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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
+----------------------------------------------------------------------------------------------------------- */
+
+/**************************  Fraunhofer IIS FDK SysLib  **********************
+
+   Author(s):   Manuel Jander
+
+******************************************************************************/
+
+/** \file   FDK_audio.h
+ *  \brief  Global audio struct and constant definitions.
+ */
+
+#ifndef FDK_AUDIO_H
+#define FDK_AUDIO_H
+
+#include "machine_type.h"
+#include "genericStds.h"
+
+#ifdef __cplusplus
+extern "C"
+{
+#endif
+
+/**
+ * File format identifiers.
+ */
+typedef enum
+{
+  FF_UNKNOWN           = -1, /**< Unknown format.        */
+  FF_RAW               = 0,  /**< No container, bit stream data conveyed "as is". */
+
+  FF_MP4_3GPP          = 3,  /**< 3GPP file format.      */
+  FF_MP4_MP4F          = 4,  /**< MPEG-4 File format.     */
+
+  FF_RAWPACKETS        = 5,  /**< Proprietary raw packet file. */
+
+  FF_DRMCT             = 12  /**< Digital Radio Mondial (DRM30/DRM+) CT proprietary file format. */
+
+} FILE_FORMAT;
+
+/**
+ * Transport type identifiers.
+ */
+typedef enum
+{
+  TT_UNKNOWN           = -1, /**< Unknown format.            */
+  TT_MP4_RAW           = 0,  /**< "as is" access units (packet based since there is obviously no sync layer) */
+  TT_MP4_ADIF          = 1,  /**< ADIF bitstream format.     */
+  TT_MP4_ADTS          = 2,  /**< ADTS bitstream format.     */
+
+  TT_MP4_LATM_MCP1     = 6,  /**< Audio Mux Elements with muxConfigPresent = 1 */
+  TT_MP4_LATM_MCP0     = 7,  /**< Audio Mux Elements with muxConfigPresent = 0, out of band StreamMuxConfig */
+
+  TT_MP4_LOAS          = 10, /**< Audio Sync Stream.         */
+
+  TT_DRM               = 12, /**< Digital Radio Mondial (DRM30/DRM+) bitstream format. */
+
+  TT_MP1_L1            = 16, /**< MPEG 1 Audio Layer 1 audio bitstream. */
+  TT_MP1_L2            = 17, /**< MPEG 1 Audio Layer 2 audio bitstream. */
+  TT_MP1_L3            = 18, /**< MPEG 1 Audio Layer 3 audio bitstream. */
+
+  TT_RSVD50            = 50 /**< */
+
+} TRANSPORT_TYPE;
+
+/**
+ * Audio Object Type definitions.
+ */
+typedef enum
+{
+  AOT_NONE             = -1,
+  AOT_NULL_OBJECT      = 0,
+  AOT_AAC_MAIN         = 1, /**< Main profile                              */
+  AOT_AAC_LC           = 2, /**< Low Complexity object                     */
+  AOT_AAC_SSR          = 3,
+  AOT_AAC_LTP          = 4,
+  AOT_SBR              = 5,
+  AOT_AAC_SCAL         = 6,
+  AOT_TWIN_VQ          = 7,
+  AOT_CELP             = 8,
+  AOT_HVXC             = 9,
+  AOT_RSVD_10          = 10, /**< (reserved)                                */
+  AOT_RSVD_11          = 11, /**< (reserved)                                */
+  AOT_TTSI             = 12, /**< TTSI Object                               */
+  AOT_MAIN_SYNTH       = 13, /**< Main Synthetic object                     */
+  AOT_WAV_TAB_SYNTH    = 14, /**< Wavetable Synthesis object                */
+  AOT_GEN_MIDI         = 15, /**< General MIDI object                       */
+  AOT_ALG_SYNTH_AUD_FX = 16, /**< Algorithmic Synthesis and Audio FX object */
+  AOT_ER_AAC_LC        = 17, /**< Error Resilient(ER) AAC Low Complexity    */
+  AOT_RSVD_18          = 18, /**< (reserved)                                */
+  AOT_ER_AAC_LTP       = 19, /**< Error Resilient(ER) AAC LTP object        */
