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, 676 insertions, 0 deletions

diff --git a/fdk-aac/libMpegTPDec/src/tpdec_latm.cpp b/fdk-aac/libMpegTPDec/src/tpdec_latm.cpp
new file mode 100644
index 0000000..2edf055
--- /dev/null
+++ b/fdk-aac/libMpegTPDec/src/tpdec_latm.cpp
@@ -0,0 +1,676 @@
+/* -----------------------------------------------------------------------------
+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):   Daniel Homm
+
+   Description:
+
+*******************************************************************************/
+
+#include "tpdec_latm.h"
+
+#include "FDK_bitstream.h"
+
+#define TPDEC_TRACKINDEX(p, l) (1 * (p) + (l))
+
+static UINT CLatmDemux_GetValue(HANDLE_FDK_BITSTREAM bs) {
+  UCHAR bytesForValue = 0, tmp = 0;
+  int value = 0;
+
+  bytesForValue = (UCHAR)FDKreadBits(bs, 2);
+
+  for (UINT i = 0; i <= bytesForValue; i++) {
+    value <<= 8;
+    tmp = (UCHAR)FDKreadBits(bs, 8);
+    value += tmp;
+  }
+
+  return value;
+}
+
+static TRANSPORTDEC_ERROR CLatmDemux_ReadAudioMuxElement(
+    HANDLE_FDK_BITSTREAM bs, CLatmDemux *pLatmDemux, int m_muxConfigPresent,
+    CSTpCallBacks *pTpDecCallbacks, CSAudioSpecificConfig *pAsc,
+    int *pfConfigFound) {
+  TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK;
+
+  if (m_muxConfigPresent) {
+    pLatmDemux->m_useSameStreamMux = FDKreadBits(bs, 1);
+
+    if (!pLatmDemux->m_useSameStreamMux) {
+      int i;
+      UCHAR configChanged = 0;
+      UCHAR configMode = 0;
+
+      FDK_BITSTREAM bsAnchor;
+
+      FDK_BITSTREAM bsAnchorDummyParse;
+
+      if (!pLatmDemux->applyAsc) {
+        bsAnchorDummyParse = *bs;
+        pLatmDemux->newCfgHasAudioPreRoll = 0;
+        /* do dummy-parsing of ASC to determine if there is an audioPreRoll */
+        configMode |= AC_CM_DET_CFG_CHANGE;
+        if (TRANSPORTDEC_OK !=
+            (ErrorStatus = CLatmDemux_ReadStreamMuxConfig(
+                 bs, pLatmDemux, pTpDecCallbacks, pAsc, pfConfigFound,
+                 configMode, configChanged))) {
+          goto bail;
+        }
+
+        /* Allow flushing only when audioPreroll functionality is enabled in
+         * current and new config otherwise the new config can be applied
+         * immediately. */
+        if (pAsc->m_sc.m_usacConfig.element[0]
+                .extElement.usacExtElementHasAudioPreRoll &&
+            pLatmDemux->newCfgHasAudioPreRoll) {
+          pLatmDemux->newCfgHasAudioPreRoll = 0;
+          /* with audioPreRoll we must flush before applying new cfg */
+          pLatmDemux->applyAsc = 0;
+        } else {
+          *bs = bsAnchorDummyParse;
+          pLatmDemux->applyAsc = 1; /* apply new config immediate */
+        }
+      }
+
+      if (pLatmDemux->applyAsc) {
+        for (i = 0; i < 2; i++) {
+          configMode = 0;
+
+          if (i == 0) {
+            configMode |= AC_CM_DET_CFG_CHANGE;
+            bsAnchor = *bs;
+          } else {
+            configMode |= AC_CM_ALLOC_MEM;
+            *bs = bsAnchor;
+          }
+
+          if (TRANSPORTDEC_OK !=
+              (ErrorStatus = CLatmDemux_ReadStreamMuxConfig(
+                   bs, pLatmDemux, pTpDecCallbacks, pAsc, pfConfigFound,
+                   configMode, configChanged))) {
+            goto bail;
+          }
+
+          if (ErrorStatus == TRANSPORTDEC_OK) {
+            if ((i == 0) && (pAsc->AacConfigChanged || pAsc->SbrConfigChanged ||
+                             pAsc->SacConfigChanged)) {
+              int errC;
+
+              configChanged = 1;
