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, 465 insertions, 0 deletions

diff --git a/fdk-aac/libAACdec/src/aacdecoder.h b/fdk-aac/libAACdec/src/aacdecoder.h
new file mode 100644
index 0000000..20f4c45
--- /dev/null
+++ b/fdk-aac/libAACdec/src/aacdecoder.h
@@ -0,0 +1,465 @@
+/* -----------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+Forschung e.V. All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software
+that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding
+scheme for digital audio. This FDK AAC Codec software is intended to be used on
+a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient
+general perceptual audio codecs. AAC-ELD is considered the best-performing
+full-bandwidth communications codec by independent studies and is widely
+deployed. AAC has been standardized by ISO and IEC as part of the MPEG
+specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including
+those of Fraunhofer) may be obtained through Via Licensing
+(www.vialicensing.com) or through the respective patent owners individually for
+the purpose of encoding or decoding bit streams in products that are compliant
+with the ISO/IEC MPEG audio standards. Please note that most manufacturers of
+Android devices already license these patent claims through Via Licensing or
+directly from the patent owners, and therefore FDK AAC Codec software may
+already be covered under those patent licenses when it is used for those
+licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions
+with enhanced sound quality, are also available from Fraunhofer. Users are
+encouraged to check the Fraunhofer website for additional applications
+information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification,
+are permitted without payment of copyright license fees provided that you
+satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of
+the FDK AAC Codec or your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation
+and/or other materials provided with redistributions of the FDK AAC Codec or
+your modifications thereto in binary form. You must make available free of
+charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived
+from this library without prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute
+the FDK AAC Codec software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating
+that you changed the software and the date of any change. For modified versions
+of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android"
+must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK
+AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without
+limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE.
+Fraunhofer provides no warranty of patent non-infringement with respect to this
+software.
+
+You may use this FDK AAC Codec software or modifications thereto only for
+purposes that are authorized by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright
+holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES,
+including but not limited to the implied warranties of merchantability and
+fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary,
+or consequential damages, including but not limited to procurement of substitute
+goods or services; loss of use, data, or profits, or business interruption,
+however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of
+this software, even if advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------- */
+
+/**************************** AAC decoder library ******************************
+
+   Author(s):   Josef Hoepfl
+
+   Description:
+
+*******************************************************************************/
+
+#ifndef AACDECODER_H
+#define AACDECODER_H
+
+#include "common_fix.h"
+
+#include "FDK_bitstream.h"
+
+#include "channel.h"
+
+#include "tpdec_lib.h"
+#include "FDK_audio.h"
+
+#include "block.h"
+
+#include "genericStds.h"
+
+#include "FDK_qmf_domain.h"
+
+#include "sbrdecoder.h"
+
+#include "aacdec_drc.h"
+
+#include "pcmdmx_lib.h"
+
+#include "FDK_drcDecLib.h"
+
+#include "limiter.h"
+
+#include "FDK_delay.h"
+
+#define TIME_DATA_FLUSH_SIZE (128)
+#define TIME_DATA_FLUSH_SIZE_SF (7)
