Include patched FDK-AAC in the repository

The initial idea was to get the DAB+ patch into upstream, but since
that follows the android source releases, there is no place for a custom
DAB+ patch there.

So instead of having to maintain a patched fdk-aac that has to have the
same .so version as the distribution package on which it is installed,
we prefer having a separate fdk-aac-dab library to avoid collision.

At that point, there's no reason to keep fdk-aac in a separate
repository, as odr-audioenc is the only tool that needs DAB+ encoding
support. Including it here simplifies installation, and makes it
consistent with toolame-dab, also shipped in this repository.

DAB+ decoding support (needed by ODR-SourceCompanion, dablin, etisnoop,
welle.io and others) can be done using upstream FDK-AAC.

1 files changed, 1090 insertions, 0 deletions

diff --git a/fdk-aac/libAACdec/include/aacdecoder_lib.h b/fdk-aac/libAACdec/include/aacdecoder_lib.h
new file mode 100644
index 0000000..5f0dd02
--- /dev/null
+++ b/fdk-aac/libAACdec/include/aacdecoder_lib.h
@@ -0,0 +1,1090 @@
+/* -----------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+Forschung e.V. All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software
+that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding
+scheme for digital audio. This FDK AAC Codec software is intended to be used on
+a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient
+general perceptual audio codecs. AAC-ELD is considered the best-performing
+full-bandwidth communications codec by independent studies and is widely
+deployed. AAC has been standardized by ISO and IEC as part of the MPEG
+specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including
+those of Fraunhofer) may be obtained through Via Licensing
+(www.vialicensing.com) or through the respective patent owners individually for
+the purpose of encoding or decoding bit streams in products that are compliant
+with the ISO/IEC MPEG audio standards. Please note that most manufacturers of
+Android devices already license these patent claims through Via Licensing or
+directly from the patent owners, and therefore FDK AAC Codec software may
+already be covered under those patent licenses when it is used for those
+licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions
+with enhanced sound quality, are also available from Fraunhofer. Users are
+encouraged to check the Fraunhofer website for additional applications
+information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification,
+are permitted without payment of copyright license fees provided that you
+satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of
+the FDK AAC Codec or your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation
+and/or other materials provided with redistributions of the FDK AAC Codec or
+your modifications thereto in binary form. You must make available free of
+charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived
+from this library without prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute
+the FDK AAC Codec software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating
+that you changed the software and the date of any change. For modified versions
+of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android"
+must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK
+AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without
+limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE.
+Fraunhofer provides no warranty of patent non-infringement with respect to this
+software.
+
+You may use this FDK AAC Codec software or modifications thereto only for
+purposes that are authorized by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright
+holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES,
+including but not limited to the implied warranties of merchantability and
+fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary,
+or consequential damages, including but not limited to procurement of substitute
+goods or services; loss of use, data, or profits, or business interruption,
+however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of
+this software, even if advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------- */
+
+/**************************** AAC decoder library ******************************
+
+   Author(s):   Manuel Jander
+
+   Description:
+
+*******************************************************************************/
+
+#ifndef AACDECODER_LIB_H
+#define AACDECODER_LIB_H
+
+/**
+ * \file   aacdecoder_lib.h
+ * \brief  FDK AAC decoder library interface header file.
+ *
+
+\page INTRO Introduction
+
+
+\section SCOPE Scope
+
+This document describes the high-level application interface and usage of the
+ISO/MPEG-2/4 AAC Decoder library developed by the Fraunhofer Institute for
+Integrated Circuits (IIS). Depending on the library configuration, decoding of
+AAC-LC (Low-Complexity), HE-AAC (High-Efficiency AAC v1 and v2), AAC-LD
+(Low-Delay) and AAC-ELD (Enhanced Low-Delay) is implemented.
+
+All references to SBR (Spectral Band Replication) are only applicable to HE-AAC
+and AAC-ELD configurations of the FDK library. All references to PS (Parametric
+Stereo) are only applicable to HE-AAC v2 decoder configuration of the library.
+
+\section DecoderBasics Decoder Basics
+
+This document can only give a rough overview about the ISO/MPEG-2, ISO/MPEG-4
+AAC audio and MPEG-D USAC coding standards. To understand all details referenced
+in this document, you are encouraged to read the following documents.
+
+- ISO/IEC 13818-7 (MPEG-2 AAC) Standard, defines the syntax of MPEG-2 AAC audio
+bitstreams.
+- ISO/IEC 14496-3 (MPEG-4 AAC, subpart 1 and 4) Standard, defines the syntax of
+MPEG-4 AAC audio bitstreams.
+- ISO/IEC 23003-3 (MPEG-D USAC), defines MPEG-D USAC unified speech and audio
+codec.
+- Lutzky, Schuller, Gayer, Krämer, Wabnik, "A guideline to audio codec
+delay", 116th AES Convention, May 8, 2004
+
+In short, MPEG Advanced Audio Coding is based on a time-to-frequency mapping of
+the signal. The signal is partitioned into overlapping time portions and
+transformed into frequency domain. The spectral components are then quantized
+and coded using a highly efficient coding scheme.\n Encoded MPEG-2 and MPEG-4
+AAC audio bitstreams are composed of frames. Contrary to MPEG-1/2 Layer-3 (mp3),
+the length of individual frames is not restricted to a fixed number of bytes,
+but can take any length between 1 and 768 bytes.
+
+In addition to the above mentioned frequency domain coding mode, MPEG-D USAC
+also employs a time domain Algebraic Code-Excited Linear Prediction (ACELP)
+speech coder core. This operating mode is selected by the encoder in order to
+achieve the optimum audio quality for different content type. Several
+enhancements allow achieving higher quality at lower bit rates compared to
+MPEG-4 HE-AAC.
+
+
+\page LIBUSE Library Usage
+
+
+\section InterfaceDescritpion API Description
+
+All API header files are located in the folder /include of the release package.
+The contents of each file is described in detail in this document. All header
+files are provided for usage in specific C/C++ programs. The main AAC decoder
+library API functions are located in aacdecoder_lib.h header file.
+
+In binary releases the decoder core resides in statically linkable libraries,
+for example libAACdec.a.
+
+
+\section Calling_Sequence Calling Sequence
+
+The following sequence is necessary for proper decoding of ISO/MPEG-2/4 AAC,
+HE-AAC v2, or MPEG-D USAC bitstreams. In the following description, input stream
+read and output write function details are left out, since they may be
+implemented in a variety of configurations depending on the user's specific
+requirements. The example implementation uses file-based input/output, and in
+such case one may call mpegFileRead_Open() to open an input file and to allocate
+memory for the required structures, and the corresponding mpegFileRead_Close()
+to close opened files and to de-allocate associated structures.
