 0x01: AAC module.
1075  *                              - 0x02: SBR module.
1076  *                              - 0x04: PS module.
1077  *                              - 0x10: Metadata module.
1078  *                              - example: (0x01|0x02|0x04|0x10) allocates all modules and is equivalent to default configuration denotet by 0x0.
1079  * \param maxChannels   Number of channels to be allocated. This parameter can be used in different ways:
1080  *                      - 0: Allocate maximum number of AAC and SBR channels as supported by the library.
1081  *                      - nChannels: Use same maximum number of channels for allocating memory in AAC and SBR module.
1082  *                      - nChannels | (nSbrCh<<8): Number of SBR channels can be different to AAC channels to save data memory.
1083  *
1084  * \return
1085  *          - AACENC_OK, on succes.
1086  *          - AACENC_INVALID_HANDLE, AACENC_MEMORY_ERROR, AACENC_INVALID_CONFIG, on failure.
1087  */
1088 AACENC_ERROR aacEncOpen(
1089         HANDLE_AACENCODER        *phAacEncoder,
1090         const UINT                encModules,
1091         const UINT                maxChannels
1092         );
1093 
1094 
1095 /**
1096  * \brief  Close the encoder instance.
1097  *
1098  * Deallocate encoder instance and free whole memory.
1099  *
1100  * \param phAacEncoder  Pointer to the encoder handle to be deallocated.
1101  *
1102  * \return
1103  *          - AACENC_OK, on success.
1104  *          - AACENC_INVALID_HANDLE, on failure.
1105  */
1106 AACENC_ERROR aacEncClose(
1107         HANDLE_AACENCODER        *phAacEncoder
1108         );
1109 
1110 
1111 /**
1112  * \brief Encode audio data.
1113  *
1114  * This function is mainly for encoding audio data. In addition the function can be used for an encoder (re)configuration
1115  * process.
1116  * - PCM input data will be retrieved from external input buffer until the fill level allows encoding a single frame.
1117  *   This functionality allows an external buffer with reduced size in comparison to the AAC or HE-AAC audio frame length.
1118  * - If the value of the input samples argument is zero, just internal reinitialization will be applied if it is
1119  *   requested.
1120  * - At the end of a file the flushing process can be triggerd via setting the value of the input samples argument to -1.
1121  *   The encoder delay lines are fully flushed when the encoder returns no valid bitstream data AACENC_OutArgs::numOutBytes.
1122  *   Furthermore the end of file is signaled by the return value AACENC_ENCODE_EOF.
1123  * - If an error occured in the previous frame or any of the encoder parameters changed, an internal reinitialization
1124  *   process will be applied before encoding the incoming audio samples.
1125  * - The function can also be used for an independent reconfiguration process without encoding. The first parameter has to be a
1126  *   valid encoder handle and all other parameters can be set to NULL.
1127  * - If the size of the external bitbuffer in outBufDesc is not sufficient for writing the whole bitstream, an internal
1128  *   error will be the return value and a reconfiguration will be triggered.
1129  *
1130  * \param hAacEncoder           A valid AAC encoder handle.
1131  * \param inBufDesc             Input buffer descriptor, see AACENC_BufDesc:
1132  *                              - At least one input buffer with audio data is expected.
1133  *                              - Optionally a second input buffer with ancillary data can be fed.
1134  * \param outBufDesc            Output buffer descriptor, see AACENC_BufDesc:
1135  *                              - Provide one output buffer for the encoded bitstream.
1136  * \param inargs                Input arguments, see AACENC_InArgs.
1137  * \param outargs               Output arguments, AACENC_OutArgs.
1138  *
1139  * \return
1140  *          - AACENC_OK, on success.
1141  *          - AACENC_INVALID_HANDLE, AACENC_ENCODE_ERROR, on failure in encoding process.
1142  *          - AACENC_INVALID_CONFIG, AACENC_INIT_ERROR, AACENC_INIT_AAC_ERROR, AACENC_INIT_SBR_ERROR, AACENC_INIT_TP_ERROR,
1143  *            AACENC_INIT_META_ERROR, on failure in encoder initialization.