+  AOT_ER_AAC_SCAL      = 20, /**< Error Resilient(ER) AAC Scalable object   */
+  AOT_ER_TWIN_VQ       = 21, /**< Error Resilient(ER) TwinVQ object         */
+  AOT_ER_BSAC          = 22, /**< Error Resilient(ER) BSAC object           */
+  AOT_ER_AAC_LD        = 23, /**< Error Resilient(ER) AAC LowDelay object   */
+  AOT_ER_CELP          = 24, /**< Error Resilient(ER) CELP object           */
+  AOT_ER_HVXC          = 25, /**< Error Resilient(ER) HVXC object           */
+  AOT_ER_HILN          = 26, /**< Error Resilient(ER) HILN object           */
+  AOT_ER_PARA          = 27, /**< Error Resilient(ER) Parametric object     */
+  AOT_RSVD_28          = 28, /**< might become SSC                          */
+  AOT_PS               = 29, /**< PS, Parametric Stereo (includes SBR)      */
+  AOT_MPEGS            = 30, /**< MPEG Surround                             */
+
+  AOT_ESCAPE           = 31, /**< Signal AOT uses more than 5 bits          */
+
+  AOT_MP3ONMP4_L1      = 32, /**< MPEG-Layer1 in mp4                        */
+  AOT_MP3ONMP4_L2      = 33, /**< MPEG-Layer2 in mp4                        */
+  AOT_MP3ONMP4_L3      = 34, /**< MPEG-Layer3 in mp4                        */
+  AOT_RSVD_35          = 35, /**< might become DST                          */
+  AOT_RSVD_36          = 36, /**< might become ALS                          */
+  AOT_AAC_SLS          = 37, /**< AAC + SLS                                 */
+  AOT_SLS              = 38, /**< SLS                                       */
+  AOT_ER_AAC_ELD       = 39, /**< AAC Enhanced Low Delay                    */
+
+  AOT_USAC             = 42, /**< USAC                                      */
+  AOT_SAOC             = 43, /**< SAOC                                      */
+  AOT_LD_MPEGS         = 44, /**< Low Delay MPEG Surround                   */
+
+  AOT_RSVD50           = 50,  /**< Interim AOT for Rsvd50                   */
+
+  /* Pseudo AOTs */
+  AOT_MP2_AAC_MAIN     = 128, /**< Virtual AOT MP2 Main profile                           */
+  AOT_MP2_AAC_LC       = 129, /**< Virtual AOT MP2 Low Complexity profile                 */
+  AOT_MP2_AAC_SSR      = 130, /**< Virtual AOT MP2 Scalable Sampling Rate profile         */
+
+  AOT_MP2_SBR          = 132, /**< Virtual AOT MP2 Low Complexity Profile with SBR        */
+
+  AOT_DAB              = 134, /**< Virtual AOT for DAB (Layer2 with scalefactor CRC)      */
+  AOT_DABPLUS_AAC_LC   = 135, /**< Virtual AOT for DAB plus AAC-LC                        */
+  AOT_DABPLUS_SBR      = 136, /**< Virtual AOT for DAB plus HE-AAC                        */
+  AOT_DABPLUS_PS       = 137, /**< Virtual AOT for DAB plus HE-AAC v2                     */
+
+  AOT_PLAIN_MP1        = 140, /**< Virtual AOT for plain mp1                              */
+  AOT_PLAIN_MP2        = 141, /**< Virtual AOT for plain mp2                              */
+  AOT_PLAIN_MP3        = 142, /**< Virtual AOT for plain mp3                              */
+
+  AOT_DRM_AAC          = 143, /**< Virtual AOT for DRM (ER-AAC-SCAL without SBR)          */
+  AOT_DRM_SBR          = 144, /**< Virtual AOT for DRM (ER-AAC-SCAL with SBR)             */
+  AOT_DRM_MPEG_PS      = 145, /**< Virtual AOT for DRM (ER-AAC-SCAL with SBR and MPEG-PS) */
+  AOT_DRM_SURROUND     = 146, /**< Virtual AOT for DRM Surround (ER-AAC-SCAL (+SBR) +MPS) */
+
+  AOT_MP2_PS           = 156, /**< Virtual AOT MP2 Low Complexity Profile with SBR and PS */