+mpegFileRead_Open() will attempt to detect the bitstream format and in case of
+MPEG-4 file format or Raw Packets file format (a proprietary Fraunhofer IIS file
+format suitable only for testing) it will read the Audio Specific Config data
+(ASC). An unsuccessful attempt to recognize the bitstream format requires the
+user to provide this information manually. For any other bitstream formats that
+are usually applicable in streaming applications, the decoder itself will try to
+synchronize and parse the given bitstream fragment using the FDK transport
+library. Hence, for streaming applications (without file access) this step is
+not necessary.
+
+
+-# Call aacDecoder_Open() to open and retrieve a handle to a new AAC decoder
+instance. \code aacDecoderInfo = aacDecoder_Open(transportType, nrOfLayers);
+\endcode
+-# If out-of-band config data (Audio Specific Config (ASC) or Stream Mux Config
+(SMC)) is available, call aacDecoder_ConfigRaw() to pass this data to the
+decoder before beginning the decoding process. If this data is not available in
+advance, the decoder will configure itself while decoding, during the
+aacDecoder_DecodeFrame() function call.
+-# Begin decoding loop.
+\code
+do {
+\endcode
+-# Read data from bitstream file or stream buffer in to the driver program
+working memory (a client-supplied input buffer "inBuffer" in framework). This
+buffer will be used to load AAC bitstream data to the decoder.  Only when all
+data in this buffer has been processed will the decoder signal an empty buffer.
+For file-based input, you may invoke mpegFileRead_Read() to acquire new
+bitstream data.
+-# Call aacDecoder_Fill() to fill the decoder's internal bitstream input buffer
+with the client-supplied bitstream input buffer. Note, if the data loaded in to
+the internal buffer is not sufficient to decode a frame,
+aacDecoder_DecodeFrame() will return ::AAC_DEC_NOT_ENOUGH_BITS until a
+sufficient amount of data is loaded in to the internal buffer. For streaming
+formats (ADTS, LOAS), it is acceptable to load more than one frame to the
+decoder. However, for RAW file format (Fraunhofer IIS proprietary format), only
+one frame may be loaded to the decoder per aacDecoder_DecodeFrame() call. For
+least amount of communication delay, fill and decode should be performed on a
+frame by frame basis. \code ErrorStatus = aacDecoder_Fill(aacDecoderInfo,
+inBuffer, bytesRead, bytesValid); \endcode
+-# Call aacDecoder_DecodeFrame(). This function decodes one frame and writes
+decoded PCM audio data to a client-supplied buffer. It is the client's
+responsibility to allocate a buffer which is large enough to hold the decoded
+output data. \code ErrorStatus = aacDecoder_DecodeFrame(aacDecoderInfo,
+TimeData, OUT_BUF_SIZE, flags); \endcode If the bitstream configuration (number
+of channels, sample rate, frame size) is not known a priori, you may call
+aacDecoder_GetStreamInfo() to retrieve a structure that contains this
+information. You may use this data to initialize an audio output device. In the
+example program, if the number of channels or the sample rate has changed since
+program start or the previously decoded frame, the audio output device is then
+re-initialized. If WAVE file output is chosen, a new WAVE file for each new
+stream configuration is be created. \code p_si =
+aacDecoder_GetStreamInfo(aacDecoderInfo); \endcode
+-# Repeat steps 5 to 7 until no data is available to decode any more, or in case
+of error. \code } while (bytesRead[0] > 0 || doFlush || doBsFlush ||
+forceContinue); \endcode
+-# Call aacDecoder_Close() to de-allocate all AAC decoder and transport layer
+structures. \code aacDecoder_Close(aacDecoderInfo); \endcode
+
+\image latex decode.png "Decode calling sequence" width=11cm
+
+\image latex change_source.png "Change data source sequence" width 5cm
+
+\image latex conceal.png "Error concealment sequence" width=14cm
+
+\subsection Error_Concealment_Sequence Error Concealment Sequence
+
+There are different strategies to handle bit stream errors. Depending on the
+system properties the product designer might choose to take different actions in
+case a bit error occurs. In many cases the decoder might be able to do
+reasonable error concealment without the need of any additional actions from the
+system. But in some cases its not even possible to know how many decoded PCM
+output samples are required to fill the gap due to the data error, then the
+software surrounding the decoder must deal with the situation. The most simple
+way would be to just stop audio playback and resume once enough bit stream data
+and/or buffered output samples are available. More sophisticated designs might
+also be able to deal with sender/receiver clock drifts or data drop outs by
+using a closed loop control of FIFO fulness levels. The chosen strategy depends
+on the final product requirements.
+
+The error concealment sequence diagram illustrates the general execution paths
+for error handling.
+
+The macro IS_OUTPUT_VALID(err) can be used to identify if the audio output
+buffer contains valid audio either from error free bit stream data or successful
+error concealment. In case the result is false, the decoder output buffer does
+not contain meaningful audio samples and should not be passed to any output as
+it is. Most likely in case that a continuous audio output PCM stream is
+required, the output buffer must be filled with audio data from the calling
+framework. This might be e.g. an appropriate number of samples all zero.
+
+If error code ::AAC_DEC_TRANSPORT_SYNC_ERROR is returned by the decoder, under
+some particular conditions it is possible to estimate lost frames due to the bit
+stream error. In that case the bit stream is required to have a constant
+bitrate, and compatible transport type. Audio samples for the lost frames can be
+obtained by calling aacDecoder_DecodeFrame() with flag ::AACDEC_CONCEAL set
+n-times where n is the count of lost frames. Please note that the decoder has to
+have encountered valid configuration data at least once to be able to generate
+concealed data, because at the minimum the sampling rate, frame size and amount
+of audio channels needs to be known.
+
+If it is not possible to get an estimation of lost frames then a constant
+fullness of the audio output buffer can be achieved by implementing different
+FIFO control techniques e.g. just stop taking of samples from the buffer to
+avoid underflow or stop filling new data to the buffer to avoid overflow. But
+this techniques are out of scope of this document.
+
+For a detailed description of a specific error code please refer also to
+::AAC_DECODER_ERROR.
+
+\section BufferSystem Buffer System
+
+There are three main buffers in an AAC decoder application. One external input
+buffer to hold bitstream data from file I/O or elsewhere, one decoder-internal
+input buffer, and one to hold the decoded output PCM sample data. In resource
+limited applications, the output buffer may be reused as an external input
+buffer prior to the subsequence aacDecoder_Fill() function call.
