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+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2012 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
+----------------------------------------------------------------------------------------------------------- */
+
+/*****************************  MPEG-4 AAC Decoder  **************************
+
+   Author(s):   Manuel Jander
+
+******************************************************************************/
+
+/**
+ * \file   aacdecoder_lib.h
+ * \brief  FDK AAC decoder library interface header file.
+ *
+
+\page INTRO Introduction
+
+\section SCOPE Scope
+
+This document describes the high-level 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, it implements decoding of AAC-LC (Low-Complexity),
+HE-AAC (High-Efficiency AAC, v1 and v2), AAC-LD (Low-Delay) and AAC-ELD (Enhanced Low-Delay).
+
+All references to SBR (Spectral Band Replication) are only applicable to HE-AAC and AAC-ELD
+versions of the library. All references to PS (Parametric Stereo) are only applicable to
+HE-AAC v2 versions of the library.
+
+\section DecoderBasics Decoder Basics
+
+This document can only give a rough overview about the ISO/MPEG-2 and ISO/MPEG-4 AAC audio
+coding standard. To understand all the terms in this document, you are encouraged to read
+the following documents.
+
+- ISO/IEC 13818-7 (MPEG-2 AAC), which defines the syntax of MPEG-2 AAC audio bitstreams.
+- ISO/IEC 14496-3 (MPEG-4 AAC, subpart 1 and 4), which defines the syntax of MPEG-4 AAC audio bitstreams.
+- Lutzky, Schuller, Gayer, Krämer, Wabnik, "A guideline to audio codec delay", 116th AES Convention, May 8, 2004
+
+MPEG Advanced Audio Coding is based on a time-to-frequency mapping of the signal. The signal
+is partitioned into overlapping portions and transformed into frequency domain. The spectral
+components are then quantized and coded.\n
+An MPEG2 or MPEG4 AAC audio bitstream is 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 on
+any length between 1 and 768 bytes.
+
+
+\page LIBUSE Library Usage
+
+\section InterfaceDescritpion API Description
+
+All API header files are located in the folder /include of the release package. They are described in
+detail in this document. All header files are provided for usage in C/C++ programs. The AAC decoder library
+API functions are located at aacdecoder_lib.h.
+
+In binary releases the decoder core resides in statically linkable libraries called for example libAACdec.a,
+(Linux) or FDK_aacDec_lib (Microsoft Visual C++).
+
+\section Calling_Sequence Calling Sequence
+
+For decoding of ISO/MPEG-2/4 AAC or HE-AAC v2 bitstreams the following sequence is mandatory. Input read
+and output write functions as well as the corresponding open and close functions are left out, since they
+may be implemented differently according to the user's specific requirements. The example implementation in
+main.cpp uses file-based input/output, and in such case 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() tries to detect the bitstream format and
+in case of MPEG-4 file format or Raw Packets file format (a Fraunhofer IIS proprietary format) reads the Audio
+Specific Config data (ASC). An unsuccessful attempt to recognize the bitstream format requires the user to
+provide this information manually (see \ref CommandLineUsage). 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.
+\dontinclude main.cpp
+\skipline aacDecoder_Open
+-# If out-of-band config data (Audio Specific Config (ASC) or Stream Mux Config (SMC)) is available, call
+aacDecoder_ConfigRaw() to pass it to the decoder and before the decoding process starts. If this data is
+not available in advance, the decoder will get it from the bitstream  and configure itself while decoding
+with aacDecoder_DecodeFrame().
+-# Begin decoding loop.
+\skipline do {
+-# Read data from bitstream file or stream into a client-supplied input buffer ("inBuffer" in main.cpp).
+If it is very small like just 4, aacDecoder_DecodeFrame() will
+repeatedly return ::AAC_DEC_NOT_ENOUGH_BITS until enough bits were fed by aacDecoder_Fill(). Only read data
+when this buffer has completely been processed and is then empty. For file-based input execute
+mpegFileRead_Read() or any other implementation with similar functionality.
+-# Call aacDecoder_Fill() to fill the decoder's internal bitstream input buffer with the client-supplied
+external bitstream input buffer.
