Avoid decoder internal clipping by converting the whole audio sample data path from 16 to 32 bit data width (FDKdec v3.2.0).

Bug: 149514474
Test: atest DecoderTestXheAac DecoderTestAacDrc
Change-Id: I8a504ab709e42e27a61fe29840212953742283a5

41 files changed, 488 insertions, 442 deletions

diff --git a/documentation/aacDecoder.pdf b/documentation/aacDecoder.pdf
index d04a7d6..cc7cf41 100644
--- a/documentation/aacDecoder.pdf
+++ b/documentation/aacDecoder.pdf
diff --git a/libAACdec/src/FDK_delay.cpp b/libAACdec/src/FDK_delay.cpp
index 0ab1a66..0cc869c 100644
--- a/libAACdec/src/FDK_delay.cpp
+++ b/libAACdec/src/FDK_delay.cpp
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -113,7 +113,7 @@ INT FDK_Delay_Create(FDK_SignalDelay* data, const USHORT delay,
 
   if (delay > 0) {
     data->delay_line =
-        (INT_PCM*)FDKcalloc(num_channels * delay, sizeof(INT_PCM));
+        (PCM_DEC*)FDKcalloc(num_channels * delay, sizeof(PCM_DEC));
     if (data->delay_line == NULL) {
       return -1;
     }
@@ -126,36 +126,36 @@ INT FDK_Delay_Create(FDK_SignalDelay* data, const USHORT delay,
   return 0;
 }
 
-void FDK_Delay_Apply(FDK_SignalDelay* data, FIXP_PCM* time_buffer,
+void FDK_Delay_Apply(FDK_SignalDelay* data, PCM_DEC* time_buffer,
                      const UINT frame_length, const UCHAR channel) {
   FDK_ASSERT(data != NULL);
 
   if (data->delay > 0) {
-    C_ALLOC_SCRATCH_START(tmp, FIXP_PCM, MAX_FRAME_LENGTH)
+    C_ALLOC_SCRATCH_START(tmp, PCM_DEC, MAX_FRAME_LENGTH)
     FDK_ASSERT(frame_length <= MAX_FRAME_LENGTH);
     FDK_ASSERT(channel < data->num_channels);
     FDK_ASSERT(time_buffer != NULL);
     if (frame_length >= data->delay) {
       FDKmemcpy(tmp, &time_buffer[frame_length - data->delay],
-                data->delay * sizeof(FIXP_PCM));
+                data->delay * sizeof(PCM_DEC));
       FDKmemmove(&time_buffer[data->delay], &time_buffer[0],
-                 (frame_length - data->delay) * sizeof(FIXP_PCM));
+                 (frame_length - data->delay) * sizeof(PCM_DEC));
       FDKmemcpy(&time_buffer[0], &data->delay_line[channel * data->delay],
-                data->delay * sizeof(FIXP_PCM));
+                data->delay * sizeof(PCM_DEC));
       FDKmemcpy(&data->delay_line[channel * data->delay], tmp,
-                data->delay * sizeof(FIXP_PCM));
+                data->delay * sizeof(PCM_DEC));
     } else {
-      FDKmemcpy(tmp, &time_buffer[0], frame_length * sizeof(FIXP_PCM));
+      FDKmemcpy(tmp, &time_buffer[0], frame_length * sizeof(PCM_DEC));
       FDKmemcpy(&time_buffer[0], &data->delay_line[channel * data->delay],
-                frame_length * sizeof(FIXP_PCM));
+                frame_length * sizeof(PCM_DEC));
       FDKmemcpy(&data->delay_line[channel * data->delay],
                 &data->delay_line[channel * data->delay + frame_length],
-                (data->delay - frame_length) * sizeof(FIXP_PCM));
+                (data->delay - frame_length) * sizeof(PCM_DEC));
       FDKmemcpy(&data->delay_line[channel * data->delay +
                                   (data->delay - frame_length)],
-                tmp, frame_length * sizeof(FIXP_PCM));
+                tmp, frame_length * sizeof(PCM_DEC));
     }
-    C_ALLOC_SCRATCH_END(tmp, FIXP_PCM, MAX_FRAME_LENGTH)
+    C_ALLOC_SCRATCH_END(tmp, PCM_DEC, MAX_FRAME_LENGTH)
   }
 
   return;
diff --git a/libAACdec/src/FDK_delay.h b/libAACdec/src/FDK_delay.h
index f89c3a2..6317d9d 100644
--- a/libAACdec/src/FDK_delay.h
+++ b/libAACdec/src/FDK_delay.h
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -109,7 +109,7 @@ amm-info@iis.fraunhofer.de
  * Structure representing one delay element for multiple channels.
  */
 typedef struct {
-  INT_PCM* delay_line; /*!< Pointer which stores allocated delay line. */
+  PCM_DEC* delay_line; /*!< Pointer which stores allocated delay line. */
   USHORT delay;        /*!< Delay required in samples (per channel). */
   UCHAR num_channels;  /*!< Number of channels to delay. */
 } FDK_SignalDelay;
@@ -137,7 +137,7 @@ INT FDK_Delay_Create(FDK_SignalDelay* data, const USHORT delay,
  *
  * \return void
  */
-void FDK_Delay_Apply(FDK_SignalDelay* data, FIXP_PCM* time_buffer,
+void FDK_Delay_Apply(FDK_SignalDelay* data, PCM_DEC* time_buffer,
                      const UINT frame_length, const UCHAR channel);
 
 /**
diff --git a/libAACdec/src/aac_rom.h b/libAACdec/src/aac_rom.h
index ffaf951..7a1597c 100644
--- a/libAACdec/src/aac_rom.h
+++ b/libAACdec/src/aac_rom.h
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -108,6 +108,7 @@ amm-info@iis.fraunhofer.de
 #include "aacdec_hcr_types.h"
 #include "aacdec_hcrs.h"
 
+#define PCM_AAC LONG
 #define PCM_DEC FIXP_DBL
 #define MAXVAL_PCM_DEC MAXVAL_DBL
 #define MINVAL_PCM_DEC MINVAL_DBL
diff --git a/libAACdec/src/aacdecoder.cpp b/libAACdec/src/aacdecoder.cpp
index f747b2d..6a0254d 100644
--- a/libAACdec/src/aacdecoder.cpp
+++ b/libAACdec/src/aacdecoder.cpp
@@ -1281,6 +1281,7 @@ LINKSPEC_CPP HANDLE_AACDECODER CAacDecoder_Open(
   /* Set default frame delay */
   aacDecoder_drcSetParam(self->hDrcInfo, DRC_BS_DELAY,
                          CConcealment_GetDelay(&self->concealCommonData));
+  self->workBufferCore1 = (FIXP_DBL *)GetWorkBufferCore1();
 
   self->workBufferCore2 = GetWorkBufferCore2();
   if (self->workBufferCore2 == NULL) goto bail;
@@ -1456,6 +1457,10 @@ LINKSPEC_CPP void CAacDecoder_Close(HANDLE_AACDECODER self) {
     FreeDrcInfo(&self->hDrcInfo);
   }
 
+  if (self->workBufferCore1 != NULL) {
+    FreeWorkBufferCore1((CWorkBufferCore1 **)&self->workBufferCore1);
+  }
+
   /* Free WorkBufferCore2 */
   if (self->workBufferCore2 != NULL) {
     FreeWorkBufferCore2(&self->workBufferCore2);
@@ -1493,6 +1498,8 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc,
   UCHAR downscaleFactor = self->downscaleFactor;
   UCHAR downscaleFactorInBS = self->downscaleFactorInBS;
 
+  self->aacOutDataHeadroom = (3);
+
   // set profile and check for supported aot
   // leave profile on default (=-1) for all other supported MPEG-4 aot's except
   // aot=2 (=AAC-LC)
@@ -2394,7 +2401,7 @@ bail:
 }
 
 LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
-    HANDLE_AACDECODER self, const UINT flags, FIXP_PCM *pTimeData,
+    HANDLE_AACDECODER self, const UINT flags, PCM_DEC *pTimeData,
     const INT timeDataSize, const int timeDataChannelOffset) {
   AAC_DECODER_ERROR ErrorStatus = AAC_DEC_OK;
 
@@ -3170,10 +3177,8 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
       FDKmemcpy(drcChMap, self->chMapping, (8) * sizeof(UCHAR));
     }
 
-    /* Turn on/off DRC modules level normalization in digital domain depending
-     * on the limiter status. */
-    aacDecoder_drcSetParam(self->hDrcInfo, APPLY_NORMALIZATION,
-                           (self->limiterEnableCurr) ? 0 : 1);
+    /* Turn off DRC modules level normalization in digital domain. */
+    aacDecoder_drcSetParam(self->hDrcInfo, APPLY_NORMALIZATION, 0);
 
     /* deactivate legacy DRC in case uniDrc is active, i.e. uniDrc payload is
      * present and one of DRC or Loudness Normalization is switched on */
@@ -3325,9 +3330,11 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
                 &pAacDecoderStaticChannelInfo->drcData);
           }
         }
+
         /* The DRC module demands to be called with the gain field holding the
          * gain scale. */
-        self->extGain[0] = (FIXP_DBL)TDL_GAIN_SCALING;
+        self->extGain[0] = (FIXP_DBL)AACDEC_DRC_GAIN_SCALING;
+
         /* DRC processing */
         aacDecoder_drcApply(
             self->hDrcInfo, self->hSbrDecoder, pAacDecoderChannelInfo,
@@ -3343,7 +3350,7 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
         if (self->flushStatus && (self->flushCnt > 0) &&
             !(flags & AACDEC_CONCEAL)) {
           FDKmemclear(pTimeData + offset,
-                      sizeof(FIXP_PCM) * self->streamInfo.aacSamplesPerFrame);
+                      sizeof(PCM_DEC) * self->streamInfo.aacSamplesPerFrame);
         } else
           switch (pAacDecoderChannelInfo->renderMode) {
             case AACDEC_RENDER_IMDCT:
@@ -3355,7 +3362,7 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
                    !frameOk_butConceal),
                   pAacDecoderChannelInfo->pComStaticData->pWorkBufferCore1
                       ->mdctOutTemp,
-                  self->elFlags[el], elCh);
+                  self->aacOutDataHeadroom, self->elFlags[el], elCh);
 
               self->extGainDelay = self->streamInfo.aacSamplesPerFrame;
               break;
@@ -3376,7 +3383,7 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
                   &self->samplingRateInfo[streamIndex],
                   (self->frameOK && !(flags & AACDEC_CONCEAL) &&
                    !frameOk_butConceal),
-                  flags, self->flags[streamIndex]);
+                  self->aacOutDataHeadroom, flags, self->flags[streamIndex]);
 
               self->extGainDelay = self->streamInfo.aacSamplesPerFrame;
               break;
@@ -3388,7 +3395,8 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
         if (!CConceal_TDFading_Applied[c]) {
           CConceal_TDFading_Applied[c] = CConcealment_TDFading(
               self->streamInfo.aacSamplesPerFrame,
-              &self->pAacDecoderStaticChannelInfo[c], pTimeData + offset, 0);
+              &self->pAacDecoderStaticChannelInfo[c], self->aacOutDataHeadroom,
+              pTimeData + offset, 0);
           if (c + 1 < (8) && c < aacChannels - 1) {
             /* update next TDNoise Seed to avoid muting in case of Parametric
              * Stereo */
@@ -3409,27 +3417,18 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
        * LR) */
       if ((aacChannels == 2) && bsPseudoLr) {
         int i, offset2;
-        const FIXP_SGL invSqrt2 =
-            FL2FXCONST_SGL(0.353553390593273f); /* scaled by -1 */
-        FIXP_PCM *pTD = pTimeData;
+        const FIXP_SGL invSqrt2 = FL2FXCONST_SGL(0.707106781186547f);
+        PCM_DEC *pTD = pTimeData;
 
         offset2 = timeDataChannelOffset;
 
         for (i = 0; i < self->streamInfo.aacSamplesPerFrame; i++) {
-          FIXP_DBL L = FX_PCM2FX_DBL(pTD[0]);
-          FIXP_DBL R = FX_PCM2FX_DBL(pTD[offset2]);
+          FIXP_DBL L = PCM_DEC2FIXP_DBL(pTD[0]);
+          FIXP_DBL R = PCM_DEC2FIXP_DBL(pTD[offset2]);
           L = fMult(L, invSqrt2);
           R = fMult(R, invSqrt2);
-#if (SAMPLE_BITS == 16)
-          pTD[0] = (FIXP_SGL)SATURATE_RIGHT_SHIFT(L + R + (FIXP_DBL)(1 << 14),
-                                                  15, FRACT_BITS);
-          pTD[offset2] = (FIXP_SGL)SATURATE_RIGHT_SHIFT(
-              L - R + (FIXP_DBL)(1 << 14), 15, FRACT_BITS);
-#else
-          pTD[0] = SATURATE_LEFT_SHIFT(FX_DBL2FX_PCM(L + R), 1, DFRACT_BITS);
-          pTD[offset2] =
-              SATURATE_LEFT_SHIFT(FX_DBL2FX_PCM(L - R), 1, DFRACT_BITS);
-#endif
+          pTD[0] = L + R;
+          pTD[offset2] = L - R;
           pTD++;
         }
       }
diff --git a/libAACdec/src/aacdecoder.h b/libAACdec/src/aacdecoder.h
index 20f4c45..bd1f38f 100644
--- a/libAACdec/src/aacdecoder.h
+++ b/libAACdec/src/aacdecoder.h
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -191,6 +191,9 @@ struct AAC_DECODER_INSTANCE {
   INT outputInterleaved; /*!< PCM output format (interleaved/none interleaved).
                           */
 
+  INT aacOutDataHeadroom; /*!< Headroom of the output time signal to prevent
+                             clipping */
+
   HANDLE_TRANSPORTDEC hInput; /*!< Transport layer handle. */
 