+
+The external input buffer is set in the example program and its size is defined
+by ::IN_BUF_SIZE. You may freely choose different buffer sizes. To feed the data
+to the decoder-internal input buffer, use the function aacDecoder_Fill(). This
+function returns important information regarding the number of bytes in the
+external input buffer that have not yet been copied into the internal input
+buffer (variable bytesValid). Once the external buffer has been fully copied, it
+can be completely re-filled again. In case you wish to refill the buffer while
+there are unprocessed bytes (bytesValid is unequal 0), you should preserve the
+unconsumed data. However, we recommend to refill the buffer only when bytesValid
+returns 0.
+
+The bytesValid parameter is an input and output parameter to the FDK decoder. As
+an input, it signals how many valid bytes are available in the external buffer.
+After consumption of the external buffer using aacDecoder_Fill() function, the
+bytesValid parameter indicates if any of the bytes in the external buffer were
+not consumed.
+
+\image latex dec_buffer.png "Life cycle of the external input buffer" width=9cm
+
+\page OutputFormat Decoder audio output
+
+\section OutputFormatObtaining Obtaining channel mapping information
+
+The decoded audio output format is indicated by a set of variables of the
+CStreamInfo structure. While the struct members sampleRate, frameSize and
+numChannels might be self explanatory, pChannelType and pChannelIndices require
+some further explanation.
+
+These two arrays indicate the configuration of channel data within the output
+buffer. Both arrays have CStreamInfo::numChannels number of cells. Each cell of
+pChannelType indicates the channel type, which is described in the enum
+::AUDIO_CHANNEL_TYPE (defined in FDK_audio.h). The cells of pChannelIndices
+indicate the sub index among the channels starting with 0 among channels of the
+same audio channel type.
+
+The indexing scheme is structured as defined in MPEG-2/4 Standards. Indices
+start from the front direction (a center channel if available, will always be
+index 0) and increment, starting with the left side, pairwise (e.g. L, R) and
+from front to back (Front L, Front R, Surround L, Surround R). For detailed
+explanation, please refer to ISO/IEC 13818-7:2005(E), chapter 8.5.3.2.
+
+In case a Program Config is included in the audio configuration, the channel
+mapping described within it will be adopted.
+
+In case of MPEG-D Surround the channel mapping will follow the same criteria
+described in ISO/IEC 13818-7:2005(E), but adding corresponding top channels (if
+available) to the channel types in order to avoid ambiguity. The examples below
+explain these aspects in detail.
+
+\section OutputFormatChange Changing the audio output format
+
+For MPEG-4 audio the channel order can be changed at runtime through the
+parameter
+::AAC_PCM_OUTPUT_CHANNEL_MAPPING. See the description of those
+parameters and the decoder library function aacDecoder_SetParam() for more
+detail.
+
+\section OutputFormatExample Channel mapping examples
+
+The following examples illustrate the location of individual audio samples in
+the audio buffer that is passed to aacDecoder_DecodeFrame() and the expected
+data in the CStreamInfo structure which can be obtained by calling
+aacDecoder_GetStreamInfo().
+
+\subsection ExamplesStereo Stereo
+
+In case of ::AAC_PCM_OUTPUT_CHANNEL_MAPPING set to 1,
+a AAC-LC bit stream which has channelConfiguration = 2 in its audio specific
+config would lead to the following values in CStreamInfo:
+
+CStreamInfo::numChannels = 2
+
+CStreamInfo::pChannelType = { ::ACT_FRONT, ::ACT_FRONT }
+
+CStreamInfo::pChannelIndices = { 0, 1 }
+
+The output buffer will be formatted as follows:
+
+\verbatim
+  <left sample 0>  <left sample 1>  <left sample 2>  ... <left sample N>
+  <right sample 0> <right sample 1> <right sample 2> ... <right sample N>
+\endverbatim
+
+Where N equals to CStreamInfo::frameSize .
+
+\subsection ExamplesSurround Surround 5.1
+
+In case of ::AAC_PCM_OUTPUT_CHANNEL_MAPPING set to 1,
+a AAC-LC bit stream which has channelConfiguration = 6 in its audio specific
+config, would lead to the following values in CStreamInfo:
+
+CStreamInfo::numChannels = 6
+
+CStreamInfo::pChannelType = { ::ACT_FRONT, ::ACT_FRONT, ::ACT_FRONT, ::ACT_LFE,
+::ACT_BACK, ::ACT_BACK }
+
+CStreamInfo::pChannelIndices = { 1, 2, 0, 0, 0, 1 }
+
+Since ::AAC_PCM_OUTPUT_CHANNEL_MAPPING is 1, WAV file channel ordering will be
+used. For a 5.1 channel scheme, thus the channels would be: front left, front
+right, center, LFE, surround left, surround right. Thus the third channel is the
+center channel, receiving the index 0. The other front channels are front left,
+front right being placed as first and second channels with indices 1 and 2
+correspondingly. There is only one LFE, placed as the fourth channel and index
+0. Finally both surround channels get the type definition ACT_BACK, and the
+indices 0 and 1.
+
+The output buffer will be formatted as follows:
+
+\verbatim
+<front left sample 0> <front right sample 0>
+<center sample 0> <LFE sample 0>
+<surround left sample 0> <surround right sample 0>
+
+<front left sample 1> <front right sample 1>
+<center sample 1> <LFE sample 1>
+<surround left sample 1> <surround right sample 1>
+
+...
+
+<front left sample N> <front right sample N>
+<center sample N> <LFE sample N>
+<surround left sample N> <surround right sample N>
+\endverbatim
+
+Where N equals to CStreamInfo::frameSize .
+
+\subsection ExamplesArib ARIB coding mode 2/1
+
+In case of ::AAC_PCM_OUTPUT_CHANNEL_MAPPING set to 1,
+in case of a ARIB bit stream using coding mode 2/1 as described in ARIB STD-B32
+Part 2 Version 2.1-E1, page 61, would lead to the following values in
+CStreamInfo:
+
+CStreamInfo::numChannels = 3
+
+CStreamInfo::pChannelType = { ::ACT_FRONT, ::ACT_FRONT, ::ACT_BACK }
+
+CStreamInfo::pChannelIndices = { 0, 1, 0 }
+
+The audio channels will be placed as follows in the audio output buffer:
+
+\verbatim
+<front left sample 0> <front right sample 0>  <mid surround sample 0>
+
+<front left sample 1> <front right sample 1> <mid surround sample 1>
+
+...
+
+<front left sample N> <front right sample N> <mid surround sample N>
+
+Where N equals to CStreamInfo::frameSize .
+
+\endverbatim
+
+*/
+
+#include "machine_type.h"
+#include "FDK_audio.h"
+
+#include "genericStds.h"
+
+#define AACDECODER_LIB_VL0 3
+#define AACDECODER_LIB_VL1 0
+#define AACDECODER_LIB_VL2 0
+
+/**
+ * \brief  AAC decoder error codes.