+
+  AOT_MPEGS_RESIDUALS  = 256  /**< Virtual AOT for MPEG Surround residuals                */
+
+} AUDIO_OBJECT_TYPE;
+
+/** Channel Mode ( 1-7 equals MPEG channel configurations, others are arbitrary). */
+typedef enum {
+  MODE_INVALID    = -1,
+  MODE_UNKNOWN    = 0,
+  MODE_1          = 1,           /**< SCE                 */
+  MODE_2          = 2,           /**< CPE                 */
+  MODE_1_2        = 3,           /**< SCE,CPE             */
+  MODE_1_2_1      = 4,           /**< SCE,CPE,SCE         */
+  MODE_1_2_2      = 5,           /**< SCE,CPE,CPE         */
+  MODE_1_2_2_1    = 6,           /**< SCE,CPE,CPE,LFE     */
+  MODE_1_2_2_2_1  = 7,           /**< SCE,CPE,CPE,CPE,LFE */
+
+  MODE_1_1                      = 16,          /**< 2 SCEs (dual mono) */
+  MODE_1_1_1_1                  = 17,          /**< 4 SCEs */
+  MODE_1_1_1_1_1_1              = 18,          /**< 6 SCEs */
+  MODE_1_1_1_1_1_1_1_1          = 19,          /**< 8 SCEs */
+  MODE_1_1_1_1_1_1_1_1_1_1_1_1  = 20,          /**< 12 SCEs */
+
+  MODE_2_2                      = 21,          /**< 2 CPEs */
+  MODE_2_2_2                    = 22,          /**< 3 CPEs */
+  MODE_2_2_2_2                  = 23,          /**< 4 CPEs */
+  MODE_2_2_2_2_2_2              = 24,          /**< 6 CPEs */
+
+  MODE_2_1                      = 30           /**< CPE,SCE (ARIB standard) */
+
+} CHANNEL_MODE;
+
+/** Speaker description tags */
+typedef enum {
+  ACT_NONE,
+  ACT_FRONT,
+  ACT_SIDE,
+  ACT_BACK,
+  ACT_LFE,
+  ACT_FRONT_TOP,
+  ACT_SIDE_TOP,
+  ACT_BACK_TOP,
+  ACT_TOP /* Ts */
+} AUDIO_CHANNEL_TYPE;
+
+/**
+ * Audio Codec flags.
+ */
+#define AC_ER_VCB11     0x000001  /*!< aacSectionDataResilienceFlag     flag (from ASC): 1 means use virtual codebooks  */
+#define AC_ER_RVLC      0x000002  /*!< aacSpectralDataResilienceFlag     flag (from ASC): 1 means use huffman codeword reordering */
+#define AC_ER_HCR       0x000004  /*!< aacSectionDataResilienceFlag     flag (from ASC): 1 means use virtual codebooks  */
+#define AC_SCALABLE     0x000008  /*!< AAC Scalable*/
+#define AC_ELD          0x000010  /*!< AAC-ELD */
+#define AC_LD           0x000020  /*!< AAC-LD */
+#define AC_ER           0x000040  /*!< ER syntax */
+#define AC_BSAC         0x000080  /*!< BSAC */
+#define AC_USAC         0x000100  /*!< USAC */
+#define AC_USAC_TW      0x000200  /*!< USAC time warped filter bank is active */
+#define AC_USAC_NOISE   0x000400  /*!< USAC noise filling is active */
+#define AC_USAC_HBE     0x000800  /*!< USAC harmonic bandwidth extension is active */
+#define AC_RSVD50       0x001000  /*!< Rsvd50 */
+#define AC_SBR_PRESENT  0x002000  /*!< SBR present flag (from ASC)             */
+#define AC_SBRCRC       0x004000  /*!< SBR CRC present flag. Only relevant for AAC-ELD for now. */
+#define AC_PS_PRESENT   0x008000  /*!< PS present flag (from ASC or implicit)  */
+#define AC_MPS_PRESENT  0x010000  /*!< MPS present flag (from ASC or implicit) */
+#define AC_DRM          0x020000  /*!< DRM bit stream syntax */
+#define AC_INDEP        0x040000  /*!< Independency flag */
+#define AC_MPS_RES      0x080000  /*!< MPS residual individual channel data. */
+#define AC_DAB          0x800000  /*!< DAB bit stream syntax */
+#define AC_LD_MPS     0x01000000  /*!< Low Delay MPS. */
+
+
+/* CODER_CONFIG::flags */
+#define CC_MPEG_ID         0x00100000
+#define CC_IS_BASELAYER    0x00200000
+#define CC_PROTECTION      0x00400000