   SamplingRateInfo
@@ -235,6 +238,7 @@ struct AAC_DECODER_INSTANCE {
   CAacDecoderStaticChannelInfo
       *pAacDecoderStaticChannelInfo[(8)]; /*!< Persistent channel memory */
 
+  FIXP_DBL *workBufferCore1;
   FIXP_DBL *workBufferCore2;
   PCM_DEC *pTimeData2;
   INT timeData2Size;
@@ -311,11 +315,10 @@ This structure is allocated once for each CPE. */
   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;
@@ -427,7 +430,7 @@ LINKSPEC_H AAC_DECODER_ERROR CAacDecoder_Init(HANDLE_AACDECODER self,
   \return  error status
 */
 LINKSPEC_H AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
-    HANDLE_AACDECODER self, const UINT flags, FIXP_PCM *pTimeData,
+    HANDLE_AACDECODER self, const UINT flags, PCM_DEC *pTimeData,
     const INT timeDataSize, const int timeDataChannelOffset);
 
 /* Free config dependent AAC memory */
diff --git a/libAACdec/src/aacdecoder_lib.cpp b/libAACdec/src/aacdecoder_lib.cpp
index 2ba0e86..f5ce7e0 100644
--- a/libAACdec/src/aacdecoder_lib.cpp
+++ b/libAACdec/src/aacdecoder_lib.cpp
@@ -119,8 +119,8 @@ amm-info@iis.fraunhofer.de
 
 /* Decoder library info */
 #define AACDECODER_LIB_VL0 3
-#define AACDECODER_LIB_VL1 1
-#define AACDECODER_LIB_VL2 3
+#define AACDECODER_LIB_VL1 2
+#define AACDECODER_LIB_VL2 0
 #define AACDECODER_LIB_TITLE "AAC Decoder Lib"
 #ifdef __ANDROID__
 #define AACDECODER_LIB_BUILD_DATE ""
@@ -1131,35 +1131,31 @@ static INT aacDecoder_EstimateNumberOfLostFrames(HANDLE_AACDECODER self) {
   return n;
 }
 
-LINKSPEC_CPP AAC_DECODER_ERROR
-aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern,
-                       const INT timeDataSize_extern, const UINT flags) {
+LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_DecodeFrame(HANDLE_AACDECODER self,
+                                                      INT_PCM *pTimeData,
+                                                      const INT timeDataSize,
+                                                      const UINT flags) {
   AAC_DECODER_ERROR ErrorStatus;
   INT layer;
   INT nBits;
+  INT timeData2Size;
+  INT timeData3Size;
+  INT timeDataHeadroom;
   HANDLE_FDK_BITSTREAM hBs;
   int fTpInterruption = 0; /* Transport originated interruption detection. */
   int fTpConceal = 0;      /* Transport originated concealment. */
-  INT_PCM *pTimeData = NULL;
-  INT timeDataSize = 0;
   UINT accessUnit = 0;
   UINT numAccessUnits = 1;
   UINT numPrerollAU = 0;
-  int fEndAuNotAdjusted = 0;  /* The end of the access unit was not adjusted */
-  int applyCrossfade = 1;     /* flag indicates if flushing was possible */
-  FIXP_PCM *pTimeDataFixpPcm; /* Signal buffer for decoding process before PCM
-                                 processing */
-  INT timeDataFixpPcmSize;
-  PCM_DEC *pTimeDataPcmPost; /* Signal buffer for PCM post-processing */
-  INT timeDataPcmPostSize;
+  int fEndAuNotAdjusted = 0; /* The end of the access unit was not adjusted */
+  int applyCrossfade = 1;    /* flag indicates if flushing was possible */
+  PCM_DEC *pTimeData2;
+  PCM_AAC *pTimeData3;
 
   if (self == NULL) {
     return AAC_DEC_INVALID_HANDLE;
   }
 
-  pTimeData = self->pcmOutputBuffer;
-  timeDataSize = sizeof(self->pcmOutputBuffer) / sizeof(*self->pcmOutputBuffer);
-
   if (flags & AACDEC_INTR) {
     self->streamInfo.numLostAccessUnits = 0;
   }
@@ -1315,19 +1311,23 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern,
       /* Use limiter configuration as requested. */
       self->limiterEnableCurr = self->limiterEnableUser;
     }
-    /* reset limiter gain on a per frame basis */
-    self->extGain[0] = FL2FXCONST_DBL(1.0f / (float)(1 << TDL_GAIN_SCALING));
 
-    pTimeDataFixpPcm = pTimeData;
-    timeDataFixpPcmSize = timeDataSize;
+    /* reset DRC level normalization gain on a per frame basis */
+    self->extGain[0] = AACDEC_DRC_GAIN_INIT_VALUE;
+
+    pTimeData2 = self->pTimeData2;
+    timeData2Size = self->timeData2Size / sizeof(PCM_DEC);
+    pTimeData3 = (PCM_AAC *)self->pTimeData2;
+    timeData3Size = self->timeData2Size / sizeof(PCM_AAC);
 
     ErrorStatus = CAacDecoder_DecodeFrame(
         self,
         flags | (fTpConceal ? AACDEC_CONCEAL : 0) |
             ((self->flushStatus && !(flags & AACDEC_CONCEAL)) ? AACDEC_FLUSH
                                                               : 0),
-        pTimeDataFixpPcm + 0, timeDataFixpPcmSize,
-        self->streamInfo.aacSamplesPerFrame + 0);
+        pTimeData2 + 0, timeData2Size, self->streamInfo.aacSamplesPerFrame + 0);
+
+    timeDataHeadroom = self->aacOutDataHeadroom;
 
     /* if flushing for USAC DASH IPF was not possible go on with decoding
      * preroll */
@@ -1352,7 +1352,7 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern,
         }
       }
 
-      /* If the current pTimeDataFixpPcm does not contain a valid signal, there
+      /* If the current pTimeData2 does not contain a valid signal, there
        * nothing else we can do, so bail. */
       if (!IS_OUTPUT_VALID(ErrorStatus)) {
         goto bail;
@@ -1366,10 +1366,10 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern,
       self->streamInfo.numChannels = self->streamInfo.aacNumChannels;
 
       {
-        FDK_Delay_Apply(&self->usacResidualDelay,
-                        pTimeDataFixpPcm +
-                            1 * (self->streamInfo.aacSamplesPerFrame + 0) + 0,
-                        self->streamInfo.frameSize, 0);
+        FDK_Delay_Apply(
+            &self->usacResidualDelay,
+            pTimeData2 + 1 * (self->streamInfo.aacSamplesPerFrame + 0) + 0,
+            self->streamInfo.frameSize, 0);
       }
 
       /* Setting of internal MPS state; may be reset in CAacDecoder_SyncQmfMode
@@ -1416,8 +1416,6 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern,
         }
       }
 
-      self->qmfDomain.globalConf.TDinput = pTimeData;
-
       switch (FDK_QmfDomain_Configure(&self->qmfDomain)) {
         default:
         case QMF_DOMAIN_INIT_ERROR:
@@ -1474,18 +1472,18 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern,
         sbrDecoder_SetParam(self->hSbrDecoder, SBR_SKIP_QMF,
                             (self->mpsEnableCurr) ? 2 : 0);
 
-        INT_PCM *input;
-        input = (INT_PCM *)self->workBufferCore2;
-        FDKmemcpy(input, pTimeData,
-                  sizeof(INT_PCM) * (self->streamInfo.numChannels) *
+        PCM_AAC *input;
+        input = (PCM_AAC *)self->workBufferCore2;
+        FDKmemcpy(input, pTimeData3,
+                  sizeof(PCM_AAC) * (self->streamInfo.numChannels) *
                       (self->streamInfo.frameSize));
 
         /* apply SBR processing */
-        sbrError = sbrDecoder_Apply(self->hSbrDecoder, input, pTimeData,
-                                    timeDataSize, &self->streamInfo.numChannels,
-                                    &self->streamInfo.sampleRate,
-                                    &self->mapDescr, self->chMapIndex,
-                                    self->frameOK, &self->psPossible);
+        sbrError = sbrDecoder_Apply(
+            self->hSbrDecoder, input, pTimeData3, timeData3Size,
+            &self->streamInfo.numChannels, &self->streamInfo.sampleRate,
+            &self->mapDescr, self->chMapIndex, self->frameOK, &self->psPossible,
+            self->aacOutDataHeadroom, &timeDataHeadroom);
 
         if (sbrError == SBRDEC_OK) {
           /* Update data in streaminfo structure. Assume that the SBR upsampling
@@ -1564,10 +1562,11 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern,
         if (err == 0) {
           err = mpegSurroundDecoder_Apply(
               (CMpegSurroundDecoder *)self->pMpegSurroundDecoder,
-              (INT_PCM *)self->workBufferCore2, pTimeData, timeDataSize,
+              (PCM_AAC *)self->workBufferCore2, pTimeData3, timeData3Size,
               self->streamInfo.aacSamplesPerFrame, &nChannels, &frameSize,
               self->streamInfo.sampleRate, self->streamInfo.aot,
-              self->channelType, self->channelIndices, &self->mapDescr);
+              self->channelType, self->channelIndices, &self->mapDescr,
+              self->aacOutDataHeadroom, &timeDataHeadroom);
         }
 
         if (err == MPS_OUTPUT_BUFFER_TOO_SMALL) {
@@ -1590,8 +1589,8 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern,
           self->streamInfo.frameSize = self->mpsFrameSizeLast;
           /* ... and clear output buffer so that potentially corrupted data does
            * not reach the framework. */
-          FDKmemclear(pTimeData, self->mpsOutChannelsLast *
-                                     self->mpsFrameSizeLast * sizeof(INT_PCM));
+          FDKmemclear(pTimeData3, self->mpsOutChannelsLast *
+                                      self->mpsFrameSizeLast * sizeof(PCM_AAC));
           /* Additionally proclaim that this frame had errors during decoding.
            */
           ErrorStatus = AAC_DEC_DECODE_FRAME_ERROR;
@@ -1612,11 +1611,11 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern,
         sbrDecoder_SetParam(self->hSbrDecoder, SBR_SKIP_QMF, 1);
 
         /* apply SBR processing */
-        sbrError = sbrDecoder_Apply(self->hSbrDecoder, pTimeData, pTimeData,
-                                    timeDataSize, &self->streamInfo.numChannels,
-                                    &self->streamInfo.sampleRate,
-                                    &self->mapDescr, self->chMapIndex,
-                                    self->frameOK, &self->psPossible);
+        sbrError = sbrDecoder_Apply(
+            self->hSbrDecoder, pTimeData3, pTimeData3, timeData3Size,
+            &self->streamInfo.numChannels, &self->streamInfo.sampleRate,
+            &self->mapDescr, self->chMapIndex, self->frameOK, &self->psPossible,
+            self->aacOutDataHeadroom, &timeDataHeadroom);
 
         if (sbrError == SBRDEC_OK) {
           /* Update data in streaminfo structure. Assume that the SBR upsampling
@@ -1644,17 +1643,15 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern,
         }
       }
 
-      /* Use dedicated memory for PCM postprocessing */
-      pTimeDataPcmPost = self->pTimeData2;
-      timeDataPcmPostSize = self->timeData2Size;
-
       {
-        const int size =
-            self->streamInfo.frameSize * self->streamInfo.numChannels;
-        FDK_ASSERT(timeDataPcmPostSize >= size);
-        for (int i = 0; i < size; i++) {
-          pTimeDataPcmPost[i] =
-              (PCM_DEC)FX_PCM2PCM_DEC(pTimeData[i]) >> PCM_OUT_HEADROOM;
+        if ((INT)PCM_OUT_HEADROOM != timeDataHeadroom) {
+          for (int i = ((self->streamInfo.frameSize *
+                         self->streamInfo.numChannels) -
+                        1);
+               i >= 0; i--) {
+            pTimeData2[i] =
+                (PCM_DEC)pTimeData3[i] >> (PCM_OUT_HEADROOM - timeDataHeadroom);
+          }
         }
       }
 
@@ -1709,22 +1706,21 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern,
           if ((self->streamInfo.numChannels > 1) &&
               (0 || (self->sbrEnabled) || (self->mpsEnableCurr))) {
             /* interleaving/deinterleaving is performed on upper part of
-             * pTimeDataPcmPost. Check if this buffer is large enough. */
-            if (timeDataPcmPostSize <
-                (INT)(2 * self->streamInfo.numChannels *
-                      self->streamInfo.frameSize * sizeof(PCM_DEC))) {
+             * pTimeData2. Check if this buffer is large enough. */
+            if (timeData2Size < (INT)(2 * self->streamInfo.numChannels *
+                                      self->streamInfo.frameSize)) {
               ErrorStatus = AAC_DEC_UNKNOWN;
               goto bail;
             }
             needsDeinterleaving = 1;
             drcWorkBuffer =
-                (FIXP_DBL *)pTimeDataPcmPost +
+                (FIXP_DBL *)pTimeData2 +
                 self->streamInfo.numChannels * self->streamInfo.frameSize;
             FDK_deinterleave(
-                pTimeDataPcmPost, drcWorkBuffer, self->streamInfo.numChannels,
+                pTimeData2, drcWorkBuffer, self->streamInfo.numChannels,
                 self->streamInfo.frameSize, self->streamInfo.frameSize);
           } else {
-            drcWorkBuffer = (FIXP_DBL *)pTimeDataPcmPost;
+            drcWorkBuffer = pTimeData2;
           }
 