+ */
+typedef enum {
+  AAC_DEC_OK =
+      0x0000, /*!< No error occurred. Output buffer is valid and error free. */
+  AAC_DEC_OUT_OF_MEMORY =
+      0x0002, /*!< Heap returned NULL pointer. Output buffer is invalid. */
+  AAC_DEC_UNKNOWN =
+      0x0005, /*!< Error condition is of unknown reason, or from a another
+                 module. Output buffer is invalid. */
+
+  /* Synchronization errors. Output buffer is invalid. */
+  aac_dec_sync_error_start = 0x1000,
+  AAC_DEC_TRANSPORT_SYNC_ERROR = 0x1001, /*!< The transport decoder had
+                                            synchronization problems. Do not
+                                            exit decoding. Just feed new
+                                              bitstream data. */
+  AAC_DEC_NOT_ENOUGH_BITS = 0x1002, /*!< The input buffer ran out of bits. */
+  aac_dec_sync_error_end = 0x1FFF,
+
+  /* Initialization errors. Output buffer is invalid. */
+  aac_dec_init_error_start = 0x2000,
+  AAC_DEC_INVALID_HANDLE =
+      0x2001, /*!< The handle passed to the function call was invalid (NULL). */
+  AAC_DEC_UNSUPPORTED_AOT =
+      0x2002, /*!< The AOT found in the configuration is not supported. */
+  AAC_DEC_UNSUPPORTED_FORMAT =
+      0x2003, /*!< The bitstream format is not supported.  */
+  AAC_DEC_UNSUPPORTED_ER_FORMAT =
+      0x2004, /*!< The error resilience tool format is not supported. */
+  AAC_DEC_UNSUPPORTED_EPCONFIG =
+      0x2005, /*!< The error protection format is not supported. */
+  AAC_DEC_UNSUPPORTED_MULTILAYER =
+      0x2006, /*!< More than one layer for AAC scalable is not supported. */
+  AAC_DEC_UNSUPPORTED_CHANNELCONFIG =
+      0x2007, /*!< The channel configuration (either number or arrangement) is
+                 not supported. */
+  AAC_DEC_UNSUPPORTED_SAMPLINGRATE = 0x2008, /*!< The sample rate specified in
+                                                the configuration is not
+                                                supported. */
+  AAC_DEC_INVALID_SBR_CONFIG =
+      0x2009, /*!< The SBR configuration is not supported. */
+  AAC_DEC_SET_PARAM_FAIL = 0x200A,  /*!< The parameter could not be set. Either
+                                       the value was out of range or the
+                                       parameter does  not exist. */
+  AAC_DEC_NEED_TO_RESTART = 0x200B, /*!< The decoder needs to be restarted,
+                                       since the required configuration change
+                                       cannot be performed. */
+  AAC_DEC_OUTPUT_BUFFER_TOO_SMALL =
+      0x200C, /*!< The provided output buffer is too small. */
+  aac_dec_init_error_end = 0x2FFF,
+
+  /* Decode errors. Output buffer is valid but concealed. */
+  aac_dec_decode_error_start = 0x4000,
+  AAC_DEC_TRANSPORT_ERROR =
+      0x4001, /*!< The transport decoder encountered an unexpected error. */
+  AAC_DEC_PARSE_ERROR = 0x4002, /*!< Error while parsing the bitstream. Most
+                                   probably it is corrupted, or the system
+                                   crashed. */
+  AAC_DEC_UNSUPPORTED_EXTENSION_PAYLOAD =
+      0x4003, /*!< Error while parsing the extension payload of the bitstream.
+                 The extension payload type found is not supported. */
+  AAC_DEC_DECODE_FRAME_ERROR = 0x4004, /*!< The parsed bitstream value is out of
+                                          range. Most probably the bitstream is
+                                          corrupt, or the system crashed. */
+  AAC_DEC_CRC_ERROR = 0x4005,          /*!< The embedded CRC did not match. */
+  AAC_DEC_INVALID_CODE_BOOK = 0x4006,  /*!< An invalid codebook was signaled.
+                                          Most probably the bitstream is corrupt,
+                                          or the system  crashed. */
+  AAC_DEC_UNSUPPORTED_PREDICTION =
+      0x4007, /*!< Predictor found, but not supported in the AAC Low Complexity
+                 profile. Most probably the bitstream is corrupt, or has a wrong
+                 format. */
+  AAC_DEC_UNSUPPORTED_CCE = 0x4008, /*!< A CCE element was found which is not
+                                       supported. Most probably the bitstream is
+                                       corrupt, or has a wrong format. */
+  AAC_DEC_UNSUPPORTED_LFE = 0x4009, /*!< A LFE element was found which is not
+                                       supported. Most probably the bitstream is
+                                       corrupt, or has a wrong format. */
+  AAC_DEC_UNSUPPORTED_GAIN_CONTROL_DATA =
+      0x400A, /*!< Gain control data found but not supported. Most probably the
+                 bitstream is corrupt, or has a wrong format. */
+  AAC_DEC_UNSUPPORTED_SBA =
+      0x400B, /*!< SBA found, but currently not supported in the BSAC profile.
+               */
+  AAC_DEC_TNS_READ_ERROR = 0x400C, /*!< Error while reading TNS data. Most
+                                      probably the bitstream is corrupt or the
+                                      system crashed. */
+  AAC_DEC_RVLC_ERROR =
+      0x400D, /*!< Error while decoding error resilient data. */
+  aac_dec_decode_error_end = 0x4FFF,
+  /* Ancillary data errors. Output buffer is valid. */
+  aac_dec_anc_data_error_start = 0x8000,
+  AAC_DEC_ANC_DATA_ERROR =
+      0x8001, /*!< Non severe error concerning the ancillary data handling. */
+  AAC_DEC_TOO_SMALL_ANC_BUFFER = 0x8002,  /*!< The registered ancillary data
+                                             buffer is too small to receive the
+                                             parsed data. */
+  AAC_DEC_TOO_MANY_ANC_ELEMENTS = 0x8003, /*!< More than the allowed number of
+                                             ancillary data elements should be
+                                             written to buffer. */
+  aac_dec_anc_data_error_end = 0x8FFF
+
+} AAC_DECODER_ERROR;
+
+/** Macro to identify initialization errors. Output buffer is invalid. */
+#define IS_INIT_ERROR(err)                                                    \
+  ((((err) >= aac_dec_init_error_start) && ((err) <= aac_dec_init_error_end)) \
+       ? 1                                                                    \
+       : 0)
+/** Macro to identify decode errors. Output buffer is valid but concealed. */
+#define IS_DECODE_ERROR(err)                 \
+  ((((err) >= aac_dec_decode_error_start) && \
+    ((err) <= aac_dec_decode_error_end))     \
+       ? 1                                   \
+       : 0)
+/**
+ * Macro to identify if the audio output buffer contains valid samples after
+ * calling aacDecoder_DecodeFrame(). Output buffer is valid but can be
+ * concealed.