+
+#define AACDEC_MAX_NUM_PREROLL_AU_USAC (3)
+#if (AACDEC_MAX_NUM_PREROLL_AU < 3)
+#undef AACDEC_MAX_NUM_PREROLL_AU
+#define AACDEC_MAX_NUM_PREROLL_AU AACDEC_MAX_NUM_PREROLL_AU_USAC
+#endif
+
+typedef struct AAC_DECODER_INSTANCE *HANDLE_AACDECODER;
+
+enum { L = 0, R = 1 };
+
+typedef struct {
+  unsigned char *buffer;
+  int bufferSize;
+  int offset[8];
+  int nrElements;
+} CAncData;
+
+typedef enum { NOT_DEFINED = -1, MODE_HQ = 0, MODE_LP = 1 } QMF_MODE;
+
+typedef struct {
+  int bsDelay;
+} SBR_PARAMS;
+
+enum {
+  AACDEC_FLUSH_OFF = 0,
+  AACDEC_RSV60_CFG_CHANGE_ATSC_FLUSH_ON = 1,
+  AACDEC_RSV60_DASH_IPF_ATSC_FLUSH_ON = 2,
+  AACDEC_USAC_DASH_IPF_FLUSH_ON = 3
+};
+
+enum {
+  AACDEC_BUILD_UP_OFF = 0,
+  AACDEC_RSV60_BUILD_UP_ON = 1,
+  AACDEC_RSV60_BUILD_UP_ON_IN_BAND = 2,
+  AACDEC_USAC_BUILD_UP_ON = 3,
+  AACDEC_RSV60_BUILD_UP_IDLE = 4,
+  AACDEC_RSV60_BUILD_UP_IDLE_IN_BAND = 5
+};
+
+typedef struct {
+  /* Usac Extension Elements */
+  USAC_EXT_ELEMENT_TYPE usacExtElementType[(3)];
+  UINT usacExtElementDefaultLength[(3)];
+  UCHAR usacExtElementPayloadFrag[(3)];
+} CUsacCoreExtensions;
+
+/* AAC decoder (opaque toward userland) struct declaration */
+struct AAC_DECODER_INSTANCE {
+  INT aacChannels; /*!< Amount of AAC decoder channels allocated.        */
+  INT ascChannels[(1 *
+                   1)]; /*!< Amount of AAC decoder channels signalled in ASC. */
+  INT blockNumber;      /*!< frame counter                                    */
+
+  INT nrOfLayers;
+
+  INT outputInterleaved; /*!< PCM output format (interleaved/none interleaved).
+                          */
+
+  HANDLE_TRANSPORTDEC hInput; /*!< Transport layer handle. */
+
+  SamplingRateInfo
+      samplingRateInfo[(1 * 1)]; /*!< Sampling Rate information table */
+
+  UCHAR
+  frameOK; /*!< Will be unset if a consistency check, e.g. CRC etc. fails */
+
+  UINT flags[(1 * 1)]; /*!< Flags for internal decoder use. DO NOT USE
+                          self::streaminfo::flags ! */
+  UINT elFlags[(3 * ((8) * 2) + (((8) * 2)) / 2 + 4 * (1) +
+                1)]; /*!< Flags for internal decoder use (element specific). DO
+                        NOT USE self::streaminfo::flags ! */
+
+  MP4_ELEMENT_ID elements[(3 * ((8) * 2) + (((8) * 2)) / 2 + 4 * (1) +
+                           1)]; /*!< Table where the element Id's are listed */
+  UCHAR elTags[(3 * ((8) * 2) + (((8) * 2)) / 2 + 4 * (1) +
+                1)]; /*!< Table where the elements id Tags are listed      */
+  UCHAR chMapping[((8) * 2)]; /*!< Table of MPEG canonical order to bitstream
+                                 channel order mapping. */
+
+  AUDIO_CHANNEL_TYPE channelType[(8)]; /*!< Audio channel type of each output
+                                          audio channel (from 0 upto
+                                          numChannels).           */
+  UCHAR channelIndices[(8)]; /*!< Audio channel index for each output audio
+                                channel (from 0 upto numChannels).         */
+  /* See ISO/IEC 13818-7:2005(E), 8.5.3.2 Explicit channel mapping using a
+   * program_config_element() */
+
+  FDK_channelMapDescr mapDescr; /*!< Describes the output channel mapping. */
+  UCHAR chMapIndex; /*!< Index to access one line of the channelOutputMapping
+                       table. This is required because not all 8 channel
+                       configurations have the same output mapping. */
+  INT sbrDataLen;   /*!< Expected length of the SBR remaining in bitbuffer after
+                         the AAC payload has been pared.   */
+
+  CProgramConfig pce;
+  CStreamInfo
+      streamInfo; /*!< Pointer to StreamInfo data (read from the bitstream) */
+  CAacDecoderChannelInfo
+      *pAacDecoderChannelInfo[(8)]; /*!< Temporal channel memory */
+  CAacDecoderStaticChannelInfo
+      *pAacDecoderStaticChannelInfo[(8)]; /*!< Persistent channel memory */
+
+  FIXP_DBL *workBufferCore2;
+  PCM_DEC *pTimeData2;
+  INT timeData2Size;
+
+  CpePersistentData *cpeStaticData[(
+      3 * ((8) * 2) + (((8) * 2)) / 2 + 4 * (1) +
+      1)]; /*!< Pointer to persistent data shared by both channels of a CPE.