+              errC = pTpDecCallbacks->cbFreeMem(pTpDecCallbacks->cbFreeMemData,
+                                                pAsc);
+              if (errC != 0) {
+                ErrorStatus = TRANSPORTDEC_PARSE_ERROR;
+                goto bail;
+              }
+            }
+          }
+        }
+      }
+    }
+  }
+
+  /* If there was no configuration read, its not possible to parse
+   * PayloadLengthInfo below. */
+  if (!*pfConfigFound) {
+    ErrorStatus = TRANSPORTDEC_SYNC_ERROR;
+    goto bail;
+  }
+
+  if (pLatmDemux->m_AudioMuxVersionA == 0) {
+    /* Do only once per call, because parsing and decoding is done in-line. */
+    if (TRANSPORTDEC_OK !=
+        (ErrorStatus = CLatmDemux_ReadPayloadLengthInfo(bs, pLatmDemux))) {
+      *pfConfigFound = 0;
+      goto bail;
+    }
+  } else {
+    /* audioMuxVersionA > 0 is reserved for future extensions */
+    ErrorStatus = TRANSPORTDEC_UNSUPPORTED_FORMAT;
+    *pfConfigFound = 0;
+    goto bail;
+  }
+
+bail:
+  if (ErrorStatus != TRANSPORTDEC_OK) {
+    pLatmDemux->applyAsc = 1;
+  }
+
+  return (ErrorStatus);
+}
+
+TRANSPORTDEC_ERROR CLatmDemux_Read(HANDLE_FDK_BITSTREAM bs,
+                                   CLatmDemux *pLatmDemux, TRANSPORT_TYPE tt,
+                                   CSTpCallBacks *pTpDecCallbacks,
+                                   CSAudioSpecificConfig *pAsc,
+                                   int *pfConfigFound,
+                                   const INT ignoreBufferFullness) {
+  UINT cntBits;
+  UINT cmpBufferFullness;
+  UINT audioMuxLengthBytesLast = 0;
+  TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK;
+
+  cntBits = FDKgetValidBits(bs);
+
+  if ((INT)cntBits < MIN_LATM_HEADERLENGTH) {
+    return TRANSPORTDEC_NOT_ENOUGH_BITS;
+  }
+
+  if (TRANSPORTDEC_OK != (ErrorStatus = CLatmDemux_ReadAudioMuxElement(
+                              bs, pLatmDemux, (tt != TT_MP4_LATM_MCP0),
+                              pTpDecCallbacks, pAsc, pfConfigFound)))
+    return (ErrorStatus);
+
+  if (!ignoreBufferFullness) {
+    cmpBufferFullness =
+        24 + audioMuxLengthBytesLast * 8 +
+        pLatmDemux->m_linfo[0][0].m_bufferFullness *
+            pAsc[TPDEC_TRACKINDEX(0, 0)].m_channelConfiguration * 32;
+
+    /* evaluate buffer fullness */
+
+    if (pLatmDemux->m_linfo[0][0].m_bufferFullness != 0xFF) {
+      if (!pLatmDemux->BufferFullnessAchieved) {
+        if (cntBits < cmpBufferFullness) {
+          /* condition for start of decoding is not fulfilled */
+
+          /* the current frame will not be decoded */
+          return TRANSPORTDEC_NOT_ENOUGH_BITS;
+        } else {
+          pLatmDemux->BufferFullnessAchieved = 1;
+        }
+      }
+    }
+  }
+
+  return (ErrorStatus);
+}
+
+TRANSPORTDEC_ERROR CLatmDemux_ReadStreamMuxConfig(
+    HANDLE_FDK_BITSTREAM bs, CLatmDemux *pLatmDemux,
+    CSTpCallBacks *pTpDecCallbacks, CSAudioSpecificConfig *pAsc,
+    int *pfConfigFound, UCHAR configMode, UCHAR configChanged) {
+  CSAudioSpecificConfig ascDummy; /* the actual config is needed for flushing,
+                                     after that new config can be parsed */
+  CSAudioSpecificConfig *pAscDummy;
+  pAscDummy = &ascDummy;
+  pLatmDemux->usacExplicitCfgChanged = 0;
+  LATM_LAYER_INFO *p_linfo = NULL;
+  TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK;
+  UCHAR updateConfig[1 * 1] = {0};
+
+  pLatmDemux->m_AudioMuxVersion = FDKreadBits(bs, 1);
+
+  if (pLatmDemux->m_AudioMuxVersion == 0) {
+    pLatmDemux->m_AudioMuxVersionA = 0;