+#define CC_SBR             0x00800000
+#define CC_SBRCRC          0x00010000
+#define CC_RVLC            0x01000000
+#define CC_VCB11           0x02000000
+#define CC_HCR             0x04000000
+#define CC_PSEUDO_SURROUND 0x08000000
+#define CC_USAC_NOISE      0x10000000
+#define CC_USAC_TW         0x20000000
+#define CC_USAC_HBE        0x40000000
+
+/** Generic audio coder configuration structure. */
+typedef struct {
+  AUDIO_OBJECT_TYPE aot;         /**< Audio Object Type (AOT).           */
+  AUDIO_OBJECT_TYPE extAOT;      /**< Extension Audio Object Type (SBR). */
+  CHANNEL_MODE channelMode;      /**< Channel mode.                      */
+  INT        samplingRate;       /**< Sampling rate.                     */
+  INT        extSamplingRate;    /**< Extended samplerate (SBR).         */
+  INT        bitRate;            /**< Average bitrate.                   */
+  int        samplesPerFrame;    /**< Number of PCM samples per codec frame and audio channel. */
+  int        noChannels;         /**< Number of audio channels.          */
+  int        bitsFrame;
+  int        nSubFrames;         /**< Amount of encoder subframes. 1 means no subframing.                                     */
+  int        BSACnumOfSubFrame;  /**< The number of the sub-frames which are grouped and transmitted in a super-frame (BSAC). */
+  int        BSAClayerLength;    /**< The average length of the large-step layers in bytes (BSAC).                            */
+  UINT       flags;              /**< flags */
+  UCHAR      matrixMixdownA;     /**< Matrix mixdown index to put into PCE. Default value 0 means no mixdown coefficient,
+                                      valid values are 1-4 which correspond to matrix_mixdown_idx 0-3. */
+  UCHAR      headerPeriod;       /**< Frame period for sending in band configuration buffers in the transport layer. */
+
+  UCHAR      stereoConfigIndex; /**< USAC MPS stereo mode */
+  UCHAR      sbrMode;           /**< USAC SBR mode */
+} CODER_CONFIG;
+
+/** MP4 Element IDs. */
+typedef enum
+{
+  ID_NONE = -1,   /**< Invalid Element helper ID.             */
+  ID_SCE = 0,     /**< Single Channel Element.                */
+  ID_CPE = 1,     /**< Channel Pair Element.                  */
+  ID_CCE = 2,     /**< Coupling Channel Element.              */
+  ID_LFE = 3,     /**< LFE Channel Element.                   */
+  ID_DSE = 4,     /**< Currently one Data Stream Element for ancillary data is supported. */
+  ID_PCE = 5,     /**< Program Config Element.                */
+  ID_FIL = 6,     /**< Fill Element.                          */
+  ID_END = 7,     /**< Arnie (End Element = Terminator).      */
+  ID_EXT = 8,     /**< Extension Payload (ER only).           */
+  ID_SCAL = 9,    /**< AAC scalable element (ER only).        */
+  ID_LAST
+} MP4_ELEMENT_ID;
+
+#define IS_CHANNEL_ELEMENT(elementId) \
+  ((elementId) == ID_SCE \
+|| (elementId) == ID_CPE \
+|| (elementId) == ID_LFE)
+
+#define EXT_ID_BITS 4  /**< Size in bits of extension payload type tags. */
+
+/** Extension payload types. */
+typedef enum {
+  EXT_FIL           = 0x00,
+  EXT_FILL_DATA     = 0x01,
+  EXT_DATA_ELEMENT  = 0x02,
+  EXT_DATA_LENGTH   = 0x03,
+  EXT_LDSAC_DATA    = 0x09,
+  EXT_SAOC_DATA     = 0x0a,
+  EXT_DYNAMIC_RANGE = 0x0b,
+  EXT_SAC_DATA      = 0x0c,
+  EXT_SBR_DATA      = 0x0d,
+  EXT_SBR_DATA_CRC  = 0x0e
+} EXT_PAYLOAD_TYPE;
+
+
+/**
+ * Proprietary raw packet file configuration data type identifier.