+This structure is allocated once for each CPE. */
+
+  CConcealParams concealCommonData;
+  CConcealmentMethod concealMethodUser;
+
+  CUsacCoreExtensions usacCoreExt; /*!< Data and handles to extend USAC FD/LPD
+                                      core decoder (SBR, MPS, ...) */
+  UINT numUsacElements[(1 * 1)];
+  UCHAR usacStereoConfigIndex[(3 * ((8) * 2) + (((8) * 2)) / 2 + 4 * (1) + 1)];
+  const CSUsacConfig *pUsacConfig[(1 * 1)];
+  INT nbDiv; /*!< number of frame divisions in LPD-domain */
+
+  UCHAR useLdQmfTimeAlign;
+
+  INT aacChannelsPrev; /*!< The amount of AAC core channels of the last
+                          successful decode call.         */
+  AUDIO_CHANNEL_TYPE channelTypePrev[(8)]; /*!< Array holding the channelType
+                                              values of the last successful
+                                              decode call.    */
+  UCHAR
+  channelIndicesPrev[(8)]; /*!< Array holding the channelIndices values of
+                              the last successful decode call. */
+
+  UCHAR
+  downscaleFactor; /*!< Variable to store a supported ELD downscale factor
+                      of 1, 2, 3 or 4 */
+  UCHAR downscaleFactorInBS; /*!< Variable to store the (not necessarily
+                                supported) ELD downscale factor discovered in
+                                the bitstream */
+
+  HANDLE_SBRDECODER hSbrDecoder; /*!< SBR decoder handle. */
+  UCHAR sbrEnabled;     /*!< flag to store if SBR has been detected     */
+  UCHAR sbrEnabledPrev; /*!< flag to store if SBR has been detected from
+                           previous frame */
+  UCHAR psPossible;     /*!< flag to store if PS is possible            */
+  SBR_PARAMS sbrParams; /*!< struct to store all sbr parameters         */
+
+  UCHAR *pDrmBsBuffer; /*!< Pointer to dynamic buffer which is used to reverse
+                          the bits of the DRM SBR payload */
+  USHORT drmBsBufferSize; /*!< Size of the dynamic buffer which is used to
+                             reverse the bits of the DRM SBR payload */
+  FDK_QMF_DOMAIN
+  qmfDomain; /*!< Instance of module for QMF domain data handling */
+
+  QMF_MODE qmfModeCurr; /*!< The current QMF mode                       */
+  QMF_MODE qmfModeUser; /*!< The QMF mode requested by the library user */
+
+  HANDLE_AAC_DRC hDrcInfo; /*!< handle to DRC data structure               */
+  INT metadataExpiry;      /*!< Metadata expiry time in milli-seconds.     */
+
+  void *pMpegSurroundDecoder; /*!< pointer to mpeg surround decoder structure */
+  UCHAR mpsEnableUser;        /*!< MPS enable user flag                       */
+  UCHAR mpsEnableCurr;        /*!< MPS enable decoder state                   */
+  UCHAR mpsApplicable;        /*!< MPS applicable                             */
+  SCHAR mpsOutputMode; /*!< setting: normal = 0, binaural = 1, stereo = 2, 5.1ch
+                          = 3 */
+  INT mpsOutChannelsLast; /*!< The amount of channels returned by the last
+                             successful MPS decoder call. */
+  INT mpsFrameSizeLast;   /*!< The frame length returned by the last successful