+  } else {
+    pLatmDemux->m_AudioMuxVersionA = FDKreadBits(bs, 1);
+  }
+
+  if (pLatmDemux->m_AudioMuxVersionA == 0) {
+    if (pLatmDemux->m_AudioMuxVersion == 1) {
+      pLatmDemux->m_taraBufferFullness = CLatmDemux_GetValue(bs);
+    }
+    pLatmDemux->m_allStreamsSameTimeFraming = FDKreadBits(bs, 1);
+    pLatmDemux->m_noSubFrames = FDKreadBits(bs, 6) + 1;
+    pLatmDemux->m_numProgram = FDKreadBits(bs, 4) + 1;
+
+    if (pLatmDemux->m_numProgram > LATM_MAX_PROG) {
+      ErrorStatus = TRANSPORTDEC_UNSUPPORTED_FORMAT;
+      goto bail;
+    }
+
+    int idCnt = 0;
+    for (UINT prog = 0; prog < pLatmDemux->m_numProgram; prog++) {
+      pLatmDemux->m_numLayer[prog] = FDKreadBits(bs, 3) + 1;
+      if (pLatmDemux->m_numLayer[prog] > LATM_MAX_LAYER) {
+        ErrorStatus = TRANSPORTDEC_UNSUPPORTED_FORMAT;
+        goto bail;
+      }
+
+      for (UINT lay = 0; lay < pLatmDemux->m_numLayer[prog]; lay++) {
+        int useSameConfig;
+        p_linfo = &pLatmDemux->m_linfo[prog][lay];
+
+        p_linfo->m_streamID = idCnt++;
+        p_linfo->m_frameLengthInBits = 0;
+
+        if ((prog == 0) && (lay == 0)) {
+          useSameConfig = 0;
+        } else {
+          useSameConfig = FDKreadBits(bs, 1);
+        }
+
+        if (useSameConfig) {
+          if (lay > 0) {
+            FDKmemcpy(&pAsc[TPDEC_TRACKINDEX(prog, lay)],
+                      &pAsc[TPDEC_TRACKINDEX(prog, lay - 1)],
+                      sizeof(CSAudioSpecificConfig));
+          } else {
+            ErrorStatus = TRANSPORTDEC_PARSE_ERROR;
+            goto bail;
+          }
+        } else {
+          UINT usacConfigLengthPrev = 0;
+          UCHAR usacConfigPrev[TP_USAC_MAX_CONFIG_LEN];
+
+          if (!(pLatmDemux->applyAsc) &&
+              (pAsc[TPDEC_TRACKINDEX(prog, lay)].m_aot == AOT_USAC)) {
+            usacConfigLengthPrev =
+                (UINT)(pAsc[TPDEC_TRACKINDEX(prog, lay)]
+                           .m_sc.m_usacConfig.UsacConfigBits +
+                       7) >>
+                3; /* store previous USAC config length */
+            if (usacConfigLengthPrev > TP_USAC_MAX_CONFIG_LEN) {
+              ErrorStatus = TRANSPORTDEC_PARSE_ERROR;
+              goto bail;
+            }
+            FDKmemclear(usacConfigPrev, TP_USAC_MAX_CONFIG_LEN);
+            FDKmemcpy(
+                usacConfigPrev,
+                &pAsc[TPDEC_TRACKINDEX(prog, lay)].m_sc.m_usacConfig.UsacConfig,
+                usacConfigLengthPrev); /* store previous USAC config */
+          }
+          if (pLatmDemux->m_AudioMuxVersion == 1) {
+            FDK_BITSTREAM tmpBs;
+            UINT ascLen = 0;
+            ascLen = CLatmDemux_GetValue(bs);
+            /* The ascLen could be wrong, so check if validBits<=bufBits*/
+            if (ascLen > FDKgetValidBits(bs)) {
+              ErrorStatus = TRANSPORTDEC_PARSE_ERROR;
+              goto bail;
+            }
+            FDKsyncCache(bs);
+            tmpBs = *bs;
+            tmpBs.hBitBuf.ValidBits = ascLen;
+
+            /* Read ASC */
+            if (pLatmDemux->applyAsc) {
+              if (TRANSPORTDEC_OK !=
+                  (ErrorStatus = AudioSpecificConfig_Parse(
+                       &pAsc[TPDEC_TRACKINDEX(prog, lay)], &tmpBs, 1,
+                       pTpDecCallbacks, configMode, configChanged,
+                       AOT_NULL_OBJECT)))
+                goto bail;
+            } else {
+              if (TRANSPORTDEC_OK !=
+                  (ErrorStatus = AudioSpecificConfig_Parse(