+ */
+#define IS_OUTPUT_VALID(err) (((err) == AAC_DEC_OK) || IS_DECODE_ERROR(err))
+
+/*! \enum  AAC_MD_PROFILE
+ *  \brief The available metadata profiles which are mostly related to downmixing. The values define the arguments
+ *         for the use with parameter ::AAC_METADATA_PROFILE.
+ */
+typedef enum {
+  AAC_MD_PROFILE_MPEG_STANDARD =
+      0, /*!< The standard profile creates a mixdown signal based on the
+            advanced downmix metadata (from a DSE). The equations and default
+            values are defined in ISO/IEC 14496:3 Ammendment 4. Any other
+            (legacy) downmix metadata will be ignored. No other parameter will
+            be modified.         */
+  AAC_MD_PROFILE_MPEG_LEGACY =
+      1, /*!< This profile behaves identical to the standard profile if advanced
+              downmix metadata (from a DSE) is available. If not, the
+            matrix_mixdown information embedded in the program configuration
+            element (PCE) will be applied. If neither is the case, the module
+            creates a mixdown using the default coefficients as defined in
+            ISO/IEC 14496:3 AMD 4. The profile can be used to support legacy
+            digital TV (e.g. DVB) streams.           */
+  AAC_MD_PROFILE_MPEG_LEGACY_PRIO =
+      2, /*!< Similar to the ::AAC_MD_PROFILE_MPEG_LEGACY profile but if both
+            the advanced (ISO/IEC 14496:3 AMD 4) and the legacy (PCE) MPEG
+            downmix metadata are available the latter will be applied.
+          */
+  AAC_MD_PROFILE_ARIB_JAPAN =
+      3 /*!< Downmix creation as described in ABNT NBR 15602-2. But if advanced
+             downmix metadata (ISO/IEC 14496:3 AMD 4) is available it will be
+             preferred because of the higher resolutions. In addition the
+           metadata expiry time will be set to the value defined in the ARIB
+           standard (see ::AAC_METADATA_EXPIRY_TIME).
+         */
+} AAC_MD_PROFILE;
+
+/*! \enum  AAC_DRC_DEFAULT_PRESENTATION_MODE_OPTIONS
+ *  \brief Options for handling of DRC parameters, if presentation mode is not indicated in bitstream
+ */
+typedef enum {
+  AAC_DRC_PARAMETER_HANDLING_DISABLED = -1, /*!< DRC parameter handling
+                                               disabled, all parameters are
+                                               applied as requested. */
+  AAC_DRC_PARAMETER_HANDLING_ENABLED =
+      0, /*!< Apply changes to requested DRC parameters to prevent clipping. */
+  AAC_DRC_PRESENTATION_MODE_1_DEFAULT =
+      1, /*!< Use DRC presentation mode 1 as default (e.g. for Nordig) */
+  AAC_DRC_PRESENTATION_MODE_2_DEFAULT =
+      2 /*!< Use DRC presentation mode 2 as default (e.g. for DTG DBook) */
+} AAC_DRC_DEFAULT_PRESENTATION_MODE_OPTIONS;
+
+/**
+ * \brief AAC decoder setting parameters
+ */
+typedef enum {
+  AAC_PCM_DUAL_CHANNEL_OUTPUT_MODE =
+      0x0002, /*!< Defines how the decoder processes two channel signals: \n
+                   0: Leave both signals as they are (default). \n
+                   1: Create a dual mono output signal from channel 1. \n
+                   2: Create a dual mono output signal from channel 2. \n
+                   3: Create a dual mono output signal by mixing both channels
+                 (L' = R' = 0.5*Ch1 + 0.5*Ch2). */
+  AAC_PCM_OUTPUT_CHANNEL_MAPPING =
+      0x0003, /*!< Output buffer channel ordering. 0: MPEG PCE style order, 1:
+                 WAV file channel order (default). */
+  AAC_PCM_LIMITER_ENABLE =
+      0x0004,                           /*!< Enable signal level limiting. \n
+                                             -1: Auto-config. Enable limiter for all
+                                           non-lowdelay configurations by default. \n
+                                              0: Disable limiter in general. \n
+                                              1: Enable limiter always.
+                                             It is recommended to call the decoder
+                                           with a AACDEC_CLRHIST flag to reset all
+                                           states when      the limiter switch is changed
+                                           explicitly. */
+  AAC_PCM_LIMITER_ATTACK_TIME = 0x0005, /*!< Signal level limiting attack time
+                                           in ms. Default configuration is 15
+                                           ms. Adjustable range from 1 ms to 15
+                                           ms. */
+  AAC_PCM_LIMITER_RELEAS_TIME = 0x0006, /*!< Signal level limiting release time
+                                           in ms. Default configuration is 50
+                                           ms. Adjustable time must be larger
+                                           than 0 ms. */
+  AAC_PCM_MIN_OUTPUT_CHANNELS =
+      0x0011, /*!< Minimum number of PCM output channels. If higher than the
+                 number of encoded audio channels, a simple channel extension is
+                 applied (see note 4 for exceptions). \n -1, 0: Disable channel
+                 extension feature. The decoder output contains the same number
+                 of channels as the encoded bitstream. \n 1:    This value is
+                 currently needed only together with the mix-down feature. See
+                          ::AAC_PCM_MAX_OUTPUT_CHANNELS and note 2 below. \n
+                    2:    Encoded mono signals will be duplicated to achieve a
+                 2/0/0.0 channel output configuration. \n 6:    The decoder
+                 tries to reorder encoded signals with less than six channels to
+                 achieve a 3/0/2.1 channel output signal. Missing channels will
+                 be filled with a zero signal. If reordering is not possible the
+                 empty channels will simply be appended. Only available if
+                 instance is configured to support multichannel output. \n 8:
+                 The decoder tries to reorder encoded signals with less than
+                 eight channels to achieve a 3/0/4.1 channel output signal.
+                 Missing channels will be filled with a zero signal. If
+                 reordering is not possible the empty channels will simply be
+                          appended. Only available if instance is configured to
+                 support multichannel output.\n NOTE: \n
+                     1. The channel signaling (CStreamInfo::pChannelType and
+                 CStreamInfo::pChannelIndices) will not be modified. Added empty
+                 channels will be signaled with channel type
+                        AUDIO_CHANNEL_TYPE::ACT_NONE. \n
+                     2. If the parameter value is greater than that of
+                 ::AAC_PCM_MAX_OUTPUT_CHANNELS both will be set to the same
+                 value. \n
+                     3. This parameter does not affect MPEG Surround processing.