+                             MPS decoder call. */
+
+  CAncData ancData; /*!< structure to handle ancillary data         */
+
+  HANDLE_PCM_DOWNMIX hPcmUtils; /*!< privat data for the PCM utils. */
+
+  TDLimiterPtr hLimiter;   /*!< Handle of time domain limiter.             */
+  UCHAR limiterEnableUser; /*!< The limiter configuration requested by the
+                              library user */
+  UCHAR limiterEnableCurr; /*!< The current limiter configuration.         */
+  FIXP_DBL extGain[1]; /*!< Gain that must be applied to the output signal. */
+  UINT extGainDelay;   /*!< Delay that must be accounted for extGain. */
+
+  INT_PCM pcmOutputBuffer[(8) * (1024 * 2)];
+
+  HANDLE_DRC_DECODER hUniDrcDecoder;
+  UCHAR multibandDrcPresent;
+  UCHAR numTimeSlots;
+  UINT loudnessInfoSetPosition[3];
+  SCHAR defaultTargetLoudness;
+
+  INT_PCM
+  *pTimeDataFlush[((8) * 2)]; /*!< Pointer to the flushed time data which
+                                 will be used for the crossfade in case of
+                                 an USAC DASH IPF config change */
+
+  UCHAR flushStatus;     /*!< Indicates flush status: on|off */
+  SCHAR flushCnt;        /*!< Flush frame counter */
+  UCHAR buildUpStatus;   /*!< Indicates build up status: on|off */
+  SCHAR buildUpCnt;      /*!< Build up frame counter */
+  UCHAR hasAudioPreRoll; /*!< Indicates preRoll status: on|off */
+  UINT prerollAULength[AACDEC_MAX_NUM_PREROLL_AU + 1]; /*!< Relative offset of
+                                                          the prerollAU end
+                                                          position to the AU
+                                                          start position in the
+                                                          bitstream */
+  INT accessUnit; /*!< Number of the actual processed preroll accessUnit */
+  UCHAR applyCrossfade; /*!< if set crossfade for seamless stream switching is
+                           applied */
+
+  FDK_SignalDelay usacResidualDelay; /*!< Delay residual signal to compensate
+                                        for eSBR delay of DMX signal in case of
+                                        stereoConfigIndex==2. */
+};
+
+#define AAC_DEBUG_EXTHLP \
+  "\
+--- AAC-Core ---\n\
+    0x00010000 Header data\n\
+    0x00020000 CRC data\n\
+    0x00040000 Channel info\n\
+    0x00080000 Section data\n\
+    0x00100000 Scalefactor data\n\
+    0x00200000 Pulse data\n\
+    0x00400000 Tns data\n\
+    0x00800000 Quantized spectrum\n\
+    0x01000000 Requantized spectrum\n\
+    0x02000000 Time output\n\
+    0x04000000 Fatal errors\n\
+    0x08000000 Buffer fullness\n\
+    0x10000000 Average bitrate\n\
+    0x20000000 Synchronization\n\
+    0x40000000 Concealment\n\
+    0x7FFF0000 all AAC-Core-Info\n\
+"
+
+/**
+ * \brief Synchronise QMF mode for all modules using QMF data.
+ * \param self decoder handle
+ */
+void CAacDecoder_SyncQmfMode(HANDLE_AACDECODER self);
+
+/**
+ * \brief Signal a bit stream interruption to the decoder
+ * \param self decoder handle
+ */
+void CAacDecoder_SignalInterruption(HANDLE_AACDECODER self);
+
+/*!
