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, 466 insertions, 0 deletions

diff --git a/fdk-aac/libMpegTPDec/include/tp_data.h b/fdk-aac/libMpegTPDec/include/tp_data.h
new file mode 100644
index 0000000..b015332
--- /dev/null
+++ b/fdk-aac/libMpegTPDec/include/tp_data.h
@@ -0,0 +1,466 @@
+/* -----------------------------------------------------------------------------
+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
+----------------------------------------------------------------------------- */
+
+/******************* MPEG transport format decoder library *********************
+
+   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_USAC_MAX_SPEAKERS (24)
+
+#define TP_USAC_MAX_EXT_ELEMENTS ((24))
+
+#define TP_USAC_MAX_ELEMENTS ((24) + TP_USAC_MAX_EXT_ELEMENTS)
+
+#define TP_USAC_MAX_CONFIG_LEN                                         \
+  512 /* next power of two of maximum of escapedValue(hBs, 4, 4, 8) in \
+         AudioPreRoll() (285) */
+
+#define TPDEC_USAC_NUM_CONFIG_CHANGE_FRAMES \
+  (1) /* Number of frames for config change in USAC */
+
+enum {
+  TPDEC_FLUSH_OFF = 0,
+  TPDEC_RSV60_CFG_CHANGE_ATSC_FLUSH_ON = 1,
+  TPDEC_RSV60_DASH_IPF_ATSC_FLUSH_ON = 2,
+  TPDEC_USAC_DASH_IPF_FLUSH_ON = 3
+};
+
+enum {
+  TPDEC_BUILD_UP_OFF = 0,
+  TPDEC_RSV60_BUILD_UP_ON = 1,
+  TPDEC_RSV60_BUILD_UP_ON_IN_BAND = 2,
+  TPDEC_USAC_BUILD_UP_ON = 3,
+  TPDEC_RSV60_BUILD_UP_IDLE = 4,
+  TPDEC_RSV60_BUILD_UP_IDLE_IN_BAND = 5
+};
+
+/**
+ * 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
+#define PC_NUM_HEIGHT_LAYER 3
+
+typedef struct {
+  /* 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 FrontElementHeightInfo[PC_FSB_CHANNELS_MAX];
+
+  UCHAR SideElementIsCpe[PC_FSB_CHANNELS_MAX];
+  UCHAR SideElementTagSelect[PC_FSB_CHANNELS_MAX];
+  UCHAR SideElementHeightInfo[PC_FSB_CHANNELS_MAX];
+
+  UCHAR BackElementIsCpe[PC_FSB_CHANNELS_MAX];
+  UCHAR BackElementTagSelect[PC_FSB_CHANNELS_MAX];
+  UCHAR BackElementHeightInfo[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];
+
+  /* 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;
+
+/**
+ * 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;
+
+typedef enum {
+  ELDEXT_TERM = 0x0,         /* Termination tag */
+  ELDEXT_SAOC = 0x1,         /* SAOC config */
+  ELDEXT_LDSAC = 0x2,        /* LD MPEG Surround config */
+  ELDEXT_DOWNSCALEINFO = 0x3 /* ELD sample rate adaptation */
+  /* 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;
+  UINT m_downscaledSamplingFrequency;
+
+} CSEldSpecificConfig;
+
+typedef struct {
+  USAC_EXT_ELEMENT_TYPE usacExtElementType;
+  USHORT usacExtElementConfigLength;
+  USHORT usacExtElementDefaultLength;
+  UCHAR usacExtElementPayloadFrag;
+  UCHAR usacExtElementHasAudioPreRoll;
+} CSUsacExtElementConfig;
+
+typedef struct {
+  MP4_ELEMENT_ID usacElementType;
+  UCHAR m_noiseFilling;
+  UCHAR m_harmonicSBR;
+  UCHAR m_interTes;
+  UCHAR m_pvc;
+  UCHAR m_stereoConfigIndex;
+  CSUsacExtElementConfig extElement;
+} CSUsacElementConfig;
+
+typedef struct {
+  UCHAR m_frameLengthFlag;
+  UCHAR m_coreSbrFrameLengthIndex;
+  UCHAR m_sbrRatioIndex;
+  UCHAR m_nUsacChannels; /* number of audio channels signaled in
+                            UsacDecoderConfig() / rsv603daDecoderConfig() via
+                            numElements and usacElementType */
+  UCHAR m_channelConfigurationIndex;
+  UINT m_usacNumElements;
+  CSUsacElementConfig element[TP_USAC_MAX_ELEMENTS];
+
+  UCHAR numAudioChannels;
+  UCHAR m_usacConfigExtensionPresent;
+  UCHAR elementLengthPresent;
+  UCHAR UsacConfig[TP_USAC_MAX_CONFIG_LEN];
+  USHORT UsacConfigBits;
+} CSUsacConfig;
+
+/**
+ * Audio configuration struct, suitable for encoder and decoder configuration.