+                       pAscDummy, &tmpBs, 1, pTpDecCallbacks, configMode,
+                       configChanged, AOT_NULL_OBJECT)))
+                goto bail;
+            }
+
+            /* The field p_linfo->m_ascLen could be wrong, so check if */
+            if (0 > (INT)FDKgetValidBits(&tmpBs)) {
+              ErrorStatus = TRANSPORTDEC_PARSE_ERROR;
+              goto bail;
+            }
+            FDKpushFor(bs, ascLen); /* position bitstream after ASC */
+          } else {
+            /* Read ASC */
+            if (pLatmDemux->applyAsc) {
+              if (TRANSPORTDEC_OK != (ErrorStatus = AudioSpecificConfig_Parse(
+                                          &pAsc[TPDEC_TRACKINDEX(prog, lay)],
+                                          bs, 0, pTpDecCallbacks, configMode,
+                                          configChanged, AOT_NULL_OBJECT)))
+                goto bail;
+            } else {
+              if (TRANSPORTDEC_OK !=
+                  (ErrorStatus = AudioSpecificConfig_Parse(
+                       pAscDummy, bs, 0, pTpDecCallbacks, configMode,
+                       configChanged, AOT_NULL_OBJECT)))
+                goto bail;
+            }
+          }
+          if (!pLatmDemux->applyAsc) {
+            updateConfig[TPDEC_TRACKINDEX(prog, lay)] = 0;
+          } else {
+            updateConfig[TPDEC_TRACKINDEX(prog, lay)] = 1;
+          }
+
+          if (!pLatmDemux->applyAsc) {
+            if (pAscDummy[TPDEC_TRACKINDEX(prog, lay)].m_aot ==
+                AOT_USAC) { /* flush in case SMC has changed */
+              const UINT usacConfigLength =
+                  (UINT)(pAscDummy->m_sc.m_usacConfig.UsacConfigBits + 7) >> 3;
+              if (usacConfigLength > TP_USAC_MAX_CONFIG_LEN) {
+                ErrorStatus = TRANSPORTDEC_PARSE_ERROR;
+                goto bail;
+              }
+              if (usacConfigLength != usacConfigLengthPrev) {
+                FDKmemclear(&pAsc[TPDEC_TRACKINDEX(prog, lay)]
+                                 .m_sc.m_usacConfig.UsacConfig,
+                            TP_USAC_MAX_CONFIG_LEN);
+                FDKmemcpy(&pAsc[TPDEC_TRACKINDEX(prog, lay)]
+                               .m_sc.m_usacConfig.UsacConfig,
+                          &pAscDummy->m_sc.m_usacConfig.UsacConfig,
+                          usacConfigLength); /* store new USAC config */
+                pAsc[TPDEC_TRACKINDEX(prog, lay)]
+                    .m_sc.m_usacConfig.UsacConfigBits =
+                    pAscDummy->m_sc.m_usacConfig.UsacConfigBits;
+                pLatmDemux->usacExplicitCfgChanged = 1;
+              } else {
+                if (FDKmemcmp(usacConfigPrev,
+                              pAscDummy->m_sc.m_usacConfig.UsacConfig,
+                              usacConfigLengthPrev)) {
+                  FDKmemclear(&pAsc[TPDEC_TRACKINDEX(prog, lay)]
+                                   .m_sc.m_usacConfig.UsacConfig,
+                              TP_USAC_MAX_CONFIG_LEN);
+                  FDKmemcpy(&pAsc[TPDEC_TRACKINDEX(prog, lay)]
+                                 .m_sc.m_usacConfig.UsacConfig,
+                            &pAscDummy->m_sc.m_usacConfig.UsacConfig,
+                            usacConfigLength); /* store new USAC config */
+                  pAsc[TPDEC_TRACKINDEX(prog, lay)]
+                      .m_sc.m_usacConfig.UsacConfigBits =
+                      pAscDummy->m_sc.m_usacConfig.UsacConfigBits;
+                  pLatmDemux->usacExplicitCfgChanged = 1;