+                 \n
+                     4. This parameter will be ignored if the number of encoded
+                 audio channels is greater than 8. */
+  AAC_PCM_MAX_OUTPUT_CHANNELS =
+      0x0012, /*!< Maximum number of PCM output channels. If lower than the
+                 number of encoded audio channels, downmixing is applied
+                 accordingly (see note 5 for exceptions). If dedicated metadata
+                 is available in the stream it will be used to achieve better
+                 mixing results. \n -1, 0: Disable downmixing feature. The
+                 decoder output contains the same number of channels as the
+                 encoded bitstream. \n 1:    All encoded audio configurations
+                 with more than one channel will be mixed down to one mono
+                 output signal. \n 2:    The decoder performs a stereo mix-down
+                 if the number encoded audio channels is greater than two. \n 6:
+                 If the number of encoded audio channels is greater than six the
+                 decoder performs a mix-down to meet the target output
+                 configuration of 3/0/2.1 channels. Only available if instance
+                 is configured to support multichannel output. \n 8:    This
+                 value is currently needed only together with the channel
+                 extension feature. See ::AAC_PCM_MIN_OUTPUT_CHANNELS and note 2
+                 below. Only available if instance is configured to support
+                 multichannel output. \n NOTE: \n
+                     1. Down-mixing of any seven or eight channel configuration
+                 not defined in ISO/IEC 14496-3 PDAM 4 is not supported by this
+                 software version. \n
+                     2. If the parameter value is greater than zero but smaller
+                 than ::AAC_PCM_MIN_OUTPUT_CHANNELS both will be set to same
+                 value. \n
+                     3. The operating mode of the MPEG Surround module will be
+                 set accordingly. \n
+                     4. Setting this parameter with any value will disable the
+                 binaural processing of the MPEG Surround module
+                     5. This parameter will be ignored if the number of encoded
+                 audio channels is greater than 8. */
+  AAC_METADATA_PROFILE =
+      0x0020, /*!< See ::AAC_MD_PROFILE for all available values. */
+  AAC_METADATA_EXPIRY_TIME = 0x0021, /*!< Defines the time in ms after which all
+                                        the bitstream associated meta-data (DRC,
+                                        downmix coefficients, ...) will be reset
+                                        to default if no update has been
+                                        received. Negative values disable the
+                                        feature. */
+
+  AAC_CONCEAL_METHOD = 0x0100, /*!< Error concealment: Processing method. \n
+                                    0: Spectral muting. \n
+                                    1: Noise substitution (see ::CONCEAL_NOISE).
+                                  \n 2: Energy interpolation (adds additional
+                                  signal delay of one frame, see
+                                  ::CONCEAL_INTER. only some AOTs are
+                                  supported). \n */
+  AAC_DRC_BOOST_FACTOR =
+      0x0200, /*!< Dynamic Range Control: Scaling factor for boosting gain
+                 values. Defines how the boosting DRC factors (conveyed in the
+                 bitstream) will be applied to the decoded signal. The valid
+                 values range from 0 (don't apply boost factors) to 127 (fully
+                   apply boost factors). Default value is 0. */
+  AAC_DRC_ATTENUATION_FACTOR =
+      0x0201, /*!< Dynamic Range Control: Scaling factor for attenuating gain
+                 values. Same as
+                   ::AAC_DRC_BOOST_FACTOR but for attenuating DRC factors. */
+  AAC_DRC_REFERENCE_LEVEL =
+      0x0202, /*!< Dynamic Range Control (DRC): Target reference level. Defines
+                 the level below full-scale (quantized in steps of 0.25dB) to
+                 which the output audio signal will be normalized to by the DRC
+                 module. The parameter controls loudness normalization for both
+                 MPEG-4 DRC and MPEG-D DRC. The valid values range from 40 (-10
+                 dBFS) to 127 (-31.75 dBFS). Any value smaller than 0 switches
+                 off loudness normalization and MPEG-4 DRC. By default, loudness
+                 normalization and MPEG-4 DRC is switched off. */
+  AAC_DRC_HEAVY_COMPRESSION =
+      0x0203, /*!< Dynamic Range Control: En-/Disable DVB specific heavy
+                 compression (aka RF mode). If set to 1, the decoder will apply
+                 the compression values from the DVB specific ancillary data
+                 field. At the same time the MPEG-4 Dynamic Range Control tool
+                 will be disabled. By default, heavy compression is disabled. */
+  AAC_DRC_DEFAULT_PRESENTATION_MODE =
+      0x0204, /*!< Dynamic Range Control: Default presentation mode (DRC
+                 parameter handling). \n Defines the handling of the DRC
+                 parameters boost factor, attenuation factor and heavy
+                 compression, if no presentation mode is indicated in the
+                 bitstream.\n For options, see
+                 ::AAC_DRC_DEFAULT_PRESENTATION_MODE_OPTIONS.\n Default:
+                 ::AAC_DRC_PARAMETER_HANDLING_DISABLED */
+  AAC_DRC_ENC_TARGET_LEVEL =
+      0x0205, /*!< Dynamic Range Control: Encoder target level for light (i.e.
+                 not heavy) compression.\n If known, this declares the target
+                 reference level that was assumed at the encoder for calculation
+                   of limiting gains. The valid values range from 0 (full-scale)
+                 to 127 (31.75 dB below full-scale). This parameter is used only
+                 with ::AAC_DRC_PARAMETER_HANDLING_ENABLED and ignored
+                 otherwise.\n Default: 127 (worst-case assumption).\n */
+  AAC_QMF_LOWPOWER = 0x0300, /*!< Quadrature Mirror Filter (QMF) Bank processing
+                                mode. \n -1: Use internal default. Implies MPEG
+                                Surround partially complex accordingly. \n 0:
+                                Use complex QMF data mode. \n 1: Use real (low
+                                power) QMF data mode. \n */
+  AAC_TPDEC_CLEAR_BUFFER =
+      0x0603, /*!< Clear internal bit stream buffer of transport layers. The
+                 decoder will start decoding at new data passed after this event
+                 and any previous data is discarded. */
+  AAC_UNIDRC_SET_EFFECT = 0x0903 /*!<  MPEG-D DRC: Request a DRC effect type for
+                                    selection of a DRC set.\n Supported indices
+                                    are:\n -1: DRC off. Completely disables
+                                    MPEG-D DRC.\n 0: None (default). Disables
+                                    MPEG-D DRC, but automatically enables DRC if
+                                    necessary to prevent clipping.\n 1: Late
+                                    night\n 2: Noisy environment\n 3: Limited
+                                    playback range\n 4: Low playback level\n 5:
+                                    Dialog enhancement\n 6: General compression.
