1 files changed, 6 insertions, 39 deletions

diff --git a/libAACdec/src/aacdecoder_lib.cpp b/libAACdec/src/aacdecoder_lib.cpp
index 50c9563..5650dd1 100644
--- a/libAACdec/src/aacdecoder_lib.cpp
+++ b/libAACdec/src/aacdecoder_lib.cpp
@@ -306,13 +306,6 @@ static INT aacDecoder_FreeMemCallback(void *handle,
     errTp = TRANSPORTDEC_UNKOWN_ERROR;
   }
 
-  /* free Ram_SbrDecoder and Ram_SbrDecChannel */
-  if (self->hSbrDecoder != NULL) {
-    if (sbrDecoder_FreeMem(&self->hSbrDecoder) != SBRDEC_OK) {
-      errTp = TRANSPORTDEC_UNKOWN_ERROR;
-    }
-  }
-
   /* free pSpatialDec and mpsData */
   if (self->pMpegSurroundDecoder != NULL) {
     if (mpegSurroundDecoder_FreeMem(
@@ -344,23 +337,6 @@ static INT aacDecoder_CtrlCFGChangeCallback(
   return errTp;
 }
 
-static INT aacDecoder_SbrCallback(
-    void *handle, 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) {
-  HANDLE_SBRDECODER self = (HANDLE_SBRDECODER)handle;
-
-  INT errTp = sbrDecoder_Header(self, hBs, sampleRateIn, sampleRateOut,
-                                samplesPerFrame, coreCodec, elementID,
-                                elementIndex, harmonicSBR, stereoConfigIndex,
-                                configMode, configChanged, downscaleFactor);
-
-  return errTp;
-}
-
 static INT aacDecoder_SscCallback(void *handle, HANDLE_FDK_BITSTREAM hBs,
                                   const AUDIO_OBJECT_TYPE coreCodec,
                                   const INT samplingRate, const INT frameSize,
@@ -806,9 +782,6 @@ LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_SetParam(
          packed into a helper function which keeps all modules and libs in a
          consistent state even in the case an error occures. */
       errorStatus = setConcealMethod(self, value);
-      if (errorStatus == AAC_DEC_OK) {
-        self->concealMethodUser = (CConcealmentMethod)value;
-      }
       break;
 
     default:
@@ -1302,9 +1275,7 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern,
 
       if (!self->qmfDomain.globalConf.qmfDomainExplicitConfig &&
           self->mpsEnableCurr) {
-        SAC_INPUT_CONFIG sac_interface = (self->sbrEnabled && self->hSbrDecoder)
-                                             ? SAC_INTERFACE_QMF
-                                             : SAC_INTERFACE_TIME;
+        SAC_INPUT_CONFIG sac_interface = SAC_INTERFACE_TIME;
         /* needs to be done before first SBR apply. */
         mpegSurroundDecoder_ConfigureQmfDomain(
             (CMpegSurroundDecoder *)self->pMpegSurroundDecoder, sac_interface,
@@ -1344,8 +1315,6 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern,
         frameSize = self->streamInfo.frameSize;
         sac_interface = SAC_INTERFACE_TIME;
 
-        if (self->sbrEnabled && self->hSbrDecoder)
-          sac_interface = SAC_INTERFACE_QMF;
         if (self->streamInfo.aot == AOT_USAC) {
           if (self->flags[0] & AC_USAC_SCFGI3) {
             sac_interface = SAC_INTERFACE_TIME;
@@ -1429,11 +1398,9 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern,
           /* If SBR and/or MPS is active, the DRC gains are aligned to the QMF
              domain signal before the QMF synthesis. Therefore the DRC gains
              need to be delayed by the QMF synthesis delay. */
-          if (self->sbrEnabled) drcDelay = 257;
           if (self->mpsEnableCurr) drcDelay = 257;
           /* Take into account concealment delay */
-          drcDelay += CConcealment_GetDelay(&self->concealCommonData) *
-                      self->streamInfo.frameSize;
+          drcDelay += 0 * self->streamInfo.frameSize;
 
           for (ch = 0; ch < self->streamInfo.numChannels; ch++) {
             UCHAR mapValue = FDK_chMapDescr_getMapValue(
@@ -1450,7 +1417,7 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern,
            * necessary for FDK_drcDec_ProcessTime, which accepts deinterleaved
            * audio only. */
           if ((self->streamInfo.numChannels > 1) &&
-              (0 || (self->sbrEnabled) || (self->mpsEnableCurr))) {
+              (0 || (self->mpsEnableCurr))) {
             /* interleaving/deinterleaving is performed on upper part of
              * pTimeDataPcmPost. Check if this buffer is large enough. */
             if (timeDataPcmPostSize <
@@ -1523,7 +1490,7 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern,
         }
 
         INT interleaved = 0;
-        interleaved |= (self->sbrEnabled) ? 1 : 0;
+        interleaved |= 0;
         interleaved |= (self->mpsEnableCurr) ? 1 : 0;
 
         /* do PCM post processing */
@@ -1561,7 +1528,7 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern,
           pcmLimiter_SetSampleRate(self->hLimiter, self->streamInfo.sampleRate);
           pcmLimiterScale += PCM_OUT_HEADROOM;
 
-          if ((self->streamInfo.numChannels == 1) || (self->sbrEnabled) ||
+          if ((self->streamInfo.numChannels == 1) ||
               (self->mpsEnableCurr)) {
             pInterleaveBuffer = (PCM_LIM *)pTimeDataPcmPost;
           } else {
@@ -1584,7 +1551,7 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern,
           /* If numChannels = 1 we do not need interleaving. The same applies if
           SBR or MPS are used, since their output is interleaved already
           (resampled or not) */
-          if ((self->streamInfo.numChannels == 1) || (self->sbrEnabled) ||
+          if ((self->streamInfo.numChannels == 1) ||
               (self->mpsEnableCurr)) {
             scaleValuesSaturate(
                 pTimeData, pTimeDataPcmPost,