           /* prepare Loudness Normalisation gain */
@@ -1759,7 +1755,7 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern,
 
           if (needsDeinterleaving) {
             FDK_interleave(
-                drcWorkBuffer, pTimeDataPcmPost, self->streamInfo.numChannels,
+                drcWorkBuffer, pTimeData2, self->streamInfo.numChannels,
                 self->streamInfo.frameSize, self->streamInfo.frameSize);
           }
         }
@@ -1799,6 +1795,9 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern,
 
       if (self->streamInfo.extAot != AOT_AAC_SLS) {
         INT pcmLimiterScale = 0;
+        INT interleaved = 0;
+        interleaved |= (self->sbrEnabled) ? 1 : 0;
+        interleaved |= (self->mpsEnableCurr) ? 1 : 0;
         PCMDMX_ERROR dmxErr = PCMDMX_OK;
         if ((flags & AACDEC_INTR) && (accessUnit == 0)) {
           /* delete data from the past (e.g. mixdown coeficients) */
@@ -1811,17 +1810,12 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern,
           }
         }
 
-        INT interleaved = 0;
-        interleaved |= (self->sbrEnabled) ? 1 : 0;
-        interleaved |= (self->mpsEnableCurr) ? 1 : 0;
-
         /* do PCM post processing */
-        dmxErr = pcmDmx_ApplyFrame(
-            self->hPcmUtils, pTimeDataPcmPost, timeDataFixpPcmSize,
-            self->streamInfo.frameSize, &self->streamInfo.numChannels,
-            interleaved, self->channelType, self->channelIndices,
-            &self->mapDescr,
-            (self->limiterEnableCurr) ? &pcmLimiterScale : NULL);
+        dmxErr = pcmDmx_ApplyFrame(self->hPcmUtils, pTimeData2, timeData2Size,
+                                   self->streamInfo.frameSize,
+                                   &self->streamInfo.numChannels, interleaved,
+                                   self->channelType, self->channelIndices,
+                                   &self->mapDescr, &pcmLimiterScale);
         if (dmxErr == PCMDMX_OUTPUT_BUFFER_TOO_SMALL) {
           ErrorStatus = AAC_DEC_OUTPUT_BUFFER_TOO_SMALL;
           goto bail;
@@ -1833,13 +1827,35 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern,
           ErrorStatus = AAC_DEC_DECODE_FRAME_ERROR;
         }
 
+        pcmLimiterScale += PCM_OUT_HEADROOM;
+
         if (flags & AACDEC_CLRHIST) {
           if (!(self->flags[0] & AC_USAC)) {
+            /* Reset DRC data */
+            aacDecoder_drcReset(self->hDrcInfo);
             /* Delete the delayed signal. */
             pcmLimiter_Reset(self->hLimiter);
           }
         }
 
+        /* Set applyExtGain if DRC processing is enabled and if
+           progRefLevelPresent is present for the first time. Consequences: The
+           headroom of the output signal can be set to AACDEC_DRC_GAIN_SCALING
+           only for audio formats which support legacy DRC Level Normalization.
+                         For all other audio formats the headroom of the output
+           signal is set to PCM_OUT_HEADROOM. */
+        if (self->hDrcInfo->enable &&
+            (self->hDrcInfo->progRefLevelPresent == 1)) {
+          self->hDrcInfo->applyExtGain |= 1;
+        }
+
+        /* Check whether time data buffer is large enough. */
+        if (timeDataSize <
+            (self->streamInfo.numChannels * self->streamInfo.frameSize)) {
+          ErrorStatus = AAC_DEC_OUTPUT_BUFFER_TOO_SMALL;
+          goto bail;
+        }
+
         if (self->limiterEnableCurr) {
           /* use workBufferCore2 buffer for interleaving */
           PCM_LIM *pInterleaveBuffer;
@@ -1848,44 +1864,72 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern,
           /* Set actual signal parameters */
           pcmLimiter_SetNChannels(self->hLimiter, self->streamInfo.numChannels);
           pcmLimiter_SetSampleRate(self->hLimiter, self->streamInfo.sampleRate);
-          pcmLimiterScale += PCM_OUT_HEADROOM;
 
           if ((self->streamInfo.numChannels == 1) || (self->sbrEnabled) ||
               (self->mpsEnableCurr)) {
-            pInterleaveBuffer = (PCM_LIM *)pTimeDataPcmPost;
+            pInterleaveBuffer = (PCM_LIM *)pTimeData2;
           } else {
-            pInterleaveBuffer = (PCM_LIM *)pTimeData;
+            pInterleaveBuffer = (PCM_LIM *)self->workBufferCore2;
+
             /* applyLimiter requests for interleaved data */
             /* Interleave ouput buffer */
-            FDK_interleave(pTimeDataPcmPost, pInterleaveBuffer,
+            FDK_interleave(pTimeData2, pInterleaveBuffer,
                            self->streamInfo.numChannels, blockLength,
                            self->streamInfo.frameSize);
           }
 
+          FIXP_DBL *pGainPerSample = NULL;
+
+          if (self->hDrcInfo->enable && self->hDrcInfo->applyExtGain) {
+            pGainPerSample = self->workBufferCore1;
+
+            if ((INT)GetRequiredMemWorkBufferCore1() <
+                (INT)(self->streamInfo.frameSize * sizeof(FIXP_DBL))) {
+              ErrorStatus = AAC_DEC_UNKNOWN;
+              goto bail;
+            }
+
+            pcmLimiterScale = applyDrcLevelNormalization(
+                self->hDrcInfo, (PCM_DEC *)pInterleaveBuffer, self->extGain,
+                pGainPerSample, pcmLimiterScale, self->extGainDelay,
+                self->streamInfo.frameSize, self->streamInfo.numChannels, 1, 1);
+          }
+
           pcmLimiter_Apply(self->hLimiter, pInterleaveBuffer, pTimeData,
-                           self->extGain, &pcmLimiterScale, 1,
-                           self->extGainDelay, self->streamInfo.frameSize);
+                           pGainPerSample, pcmLimiterScale,
+                           self->streamInfo.frameSize);
 
           {
             /* Announce the additional limiter output delay */
             self->streamInfo.outputDelay += pcmLimiter_GetDelay(self->hLimiter);
           }
         } else {
+          if (self->hDrcInfo->enable && self->hDrcInfo->applyExtGain) {
+            pcmLimiterScale = applyDrcLevelNormalization(
+                self->hDrcInfo, pTimeData2, self->extGain, NULL,
+                pcmLimiterScale, self->extGainDelay, self->streamInfo.frameSize,
+                self->streamInfo.numChannels,
+                (interleaved || (self->streamInfo.numChannels == 1))
+                    ? 1
+                    : self->streamInfo.frameSize,
+                0);
+          }
+
           /* 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) ||
               (self->mpsEnableCurr)) {
             scaleValuesSaturate(
-                pTimeData, pTimeDataPcmPost,
+                pTimeData, pTimeData2,
                 self->streamInfo.frameSize * self->streamInfo.numChannels,
-                PCM_OUT_HEADROOM);
+                pcmLimiterScale);
 
           } else {
             scaleValuesSaturate(
-                (INT_PCM *)self->workBufferCore2, pTimeDataPcmPost,
+                (INT_PCM *)self->workBufferCore2, pTimeData2,
                 self->streamInfo.frameSize * self->streamInfo.numChannels,
-                PCM_OUT_HEADROOM);
+                pcmLimiterScale);
             /* Interleave ouput buffer */
             FDK_interleave((INT_PCM *)self->workBufferCore2, pTimeData,
                            self->streamInfo.numChannels,
@@ -1981,20 +2025,8 @@ bail:
     ErrorStatus = AAC_DEC_UNKNOWN;
   }
 
-  /* Check whether external output buffer is large enough. */
-  if (timeDataSize_extern <
-      self->streamInfo.numChannels * self->streamInfo.frameSize) {
-    ErrorStatus = AAC_DEC_OUTPUT_BUFFER_TOO_SMALL;
-  }
-
-  /* Update external output buffer. */
-  if (IS_OUTPUT_VALID(ErrorStatus)) {
-    FDKmemcpy(pTimeData_extern, pTimeData,
-              self->streamInfo.numChannels * self->streamInfo.frameSize *
-                  sizeof(*pTimeData));
-  } else {
-    FDKmemclear(pTimeData_extern,
-                timeDataSize_extern * sizeof(*pTimeData_extern));
+  if (!IS_OUTPUT_VALID(ErrorStatus)) {
+    FDKmemclear(pTimeData, timeDataSize * sizeof(*pTimeData));
   }
 
   return ErrorStatus;
diff --git a/libAACdec/src/block.cpp b/libAACdec/src/block.cpp
index b3d09a6..0bca577 100644
--- a/libAACdec/src/block.cpp
+++ b/libAACdec/src/block.cpp
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -1015,9 +1015,9 @@ FIXP_DBL get_gain(const FIXP_DBL *x, const FIXP_DBL *y, int n) {
 
 void CBlock_FrequencyToTime(
     CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo,
-    CAacDecoderChannelInfo *pAacDecoderChannelInfo, FIXP_PCM outSamples[],
+    CAacDecoderChannelInfo *pAacDecoderChannelInfo, PCM_DEC outSamples[],
     const SHORT frameLen, const int frameOk, FIXP_DBL *pWorkBuffer1,
-    UINT elFlags, INT elCh) {
+    const INT aacOutDataHeadroom, UINT elFlags, INT elCh) {
   int fr, fl, tl, nSpec;
 
 #if defined(FDK_ASSERT_ENABLE)
@@ -1213,6 +1213,7 @@ void CBlock_FrequencyToTime(
         bass_pf_1sf_delay(p2_synth, pitch, pit_gain, frameLen,
                           (LpdSfd + 2) * L_SUBFR + BPF_SFD * L_SUBFR,
                           frameLen - (LpdSfd + 4) * L_SUBFR, outSamples,
+                          aacOutDataHeadroom,
                           pAacDecoderStaticChannelInfo->mem_bpf);
       }
 
@@ -1236,7 +1237,8 @@ void CBlock_FrequencyToTime(
                          ? MLT_FLAG_CURR_ALIAS_SYMMETRY
                          : 0);
 
-      scaleValuesSaturate(outSamples, tmp, frameLen, MDCT_OUT_HEADROOM);
+      scaleValuesSaturate(outSamples, tmp, frameLen,
+                          MDCT_OUT_HEADROOM - aacOutDataHeadroom);
     }
   }
 
@@ -1251,7 +1253,7 @@ void CBlock_FrequencyToTime(
 #include "ldfiltbank.h"
 void CBlock_FrequencyToTimeLowDelay(
     CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo,
-    CAacDecoderChannelInfo *pAacDecoderChannelInfo, FIXP_PCM outSamples[],
+    CAacDecoderChannelInfo *pAacDecoderChannelInfo, PCM_DEC outSamples[],
     const short frameLen) {
   InvMdctTransformLowDelay_fdk(
       SPEC_LONG(pAacDecoderChannelInfo->pSpectralCoefficient),
diff --git a/libAACdec/src/block.h b/libAACdec/src/block.h
index f0f56cd..f5118a2 100644
--- a/libAACdec/src/block.h
+++ b/libAACdec/src/block.h
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -218,16 +218,16 @@ void ApplyTools(CAacDecoderChannelInfo *pAacDecoderChannelInfo[],
  */
 void CBlock_FrequencyToTime(
     CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo,
-    CAacDecoderChannelInfo *pAacDecoderChannelInfo, FIXP_PCM outSamples[],
+    CAacDecoderChannelInfo *pAacDecoderChannelInfo, PCM_DEC outSamples[],
     const SHORT frameLen, const int frameOk, FIXP_DBL *pWorkBuffer1,
-    UINT elFlags, INT elCh);
+    const INT aacOutDataHeadroom, UINT elFlags, INT elCh);
 
 /**
  * \brief Transform double lapped MDCT (AAC-ELD) spectral data into time domain.
  */
 void CBlock_FrequencyToTimeLowDelay(
     CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo,
-    CAacDecoderChannelInfo *pAacDecoderChannelInfo, FIXP_PCM outSamples[],
+    CAacDecoderChannelInfo *pAacDecoderChannelInfo, PCM_DEC outSamples[],
     const short frameLen);
 
 AAC_DECODER_ERROR CBlock_InverseQuantizeSpectralData(
diff --git a/libAACdec/src/conceal.cpp b/libAACdec/src/conceal.cpp
index 379e63a..0939bb5 100644
--- a/libAACdec/src/conceal.cpp
+++ b/libAACdec/src/conceal.cpp
@@ -226,7 +226,7 @@ static void CConcealment_ApplyRandomSign(int iRandomPhase, FIXP_DBL *spec,
 
 /* TimeDomainFading */
 static void CConcealment_TDFadePcmAtt(int start, int len, FIXP_DBL fadeStart,
-                                      FIXP_DBL fadeStop, FIXP_PCM *pcmdata);
+                                      FIXP_DBL fadeStop, PCM_DEC *pcmdata);
 static void CConcealment_TDFadeFillFadingStations(FIXP_DBL *fadingStations,
                                                   int *fadingSteps,
                                                   FIXP_DBL fadeStop,
@@ -242,7 +242,9 @@ static int CConcealment_ApplyFadeOut(
 
 static int CConcealment_TDNoise_Random(ULONG *seed);
 static void CConcealment_TDNoise_Apply(CConcealmentInfo *const pConcealmentInfo,
-                                       const int len, FIXP_PCM *const pcmdata);
+                                       const int len,
+                                       const INT aacOutDataHeadroom,
+                                       PCM_DEC *const pcmdata);
 
 static BLOCK_TYPE CConcealment_GetWinSeq(int prevWinSeq) {
   BLOCK_TYPE newWinSeq = BLOCK_LONG;
@@ -1844,7 +1846,7 @@ Target fading level is determined by fading index cntFadeFrames.
 