1144  *          - AACENC_ENCODE_EOF, when flushing fully concluded.
1145  */
1146 AACENC_ERROR aacEncEncode(
1147         const HANDLE_AACENCODER   hAacEncoder,
1148         const AACENC_BufDesc     *inBufDesc,
1149         const AACENC_BufDesc     *outBufDesc,
1150         const AACENC_InArgs      *inargs,
1151         AACENC_OutArgs           *outargs
1152         );
1153 
1154 
1155 /**
1156  * \brief  Acquire info about present encoder instance.
1157  *
1158  * This function retrieves information of the encoder configuration. In addition to informative internal states,
1159  * a configuration data block of the current encoder settings will be returned. The format is either Audio Specific Config
1160  * in case of Raw Packets transport format or StreamMuxConfig in case of LOAS/LATM transport format. The configuration
1161  * data block is binary coded as specified in ISO/IEC 14496-3 (MPEG-4 audio), to be used directly for MPEG-4 File Format
1162  * or RFC3016 or RFC3640 applications.
1163  *
1164  * \param hAacEncoder           A valid AAC encoder handle.
1165  * \param pInfo                 Pointer to AACENC_InfoStruct. Filled on return.
1166  *
1167  * \return
1168  *          - AACENC_OK, on succes.
1169  *          - AACENC_INIT_ERROR, on failure.
1170  */
1171 AACENC_ERROR aacEncInfo(
1172         const HANDLE_AACENCODER   hAacEncoder,
1173         AACENC_InfoStruct        *pInfo
1174         );
1175 
1176 
1177 /**
1178  * \brief  Set one single AAC encoder parameter.
1179  *
1180  * This function allows configuration of all encoder parameters specified in ::AACENC_PARAM. Each parameter must be
1181  * set with a separate function call. An internal validation of the configuration value range will be done and an
1182  * internal reconfiguration will be signaled. The actual configuration adoption is part of the subsequent aacEncEncode() call.
1183  *
1184  * \param hAacEncoder           A valid AAC encoder handle.
1185  * \param param                 Parameter to be set. See ::AACENC_PARAM.
1186  * \param value                 Parameter value. See parameter description in ::AACENC_PARAM.
1187  *
1188  * \return
1189  *          - AACENC_OK, on success.
1190  *          - AACENC_INVALID_HANDLE, AACENC_UNSUPPORTED_PARAMETER, AACENC_INVALID_CONFIG, on failure.
1191  */
1192 AACENC_ERROR aacEncoder_SetParam(
1193         const HANDLE_AACENCODER   hAacEncoder,
1194         const AACENC_PARAM        param,
1195         const UINT                value
1196         );
1197 
1198 
1199 /**
1200  * \brief  Get one single AAC encoder parameter.
1201  *
1202  * This function is the complement to aacEncoder_SetParam(). After encoder reinitialization with user defined settings,
1203  * the internal status can be obtained of each parameter, specified with ::AACENC_PARAM.
1204  *
1205  * \param hAacEncoder           A valid AAC encoder handle.
1206  * \param param                 Parameter to be returned. See ::AACENC_PARAM.
1207  *
1208  * \return  Internal configuration value of specifed parameter ::AACENC_PARAM.
1209  */
1210 UINT aacEncoder_GetParam(
1211         const HANDLE_AACENCODER   hAacEncoder,
1212         const AACENC_PARAM        param
1213         );
1214 
1215 
1216 /**
1217  * \brief  Get information about encoder library build.
1218  *
1219  * Fill a given LIB_INFO structure with library version information.
1220  *
1221  * \param info  Pointer to an allocated LIB_INFO struct.
1222  *
1223  * \return
1224  *          - AACENC_OK, on success.
1225  *          - AACENC_INVALID_HANDLE, AACENC_INIT_ERROR, on failure.
1226  */
1227 AACENC_ERROR aacEncGetLibInfo(
1228         LIB_INFO                 *info
1229         );
1230 
1231 
1232 #ifdef __cplusplus
1233 }
1234 #endif
1235 
1236 #endif   /* _AAC_ENC_LIB_H_ */
1237