+  \brief Initialize ancillary buffer
+
+  \ancData Pointer to ancillary data structure
+  \buffer Pointer to (external) anc data buffer
+  \size Size of the buffer pointed on by buffer
+
+  \return  Error code
+*/
+AAC_DECODER_ERROR CAacDecoder_AncDataInit(CAncData *ancData,
+                                          unsigned char *buffer, int size);
+
+/*!
+  \brief Get one ancillary data element
+
+  \ancData Pointer to ancillary data structure
+  \index Index of the anc data element to get
+  \ptr Pointer to a buffer receiving a pointer to the requested anc data element
+  \size Pointer to a buffer receiving the length of the requested anc data
+  element
+
+  \return  Error code
+*/
+AAC_DECODER_ERROR CAacDecoder_AncDataGet(CAncData *ancData, int index,
+                                         unsigned char **ptr, int *size);
+
+/* initialization of aac decoder */
+LINKSPEC_H HANDLE_AACDECODER CAacDecoder_Open(TRANSPORT_TYPE bsFormat);
+
+/* Initialization of channel elements */
+LINKSPEC_H AAC_DECODER_ERROR CAacDecoder_Init(HANDLE_AACDECODER self,
+                                              const CSAudioSpecificConfig *asc,
+                                              UCHAR configMode,
+                                              UCHAR *configChanged);
+/*!
+  \brief Decodes one aac frame
+
+  The function decodes one aac frame. The decoding of coupling channel
+  elements are not supported. The transport layer might signal, that the
+  data of the current frame is invalid, e.g. as a result of a packet
+  loss in streaming mode.
+  The bitstream position of transportDec_GetBitstream(self->hInput) must
+  be exactly the end of the access unit, including all byte alignment bits.
+  For this purpose, the variable auStartAnchor is used.
+
+  \return  error status
+*/
+LINKSPEC_H AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
+    HANDLE_AACDECODER self, const UINT flags, FIXP_PCM *pTimeData,
+    const INT timeDataSize, const int timeDataChannelOffset);
+
+/* Free config dependent AAC memory */
+LINKSPEC_H AAC_DECODER_ERROR CAacDecoder_FreeMem(HANDLE_AACDECODER self,
+                                                 const int subStreamIndex);
+
+/* Prepare crossfade for USAC DASH IPF config change */
+LINKSPEC_H AAC_DECODER_ERROR CAacDecoder_PrepareCrossFade(
+    const INT_PCM *pTimeData, INT_PCM **pTimeDataFlush, const INT numChannels,
+    const INT frameSize, const INT interleaved);
+
+/* Apply crossfade for USAC DASH IPF config change */
+LINKSPEC_H AAC_DECODER_ERROR CAacDecoder_ApplyCrossFade(
+    INT_PCM *pTimeData, INT_PCM **pTimeDataFlush, const INT numChannels,
+    const INT frameSize, const INT interleaved);
+
+/* Set flush and build up mode */
+LINKSPEC_H AAC_DECODER_ERROR CAacDecoder_CtrlCFGChange(HANDLE_AACDECODER self,
+                                                       UCHAR flushStatus,
+                                                       SCHAR flushCnt,
+                                                       UCHAR buildUpStatus,
+                                                       SCHAR buildUpCnt);
+
+/* Parse preRoll Extension Payload */
+LINKSPEC_H AAC_DECODER_ERROR CAacDecoder_PreRollExtensionPayloadParse(
+    HANDLE_AACDECODER self, UINT *numPrerollAU, UINT *prerollAUOffset,
+    UINT *prerollAULength);
+
+/* Destroy aac decoder */
+LINKSPEC_H void CAacDecoder_Close(HANDLE_AACDECODER self);
+
+/* get streaminfo handle from decoder */
+LINKSPEC_H CStreamInfo *CAacDecoder_GetStreamInfo(HANDLE_AACDECODER self);
+
+#endif /* #ifndef AACDECODER_H */