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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
+   Description: MPEG Transport data tables
+
+******************************************************************************/
+
+#ifndef __TP_DATA_H__
+#define __TP_DATA_H__
+
+#include "machine_type.h"
+#include "FDK_audio.h"
+#include "FDK_bitstream.h"
+
+/*
+ * Configuration
+ */
+#define TP_GA_ENABLE
+/* #define TP_CELP_ENABLE */
+/* #define TP_HVXC_ENABLE */
+/* #define TP_SLS_ENABLE */
+#define TP_ELD_ENABLE
+/* #define TP_USAC_ENABLE */
+/* #define TP_RSVD50_ENABLE */
+
+#if defined(TP_GA_ENABLE) || defined(TP_SLS_ENABLE)
+#define TP_PCE_ENABLE       /**< Enable full PCE support */
+#endif
+
+/**
+ * ProgramConfig struct.
+ */
+/* ISO/IEC 14496-3 4.4.1.1 Table 4.2 Program config element */
+#define PC_FSB_CHANNELS_MAX 16 /* Front/Side/Back channels */
+#define PC_LFE_CHANNELS_MAX 4
+#define PC_ASSOCDATA_MAX    8
+#define PC_CCEL_MAX         16 /* CC elements */
+#define PC_COMMENTLENGTH    256
+
+typedef struct
+{
+#ifdef TP_PCE_ENABLE
+  /* PCE bitstream elements: */
+  UCHAR ElementInstanceTag;
+  UCHAR Profile;
+  UCHAR SamplingFrequencyIndex;
+  UCHAR NumFrontChannelElements;
+  UCHAR NumSideChannelElements;
+  UCHAR NumBackChannelElements;
+  UCHAR NumLfeChannelElements;
+  UCHAR NumAssocDataElements;
+  UCHAR NumValidCcElements;
+
+  UCHAR MonoMixdownPresent;
+  UCHAR MonoMixdownElementNumber;
+
+  UCHAR StereoMixdownPresent;
+  UCHAR StereoMixdownElementNumber;
+
+  UCHAR MatrixMixdownIndexPresent;
+  UCHAR MatrixMixdownIndex;
+  UCHAR PseudoSurroundEnable;
+
+  UCHAR FrontElementIsCpe[PC_FSB_CHANNELS_MAX];
+  UCHAR FrontElementTagSelect[PC_FSB_CHANNELS_MAX];
+
+  UCHAR SideElementIsCpe[PC_FSB_CHANNELS_MAX];
+  UCHAR SideElementTagSelect[PC_FSB_CHANNELS_MAX];
+
+  UCHAR BackElementIsCpe[PC_FSB_CHANNELS_MAX];
+  UCHAR BackElementTagSelect[PC_FSB_CHANNELS_MAX];
+
+  UCHAR LfeElementTagSelect[PC_LFE_CHANNELS_MAX];
+
+  UCHAR AssocDataElementTagSelect[PC_ASSOCDATA_MAX];
+
+  UCHAR CcElementIsIndSw[PC_CCEL_MAX];
+  UCHAR ValidCcElementTagSelect[PC_CCEL_MAX];
+
+  UCHAR CommentFieldBytes;
+  UCHAR Comment[PC_COMMENTLENGTH];
+#endif /* TP_PCE_ENABLE */
+
+  /* Helper variables for administration: */
+  UCHAR isValid;               /*!< Flag showing if PCE has been read successfully. */
+  UCHAR NumChannels;           /*!< Amount of audio channels summing all channel elements including LFEs */
+  UCHAR NumEffectiveChannels;  /*!< Amount of audio channels summing only SCEs and CPEs */
+  UCHAR elCounter;
+
+} CProgramConfig;
+
+typedef enum {
+  ASCEXT_UNKOWN = -1,
+  ASCEXT_SBR    = 0x2b7,
+  ASCEXT_PS     = 0x548,
+  ASCEXT_MPS    = 0x76a,
+  ASCEXT_SAOC   = 0x7cb,
+  ASCEXT_LDMPS  = 0x7cc
+
+} TP_ASC_EXTENSION_ID;
+
+#ifdef TP_GA_ENABLE
+/**
+ * GaSpecificConfig struct
+ */
+typedef struct {
+  UINT m_frameLengthFlag ;
+  UINT m_dependsOnCoreCoder ;
+  UINT m_coreCoderDelay ;
+
+  UINT m_extensionFlag ;
+  UINT m_extensionFlag3 ;
+
+  UINT m_layer;
+  UINT m_numOfSubFrame;
+  UINT m_layerLength;
+
+} CSGaSpecificConfig;
+#endif /* TP_GA_ENABLE */
+
+
+
+
+#ifdef TP_ELD_ENABLE
+
+typedef enum {
+  ELDEXT_TERM  = 0x0,   /* Termination tag */
+  ELDEXT_SAOC  = 0x1,   /* SAOC config */
+  ELDEXT_LDSAC = 0x2    /* LD MPEG Surround config */
+  /* reserved */
+} ASC_ELD_EXT_TYPE;
+
+typedef struct {
+  UCHAR m_frameLengthFlag;
+
+  UCHAR m_sbrPresentFlag;
+  UCHAR m_useLdQmfTimeAlign;  /* Use LD-MPS QMF in SBR to achive time alignment */
+  UCHAR m_sbrSamplingRate;
+  UCHAR m_sbrCrcFlag;
+
+} CSEldSpecificConfig;
+#endif /* TP_ELD_ENABLE */
+
+
+
+
+/**
+ * Audio configuration struct, suitable for encoder and decoder configuration.