+ */
+typedef enum
+{
+  TC_NOTHING = 0,     /* No configuration available -> in-band configuration.   */
+  TC_RAW_ASC,         /* Configuration data field is a raw AudioSpecificConfig. */
+  TC_RAW_SMC,         /* Configuration data field is a raw StreamMuxConfig.     */
+  TC_RAW_SDC          /* Configuration data field is a raw Drm SDC.             */
+
+} TP_CONFIG_TYPE;
+
+/*
+ * ##############################################################################################
+ * Library identification and error handling
+ * ##############################################################################################
+ */
+/* \cond */
+#define MODULE_ID_MASK  (0x000000ff)
+#define MODULE_ID_SHIFT (24)
+
+typedef enum {
+  FDK_NONE   = 0,
+  FDK_TOOLS  = 1,
+  FDK_SYSLIB = 2,
+  FDK_AACDEC = 3,
+  FDK_AACENC = 4,
+  FDK_SBRDEC = 5,
+  FDK_SBRENC = 6,
+  FDK_TPDEC  = 7,
+  FDK_TPENC  = 8,
+  FDK_MPSDEC = 9,
+  FDK_MPEGFILEREAD  = 10,
+  FDK_MPEGFILEWRITE = 11,
+  FDK_MP2DEC = 12,
+  FDK_DABDEC = 13,
+  FDK_DABPARSE = 14,
+  FDK_DRMDEC = 15,
+  FDK_DRMPARSE = 16,
+  FDK_AACLDENC = 17,
+  FDK_MP2ENC = 18,
+  FDK_MP3ENC = 19,
+  FDK_MP3DEC = 20,
+  FDK_MP3HEADPHONE = 21,
+  FDK_MP3SDEC = 22,
+  FDK_MP3SENC = 23,
+  FDK_EAEC    = 24,
+  FDK_DABENC  = 25,
+  FDK_DMBDEC  = 26,
+  FDK_FDREVERB = 27,
+  FDK_DRMENC  = 28,
+  FDK_METADATATRANSCODER = 29,
+  FDK_AC3DEC = 30,
+  FDK_PCMDMX = 31,
+
+  FDK_MODULE_LAST
+
+} FDK_MODULE_ID;
+
+/* AAC capability flags */
+#define CAPF_AAC_LC           0x00000001  /**< Support flag for AAC Low Complexity.                             */
+#define CAPF_ER_AAC_LD        0x00000002  /**< Support flag for AAC Low Delay with Error Resilience tools.      */
+#define CAPF_ER_AAC_SCAL      0x00000004  /**< Support flag for AAC Scalable.                                   */
+#define CAPF_ER_AAC_LC        0x00000008  /**< Support flag for AAC Low Complexity with Error Resilience tools. */
+#define CAPF_AAC_480          0x00000010  /**< Support flag for AAC with 480 framelength.  */
+#define CAPF_AAC_512          0x00000020  /**< Support flag for AAC with 512 framelength.  */
+#define CAPF_AAC_960          0x00000040  /**< Support flag for AAC with 960 framelength.  */
+#define CAPF_AAC_1024         0x00000080  /**< Support flag for AAC with 1024 framelength. */
+#define CAPF_AAC_HCR          0x00000100  /**< Support flag for AAC with Huffman Codeword Reordering.    */
+#define CAPF_AAC_VCB11        0x00000200  /**< Support flag for AAC Virtual Codebook 11.    */
+#define CAPF_AAC_RVLC         0x00000400  /**< Support flag for AAC Reversible Variable Length Coding.   */
+#define CAPF_AAC_MPEG4        0x00000800  /**< Support flag for MPEG file format.                           */
+#define CAPF_AAC_DRC          0x00001000  /**< Support flag for AAC Dynamic Range Control. */
+#define CAPF_AAC_CONCEALMENT  0x00002000  /**< Support flag for AAC concealment.           */
+#define CAPF_AAC_DRM_BSFORMAT 0x00004000  /**< Support flag for AAC DRM bistream format.                         */
+#define CAPF_ER_AAC_ELD       0x00008000  /**< Support flag for AAC Enhanced Low Delay with Error Resilience tools.  */
+#define CAPF_ER_AAC_BSAC      0x00010000  /**< Support flag for AAC BSAC.                           */
+#define CAPF_AAC_SUPERFRAMING 0x00020000  /**< Support flag for AAC Superframing.                           */
+
+/* Transport capability flags */