+                                    Used for generally enabling MPEG-D DRC
+                                    without particular request.\n */
+
+} AACDEC_PARAM;
+
+/**
+ * \brief This structure gives information about the currently decoded audio
+ * data. All fields are read-only.
+ */
+typedef struct {
+  /* These five members are the only really relevant ones for the user. */
+  INT sampleRate; /*!< The sample rate in Hz of the decoded PCM audio signal. */
+  INT frameSize;  /*!< The frame size of the decoded PCM audio signal. \n
+                       Typically this is: \n
+                       1024 or 960 for AAC-LC \n
+                       2048 or 1920 for HE-AAC (v2) \n
+                       512 or 480 for AAC-LD and AAC-ELD \n
+                       768, 1024, 2048 or 4096 for USAC  */
+  INT numChannels; /*!< The number of output audio channels before the rendering
+                      module, i.e. the original channel configuration. */
+  AUDIO_CHANNEL_TYPE
+  *pChannelType; /*!< Audio channel type of each output audio channel. */
+  UCHAR *pChannelIndices; /*!< Audio channel index for each output audio
+                             channel. See ISO/IEC 13818-7:2005(E), 8.5.3.2
+                             Explicit channel mapping using a
+                             program_config_element() */
+  /* Decoder internal members. */
+  INT aacSampleRate; /*!< Sampling rate in Hz without SBR (from configuration
+                        info) divided by a (ELD) downscale factor if present. */
+  INT profile; /*!< MPEG-2 profile (from file header) (-1: not applicable (e. g.
+                  MPEG-4)).               */
+  AUDIO_OBJECT_TYPE
+  aot; /*!< Audio Object Type (from ASC): is set to the appropriate value
+          for MPEG-2 bitstreams (e. g. 2 for AAC-LC). */
+  INT channelConfig; /*!< Channel configuration (0: PCE defined, 1: mono, 2:
+                        stereo, ...                       */
+  INT bitRate;       /*!< Instantaneous bit rate.                   */
+  INT aacSamplesPerFrame;   /*!< Samples per frame for the AAC core (from ASC)
+                               divided by a (ELD) downscale factor if present. \n
+                                 Typically this is (with a downscale factor of 1):
+                               \n   1024 or 960 for AAC-LC \n   512 or 480 for
+                               AAC-LD   and AAC-ELD         */
+  INT aacNumChannels;       /*!< The number of audio channels after AAC core
+                               processing (before PS or MPS processing).       CAUTION: This
+                               are not the final number of output channels! */
+  AUDIO_OBJECT_TYPE extAot; /*!< Extension Audio Object Type (from ASC)   */
+  INT extSamplingRate; /*!< Extension sampling rate in Hz (from ASC) divided by
+                          a (ELD) downscale factor if present. */
+
+  UINT outputDelay; /*!< The number of samples the output is additionally
+                       delayed by.the decoder. */
+  UINT flags; /*!< Copy of internal flags. Only to be written by the decoder,
+                 and only to be read externally. */
+
+  SCHAR epConfig; /*!< epConfig level (from ASC): only level 0 supported, -1
+                     means no ER (e. g. AOT=2, MPEG-2 AAC, etc.)  */
+  /* Statistics */
+  INT numLostAccessUnits; /*!< This integer will reflect the estimated amount of
+                             lost access units in case aacDecoder_DecodeFrame()
+                               returns AAC_DEC_TRANSPORT_SYNC_ERROR. It will be
+                             < 0 if the estimation failed. */
+
+  INT64 numTotalBytes; /*!< This is the number of total bytes that have passed
+                          through the decoder. */
+  INT64
+  numBadBytes; /*!< This is the number of total bytes that were considered
+                  with errors from numTotalBytes. */
+  INT64
+  numTotalAccessUnits;     /*!< This is the number of total access units that
+                              have passed through the decoder. */
+  INT64 numBadAccessUnits; /*!< This is the number of total access units that
+                              were considered with errors from numTotalBytes. */
+
+  /* Metadata */
+  SCHAR drcProgRefLev; /*!< DRC program reference level. Defines the reference
+                          level below full-scale. It is quantized in steps of
+                          0.25dB. The valid values range from 0 (0 dBFS) to 127
+                          (-31.75 dBFS). It is used to reflect the average
+                          loudness of the audio in LKFS according to ITU-R BS
+                          1770. If no level has been found in the bitstream the
+                          value is -1. */
+  SCHAR
+  drcPresMode; /*!< DRC presentation mode. According to ETSI TS 101 154,
+                  this field indicates whether   light (MPEG-4 Dynamic Range
+                  Control tool) or heavy compression (DVB heavy
+                  compression)   dynamic range control shall take priority
+                  on the outputs.   For details, see ETSI TS 101 154, table
+                  C.33. Possible values are: \n   -1: No corresponding
+                  metadata found in the bitstream \n   0: DRC presentation
+                  mode not indicated \n   1: DRC presentation mode 1 \n   2:
+                  DRC presentation mode 2 \n   3: Reserved */
+
+} CStreamInfo;
+
+typedef struct AAC_DECODER_INSTANCE
+    *HANDLE_AACDECODER; /*!< Pointer to a AAC decoder instance. */
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/**
+ * \brief Initialize ancillary data buffer.
+ *
+ * \param self    AAC decoder handle.
+ * \param buffer  Pointer to (external) ancillary data buffer.
+ * \param size    Size of the buffer pointed to by buffer.
+ * \return        Error code.
+ */
+LINKSPEC_H AAC_DECODER_ERROR aacDecoder_AncDataInit(HANDLE_AACDECODER self,
+                                                    UCHAR *buffer, int size);
+
+/**
+ * \brief Get one ancillary data element.
+ *
+ * \param self   AAC decoder handle.
+ * \param index  Index of the ancillary data element to get.
+ * \param ptr    Pointer to a buffer receiving a pointer to the requested
+ * ancillary data element.
+ * \param size   Pointer to a buffer receiving the length of the requested
+ * ancillary data element.
+ * \return       Error code.
+ */
+LINKSPEC_H AAC_DECODER_ERROR aacDecoder_AncDataGet(HANDLE_AACDECODER self,
+                                                   int index, UCHAR **ptr,
+                                                   int *size);
+
+/**
+ * \brief Set one single decoder parameter.
+ *
+ * \param self   AAC decoder handle.
+ * \param param  Parameter to be set.
+ * \param value  Parameter value.
+ * \return       Error code.