 INT CConcealment_TDFading(
     int len, CAacDecoderStaticChannelInfo **ppAacDecoderStaticChannelInfo,
-    FIXP_PCM *pcmdata, FIXP_PCM *pcmdata_1) {
+    const INT aacOutDataHeadroom, PCM_DEC *pcmdata, PCM_DEC *pcmdata_1) {
   /*
   Do the fading in Time domain based on concealment states and core mode
   */
@@ -1957,7 +1959,8 @@ INT CConcealment_TDFading(
       start += len;
     }
   }
-  CConcealment_TDNoise_Apply(pConcealmentInfo, len, pcmdata);
+  CConcealment_TDNoise_Apply(pConcealmentInfo, len, aacOutDataHeadroom,
+                             pcmdata);
 
   /* Save end-of-frame attenuation and fading type */
   pConcealmentInfo->lastFadingType = fadingType;
@@ -1969,12 +1972,11 @@ INT CConcealment_TDFading(
 
 /* attenuate pcmdata in Time Domain Fading process */
 static void CConcealment_TDFadePcmAtt(int start, int len, FIXP_DBL fadeStart,
-                                      FIXP_DBL fadeStop, FIXP_PCM *pcmdata) {
+                                      FIXP_DBL fadeStop, PCM_DEC *pcmdata) {
   int i;
   FIXP_DBL dStep;
   FIXP_DBL dGain;
   FIXP_DBL dGain_apply;
-  int bitshift = (DFRACT_BITS - SAMPLE_BITS);
 
   /* set start energy */
   dGain = fadeStart;
@@ -1987,7 +1989,7 @@ static void CConcealment_TDFadePcmAtt(int start, int len, FIXP_DBL fadeStart,
      */
     dGain_apply = fMax((FIXP_DBL)0, dGain);
     /* finally, attenuate samples */
-    pcmdata[i] = (FIXP_PCM)((fMult(pcmdata[i], (dGain_apply))) >> bitshift);
+    pcmdata[i] = FIXP_DBL2PCM_DEC(fMult(pcmdata[i], dGain_apply));
   }
 }
 
@@ -2050,9 +2052,11 @@ static int CConcealment_TDNoise_Random(ULONG *seed) {
 }
 
 static void CConcealment_TDNoise_Apply(CConcealmentInfo *const pConcealmentInfo,
-                                       const int len, FIXP_PCM *const pcmdata) {
-  FIXP_PCM *states = pConcealmentInfo->TDNoiseStates;
-  FIXP_PCM noiseVal;
+                                       const int len,
+                                       const INT aacOutDataHeadroom,
+                                       PCM_DEC *const pcmdata) {
+  PCM_DEC *states = pConcealmentInfo->TDNoiseStates;
+  PCM_DEC noiseVal;
   FIXP_DBL noiseValLong;
   FIXP_SGL *coef = pConcealmentInfo->TDNoiseCoef;
   FIXP_DBL TDNoiseAtt;
@@ -2070,18 +2074,20 @@ static void CConcealment_TDNoise_Apply(CConcealmentInfo *const pConcealmentInfo,
       /* create filtered noise */
       states[2] = states[1];
       states[1] = states[0];
-      states[0] = ((FIXP_PCM)CConcealment_TDNoise_Random(&seed));
+      states[0] =
+          FIXP_DBL2PCM_DEC((FIXP_DBL)CConcealment_TDNoise_Random(&seed));
       noiseValLong = fMult(states[0], coef[0]) + fMult(states[1], coef[1]) +
                      fMult(states[2], coef[2]);
-      noiseVal = FX_DBL2FX_PCM(fMult(noiseValLong, TDNoiseAtt));
+      noiseVal = FIXP_DBL2PCM_DEC(fMult(noiseValLong, TDNoiseAtt) >>
+                                  aacOutDataHeadroom);
 
       /* add filtered noise - check for clipping, before */
-      if (noiseVal > (FIXP_PCM)0 &&
-          pcmdata[ii] > (FIXP_PCM)MAXVAL_FIXP_PCM - noiseVal) {
-        noiseVal = noiseVal * (FIXP_PCM)-1;
-      } else if (noiseVal < (FIXP_PCM)0 &&
-                 pcmdata[ii] < (FIXP_PCM)MINVAL_FIXP_PCM - noiseVal) {
-        noiseVal = noiseVal * (FIXP_PCM)-1;
+      if (noiseVal > (PCM_DEC)0 &&
+          pcmdata[ii] > (PCM_DEC)MAXVAL_PCM_DEC - noiseVal) {
+        noiseVal = noiseVal * (PCM_DEC)-1;
+      } else if (noiseVal < (PCM_DEC)0 &&
+                 pcmdata[ii] < (PCM_DEC)MINVAL_PCM_DEC - noiseVal) {
+        noiseVal = noiseVal * (PCM_DEC)-1;
       }
 
       pcmdata[ii] += noiseVal;
diff --git a/libAACdec/src/conceal.h b/libAACdec/src/conceal.h
index e01a796..0c002a5 100644
--- a/libAACdec/src/conceal.h
+++ b/libAACdec/src/conceal.h
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -147,6 +147,6 @@ int CConcealment_GetLastFrameOk(CConcealmentInfo *hConcealmentInfo,
 
 INT CConcealment_TDFading(
     int len, CAacDecoderStaticChannelInfo **ppAacDecoderStaticChannelInfo,
-    FIXP_PCM *pcmdata, FIXP_PCM *pcmdata_1);
+    const INT aacOutDataHeadroom, PCM_DEC *pcmdata, PCM_DEC *pcmdata_1);
 
 #endif /* #ifndef CONCEAL_H */
diff --git a/libAACdec/src/conceal_types.h b/libAACdec/src/conceal_types.h
index d90374e..36e7dec 100644
--- a/libAACdec/src/conceal_types.h
+++ b/libAACdec/src/conceal_types.h
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -194,7 +194,7 @@ typedef struct {
   FIXP_DBL last_tcx_gain;
   INT last_tcx_gain_e;
   ULONG TDNoiseSeed;
-  FIXP_PCM TDNoiseStates[3];
+  PCM_DEC TDNoiseStates[3];
   FIXP_SGL TDNoiseCoef[3];
   FIXP_SGL TDNoiseAtt;
 
diff --git a/libAACdec/src/ldfiltbank.cpp b/libAACdec/src/ldfiltbank.cpp
index 1898557..13e61a5 100644
--- a/libAACdec/src/ldfiltbank.cpp
+++ b/libAACdec/src/ldfiltbank.cpp
@@ -112,17 +112,20 @@ amm-info@iis.fraunhofer.de
 #if defined(__arm__)
 #endif
 
-static void multE2_DinvF_fdk(FIXP_PCM *output, FIXP_DBL *x, const FIXP_WTB *fb,
+static void multE2_DinvF_fdk(PCM_DEC *output, FIXP_DBL *x, const FIXP_WTB *fb,
                              FIXP_DBL *z, const int N) {
   int i;
 
-  /*  scale for FIXP_DBL -> INT_PCM conversion. */
-  const int scale = (DFRACT_BITS - SAMPLE_BITS) - LDFB_HEADROOM;
-#if ((DFRACT_BITS - SAMPLE_BITS - LDFB_HEADROOM) > 0)
+  /*  scale for FIXP_DBL -> PCM_DEC conversion:       */
+  const int scale = (DFRACT_BITS - PCM_OUT_BITS) - LDFB_HEADROOM + (3);
+
+#if ((DFRACT_BITS - PCM_OUT_BITS - LDFB_HEADROOM + (3) - 1) > 0)
   FIXP_DBL rnd_val_wts0 = (FIXP_DBL)0;
   FIXP_DBL rnd_val_wts1 = (FIXP_DBL)0;
+#if ((DFRACT_BITS - PCM_OUT_BITS - LDFB_HEADROOM + (3) - WTS0 - 1) > 0)
   if (-WTS0 - 1 + scale)
     rnd_val_wts0 = (FIXP_DBL)(1 << (-WTS0 - 1 + scale - 1));
+#endif
   if (-WTS1 - 1 + scale)
     rnd_val_wts1 = (FIXP_DBL)(1 << (-WTS1 - 1 + scale - 1));
 #endif
@@ -141,16 +144,16 @@ static void multE2_DinvF_fdk(FIXP_PCM *output, FIXP_DBL *x, const FIXP_WTB *fb,
     tmp = (fMultDiv2(z[N / 2 + i], fb[N + N / 2 - 1 - i]) +
            fMultDiv2(z[i], fb[N + N / 2 + i]));
 
-#if ((DFRACT_BITS - SAMPLE_BITS - LDFB_HEADROOM) > 0)
+#if ((DFRACT_BITS - PCM_OUT_BITS - LDFB_HEADROOM + (3) - 1) > 0)
     FDK_ASSERT((-WTS1 - 1 + scale) >= 0);
     FDK_ASSERT(tmp <= ((FIXP_DBL)0x7FFFFFFF -
                        rnd_val_wts1)); /* rounding must not cause overflow */
-    output[(N * 3 / 4 - 1 - i)] = (FIXP_PCM)SATURATE_RIGHT_SHIFT(
+    output[(N * 3 / 4 - 1 - i)] = (PCM_DEC)SATURATE_RIGHT_SHIFT(
         tmp + rnd_val_wts1, -WTS1 - 1 + scale, PCM_OUT_BITS);
 #else
     FDK_ASSERT((WTS1 + 1 - scale) >= 0);
     output[(N * 3 / 4 - 1 - i)] =
-        (FIXP_PCM)SATURATE_LEFT_SHIFT(tmp, WTS1 + 1 - scale, PCM_OUT_BITS);
+        (PCM_DEC)SATURATE_LEFT_SHIFT(tmp, WTS1 + 1 - scale, PCM_OUT_BITS);
 #endif
 
     z[i] = z0;
@@ -173,22 +176,22 @@ static void multE2_DinvF_fdk(FIXP_PCM *output, FIXP_DBL *x, const FIXP_WTB *fb,
     tmp1 = (fMultDiv2(z[N / 2 + i], fb[N + N / 2 - 1 - i]) +
             fMultDiv2(z[i], fb[N + N / 2 + i]));
 
-#if ((DFRACT_BITS - SAMPLE_BITS - LDFB_HEADROOM) > 0)
+#if ((DFRACT_BITS - PCM_OUT_BITS - LDFB_HEADROOM + (3) - 1) > 0)
     FDK_ASSERT((-WTS0 - 1 + scale) >= 0);
     FDK_ASSERT(tmp0 <= ((FIXP_DBL)0x7FFFFFFF -
                         rnd_val_wts0)); /* rounding must not cause overflow */
     FDK_ASSERT(tmp1 <= ((FIXP_DBL)0x7FFFFFFF -
                         rnd_val_wts1)); /* rounding must not cause overflow */
-    output[(i - N / 4)] = (FIXP_PCM)SATURATE_RIGHT_SHIFT(
+    output[(i - N / 4)] = (PCM_DEC)SATURATE_RIGHT_SHIFT(
         tmp0 + rnd_val_wts0, -WTS0 - 1 + scale, PCM_OUT_BITS);
-    output[(N * 3 / 4 - 1 - i)] = (FIXP_PCM)SATURATE_RIGHT_SHIFT(
+    output[(N * 3 / 4 - 1 - i)] = (PCM_DEC)SATURATE_RIGHT_SHIFT(
         tmp1 + rnd_val_wts1, -WTS1 - 1 + scale, PCM_OUT_BITS);
 #else
     FDK_ASSERT((WTS0 + 1 - scale) >= 0);
     output[(i - N / 4)] =
-        (FIXP_PCM)SATURATE_LEFT_SHIFT(tmp0, WTS0 + 1 - scale, PCM_OUT_BITS);
+        (PCM_DEC)SATURATE_LEFT_SHIFT(tmp0, WTS0 + 1 - scale, PCM_OUT_BITS);
     output[(N * 3 / 4 - 1 - i)] =
-        (FIXP_PCM)SATURATE_LEFT_SHIFT(tmp1, WTS1 + 1 - scale, PCM_OUT_BITS);
+        (PCM_DEC)SATURATE_LEFT_SHIFT(tmp1, WTS1 + 1 - scale, PCM_OUT_BITS);
 #endif
     z[i] = z0;
     z[N + i] = z2;
@@ -198,22 +201,22 @@ static void multE2_DinvF_fdk(FIXP_PCM *output, FIXP_DBL *x, const FIXP_WTB *fb,
   for (i = 0; i < N / 4; i++) {
     FIXP_DBL tmp0 = fMultDiv2(z[i], fb[N / 2 + i]);
 
-#if ((DFRACT_BITS - SAMPLE_BITS - LDFB_HEADROOM) > 0)
+#if ((DFRACT_BITS - PCM_OUT_BITS - LDFB_HEADROOM + (3) - 1) > 0)
     FDK_ASSERT((-WTS0 - 1 + scale) >= 0);
     FDK_ASSERT(tmp0 <= ((FIXP_DBL)0x7FFFFFFF -
                         rnd_val_wts0)); /* rounding must not cause overflow */
-    output[(N * 3 / 4 + i)] = (FIXP_PCM)SATURATE_RIGHT_SHIFT(
+    output[(N * 3 / 4 + i)] = (PCM_DEC)SATURATE_RIGHT_SHIFT(
         tmp0 + rnd_val_wts0, -WTS0 - 1 + scale, PCM_OUT_BITS);
 #else
     FDK_ASSERT((WTS0 + 1 - scale) >= 0);
     output[(N * 3 / 4 + i)] =
-        (FIXP_PCM)SATURATE_LEFT_SHIFT(tmp0, WTS0 + 1 - scale, PCM_OUT_BITS);
+        (PCM_DEC)SATURATE_LEFT_SHIFT(tmp0, WTS0 + 1 - scale, PCM_OUT_BITS);
 #endif
   }
 }
 
 int InvMdctTransformLowDelay_fdk(FIXP_DBL *mdctData, const int mdctData_e,
-                                 FIXP_PCM *output, FIXP_DBL *fs_buffer,
+                                 PCM_DEC *output, FIXP_DBL *fs_buffer,
                                  const int N) {
   const FIXP_WTB *coef;
   FIXP_DBL gain = (FIXP_DBL)0;
diff --git a/libAACdec/src/ldfiltbank.h b/libAACdec/src/ldfiltbank.h
index b63da6b..02971d0 100644
--- a/libAACdec/src/ldfiltbank.h
+++ b/libAACdec/src/ldfiltbank.h
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -104,9 +104,10 @@ amm-info@iis.fraunhofer.de
 #define LDFILTBANK_H
 