+ */
+typedef struct {
+
+  /* XYZ Specific Data */
+  union {
+#ifdef TP_GA_ENABLE
+    CSGaSpecificConfig    m_gaSpecificConfig;       /**< General audio specific configuration.         */
+#endif /* TP_GA_ENABLE */
+#ifdef TP_ELD_ENABLE
+    CSEldSpecificConfig   m_eldSpecificConfig;      /**< ELD specific configuration.                   */
+#endif /* TP_ELD_ENABLE */
+  } m_sc;
+  
+  /* Common ASC parameters */
+#ifdef TP_PCE_ENABLE
+  CProgramConfig        m_progrConfigElement;     /**< Program configuration.                          */
+#endif /* TP_PCE_ENABLE */
+
+  AUDIO_OBJECT_TYPE     m_aot;                    /**< Audio Object Type.                              */
+  UINT                  m_samplingFrequency;      /**< Samplerate.                                     */
+  UINT                  m_samplesPerFrame;        /**< Amount of samples per frame.                    */
+  UINT                  m_directMapping;          /**< Document this please !!                         */
+
+  AUDIO_OBJECT_TYPE     m_extensionAudioObjectType;        /**< Audio object type                      */
+  UINT                  m_extensionSamplingFrequency;      /**< Samplerate                             */  
+
+  SCHAR                 m_channelConfiguration;   /**< Channel configuration index                     */
+
+  SCHAR                 m_epConfig;               /**< Error protection index                           */
+  SCHAR                 m_vcb11Flag;              /**< aacSectionDataResilienceFlag                     */
+  SCHAR                 m_rvlcFlag;               /**< aacScalefactorDataResilienceFlag                 */
+  SCHAR                 m_hcrFlag;                /**< aacSpectralDataResilienceFlag                    */
+
+  SCHAR                 m_sbrPresentFlag;         /**< Flag indicating the presence of SBR data in the bitstream               */
+  SCHAR                 m_psPresentFlag;          /**< Flag indicating the presence of parametric stereo data in the bitstream */
+  UCHAR                 m_samplingFrequencyIndex; /**< Samplerate index                                 */
+  UCHAR                 m_extensionSamplingFrequencyIndex; /**< Samplerate index                        */
+  SCHAR                 m_extensionChannelConfiguration;   /**< Channel configuration index             */
+
+} CSAudioSpecificConfig;
+
+typedef INT (*cbUpdateConfig_t)(void*, const CSAudioSpecificConfig*);
+typedef INT (*cbSsc_t)(
+        void*, HANDLE_FDK_BITSTREAM,
+        const AUDIO_OBJECT_TYPE coreCodec,
+        const INT samplingFrequency,
+        const INT muxMode,
+        const INT configBytes
+        );
+typedef INT (*cbSbr_t)(
+        void *                  self,
+        HANDLE_FDK_BITSTREAM    hBs,
+        const INT sampleRateIn,
+        const INT sampleRateOut,
+        const INT samplesPerFrame,
+        const AUDIO_OBJECT_TYPE coreCodec,
+        const MP4_ELEMENT_ID    elementID,
+        const INT               elementIndex
+        );
+
+typedef struct  {
+  cbUpdateConfig_t cbUpdateConfig; /*!< Function pointer for Config change notify callback.  */
+  void *cbUpdateConfigData;        /*!< User data pointer for Config change notify callback. */
+  cbSsc_t cbSsc;                   /*!< Function pointer for SSC parser callback. */
+  void *cbSscData;                 /*!< User data pointer for SSC parser callback. */
+  cbSbr_t cbSbr;                   /*!< Function pointer for SBR header parser callback. */
+  void *cbSbrData;                 /*!< User data pointer for SBR header parser callback. */
+} CSTpCallBacks;
+
+static const UINT SamplingRateTable[] = 
+{ 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, 16000, 12000, 11025, 8000, 7350, 0, 0,
+  0
+};
+
+static inline
+int getSamplingRateIndex( UINT samplingRate )
+{
+  UINT sf_index, tableSize=sizeof(SamplingRateTable)/sizeof(UINT);
+
+  for (sf_index=0; sf_index<tableSize; sf_index++) {
+    if( SamplingRateTable[sf_index] == samplingRate ) break;
+  }
+
+  if (sf_index>tableSize-1) {
+    return tableSize-1;
+  }
+
+  return sf_index;
+}
+
+/*
+ * Get Channel count from channel configuration
+ */
+static inline int getNumberOfTotalChannels(int channelConfig)
+{
+  if (channelConfig > 0 && channelConfig < 8)
+    return (channelConfig == 7)?8:channelConfig;
+  else
+    return 0;
+}
+
+static inline
+int getNumberOfEffectiveChannels(const int channelConfig)
+{
+  const int n[] = {0,1,2,3,4,5,5,7};
+  return n[channelConfig];
+}
+
+#endif /* __TP_DATA_H__ */