+ */
+LINKSPEC_H AAC_DECODER_ERROR aacDecoder_SetParam(const HANDLE_AACDECODER self,
+                                                 const AACDEC_PARAM param,
+                                                 const INT value);
+
+/**
+ * \brief              Get free bytes inside decoder internal buffer.
+ * \param self         Handle of AAC decoder instance.
+ * \param pFreeBytes   Pointer to variable receiving amount of free bytes inside
+ * decoder internal buffer.
+ * \return             Error code.
+ */
+LINKSPEC_H AAC_DECODER_ERROR
+aacDecoder_GetFreeBytes(const HANDLE_AACDECODER self, UINT *pFreeBytes);
+
+/**
+ * \brief               Open an AAC decoder instance.
+ * \param transportFmt  The transport type to be used.
+ * \param nrOfLayers    Number of transport layers.
+ * \return              AAC decoder handle.
+ */
+LINKSPEC_H HANDLE_AACDECODER aacDecoder_Open(TRANSPORT_TYPE transportFmt,
+                                             UINT nrOfLayers);
+
+/**
+ * \brief Explicitly configure the decoder by passing a raw AudioSpecificConfig
+ * (ASC) or a StreamMuxConfig (SMC), contained in a binary buffer. This is
+ * required for MPEG-4 and Raw Packets file format bitstreams as well as for
+ * LATM bitstreams with no in-band SMC. If the transport format is LATM with or
+ * without LOAS, configuration is assumed to be an SMC, for all other file
+ * formats an ASC.
+ *
+ * \param self    AAC decoder handle.
+ * \param conf    Pointer to an unsigned char buffer containing the binary
+ * configuration buffer (either ASC or SMC).
+ * \param length  Length of the configuration buffer in bytes.
+ * \return        Error code.
+ */
+LINKSPEC_H AAC_DECODER_ERROR aacDecoder_ConfigRaw(HANDLE_AACDECODER self,
+                                                  UCHAR *conf[],
+                                                  const UINT length[]);
+
+/**
+ * \brief Submit raw ISO base media file format boxes to decoder for parsing
+ * (only some box types are recognized).
+ *
+ * \param self    AAC decoder handle.
+ * \param buffer  Pointer to an unsigned char buffer containing the binary box
+ * data (including size and type, can be a sequence of multiple boxes).
+ * \param length  Length of the data in bytes.
+ * \return        Error code.
+ */
+LINKSPEC_H AAC_DECODER_ERROR aacDecoder_RawISOBMFFData(HANDLE_AACDECODER self,
+                                                       UCHAR *buffer,
+                                                       UINT length);
+
+/**
+ * \brief Fill AAC decoder's internal input buffer with bitstream data from the
+ * external input buffer. The function only copies such data as long as the
+ * decoder-internal input buffer is not full. So it grabs whatever it can from
+ * pBuffer and returns information (bytesValid) so that at a subsequent call of
+ * %aacDecoder_Fill(), the right position in pBuffer can be determined to grab
+ * the next data.
+ *
+ * \param self        AAC decoder handle.
+ * \param pBuffer     Pointer to external input buffer.
+ * \param bufferSize  Size of external input buffer. This argument is required
+ * because decoder-internally we need the information to calculate the offset to
+ * pBuffer, where the next available data is, which is then
+ * fed into the decoder-internal buffer (as much as
+ * possible). Our example framework implementation fills the
+ * buffer at pBuffer again, once it contains no available valid bytes anymore
+ * (meaning bytesValid equal 0).
+ * \param bytesValid  Number of bitstream bytes in the external bitstream buffer
+ * that have not yet been copied into the decoder's internal bitstream buffer by
+ * calling this function. The value is updated according to
+ * the amount of newly copied bytes.
+ * \return            Error code.
+ */
+LINKSPEC_H AAC_DECODER_ERROR aacDecoder_Fill(HANDLE_AACDECODER self,
+                                             UCHAR *pBuffer[],
+                                             const UINT bufferSize[],
+                                             UINT *bytesValid);
+
+#define AACDEC_CONCEAL                                                        \
+  1 /*!< Flag for aacDecoder_DecodeFrame(): Trigger the built-in error        \
+       concealment module to generate a substitute signal for one lost frame. \
+       New input data will not be considered. */
+#define AACDEC_FLUSH                                                         \
+  2 /*!< Flag for aacDecoder_DecodeFrame(): Flush all filterbanks to get all \
+       delayed audio without having new input data. Thus new input data will \
+       not be considered.*/
+#define AACDEC_INTR                                                         \
+  4 /*!< Flag for aacDecoder_DecodeFrame(): Signal an input bit stream data \
+       discontinuity. Resync any internals as necessary. */
+#define AACDEC_CLRHIST                                                        \
+  8 /*!< Flag for aacDecoder_DecodeFrame(): Clear all signal delay lines and  \
+       history buffers. CAUTION: This can cause discontinuities in the output \
+       signal. */
+
+/**
+ * \brief               Decode one audio frame
+ *
+ * \param self          AAC decoder handle.
+ * \param pTimeData     Pointer to external output buffer where the decoded PCM
+ * samples will be stored into.
+ * \param timeDataSize  Size of external output buffer.
+ * \param flags         Bit field with flags for the decoder: \n
+ *                      (flags & AACDEC_CONCEAL) == 1: Do concealment. \n
+ *                      (flags & AACDEC_FLUSH) == 2: Discard input data. Flush
+ * filter banks (output delayed audio). \n (flags & AACDEC_INTR) == 4: Input
+ * data is discontinuous. Resynchronize any internals as
+ * necessary. \n (flags & AACDEC_CLRHIST) == 8: Clear all signal delay lines and
+ * history buffers.
+ * \return              Error code.
+ */
+LINKSPEC_H AAC_DECODER_ERROR aacDecoder_DecodeFrame(HANDLE_AACDECODER self,
+                                                    INT_PCM *pTimeData,
+                                                    const INT timeDataSize,
+                                                    const UINT flags);
+
+/**
+ * \brief       De-allocate all resources of an AAC decoder instance.
+ *
+ * \param self  AAC decoder handle.
+ * \return      void.
+ */
+LINKSPEC_H void aacDecoder_Close(HANDLE_AACDECODER self);
+
+/**
+ * \brief       Get CStreamInfo handle from decoder.
+ *
+ * \param self  AAC decoder handle.
+ * \return      Reference to requested CStreamInfo.
+ */
+LINKSPEC_H CStreamInfo *aacDecoder_GetStreamInfo(HANDLE_AACDECODER self);
+
+/**
+ * \brief       Get decoder library info.
+ *
+ * \param info  Pointer to an allocated LIB_INFO structure.
+ * \return      0 on success.
+ */
+LINKSPEC_H INT aacDecoder_GetLibInfo(LIB_INFO *info);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* AACDECODER_LIB_H */