 #include "common_fix.h"
+#include "aac_rom.h"
 
 int InvMdctTransformLowDelay_fdk(FIXP_DBL *mdctdata_m, const int mdctdata_e,
-                                 FIXP_PCM *mdctOut, FIXP_DBL *fs_buffer,
+                                 PCM_DEC *mdctOut, FIXP_DBL *fs_buffer,
                                  const int frameLength);
 
 #endif
diff --git a/libAACdec/src/usacdec_lpd.cpp b/libAACdec/src/usacdec_lpd.cpp
index de0c2de..fbf6fab 100644
--- a/libAACdec/src/usacdec_lpd.cpp
+++ b/libAACdec/src/usacdec_lpd.cpp
@@ -122,18 +122,21 @@ amm-info@iis.fraunhofer.de
 
 #include "ac_arith_coder.h"
 
-void filtLP(const FIXP_DBL *syn, FIXP_PCM *syn_out, FIXP_DBL *noise,
-            const FIXP_SGL *filt, INT stop, int len) {
+void filtLP(const FIXP_DBL *syn, PCM_DEC *syn_out, FIXP_DBL *noise,
+            const FIXP_SGL *filt, const INT aacOutDataHeadroom, INT stop,
+            int len) {
   INT i, j;
   FIXP_DBL tmp;
 
+  FDK_ASSERT((aacOutDataHeadroom - 1) >= -(MDCT_OUTPUT_SCALE));
+
   for (i = 0; i < stop; i++) {
     tmp = fMultDiv2(noise[i], filt[0]);  // Filt in Q-1.16
     for (j = 1; j <= len; j++) {
       tmp += fMult((noise[i - j] >> 1) + (noise[i + j] >> 1), filt[j]);
     }
-    syn_out[i] = (FIXP_PCM)(SATURATE_SHIFT(
-        (syn[i] >> 1) - (tmp >> 1), (MDCT_OUTPUT_SCALE - 1), PCM_OUT_BITS));
+    syn_out[i] = (PCM_DEC)(
+        IMDCT_SCALE((syn[i] >> 1) - (tmp >> 1), aacOutDataHeadroom - 1));
   }
 }
 
@@ -143,8 +146,10 @@ void bass_pf_1sf_delay(
     FIXP_DBL *pit_gain,
     const int frame_length, /* (i) : frame length (should be 768|1024) */
     const INT l_frame,
-    const INT l_next, /* (i) : look ahead for symmetric filtering           */
-    FIXP_PCM *synth_out, /* (o) : filtered synthesis (with delay of 1 subfr) */
+    const INT l_next,   /* (i) : look ahead for symmetric filtering           */
+    PCM_DEC *synth_out, /* (o) : filtered synthesis (with delay of 1 subfr)   */
+    const INT aacOutDataHeadroom, /* (i) : headroom of the output time signal to
+                                     prevent clipping */
     FIXP_DBL mem_bpf[]) /* i/o : memory state [L_FILT+L_SUBFR]                */
 {
   INT i, sf, i_subfr, T, T2, lg;
@@ -370,7 +375,7 @@ void bass_pf_1sf_delay(
 
     {
       filtLP(&syn[i_subfr - L_SUBFR], &synth_out[i_subfr], noise,
-             fdk_dec_filt_lp, L_SUBFR, L_FILT);
+             fdk_dec_filt_lp, aacOutDataHeadroom, L_SUBFR, L_FILT);
     }
   }
 
@@ -383,9 +388,9 @@ void bass_pf_1sf_delay(
     /* Output scaling of the BPF memory */
     scaleValues(mem_bpf, (L_FILT + L_SUBFR), -1);
     /* Copy the rest of the signal (after the fac) */
-    scaleValuesSaturate((FIXP_PCM *)&synth_out[l_frame],
-                        (FIXP_DBL *)&syn[l_frame - L_SUBFR],
-                        (frame_length - l_frame), MDCT_OUT_HEADROOM);
+    scaleValuesSaturate(
+        (PCM_DEC *)&synth_out[l_frame], (FIXP_DBL *)&syn[l_frame - L_SUBFR],
+        (frame_length - l_frame), MDCT_OUT_HEADROOM - aacOutDataHeadroom);
   }
 
   return;
@@ -1552,9 +1557,9 @@ void CLpdChannelStream_Decode(
 
 AAC_DECODER_ERROR CLpd_RenderTimeSignal(
     CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo,
-    CAacDecoderChannelInfo *pAacDecoderChannelInfo, FIXP_PCM *pTimeData,
-    INT lFrame, SamplingRateInfo *pSamplingRateInfo, UINT frameOk, UINT flags,
-    UINT strmFlags) {
+    CAacDecoderChannelInfo *pAacDecoderChannelInfo, PCM_DEC *pTimeData,
+    INT lFrame, SamplingRateInfo *pSamplingRateInfo, UINT frameOk,
+    const INT aacOutDataHeadroom, UINT flags, UINT strmFlags) {
   UCHAR *mod = pAacDecoderChannelInfo->data.usac.mod;
   AAC_DECODER_ERROR error = AAC_DEC_OK;
   int k, i_offset;
@@ -2017,7 +2022,8 @@ AAC_DECODER_ERROR CLpd_RenderTimeSignal(
     {
       bass_pf_1sf_delay(p2_synth, pitch, pit_gain, lFrame, lFrame / facFB,
                         mod[nbDiv - 1] ? (SynDelay - (lDiv / 2)) : SynDelay,
-                        pTimeData, pAacDecoderStaticChannelInfo->mem_bpf);
+                        pTimeData, aacOutDataHeadroom,
+                        pAacDecoderStaticChannelInfo->mem_bpf);
     }
   }
 
diff --git a/libAACdec/src/usacdec_lpd.h b/libAACdec/src/usacdec_lpd.h
index 3e7938d..448dc55 100644
--- a/libAACdec/src/usacdec_lpd.h
+++ b/libAACdec/src/usacdec_lpd.h
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -140,13 +140,14 @@ void CLpdChannelStream_Decode(
  * \param pTimeData pointer to output buffer
  * \param samplesPerFrame amount of output samples
  * \param pSamplingRateInfo holds the sampling rate information
- * \param pWorkBuffer1 pointer to work buffer for temporal data
+ * \param aacOutDataHeadroom headroom of the output time signal to prevent
+ * clipping
  */
 AAC_DECODER_ERROR CLpd_RenderTimeSignal(
     CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo,
-    CAacDecoderChannelInfo *pAacDecoderChannelInfo, FIXP_PCM *pTimeData,
+    CAacDecoderChannelInfo *pAacDecoderChannelInfo, PCM_DEC *pTimeData,
     INT samplesPerFrame, SamplingRateInfo *pSamplingRateInfo, UINT frameOk,
-    UINT flags, UINT strmFlags);
+    const INT aacOutDataHeadroom, UINT flags, UINT strmFlags);
 
 static inline INT CLpd_FAC_getLength(int fNotShortBlock, int fac_length_long) {
   if (fNotShortBlock) {
@@ -156,8 +157,9 @@ static inline INT CLpd_FAC_getLength(int fNotShortBlock, int fac_length_long) {
   }
 }
 
-void filtLP(const FIXP_DBL *syn, FIXP_PCM *syn_out, FIXP_DBL *noise,
-            const FIXP_SGL *filt, INT stop, int len);
+void filtLP(const FIXP_DBL *syn, PCM_DEC *syn_out, FIXP_DBL *noise,
+            const FIXP_SGL *filt, const INT aacOutDataHeadroom, INT stop,
+            int len);
 
 /**
  * \brief perform a low-frequency pitch enhancement on time domain signal
@@ -171,13 +173,14 @@ void filtLP(const FIXP_DBL *syn, FIXP_PCM *syn_out, FIXP_DBL *noise,
  * \param[in] l_frame length of filtering, must be multiple of L_SUBFR
  * \param[in] l_next length of allowed look ahead on syn[i], i < l_frame+l_next
  * \param[out] synth_out pointer to time domain output signal
+ * \param[in] headroom of the output time signal to prevent clipping
  * \param[in,out] mem_bpf pointer to filter memory (L_FILT+L_SUBFR)
  */
 
 void bass_pf_1sf_delay(FIXP_DBL syn[], const INT T_sf[], FIXP_DBL *pit_gain,
                        const int frame_length, const INT l_frame,
-                       const INT l_next, FIXP_PCM *synth_out,
-                       FIXP_DBL mem_bpf[]);
+                       const INT l_next, PCM_DEC *synth_out,
+                       const INT aacOutDataHeadroom, FIXP_DBL mem_bpf[]);
 
 /**
  * \brief random sign generator for FD and TCX noise filling
diff --git a/libFDK/include/FDK_qmf_domain.h b/libFDK/include/FDK_qmf_domain.h
index aa247e9..0e83da3 100644
--- a/libFDK/include/FDK_qmf_domain.h
+++ b/libFDK/include/FDK_qmf_domain.h
@@ -160,8 +160,6 @@ H_ALLOC_MEM(QmfSlotsImag32, FIXP_DBL *)
 H_ALLOC_MEM(QmfOverlapBuffer16, FIXP_DBL)
 H_ALLOC_MEM(QmfOverlapBuffer32, FIXP_DBL)
 
-#define QDOM_PCM INT_PCM
-
 /**
  * Structure to hold the configuration data which is global whithin a QMF domain
  * instance.
@@ -181,9 +179,6 @@ typedef struct {
                   park a channel if only one processing channel is
                   available. */
   UCHAR parkChannel_requested;
-  QDOM_PCM
-  *TDinput; /*!< Pointer to time domain data used as input for the QMF
-               analysis. */
   FIXP_DBL *
       pWorkBuffer[QMF_MAX_WB_SECTIONS]; /*!< Pointerarray to volatile memory. */
   UINT flags; /*!< Flags to be set on all QMF analysis/synthesis filter
diff --git a/libFDK/include/mdct.h b/libFDK/include/mdct.h
index 1382374..e7cf3ad 100644
--- a/libFDK/include/mdct.h
+++ b/libFDK/include/mdct.h
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -106,18 +106,16 @@ amm-info@iis.fraunhofer.de
 #include "common_fix.h"
 
 #define MDCT_OUT_HEADROOM 2 /* Output additional headroom */
-#define PCM_OUT_BITS SAMPLE_BITS
+
+#define PCM_OUT_BITS DFRACT_BITS
 #define PCM_OUT_HEADROOM 8 /* Must have the same values as DMXH_HEADROOM */
 
-#define MDCT_OUTPUT_SCALE (-MDCT_OUT_HEADROOM + (DFRACT_BITS - SAMPLE_BITS))
+#define MDCT_OUTPUT_SCALE (-MDCT_OUT_HEADROOM + (DFRACT_BITS - PCM_OUT_BITS))
 /* Refer to "Output word length" in ISO/IEC 14496-3:2008(E) 23.2.3.6 */
 #define MDCT_OUTPUT_GAIN 16
 
-#if (MDCT_OUTPUT_SCALE >= 0)
-#define IMDCT_SCALE(x) SATURATE_RIGHT_SHIFT(x, MDCT_OUTPUT_SCALE, PCM_OUT_BITS)
-#else
-#define IMDCT_SCALE(x) SATURATE_LEFT_SHIFT(x, -MDCT_OUTPUT_SCALE, PCM_OUT_BITS)
-#endif
+#define IMDCT_SCALE(x, s) \
+  SATURATE_RIGHT_SHIFT((x), ((s) + MDCT_OUTPUT_SCALE), PCM_OUT_BITS)
 #define IMDCT_SCALE_DBL(x) (FIXP_DBL)(x)
 #define IMDCT_SCALE_DBL_LSH1(x) SATURATE_LEFT_SHIFT_ALT((x), 1, DFRACT_BITS)
 
diff --git a/libFDK/src/FDK_core.cpp b/libFDK/src/FDK_core.cpp
index 75ea8a2..2f77179 100644
--- a/libFDK/src/FDK_core.cpp
+++ b/libFDK/src/FDK_core.cpp
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -104,7 +104,7 @@ amm-info@iis.fraunhofer.de
 