+                }
+              }
+
+              if (pAscDummy[TPDEC_TRACKINDEX(prog, lay)]
+                      .m_sc.m_usacConfig.m_usacNumElements) {
+                if (pAscDummy[TPDEC_TRACKINDEX(prog, lay)]
+                        .m_sc.m_usacConfig.element[0]
+                        .extElement.usacExtElementHasAudioPreRoll) {
+                  pLatmDemux->newCfgHasAudioPreRoll =
+                      1; /* if dummy parsed cfg has audioPreRoll we first flush
+                            before applying new cfg */
+                }
+              }
+            }
+          }
+        }
+
+        p_linfo->m_frameLengthType = FDKreadBits(bs, 3);
+        switch (p_linfo->m_frameLengthType) {
+          case 0:
+            p_linfo->m_bufferFullness = FDKreadBits(bs, 8);
+
+            if (!pLatmDemux->m_allStreamsSameTimeFraming) {
+              if ((lay > 0) &&
+                  (pAsc[TPDEC_TRACKINDEX(prog, lay)].m_aot == AOT_AAC_SCAL ||
+                   pAsc[TPDEC_TRACKINDEX(prog, lay)].m_aot ==
+                       AOT_ER_AAC_SCAL) &&
+                  (pAsc[TPDEC_TRACKINDEX(prog, lay - 1)].m_aot == AOT_CELP ||
+                   pAsc[TPDEC_TRACKINDEX(prog, lay - 1)].m_aot ==
+                       AOT_ER_CELP)) { /* The layer maybe
+                                          ignored later so
+                                          read it anyway: */
+                /* coreFrameOffset = */ FDKreadBits(bs, 6);
+              }
+            }
+            break;
+          case 1:
+            p_linfo->m_frameLengthInBits = FDKreadBits(bs, 9);
+            break;
+          case 3:
+          case 4:
+          case 5:
+            /* CELP */
+          case 6:
+          case 7:
+            /* HVXC */
+          default:
+            ErrorStatus = TRANSPORTDEC_PARSE_ERROR;
+            goto bail;
+        } /* switch framelengthtype*/
+
+      } /* layer loop */
+    }   /* prog loop */
+
+    pLatmDemux->m_otherDataPresent = FDKreadBits(bs, 1);
+    pLatmDemux->m_otherDataLength = 0;
+
+    if (pLatmDemux->m_otherDataPresent) {
+      if (pLatmDemux->m_AudioMuxVersion == 1) {
+        pLatmDemux->m_otherDataLength = CLatmDemux_GetValue(bs);
+      } else {
+        int otherDataLenEsc = 0;
+        do {
+          pLatmDemux->m_otherDataLength <<= 8;  // *= 256
+          otherDataLenEsc = FDKreadBits(bs, 1);
+          pLatmDemux->m_otherDataLength += FDKreadBits(bs, 8);
+        } while (otherDataLenEsc);
+      }
+      if (pLatmDemux->m_audioMuxLengthBytes <
+          (pLatmDemux->m_otherDataLength >> 3)) {
+        ErrorStatus = TRANSPORTDEC_PARSE_ERROR;
+        goto bail;
+      }
+    }
+
+    pLatmDemux->m_crcCheckPresent = FDKreadBits(bs, 1);
+
+    if (pLatmDemux->m_crcCheckPresent) {
+      FDKreadBits(bs, 8);
+    }
+
+  } else {
+    /* audioMuxVersionA > 0 is reserved for future extensions */
+    ErrorStatus = TRANSPORTDEC_UNSUPPORTED_FORMAT;
+  }
+
+  /* Configure source decoder: */
+  if (ErrorStatus == TRANSPORTDEC_OK) {
+    UINT prog;
+    for (prog = 0; prog < pLatmDemux->m_numProgram; prog++) {
+      UINT lay;
+      for (lay = 0; lay < pLatmDemux->m_numLayer[prog]; lay++) {
+        if (updateConfig[TPDEC_TRACKINDEX(prog, lay)] != 0) {
+          int cbError;
+          cbError = pTpDecCallbacks->cbUpdateConfig(
+              pTpDecCallbacks->cbUpdateConfigData,
+              &pAsc[TPDEC_TRACKINDEX(prog, lay)],
+              pAsc[TPDEC_TRACKINDEX(prog, lay)].configMode,