+ */
+typedef struct {
+  /* XYZ Specific Data */
+  union {
+    CSGaSpecificConfig
+        m_gaSpecificConfig; /**< General audio specific configuration. */
+    CSEldSpecificConfig m_eldSpecificConfig; /**< ELD specific configuration. */
+    CSUsacConfig m_usacConfig; /**< USAC specific configuration               */
+  } m_sc;
+
+  /* Common ASC parameters */
+  CProgramConfig m_progrConfigElement; /**< Program configuration. */
+
+  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   */
+
+  UCHAR
+  configMode; /**< The flag indicates if the callback shall work in memory
+                 allocation mode or in config change detection mode */
+  UCHAR AacConfigChanged; /**< The flag will be set if at least one aac config
+                             parameter has changed that requires a memory
+                             reconfiguration, otherwise it will be cleared */
+  UCHAR SbrConfigChanged; /**< The flag will be set if at least one sbr config
+                             parameter has changed that requires a memory
+                             reconfiguration, otherwise it will be cleared */
+  UCHAR SacConfigChanged; /**< The flag will be set if at least one sac config
+                             parameter has changed that requires a memory
+                             reconfiguration, otherwise it will be cleared */
+
+  UCHAR
+  config[TP_USAC_MAX_CONFIG_LEN]; /**< Configuration stored as bitstream */
+  UINT configBits;                /**< Configuration length in bits */
+
+} CSAudioSpecificConfig;
+
+typedef struct {
+  SCHAR flushCnt;      /**< Flush frame counter */
+  UCHAR flushStatus;   /**< Flag indicates flush mode: on|off */
+  SCHAR buildUpCnt;    /**< Build up frame counter */
+  UCHAR buildUpStatus; /**< Flag indicates build up mode: on|off */
+  UCHAR cfgChanged; /**< Flag indicates that the config changed and the decoder
+                       needs to be initialized again via callback. Make sure
+                       that memory is freed before initialization. */
+  UCHAR contentChanged; /**< Flag indicates that the content changed i.e. a
+                           right truncation occured before */
+  UCHAR forceCfgChange; /**< Flag indicates if config change has to be forced
+                           even if new config is the same */
+} CCtrlCFGChange;
+
+typedef INT (*cbUpdateConfig_t)(void *, const CSAudioSpecificConfig *,
+                                const UCHAR configMode, UCHAR *configChanged);
+typedef INT (*cbFreeMem_t)(void *, const CSAudioSpecificConfig *);
+typedef INT (*cbCtrlCFGChange_t)(void *, const CCtrlCFGChange *);
+typedef INT (*cbSsc_t)(void *, HANDLE_FDK_BITSTREAM,
+                       const AUDIO_OBJECT_TYPE coreCodec,
+                       const INT samplingRate, const INT frameSize,
+                       const INT stereoConfigIndex,
+                       const INT coreSbrFrameLengthIndex, const INT configBytes,
+                       const UCHAR configMode, UCHAR *configChanged);
+
+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,
+                       const UCHAR harmonicSbr, const UCHAR stereoConfigIndex,
+                       const UCHAR configMode, UCHAR *configChanged,
+                       const INT downscaleFactor);
+
+typedef INT (*cbUsac_t)(void *self, HANDLE_FDK_BITSTREAM hBs);
+
+typedef INT (*cbUniDrc_t)(void *self, HANDLE_FDK_BITSTREAM hBs,
+                          const INT fullPayloadLength, const INT payloadType,
+                          const INT subStreamIndex, const INT payloadStart,
+                          const AUDIO_OBJECT_TYPE);
+
+typedef struct {
+  cbUpdateConfig_t cbUpdateConfig; /*!< Function pointer for Config change
+                                      notify callback.  */
+  void *cbUpdateConfigData; /*!< User data pointer for Config change notify
+                               callback. */
+  cbFreeMem_t cbFreeMem;    /*!< Function pointer for free memory callback.  */
+  void *cbFreeMemData;      /*!< User data pointer for free memory callback. */
+  cbCtrlCFGChange_t cbCtrlCFGChange; /*!< Function pointer for config change
+                                        control callback. */
+  void *cbCtrlCFGChangeData; /*!< User data pointer for config change control
+                                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. */
+  cbUsac_t cbUsac;
+  void *cbUsacData;
+  cbUniDrc_t cbUniDrc; /*!< Function pointer for uniDrcConfig and
+                          loudnessInfoSet parser callback. */
+  void *cbUniDrcData;  /*!< User data pointer for uniDrcConfig and
+                          loudnessInfoSet parser callback. */
+} CSTpCallBacks;
+
+static const UINT SamplingRateTable[] = {
+    96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, 16000, 12000, 11025,
+    8000,  7350,  0,     0,     57600, 51200, 40000, 38400, 34150, 28800, 25600,
+    20000, 19200, 17075, 14400, 12800, 9600,  0,     0,     0,     0};
+
+static inline int getSamplingRateIndex(UINT samplingRate, UINT nBits) {
+  UINT sf_index;
+  UINT tableSize = (1 << nBits) - 1;
+
+  for (sf_index = 0; sf_index < tableSize; sf_index++) {
+    if (SamplingRateTable[sf_index] == samplingRate) break;
+  }
+
+  if (sf_index > tableSize) {
+    return tableSize - 1;
+  }
+
+  return sf_index;
+}
+
+/*
+ * Get Channel count from channel configuration
+ */
+static inline int getNumberOfTotalChannels(int channelConfig) {
+  switch (channelConfig) {
+    case 1:
+    case 2:
+    case 3:
+    case 4:
+    case 5:
+    case 6:
+      return channelConfig;
+    case 7:
+    case 12:
+    case 14:
+      return 8;
+    case 11:
+      return 7;
+    case 13:
+      return 24;
+    default:
+      return 0;
+  }
+}
+
+static inline int getNumberOfEffectiveChannels(
+    const int
+        channelConfig) { /* index: 0,1,2,3,4,5,6,7,8,9,10,11,12,13,14,15 */
+  const int n[] = {0, 1, 2, 3, 4, 5, 5, 7, 0, 0, 0, 6, 7, 22, 7, 0};
+  return n[channelConfig];
+}
+
+#endif /* TP_DATA_H */