 /* FDK tools library info */
 #define FDK_TOOLS_LIB_VL0 3
-#define FDK_TOOLS_LIB_VL1 0
+#define FDK_TOOLS_LIB_VL1 1
 #define FDK_TOOLS_LIB_VL2 0
 #define FDK_TOOLS_LIB_TITLE "FDK Tools"
 #ifdef __ANDROID__
diff --git a/libFDK/src/FDK_qmf_domain.cpp b/libFDK/src/FDK_qmf_domain.cpp
index 71e2ef5..77c5ca2 100644
--- a/libFDK/src/FDK_qmf_domain.cpp
+++ b/libFDK/src/FDK_qmf_domain.cpp
@@ -637,10 +637,10 @@ void FDK_QmfDomain_GetSlot(const HANDLE_FDK_QMF_DOMAIN_IN qd_ch, const int ts,
 
   if (pQmfOutImag == NULL) {
     for (; b < fMin(lsb, stop_band); b++) {
-      pQmfOutReal[b] = scaleValue(real[b], lb_sf);
+      pQmfOutReal[b] = scaleValueSaturate(real[b], lb_sf);
     }
     for (; b < fMin(usb, stop_band); b++) {
-      pQmfOutReal[b] = scaleValue(real[b], hb_sf);
+      pQmfOutReal[b] = scaleValueSaturate(real[b], hb_sf);
     }
     for (; b < stop_band; b++) {
       pQmfOutReal[b] = (FIXP_DBL)0;
@@ -648,12 +648,12 @@ void FDK_QmfDomain_GetSlot(const HANDLE_FDK_QMF_DOMAIN_IN qd_ch, const int ts,
   } else {
     FDK_ASSERT(imag != NULL);
     for (; b < fMin(lsb, stop_band); b++) {
-      pQmfOutReal[b] = scaleValue(real[b], lb_sf);
-      pQmfOutImag[b] = scaleValue(imag[b], lb_sf);
+      pQmfOutReal[b] = scaleValueSaturate(real[b], lb_sf);
+      pQmfOutImag[b] = scaleValueSaturate(imag[b], lb_sf);
     }
     for (; b < fMin(usb, stop_band); b++) {
-      pQmfOutReal[b] = scaleValue(real[b], hb_sf);
-      pQmfOutImag[b] = scaleValue(imag[b], hb_sf);
+      pQmfOutReal[b] = scaleValueSaturate(real[b], hb_sf);
+      pQmfOutImag[b] = scaleValueSaturate(imag[b], hb_sf);
     }
     for (; b < stop_band; b++) {
       pQmfOutReal[b] = (FIXP_DBL)0;
diff --git a/libPCMutils/include/limiter.h b/libPCMutils/include/limiter.h
index fab7226..419e891 100644
--- a/libPCMutils/include/limiter.h
+++ b/libPCMutils/include/limiter.h
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -109,8 +109,6 @@ amm-info@iis.fraunhofer.de
 #define TDL_ATTACK_DEFAULT_MS (15)  /* default attack  time in ms */
 #define TDL_RELEASE_DEFAULT_MS (50) /* default release time in ms */
 
-#define TDL_GAIN_SCALING (15) /* scaling of gain value. */
-
 #ifdef __cplusplus
 extern "C" {
 #endif
@@ -128,10 +126,7 @@ struct TDLimiter {
   unsigned int maxBufIdx, delayBufIdx;
   FIXP_DBL smoothState0;
   FIXP_DBL minGain;
-
-  FIXP_DBL additionalGainPrev;
-  FIXP_DBL additionalGainFilterState;
-  FIXP_DBL additionalGainFilterState1;
+  INT scaling;
 };
 
 typedef enum {
@@ -255,27 +250,16 @@ TDLIMITER_ERROR pcmLimiter_SetThreshold(TDLimiterPtr limiter,
 
 /******************************************************************************
  * pcmLimiter_Apply                                                            *
- * limiter:    limiter handle                                                  *
- * pGain :     pointer to gains to be applied to the signal before limiting,   *
- *             which are downscaled by TDL_GAIN_SCALING bit.                   *
- *             These gains are delayed by gain_delay, and smoothed.            *
- *             Smoothing is done by a butterworth lowpass filter with a cutoff *
- *             frequency which is fixed with respect to the sampling rate.     *
- *             It is a substitute for the smoothing due to windowing and       *
- *             overlap/add, if a gain is applied in frequency domain.          *
- * gain_scale: pointer to scaling exponents to be applied to the signal before *
- *             limiting, without delay and without smoothing                   *
- * gain_size:  number of elements in pGain, currently restricted to 1          *
- * gain_delay: delay [samples] with which the gains in pGain shall be applied  *
- *             gain_delay <= nSamples                                          *
- * samples:    input/output buffer containing interleaved samples              *
- *             precision of output will be DFRACT_BITS-TDL_GAIN_SCALING bits   *
- * nSamples:   number of samples per channel                                   *
+ * limiter:        limiter handle                                              *
+ * samplesIn:      pointer to input buffer containing interleaved samples      *
+ * samplesOut:     pointer to output buffer containing interleaved samples     *
+ * pGainPerSample: pointer to gains for each sample                            *
+ * scaling:        scaling of output samples                                   *
+ * nSamples:       number of samples per channel                               *
  * returns:    error code                                                      *
  ******************************************************************************/
 TDLIMITER_ERROR pcmLimiter_Apply(TDLimiterPtr limiter, PCM_LIM* samplesIn,
-                                 INT_PCM* samplesOut, FIXP_DBL* pGain,
-                                 const INT* gain_scale, const UINT gain_size,
-                                 const UINT gain_delay, const UINT nSamples);
+                                 INT_PCM* samplesOut, FIXP_DBL* pGainPerSample,
+                                 const INT scaling, const UINT nSamples);
 
 #endif /* #ifndef LIMITER_H */
diff --git a/libPCMutils/src/limiter.cpp b/libPCMutils/src/limiter.cpp
index 60e3e3b..598dc0c 100644
--- a/libPCMutils/src/limiter.cpp
+++ b/libPCMutils/src/limiter.cpp
@@ -152,7 +152,7 @@ TDLimiterPtr pcmLimiter_Create(unsigned int maxAttackMs, unsigned int releaseMs,
   limiter->attack = attack;
   limiter->attackConst = attackConst;
   limiter->releaseConst = releaseConst;
-  limiter->threshold = threshold >> TDL_GAIN_SCALING;
+  limiter->threshold = threshold;
   limiter->channels = maxChannels;
   limiter->maxChannels = maxChannels;
   limiter->sampleRate = maxSampleRate;
@@ -165,18 +165,13 @@ TDLimiterPtr pcmLimiter_Create(unsigned int maxAttackMs, unsigned int releaseMs,
 
 /* apply limiter */
 TDLIMITER_ERROR pcmLimiter_Apply(TDLimiterPtr limiter, PCM_LIM* samplesIn,
-                                 INT_PCM* samplesOut, FIXP_DBL* RESTRICT pGain,
-                                 const INT* RESTRICT gain_scale,
-                                 const UINT gain_size, const UINT gain_delay,
-                                 const UINT nSamples) {
+                                 INT_PCM* samplesOut, FIXP_DBL* pGainPerSample,
+                                 const INT scaling, const UINT nSamples) {
   unsigned int i, j;
-  FIXP_DBL tmp1;
   FIXP_DBL tmp2;
-  FIXP_DBL tmp, old, gain, additionalGain = 0, additionalGainUnfiltered;
+  FIXP_DBL tmp, old, gain, additionalGain = 0;
   FIXP_DBL minGain = FL2FXCONST_DBL(1.0f / (1 << 1));
-
-  FDK_ASSERT(gain_size == 1);
-  FDK_ASSERT(gain_delay <= nSamples);
+  UINT additionalGainAvailable = 1;
 
   if (limiter == NULL) return TDLIMIT_INVALID_HANDLE;
 
@@ -185,7 +180,7 @@ TDLIMITER_ERROR pcmLimiter_Apply(TDLimiterPtr limiter, PCM_LIM* samplesIn,
     unsigned int attack = limiter->attack;
     FIXP_DBL attackConst = limiter->attackConst;
     FIXP_DBL releaseConst = limiter->releaseConst;
-    FIXP_DBL threshold = limiter->threshold;
+    FIXP_DBL threshold = limiter->threshold >> scaling;
 
     FIXP_DBL max = limiter->max;
     FIXP_DBL* maxBuf = limiter->maxBuf;
@@ -195,55 +190,34 @@ TDLIMITER_ERROR pcmLimiter_Apply(TDLimiterPtr limiter, PCM_LIM* samplesIn,
     unsigned int delayBufIdx = limiter->delayBufIdx;
 
     FIXP_DBL smoothState0 = limiter->smoothState0;
-    FIXP_DBL additionalGainSmoothState = limiter->additionalGainFilterState;
-    FIXP_DBL additionalGainSmoothState1 = limiter->additionalGainFilterState1;
 
-    if (!gain_delay) {
-      additionalGain = pGain[0];
-      if (gain_scale[0] > 0) {
-        additionalGain <<= gain_scale[0];
-      } else {
-        additionalGain >>= -gain_scale[0];
-      }
+    if (limiter->scaling != scaling) {
+      scaleValuesSaturate(delayBuf, attack * channels,
+                          limiter->scaling - scaling);
+      scaleValuesSaturate(maxBuf, attack + 1, limiter->scaling - scaling);
+      max = scaleValueSaturate(max, limiter->scaling - scaling);
+      limiter->scaling = scaling;
     }
 
-    for (i = 0; i < nSamples; i++) {
-      if (gain_delay) {
-        if (i < gain_delay) {
-          additionalGainUnfiltered = limiter->additionalGainPrev;
-        } else {
-          additionalGainUnfiltered = pGain[0];
-        }
+    if (pGainPerSample == NULL) {
+      additionalGainAvailable = 0;
+    }
 
-        /* Smooth additionalGain */
-        /* [b,a] = butter(1, 0.01) */
-        static const FIXP_SGL b[] = {FL2FXCONST_SGL(0.015466 * 2.0),
-                                     FL2FXCONST_SGL(0.015466 * 2.0)};
-        static const FIXP_SGL a[] = {(FIXP_SGL)MAXVAL_SGL,
-                                     FL2FXCONST_SGL(-0.96907)};
-        additionalGain = -fMult(additionalGainSmoothState, a[1]) +
-                         fMultDiv2(additionalGainUnfiltered, b[0]) +
-                         fMultDiv2(additionalGainSmoothState1, b[1]);
-        additionalGainSmoothState1 = additionalGainUnfiltered;
-        additionalGainSmoothState = additionalGain;
-
-        /* Apply the additional scaling that has no delay and no smoothing */
-        if (gain_scale[0] > 0) {
-          additionalGain <<= gain_scale[0];
-        } else {
-          additionalGain >>= -gain_scale[0];
-        }
-      }
+    for (i = 0; i < nSamples; i++) {
       /* get maximum absolute sample value of all channels, including the
        * additional gain. */
-      tmp1 = (FIXP_DBL)0;
+      tmp = (FIXP_DBL)0;
       for (j = 0; j < channels; j++) {
         tmp2 = PCM_LIM2FIXP_DBL(samplesIn[j]);
         tmp2 =
             (tmp2 == (FIXP_DBL)MINVAL_DBL) ? (FIXP_DBL)MAXVAL_DBL : fAbs(tmp2);
-        tmp1 = fMax(tmp1, tmp2);
+        tmp = fMax(tmp, tmp2);
+      }
+
+      if (additionalGainAvailable) {
+        additionalGain = pGainPerSample[i];
+        tmp = fMult(tmp, additionalGain);
       }
-      tmp = fMult(tmp1, additionalGain);
 
       /* set threshold as lower border to save calculations in running maximum
        * algorithm */
@@ -314,22 +288,42 @@ TDLIMITER_ERROR pcmLimiter_Apply(TDLimiterPtr limiter, PCM_LIM* samplesIn,
         /* lookahead delay, apply gain */
         for (j = 0; j < channels; j++) {
           tmp = p_delayBuf[j];
-          p_delayBuf[j] = fMult((FIXP_PCM_LIM)samplesIn[j], additionalGain);
+
+          if (additionalGainAvailable) {
+            p_delayBuf[j] = fMult((FIXP_PCM_LIM)samplesIn[j], additionalGain);
+          } else {
+            p_delayBuf[j] = PCM_LIM2FIXP_DBL(samplesIn[j]);
+          }
 
           /* Apply gain to delayed signal */
           tmp = fMultDiv2(tmp, gain);
-
+#if (SAMPLE_BITS == DFRACT_BITS)
+          samplesOut[j] = (INT_PCM)FX_DBL2FX_PCM(
+              (FIXP_DBL)SATURATE_LEFT_SHIFT(tmp, scaling + 1, DFRACT_BITS));
+#else
           samplesOut[j] = (INT_PCM)FX_DBL2FX_PCM((FIXP_DBL)SATURATE_LEFT_SHIFT(
-              tmp, TDL_GAIN_SCALING + 1, DFRACT_BITS));
+              tmp + ((FIXP_DBL)0x8000 >> (scaling + 1)), scaling + 1,
+              DFRACT_BITS));
+#endif
         }
         gain >>= 1;
       } else {
         /* lookahead delay, apply gain=1.0f */
         for (j = 0; j < channels; j++) {
           tmp = p_delayBuf[j];
-          p_delayBuf[j] = fMult((FIXP_PCM_LIM)samplesIn[j], additionalGain);
+          if (additionalGainAvailable) {
+            p_delayBuf[j] = fMult((FIXP_PCM_LIM)samplesIn[j], additionalGain);
+          } else {
+            p_delayBuf[j] = PCM_LIM2FIXP_DBL(samplesIn[j]);
+          }
+
+#if (SAMPLE_BITS == DFRACT_BITS)
+          samplesOut[j] = (INT_PCM)FX_DBL2FX_PCM(
+              (FIXP_DBL)SATURATE_LEFT_SHIFT(tmp, scaling, DFRACT_BITS));
+#else
           samplesOut[j] = (INT_PCM)FX_DBL2FX_PCM((FIXP_DBL)SATURATE_LEFT_SHIFT(
-              tmp, TDL_GAIN_SCALING, DFRACT_BITS));
+              tmp + ((FIXP_DBL)0x8000 >> scaling), scaling, DFRACT_BITS));
+#endif
         }
       }
 
@@ -354,13 +348,9 @@ TDLIMITER_ERROR pcmLimiter_Apply(TDLimiterPtr limiter, PCM_LIM* samplesIn,
     limiter->delayBufIdx = delayBufIdx;
 
     limiter->smoothState0 = smoothState0;
-    limiter->additionalGainFilterState = additionalGainSmoothState;
-    limiter->additionalGainFilterState1 = additionalGainSmoothState1;
 
     limiter->minGain = minGain;
 
-    limiter->additionalGainPrev = pGain[0];
-
     return TDLIMIT_OK;
   }
 }
@@ -370,7 +360,7 @@ TDLIMITER_ERROR pcmLimiter_SetThreshold(TDLimiterPtr limiter,
                                         FIXP_DBL threshold) {
   if (limiter == NULL) return TDLIMIT_INVALID_HANDLE;
 