+              &pAsc[TPDEC_TRACKINDEX(prog, lay)].AacConfigChanged);
+          if (cbError == TRANSPORTDEC_NEED_TO_RESTART) {
+            *pfConfigFound = 0;
+            ErrorStatus = TRANSPORTDEC_NEED_TO_RESTART;
+            goto bail;
+          }
+          if (cbError != 0) {
+            *pfConfigFound = 0;
+            if (lay == 0) {
+              ErrorStatus = TRANSPORTDEC_SYNC_ERROR;
+              goto bail;
+            }
+          } else {
+            *pfConfigFound = 1;
+          }
+        } else {
+          *pfConfigFound = 1;
+        }
+      }
+    }
+  }
+
+bail:
+  if (ErrorStatus != TRANSPORTDEC_OK) {
+    UCHAR applyAsc = pLatmDemux->applyAsc;
+    FDKmemclear(pLatmDemux, sizeof(CLatmDemux)); /* reset structure */
+    pLatmDemux->applyAsc = applyAsc;
+  } else {
+    /* no error and config parsing is finished */
+    if (configMode == AC_CM_ALLOC_MEM) pLatmDemux->applyAsc = 0;
+  }
+
+  return (ErrorStatus);
+}
+
+TRANSPORTDEC_ERROR CLatmDemux_ReadPayloadLengthInfo(HANDLE_FDK_BITSTREAM bs,
+                                                    CLatmDemux *pLatmDemux) {
+  TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK;
+  int totalPayloadBits = 0;
+
+  if (pLatmDemux->m_allStreamsSameTimeFraming == 1) {
+    FDK_ASSERT(pLatmDemux->m_numProgram <= LATM_MAX_PROG);
+    for (UINT prog = 0; prog < pLatmDemux->m_numProgram; prog++) {
+      FDK_ASSERT(pLatmDemux->m_numLayer[prog] <= LATM_MAX_LAYER);
+      for (UINT lay = 0; lay < pLatmDemux->m_numLayer[prog]; lay++) {
+        LATM_LAYER_INFO *p_linfo = &pLatmDemux->m_linfo[prog][lay];
+
+        switch (p_linfo->m_frameLengthType) {
+          case 0:
+            p_linfo->m_frameLengthInBits = CLatmDemux_ReadAuChunkLengthInfo(bs);
+            totalPayloadBits += p_linfo->m_frameLengthInBits;
+            break;
+          case 3:
+          case 5:
+          case 7:
+          default:
+            return TRANSPORTDEC_PARSE_ERROR;  // AAC_DEC_LATM_INVALIDFRAMELENGTHTYPE;
+        }
+      }
+    }
+  } else {
+    ErrorStatus = TRANSPORTDEC_PARSE_ERROR;  // AAC_DEC_LATM_TIMEFRAMING;
+  }
+  if (pLatmDemux->m_audioMuxLengthBytes > (UINT)0 &&
+      totalPayloadBits > (int)pLatmDemux->m_audioMuxLengthBytes * 8) {
+    return TRANSPORTDEC_PARSE_ERROR;
+  }
+
+  return (ErrorStatus);
+}
+
+int CLatmDemux_ReadAuChunkLengthInfo(HANDLE_FDK_BITSTREAM bs) {
+  UCHAR endFlag;
+  int len = 0;
+
+  do {
+    UCHAR tmp = (UCHAR)FDKreadBits(bs, 8);
+    endFlag = (tmp < 255);
+
+    len += tmp;
+
+  } while (endFlag == 0);
+
+  len <<= 3; /* convert from bytes to bits */
+
+  return len;
+}
+
+UINT CLatmDemux_GetFrameLengthInBits(CLatmDemux *pLatmDemux, const UINT prog,
+                                     const UINT layer) {
+  UINT nFrameLenBits = 0;
+  if (prog < pLatmDemux->m_numProgram) {
+    if (layer < pLatmDemux->m_numLayer[prog]) {
+      nFrameLenBits = pLatmDemux->m_linfo[prog][layer].m_frameLengthInBits;
+    }
+  }
+  return nFrameLenBits;
+}
+
+UINT CLatmDemux_GetOtherDataPresentFlag(CLatmDemux *pLatmDemux) {
+  return pLatmDemux->m_otherDataPresent ? 1 : 0;
+}
+
+UINT CLatmDemux_GetOtherDataLength(CLatmDemux *pLatmDemux) {
+  return pLatmDemux->m_otherDataLength;
+}
+
+UINT CLatmDemux_GetNrOfSubFrames(CLatmDemux *pLatmDemux) {
+  return pLatmDemux->m_noSubFrames;
+}
+
+UINT CLatmDemux_GetNrOfLayers(CLatmDemux *pLatmDemux, const UINT prog) {
+  UINT numLayer = 0;
+  if (prog < pLatmDemux->m_numProgram) {
+    numLayer = pLatmDemux->m_numLayer[prog];
+  }
+  return numLayer;
+}