+\skipline aacDecoder_Fill
+-# Call aacDecoder_DecodeFrame() which 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 this output data.
+\skipline aacDecoder_DecodeFrame
+If the bitstream's configuration (number of channels, sample rate, frame size) is not known in advance, you may
+call aacDecoder_GetStreamInfo() to retrieve a structure containing this information and then initialize an audio
+output device. In the example main.cpp, if the number of channels or the sample rate has changed since program
+start or since the previously decoded frame, the audio output device will be re-initialized. If WAVE file output
+is chosen, a new WAVE file for each new configuration will be created.
+\skipline aacDecoder_GetStreamInfo
+-# Repeat steps 5 to 7 until no data to decode is available anymore, or if an error occured.
+\skipline } while
+-# Call aacDecoder_Close() to de-allocate all AAC decoder and transport layer structures.
+\skipline aacDecoder_Close
+
+\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, whereas this output buffer may overlap with the external input buffer.
+
+The external input buffer is set in the example framework main.cpp and its size is defined by ::IN_BUF_SIZE.
+You may freely choose different sizes here. To feed the data to the decoder-internal input buffer, use the
+function aacDecoder_Fill(). This function returns important information about how many bytes in the
+external input buffer have not yet been copied into the internal input buffer (variable bytesValid).
+Once the external buffer has been fully copied, it can be re-filled again.
+In case you want to re-fill it when there are still unprocessed bytes (bytesValid is unequal 0), you
+would have to additionally perform a memcpy(), so that just means unnecessary computational overhead
+and therefore we recommend to re-fill the buffer only when bytesValid is 0.
+
+\image latex dec_buffer.png "Lifecycle of the external input buffer" width=9cm
+
+The size of the decoder-internal input buffer is set in tpdec_lib.h (see define ::TRANSPORTDEC_INBUF_SIZE).
+You may choose a smaller size under the following considerations:
+
+- each input channel requires 768 bytes
+- the whole buffer must be of size 2^n
+
+So for example a stereo decoder:
+
+\f[
+TRANSPORTDEC\_INBUF\_SIZE = 2 * 768 = 1536 => 2048
+\f]
+
+tpdec_lib.h and TRANSPORTDEC_INBUF_SIZE are not part of the decoder's library interface. Therefore
+only source-code clients may change this setting. If you received a library release, please ask us and
+we can change this in order to meet your memory requirements.
+
+\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 members sampleRate, frameSize and numChannels might be quite self explaining,
+pChannelType and pChannelIndices might require some more detailed explanation.
+
+These two arrays indicate what is each output channel supposed to be. Both array have
+CStreamInfo::numChannels cells. Each cell of pChannelType indicates the channel type, 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 all channels of the same audio channel type.
+
+The indexing scheme is the same as for MPEG-2/4. Thus indices are counted upwards starting from the front
+direction (thus a center channel if any, will always be index 0). Then the indices count up, starting always
+with the left side, pairwise from front toward back. 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 to the channel types front, side and back, in order to avoid any
+loss of information.
+
+\section OutputFormatChange Changing the audio output format
+
+The channel interleaving scheme and the actual channel order can be changed at runtime through the
+parameters ::AAC_PCM_OUTPUT_INTERLEAVED and ::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_INTERLEAVED set to 0 and ::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 }
+
+Since ::AAC_PCM_OUTPUT_INTERLEAVED is set to 0, the audio channels will be located as contiguous blocks
+in the output buffer 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_INTERLEAVED set to 1 and ::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.
+
+Since ::AAC_PCM_OUTPUT_INTERLEAVED is set to 1, the audio channels will be placed in the output buffer
+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_INTERLEAVED set to 1 and ::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
+
+*/
+
+#ifndef AACDECODER_LIB_H
+#define AACDECODER_LIB_H
+
+#include "machine_type.h"
+#include "FDK_audio.h"
+
+#include "genericStds.h"
+
+/**
+ * \brief  AAC decoder error codes.