-  limiter->threshold = threshold >> TDL_GAIN_SCALING;
+  limiter->threshold = threshold;
 
   return TDLIMIT_OK;
 }
@@ -384,13 +374,7 @@ TDLIMITER_ERROR pcmLimiter_Reset(TDLimiterPtr limiter) {
     limiter->cor = FL2FXCONST_DBL(1.0f / (1 << 1));
     limiter->smoothState0 = FL2FXCONST_DBL(1.0f / (1 << 1));
     limiter->minGain = FL2FXCONST_DBL(1.0f / (1 << 1));
-
-    limiter->additionalGainPrev =
-        FL2FXCONST_DBL(1.0f / (1 << TDL_GAIN_SCALING));
-    limiter->additionalGainFilterState =
-        FL2FXCONST_DBL(1.0f / (1 << TDL_GAIN_SCALING));
-    limiter->additionalGainFilterState1 =
-        FL2FXCONST_DBL(1.0f / (1 << TDL_GAIN_SCALING));
+    limiter->scaling = 0;
 
     FDKmemset(limiter->maxBuf, 0, (limiter->attack + 1) * sizeof(FIXP_DBL));
     FDKmemset(limiter->delayBuf, 0,
diff --git a/libPCMutils/src/version.h b/libPCMutils/src/version.h
index fa31af1..05371f8 100644
--- a/libPCMutils/src/version.h
+++ b/libPCMutils/src/version.h
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -105,7 +105,7 @@ amm-info@iis.fraunhofer.de
 
 /* library info */
 #define PCMUTIL_LIB_VL0 3
-#define PCMUTIL_LIB_VL1 0
+#define PCMUTIL_LIB_VL1 1
 #define PCMUTIL_LIB_VL2 0
 #define PCMUTIL_LIB_TITLE "PCM Utility Lib"
 #ifdef __ANDROID__
diff --git a/libSACdec/include/sac_dec_lib.h b/libSACdec/include/sac_dec_lib.h
index 9913279..1827504 100644
--- a/libSACdec/include/sac_dec_lib.h
+++ b/libSACdec/include/sac_dec_lib.h
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -252,7 +252,7 @@ typedef enum {
                 scenario. Default parameter value is 3 frames. */
 } SACDEC_PARAM;
 
-#define PCM_MPS INT_PCM
+#define PCM_MPS LONG
 
 /**
  * \brief MPEG Surround decoder handle.
@@ -401,17 +401,22 @@ int mpegSurroundDecoder_Parse(CMpegSurroundDecoder *pMpegSurroundDecoder,
  * for each output audio channel is stored into.
  * \param mapDescr             Channep map descriptor for output channel mapping
  * to be used (From MPEG PCE ordering to whatever is required).
+ * \param inDataHeadroom       Headroom of SAC input time signal to prevent
+ * clipping.
+ * \param outDataHeadroom      Pointer to headroom of SAC output time signal to
+ * prevent clipping.
  *
  * \return  Error code.
  */
 int mpegSurroundDecoder_Apply(CMpegSurroundDecoder *pMpegSurroundDecoder,
-                              INT_PCM *input, PCM_MPS *pTimeData,
+                              PCM_MPS *input, PCM_MPS *pTimeData,
                               const int timeDataSize, int timeDataFrameSize,
                               int *nChannels, int *frameSize, int sampleRate,
                               AUDIO_OBJECT_TYPE coreCodec,
                               AUDIO_CHANNEL_TYPE channelType[],
                               UCHAR channelIndices[],
-                              const FDK_channelMapDescr *const mapDescr);
+                              const FDK_channelMapDescr *const mapDescr,
+                              const INT inDataHeadroom, INT *outDataHeadroom);
 
 /**
  * \brief                       Deallocate a MPEG Surround decoder instance.
diff --git a/libSACdec/src/sac_dec.cpp b/libSACdec/src/sac_dec.cpp
index c1832f1..a7b50df 100644
--- a/libSACdec/src/sac_dec.cpp
+++ b/libSACdec/src/sac_dec.cpp
@@ -766,7 +766,7 @@ SACDEC_ERROR FDK_SpatialDecInit(spatialDec *self, SPATIAL_BS_FRAME *frame,
 
     /* output scaling */
     for (nCh = 0; nCh < self->numOutputChannelsAT; nCh++) {
-      int outputScale = 0, outputGain_e = 0, scale = 0;
+      int outputScale = 0, outputGain_e = 0, scale = -(8) + (1);
       FIXP_DBL outputGain_m = getChGain(self, nCh, &outputGain_e);
 
       if (!isTwoChMode(self->upmixType) && !bypassMode) {
@@ -775,7 +775,7 @@ SACDEC_ERROR FDK_SpatialDecInit(spatialDec *self, SPATIAL_BS_FRAME *frame,
                                              synthesis qmf */
       }
 
-      scale = outputScale;
+      scale += outputScale;
 
       qmfChangeOutScalefactor(&self->pQmfDomain->QmfDomainOut[nCh].fb, scale);
       qmfChangeOutGain(&self->pQmfDomain->QmfDomainOut[nCh].fb, outputGain_m,
@@ -1223,18 +1223,24 @@ static SACDEC_ERROR SpatialDecApplyParameterSets(
               !(self->stereoConfigIndex == 3)) {
             for (i = 0; i < self->qmfBands; i++) {
               self_qmfResidualReal__FDK_0_0[i] =
-                  fMult(self_qmfResidualReal__FDK_0_0[i] << 1,
+                  fMult(scaleValueSaturate(self_qmfResidualReal__FDK_0_0[i],
+                                           1 + self->sacInDataHeadroom - (1)),
                         self->clipProtectGain__FDK);
               self_qmfResidualImag__FDK_0_0[i] =
-                  fMult(self_qmfResidualImag__FDK_0_0[i] << 1,
+                  fMult(scaleValueSaturate(self_qmfResidualImag__FDK_0_0[i],
+                                           1 + self->sacInDataHeadroom - (1)),
                         self->clipProtectGain__FDK);
             }
           } else {
             for (i = 0; i < self->qmfBands; i++) {
-              self_qmfResidualReal__FDK_0_0[i] = fMult(
-                  self_qmfResidualReal__FDK_0_0[i], self->clipProtectGain__FDK);
-              self_qmfResidualImag__FDK_0_0[i] = fMult(
-                  self_qmfResidualImag__FDK_0_0[i], self->clipProtectGain__FDK);
+              self_qmfResidualReal__FDK_0_0[i] =
+                  fMult(scaleValueSaturate(self_qmfResidualReal__FDK_0_0[i],
+                                           self->sacInDataHeadroom - (1)),
+                        self->clipProtectGain__FDK);
+              self_qmfResidualImag__FDK_0_0[i] =
+                  fMult(scaleValueSaturate(self_qmfResidualImag__FDK_0_0[i],
+                                           self->sacInDataHeadroom - (1)),
+                        self->clipProtectGain__FDK);
             }
           }
         }
@@ -1416,6 +1422,7 @@ SACDEC_ERROR SpatialDecApplyFrame(
   FDK_ASSERT(self != NULL);
   FDK_ASSERT(pControlFlags != NULL);
   FDK_ASSERT(pcmOutBuf != NULL);
+  FDK_ASSERT(self->sacInDataHeadroom >= (1));
 
   self->errInt = err; /* Init internal error */
 
diff --git a/libSACdec/src/sac_dec.h b/libSACdec/src/sac_dec.h
index 992acad..1c3df71 100644
--- a/libSACdec/src/sac_dec.h
+++ b/libSACdec/src/sac_dec.h
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -523,6 +523,9 @@ struct spatialDec_struct {
                              new frame after SSC change (aka
                              decodeAfterConfigHasChangedFlag). */
   SpatialDecConcealmentInfo concealInfo;
+
+  INT sacInDataHeadroom; /* Headroom of the SAC input time signal to prevent
+                            clipping */
 };
 
 #define SACDEC_SYNTAX_MPS 1
diff --git a/libSACdec/src/sac_dec_interface.h b/libSACdec/src/sac_dec_interface.h
index a2eea92..05a9a75 100644
--- a/libSACdec/src/sac_dec_interface.h
+++ b/libSACdec/src/sac_dec_interface.h
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -249,10 +249,10 @@ typedef struct {
 
 } MEM_REQUIREMENTS;
 
-#define PCM_MPS INT_PCM
-#define PCM_MPSF FIXP_PCM
+#define PCM_MPS LONG
+#define PCM_MPSF FIXP_DBL
 
-#define FIXP_DBL2PCM_MPS(x) ((INT_PCM)FX_DBL2FX_PCM(x))
+#define FIXP_DBL2PCM_MPS(x) ((LONG)(x))
 
 /* exposed functions (library interface) */
 
diff --git a/libSACdec/src/sac_dec_lib.cpp b/libSACdec/src/sac_dec_lib.cpp
index bf6dedf..856a923 100644
--- a/libSACdec/src/sac_dec_lib.cpp
+++ b/libSACdec/src/sac_dec_lib.cpp
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -1507,15 +1507,17 @@ bail:
 }
 
 int mpegSurroundDecoder_Apply(CMpegSurroundDecoder *pMpegSurroundDecoder,
-                              INT_PCM *input, PCM_MPS *pTimeData,
+                              PCM_MPS *input, PCM_MPS *pTimeData,
                               const int timeDataSize, int timeDataFrameSize,
                               int *nChannels, int *frameSize, int sampleRate,
                               AUDIO_OBJECT_TYPE coreCodec,
                               AUDIO_CHANNEL_TYPE channelType[],
                               UCHAR channelIndices[],
-                              const FDK_channelMapDescr *const mapDescr) {
+                              const FDK_channelMapDescr *const mapDescr,
+                              const INT inDataHeadroom, INT *outDataHeadroom) {
   SACDEC_ERROR err = MPS_OK;
   PCM_MPS *pTimeOut = pTimeData;
+  PCM_MPS *TDinput = NULL;
   UINT initControlFlags = 0, controlFlags = 0;
   int timeDataRequiredSize = 0;
   int newData;
@@ -1534,6 +1536,9 @@ int mpegSurroundDecoder_Apply(CMpegSurroundDecoder *pMpegSurroundDecoder,
     return MPS_NOTOK;
   }
 
+  pMpegSurroundDecoder->pSpatialDec->sacInDataHeadroom = inDataHeadroom;
+  *outDataHeadroom = (INT)(8);
+
   pMpegSurroundDecoder->pSpatialDec->pConfigCurrent =
       &pMpegSurroundDecoder
            ->spatialSpecificConfig[pMpegSurroundDecoder->bsFrameDecode];
@@ -1682,8 +1687,7 @@ int mpegSurroundDecoder_Apply(CMpegSurroundDecoder *pMpegSurroundDecoder,
         (timeDataFrameSize *
          pMpegSurroundDecoder->pQmfDomain->globalConf.nBandsSynthesis) /
         pMpegSurroundDecoder->pQmfDomain->globalConf.nBandsAnalysis;
-    pMpegSurroundDecoder->pQmfDomain->globalConf.TDinput =
-        pTimeData + timeDataFrameSizeOut - timeDataFrameSize;
+    TDinput = pTimeData + timeDataFrameSizeOut - timeDataFrameSize;
     for (int i = *nChannels - 1; i >= 0; i--) {
       FDKmemmove(pTimeData + (i + 1) * timeDataFrameSizeOut - timeDataFrameSize,
                  pTimeData + timeDataFrameSize * i,
@@ -1694,8 +1698,8 @@ int mpegSurroundDecoder_Apply(CMpegSurroundDecoder *pMpegSurroundDecoder,
   } else {
     if (pMpegSurroundDecoder->mpegSurroundUseTimeInterface) {
       FDKmemcpy(input, pTimeData,
-                sizeof(INT_PCM) * (*nChannels) * (*frameSize));
-      pMpegSurroundDecoder->pQmfDomain->globalConf.TDinput = input;
+                sizeof(PCM_MPS) * (*nChannels) * (*frameSize));
+      TDinput = input;
     }
   }
 
@@ -1707,8 +1711,8 @@ int mpegSurroundDecoder_Apply(CMpegSurroundDecoder *pMpegSurroundDecoder,
       &pMpegSurroundDecoder->bsFrames[pMpegSurroundDecoder->bsFrameDecode],
       pMpegSurroundDecoder->mpegSurroundUseTimeInterface ? INPUTMODE_TIME
                                                          : INPUTMODE_QMF_SBR,
-      pMpegSurroundDecoder->pQmfDomain->globalConf.TDinput, NULL, NULL,
-      pTimeOut, *frameSize, &controlFlags, *nChannels, mapDescr);
+      TDinput, NULL, NULL, pTimeOut, *frameSize, &controlFlags, *nChannels,
+      mapDescr);
   *nChannels = pMpegSurroundDecoder->pSpatialDec->numOutputChannelsAT;
 
   if (err !=
@@ -1781,7 +1785,7 @@ void mpegSurroundDecoder_Close(CMpegSurroundDecoder *pMpegSurroundDecoder) {
 }
 