+#define CAPF_ADTS             0x00000001  /**< Support flag for ADTS transport format.        */
+#define CAPF_ADIF             0x00000002  /**< Support flag for ADIF transport format.        */
+#define CAPF_LATM             0x00000004  /**< Support flag for LATM transport format.        */
+#define CAPF_LOAS             0x00000008  /**< Support flag for LOAS transport format.        */
+#define CAPF_RAWPACKETS       0x00000010  /**< Support flag for RAW PACKETS transport format. */
+#define CAPF_DRM              0x00000020  /**< Support flag for DRM/DRM+ transport format.    */
+#define CAPF_RSVD50           0x00000040  /**< Support flag for RSVD50 transport format       */
+
+/* SBR capability flags */
+#define CAPF_SBR_LP           0x00000001  /**< Support flag for SBR Low Power mode.           */
+#define CAPF_SBR_HQ           0x00000002  /**< Support flag for SBR High Quality mode.        */
+#define CAPF_SBR_DRM_BS       0x00000004  /**< Support flag for                               */
+#define CAPF_SBR_CONCEALMENT  0x00000008  /**< Support flag for SBR concealment.              */
+#define CAPF_SBR_DRC          0x00000010  /**< Support flag for SBR Dynamic Range Control.    */
+#define CAPF_SBR_PS_MPEG      0x00000020  /**< Support flag for MPEG Parametric Stereo.       */
+#define CAPF_SBR_PS_DRM       0x00000040  /**< Support flag for DRM Parametric Stereo.        */
+
+/* MP2 encoder capability flags */
+#define CAPF_MP2ENC_SS        0x00000001  /**< Support flag for Seamless Switching.           */
+#define CAPF_MP2ENC_DAB       0x00000002  /**< Support flag for Layer2 DAB.                   */
+
+/* DAB capability flags */
+#define CAPF_DAB_MP2          0x00000001  /**< Support flag for Layer2 DAB.                   */
+#define CAPF_DAB_AAC          0x00000002  /**< Support flag for DAB+ (HE-AAC v2).             */
+#define CAPF_DAB_PAD          0x00000004  /**< Support flag for PAD extraction.               */
+#define CAPF_DAB_DRC          0x00000008  /**< Support flag for Dynamic Range Control.        */
+#define CAPF_DAB_SURROUND     0x00000010  /**< Support flag for DAB Surround (MPS).           */
+
+/* DMB capability flags */
+#define CAPF_DMB_BSAC         0x00000001  /**< Support flag for ER AAC BSAC.                  */
+#define CAPF_DMB_DRC          0x00000008  /**< Support flag for Dynamic Range Control.        */
+#define CAPF_DMB_SURROUND     0x00000010  /**< Support flag for DMB Surround (MPS).           */
+
+/* PCM up/downmmix capability flags */
+#define CAPF_DMX_BLIND        0x00000001  /**< Support flag for blind downmixing.             */
+#define CAPF_DMX_PCE          0x00000002  /**< Support flag for guided downmix with data from MPEG-2/4 Program Config Elements (PCE). */
+#define CAPF_DMX_ARIB         0x00000004  /**< Support flag for PCE guided downmix with slightly different equations and levels to fulfill ARIB standard. */
+#define CAPF_DMX_DVB          0x00000008  /**< Support flag for guided downmix with data from DVB ancillary data fields. */
+#define CAPF_DMX_CH_EXP       0x00000010  /**< Support flag for simple upmixing by dublicating channels or adding zero channels. */
+/* \endcond */
+
+
+/*
+ * ##############################################################################################
+ * Library versioning
+ * ##############################################################################################
+ */
+
+/**
+ * Convert each member of version numbers to one single numeric version representation.