+ */
+typedef enum {
+  AAC_DEC_OK                             = 0x0000,  /*!< No error occured. 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 syncronisation 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 requiered configuration change cannot be
+                                                         performed. */
+  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 signalled. 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 resillient 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. */
+#define IS_INIT_ERROR(err)   ( (((err)>=aac_dec_init_error_start)   && ((err)<=aac_dec_init_error_end))   ? 1 : 0)
+/** Macro to identify decode errors. */
+#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(). */
+#define IS_OUTPUT_VALID(err) ( ((err) == AAC_DEC_OK) || IS_DECODE_ERROR(err) )
+
+/**
+ * \brief AAC decoder setting parameters
+ */
+typedef enum
+{
+  AAC_PCM_OUTPUT_INTERLEAVED              = 0x0000,  /*!< PCM output mode (1: interleaved (default); 0: not interleaved). */
+  AAC_PCM_OUTPUT_CHANNELS                 = 0x0001,  /*!< Number of PCM output channels (if different from encoded audio channels, downmixing or
+                                                          upmixing is applied). \n
+                                                          -1: Disable up-/downmixing. The decoder output contains the same number of channels as the
+                                                              encoded bitstream. \n
+                                                           1: The decoder performs a mono matrix mix-down if the encoded audio channels are greater
+                                                              than one. Thus it ouputs always exact one channel. \n
+                                                           2: The decoder performs a stereo matrix mix-down if the encoded audio channels are greater
+                                                              than two. If the encoded audio channels are smaller than two the decoder duplicates the
+                                                              output. Thus it ouputs always exact two channels. \n */
+  AAC_PCM_DUAL_CHANNEL_OUTPUT_MODE        = 0x0002,  /*!< Defines how the decoder processes two channel signals:
+                                                          0: Leave both signals as they are (default).
+                                                          1: Create a dual mono output signal from channel 1.
+                                                          2: Create a dual mono output signal from channel 2.
+                                                          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_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). \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 all boosting factors). */
+  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: 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 valid values range from 0 (full-scale) to 127 (31.75 dB below
+                                                          full-scale). The value smaller than 0 switches off normalization. */
+  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_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_MPEGS_ENABLE                        = 0x0500,  /*!< MPEG Surround: Allow/Disable decoding of MPS content. Available only for decoders with MPEG
+                                                          Surround support. */
+
+  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. */
+
+} AACDEC_PARAM;
+
+/**
+ * \brief This structure gives information about the currently decoded audio data.
+ *        All fields are read-only.
+ */
+typedef struct
+{
+  /* These three members are the only really relevant ones for the user.                                                           */
+  INT               sampleRate;          /*!< The samplerate in Hz of the fully decoded PCM audio signal (after SBR processing).   */
+  INT               frameSize;           /*!< The frame size of the decoded PCM audio signal. \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                                                    */
+  INT               numChannels;         /*!< The number of output audio channels in the decoded and interleaved PCM audio signal. */
+  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).                           */
+  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). \n
+                                              1024 or 960 for AAC-LC \n
+                                              512 or 480 for AAC-LD and AAC-ELD         */
+
+  AUDIO_OBJECT_TYPE extAot;              /*!< Extension Audio Object Type (from ASC)   */
+  INT               extSamplingRate;     /*!< Extension sampling rate in Hz (from ASC) */
+
+  UINT              flags;               /*!< Copy if 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. */
+
+  UINT              numTotalBytes;       /*!< This is the number of total bytes that have passed through the decoder. */
+  UINT              numBadBytes;         /*!< This is the number of total bytes that were considered with errors from numTotalBytes. */
+  UINT              numTotalAccessUnits; /*!< This is the number of total access units that have passed through the decoder. */
+  UINT              numBadAccessUnits;   /*!< This is the number of total access units that were considered with errors from numTotalBytes. */
+
+} CStreamInfo;
+
+
+typedef struct AAC_DECODER_INSTANCE *HANDLE_AACDECODER;
+
+#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 receving 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
+ * \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 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(): do not consider new input data. Do concealment. */
+#define AACDEC_FLUSH    2 /*!< Flag for aacDecoder_DecodeFrame(): Do not consider new input data. Flush filterbanks (output delayed audio). */
+#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 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.
+ * \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 */