+ * \param lev0  1st level of version number.
+ * \param lev1  2nd level of version number.
+ * \param lev2  3rd level of version number.
+  */
+#define LIB_VERSION(lev0, lev1, lev2) ((lev0<<24 & 0xff000000) | \
+                                       (lev1<<16 & 0x00ff0000) | \
+                                       (lev2<<8  & 0x0000ff00))
+
+/**
+ *  Build text string of version.
+ */
+#define LIB_VERSION_STRING(info) FDKsprintf((info)->versionStr, "%d.%d.%d", (((info)->version >> 24) & 0xff), (((info)->version >> 16) & 0xff), (((info)->version >> 8 ) & 0xff))
+
+/**
+ *  Library information.
+ */
+typedef struct LIB_INFO
+{
+  const char* title;
+  const char* build_date;
+  const char* build_time;
+  FDK_MODULE_ID module_id;
+  INT version;
+  UINT  flags;
+  char versionStr[32];
+} LIB_INFO;
+
+/** Initialize library info. */
+static inline void FDKinitLibInfo( LIB_INFO* info )
+{
+  int i;
+
+  for (i = 0; i < FDK_MODULE_LAST; i++) {
+    info[i].module_id = FDK_NONE;
+  }
+}
+
+/** Aquire supported features of library. */
+static inline UINT FDKlibInfo_getCapabilities( const LIB_INFO* info, FDK_MODULE_ID module_id )
+{
+  int i;
+
+  for (i=0; i<FDK_MODULE_LAST; i++) {
+    if (info[i].module_id == module_id) {
+      return info[i].flags;
+    }
+  }
+  return 0;
+}
+
+/** Search for next free tab. */
+static inline INT FDKlibInfo_lookup( const LIB_INFO* info, FDK_MODULE_ID module_id )
+{
+  int i = -1;
+
+  for (i = 0; i < FDK_MODULE_LAST; i++) {
+    if (info[i].module_id == module_id)
+      return -1;
+    if (info[i].module_id == FDK_NONE)
+      break;
+  }
+  if (i == FDK_MODULE_LAST)
+    return -1;
+
+  return i;
+}
+
+
+/*
+ * ##############################################################################################
+ * Buffer description
+ * ##############################################################################################
+ */
+
+/**
+ *  I/O buffer descriptor.
+ */
+typedef struct FDK_bufDescr
+{
+  void  **ppBase;     /*!< Pointer to an array containing buffer base addresses.
+                           Set to NULL for buffer requirement info. */
+  UINT   *pBufSize;   /*!< Pointer to an array containing the number of elements that can
+                           be placed in the specific buffer. */
+  UINT   *pEleSize;   /*!< Pointer to an array containing the element size for each buffer
+                           in bytes. That is mostly the number returned by the sizeof()
+                           operator for the data type used for the specific buffer. */
+  UINT   *pBufType;   /*!< Pointer to an array of bit fields containing a description
+                           for each buffer. See XXX below for more details.  */
+  UINT    numBufs;    /*!< Total number of buffers. */
+
+} FDK_bufDescr;
+
+/**
+ * Buffer type description field.
+ */
+#define FDK_BUF_TYPE_MASK_IO     ( 0x03 << 30 )
+#define FDK_BUF_TYPE_MASK_DESCR  ( 0x3F << 16 )
+#define FDK_BUF_TYPE_MASK_ID     ( 0xFF )
+
+#define FDK_BUF_TYPE_INPUT       ( 0x1 << 30 )
+#define FDK_BUF_TYPE_OUTPUT      ( 0x2 << 30 )
+
+#define FDK_BUF_TYPE_PCM_DATA    ( 0x1 << 16 )
+#define FDK_BUF_TYPE_ANC_DATA    ( 0x2 << 16 )
+#define FDK_BUF_TYPE_BS_DATA     ( 0x4 << 16 )
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif /* FDK_AUDIO_H */