 #define SACDEC_VL0 2
-#define SACDEC_VL1 0
+#define SACDEC_VL1 1
 #define SACDEC_VL2 0
 
 int mpegSurroundDecoder_GetLibInfo(LIB_INFO *info) {
diff --git a/libSACdec/src/sac_process.cpp b/libSACdec/src/sac_process.cpp
index 95128f3..22091a9 100644
--- a/libSACdec/src/sac_process.cpp
+++ b/libSACdec/src/sac_process.cpp
@@ -187,8 +187,12 @@ SACDEC_ERROR SpatialDecQMFAnalysis(spatialDec *self, const PCM_MPS *inData,
       if (!isTwoChMode(self->upmixType) && !bypassMode) {
         int i;
         for (i = 0; i < self->qmfBands; i++) {
-          qmfReal[ch][i] = fMult(qmfReal[ch][i], self->clipProtectGain__FDK);
-          qmfImag[ch][i] = fMult(qmfImag[ch][i], self->clipProtectGain__FDK);
+          qmfReal[ch][i] = fMult(
+              scaleValueSaturate(qmfReal[ch][i], self->sacInDataHeadroom - (1)),
+              self->clipProtectGain__FDK);
+          qmfImag[ch][i] = fMult(
+              scaleValueSaturate(qmfImag[ch][i], self->sacInDataHeadroom - (1)),
+              self->clipProtectGain__FDK);
         }
       }
     }
@@ -216,16 +220,17 @@ SACDEC_ERROR SpatialDecFeedQMF(spatialDec *self, FIXP_DBL **qmfInDataReal,
 
       /* Write Input data to pQmfRealAnalysis. */
       if (self->bShareDelayWithSBR) {
-        FDK_QmfDomain_GetSlot(
-            &self->pQmfDomain->QmfDomainIn[ch], ts + HYBRID_FILTER_DELAY, 0,
-            MAX_QMF_BANDS_TO_HYBRID, pQmfRealAnalysis, pQmfImagAnalysis, 15);
+        FDK_QmfDomain_GetSlot(&self->pQmfDomain->QmfDomainIn[ch],
+                              ts + HYBRID_FILTER_DELAY, 0,
+                              MAX_QMF_BANDS_TO_HYBRID, pQmfRealAnalysis,
+                              pQmfImagAnalysis, 15 + (1));
         FDK_QmfDomain_GetSlot(&self->pQmfDomain->QmfDomainIn[ch], ts,
                               MAX_QMF_BANDS_TO_HYBRID, self->qmfBands,
-                              pQmfRealAnalysis, pQmfImagAnalysis, 15);
+                              pQmfRealAnalysis, pQmfImagAnalysis, 15 + (1));
       } else {
         FDK_QmfDomain_GetSlot(&self->pQmfDomain->QmfDomainIn[ch], ts, 0,
                               self->qmfBands, pQmfRealAnalysis,
-                              pQmfImagAnalysis, 15);
+                              pQmfImagAnalysis, 15 + (1));
       }
       if (ts == self->pQmfDomain->globalConf.nQmfTimeSlots - 1) {
         /* Is currently also needed in case we dont have any overlap. We need to
@@ -501,8 +506,8 @@ SACDEC_ERROR SpatialDecApplyM2_Mode212_ResidualsPlusPhaseCoding(
     for (pb = 0, qs = 3; pb < 2; pb++) {
       INT s;
       FIXP_DBL maxVal;
-      FIXP_SGL mReal1;
-      FIXP_SGL mReal0, mImag0;
+      FIXP_DBL mReal1;
+      FIXP_DBL mReal0, mImag0;
       FIXP_DBL iReal0, iImag0, iReal1;
 
       iReal0 = interpolateParameter(alpha, MReal0[pb], MRealPrev0[pb]);
@@ -515,9 +520,9 @@ SACDEC_ERROR SpatialDecApplyM2_Mode212_ResidualsPlusPhaseCoding(
       s = fMax(CntLeadingZeros(maxVal) - 1, 0);
       s = fMin(s, scale_param_m2);
 
-      mReal0 = FX_DBL2FX_SGL(iReal0 << s);
-      mImag0 = FX_DBL2FX_SGL(iImag0 << s);
-      mReal1 = FX_DBL2FX_SGL(iReal1 << s);
+      mReal0 = iReal0 << s;
+      mImag0 = iImag0 << s;
+      mReal1 = iReal1 << s;
 
       s = scale_param_m2 - s;
 
@@ -934,6 +939,7 @@ SACDEC_ERROR SpatialDecSynthesis(spatialDec *self, const INT ts,
         self->pQmfDomain->QmfDomainIn[outCh].scaling.lb_scale -=
             self->clipProtectGainSF__FDK;
 
+        self->pQmfDomain->QmfDomainIn[outCh].scaling.lb_scale -= (1);
       } else {
         /* Call the QMF synthesis for dry. */
         err = CalculateSpaceSynthesisQmf(&self->pQmfDomain->QmfDomainOut[outCh],
diff --git a/libSACdec/src/sac_qmf.cpp b/libSACdec/src/sac_qmf.cpp
index a075490..fd7599d 100644
--- a/libSACdec/src/sac_qmf.cpp
+++ b/libSACdec/src/sac_qmf.cpp
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -110,7 +110,7 @@ amm-info@iis.fraunhofer.de
 
 SACDEC_ERROR CalculateSpaceSynthesisQmf(
     const HANDLE_FDK_QMF_DOMAIN_OUT hQmfDomainOutCh, const FIXP_DBL *Sr,
-    const FIXP_DBL *Si, const INT stride, INT_PCM *timeSig) {
+    const FIXP_DBL *Si, const INT stride, PCM_MPS *timeSig) {
   SACDEC_ERROR err = MPS_OK;
 
   if (hQmfDomainOutCh == NULL) {
diff --git a/libSACdec/src/sac_qmf.h b/libSACdec/src/sac_qmf.h
index d1dc837..5cd573e 100644
--- a/libSACdec/src/sac_qmf.h
+++ b/libSACdec/src/sac_qmf.h
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -124,7 +124,7 @@ amm-info@iis.fraunhofer.de
  */
 SACDEC_ERROR CalculateSpaceSynthesisQmf(
     const HANDLE_FDK_QMF_DOMAIN_OUT hQmfDomainOutCh, const FIXP_DBL *Sr,
-    const FIXP_DBL *Si, const INT stride, INT_PCM *timeSig);
+    const FIXP_DBL *Si, const INT stride, PCM_MPS *timeSig);
 
 /**
  * \brief  Convert audio input data to qmf representation.
diff --git a/libSACdec/src/sac_reshapeBBEnv.cpp b/libSACdec/src/sac_reshapeBBEnv.cpp
index b44ce4e..272d009 100644
--- a/libSACdec/src/sac_reshapeBBEnv.cpp
+++ b/libSACdec/src/sac_reshapeBBEnv.cpp
@@ -253,17 +253,17 @@ static inline void slotAmp(FIXP_DBL *RESTRICT slotAmp_dry,
 
   dry = wet = FL2FXCONST_DBL(0.0f);
   for (qs = 0; qs < cplxBands; qs++) {
-    dry = fAddSaturate(dry, fPow2Div2(pHybOutputRealDry[qs]) +
-                                fPow2Div2(pHybOutputImagDry[qs]));
-    wet = fAddSaturate(wet, fPow2Div2(pHybOutputRealWet[qs]) +
-                                fPow2Div2(pHybOutputImagWet[qs]));
+    dry = fAddSaturate(dry, fPow2Div2(pHybOutputRealDry[qs] << (1)) +
+                                fPow2Div2(pHybOutputImagDry[qs] << (1)));
+    wet = fAddSaturate(wet, fPow2Div2(pHybOutputRealWet[qs] << (1)) +
+                                fPow2Div2(pHybOutputImagWet[qs] << (1)));
   }
   for (; qs < hybBands; qs++) {
-    dry = fAddSaturate(dry, fPow2Div2(pHybOutputRealDry[qs]));
-    wet = fAddSaturate(wet, fPow2Div2(pHybOutputRealWet[qs]));
+    dry = fAddSaturate(dry, fPow2Div2(pHybOutputRealDry[qs] << (1)));
+    wet = fAddSaturate(wet, fPow2Div2(pHybOutputRealWet[qs] << (1)));
   }
-  *slotAmp_dry = dry;
-  *slotAmp_wet = wet;
+  *slotAmp_dry = dry >> (2 * (1));
+  *slotAmp_wet = wet >> (2 * (1));
 }
 
 #if defined(__aarch64__)
@@ -327,7 +327,7 @@ static void extractBBEnv(spatialDec *self, INT inp, INT start, INT channels,
 
   INT shapeActiv = 1;
   INT hybBands = fixMin(42, self->hybridBands);
-  INT staticScale = self->staticDecScale;
+  INT staticScale = self->staticDecScale + (1);
   INT cplxBands;
   cplxBands = fixMin(42, self->hybridBands);
 
diff --git a/libSACdec/src/sac_rom.h b/libSACdec/src/sac_rom.h
index 38f17f2..d317856 100644
--- a/libSACdec/src/sac_rom.h
+++ b/libSACdec/src/sac_rom.h
@@ -111,21 +111,12 @@ amm-info@iis.fraunhofer.de
 #include "machine_type.h"
 
 /* Global ROM table data type: */
-#ifndef ARCH_PREFER_MULT_32x32
-#define FIXP_CFG FIXP_SGL
-#define FX_CFG2FX_DBL FX_SGL2FX_DBL
-#define FX_CFG2FX_SGL
-#define CFG(a) (FX_DBL2FXCONST_SGL(a))
-#define FL2FXCONST_CFG FL2FXCONST_SGL
-#define FX_DBL2FX_CFG(x) FX_DBL2FX_SGL((FIXP_DBL)(x))
-#else
 #define FIXP_CFG FIXP_DBL
 #define FX_CFG2FX_DBL
 #define FX_CFG2FX_SGL FX_DBL2FX_SGL
 #define CFG(a) FIXP_DBL(a)
 #define FL2FXCONST_CFG FL2FXCONST_DBL
 #define FX_DBL2FX_CFG(x) ((FIXP_DBL)(x))
-#endif
 
 /* others  */
 #define SCALE_INV_ICC (2)
@@ -133,15 +124,9 @@ amm-info@iis.fraunhofer.de
 
 #define QCC_SCALE 1
 #define M1M2_DATA FIXP_DBL
-#ifndef ARCH_PREFER_MULT_32x32
-#define M1M2_CDATA FIXP_SGL
-#define M1M2_CDATA2FX_DBL(a) FX_SGL2FX_DBL(a)
-#define FX_DBL2M1M2_CDATA(a) FX_DBL2FX_SGL(a)
-#else
 #define M1M2_CDATA FIXP_DBL
 #define M1M2_CDATA2FX_DBL(a) (a)
 #define FX_DBL2M1M2_CDATA(a) (a)
-#endif
 
 #define CLIP_PROTECT_GAIN_0(x) FL2FXCONST_CFG(((x) / (float)(1 << 0)))
 #define CLIP_PROTECT_GAIN_1(x) FL2FXCONST_CFG(((x) / (float)(1 << 1)))
diff --git a/libSBRdec/include/sbrdecoder.h b/libSBRdec/include/sbrdecoder.h
index cc55572..c09c985 100644
--- a/libSBRdec/include/sbrdecoder.h
+++ b/libSBRdec/include/sbrdecoder.h
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -361,15 +361,20 @@ SBR_ERROR sbrDecoder_Parse(HANDLE_SBRDECODER self, HANDLE_FDK_BITSTREAM hBs,
  * error (0: core decoder found errors, 1: no errors).
  * \param psDecoded       Pointer to a buffer holding a flag. Input: PS is
  * possible, Output: PS has been rendered.
+ * \param inDataHeadroom  Headroom of the SBR input time signal to prevent
+ * clipping.
+ * \param outDataHeadroom Pointer to headroom of the SBR output time signal to
+ * prevent clipping.
  *
  * \return  Error code.
  */
-SBR_ERROR sbrDecoder_Apply(HANDLE_SBRDECODER self, INT_PCM *input,
-                           INT_PCM *timeData, const int timeDataSize,
-                           int *numChannels, int *sampleRate,
+SBR_ERROR sbrDecoder_Apply(HANDLE_SBRDECODER self, LONG *input, LONG *timeData,
+                           const int timeDataSize, int *numChannels,
+                           int *sampleRate,
                            const FDK_channelMapDescr *const mapDescr,
                            const int mapIdx, const int coreDecodedOk,
-                           UCHAR *psDecoded);
+                           UCHAR *psDecoded, const INT inDataHeadroom,
+                           INT *outDataHeadroom);
 
 /**
  * \brief       Close SBR decoder instance and free memory.
diff --git a/libSBRdec/src/HFgen_preFlat.cpp b/libSBRdec/src/HFgen_preFlat.cpp
index 268011e..ad4caba 100644
--- a/libSBRdec/src/HFgen_preFlat.cpp
+++ b/libSBRdec/src/HFgen_preFlat.cpp
@@ -897,10 +897,10 @@ void sbrDecoder_calculateGainVec(FIXP_DBL **sourceBufferReal,
     for (i = startSample; i < stopSample; i++) {
       maxVal |=
           (FIXP_DBL)((LONG)(sourceBufferReal[i][loBand]) ^
-                     ((LONG)sourceBufferReal[i][loBand] >> (SAMPLE_BITS - 1)));
+                     ((LONG)sourceBufferReal[i][loBand] >> (DFRACT_BITS - 1)));
       maxVal |=
           (FIXP_DBL)((LONG)(sourceBufferImag[i][loBand]) ^
-                     ((LONG)sourceBufferImag[i][loBand] >> (SAMPLE_BITS - 1)));
+                     ((LONG)sourceBufferImag[i][loBand] >> (DFRACT_BITS - 1)));
     }
 
     if (maxVal != FL2FX_DBL(0.0f)) {
diff --git a/libSBRdec/src/hbe.cpp b/libSBRdec/src/hbe.cpp
index 2284c42..d210bb6 100644
--- a/libSBRdec/src/hbe.cpp
+++ b/libSBRdec/src/hbe.cpp
@@ -957,7 +957,7 @@ QmfTransposerCreate(HANDLE_HBE_TRANSPOSER* hQmfTransposer, const int frameSize,
     hQmfTran->qmfOutBufSize = 2 * (hQmfTran->noCols / 2 + QMF_WIN_LEN - 1);
 
     hQmfTran->inBuf_F =
-        (INT_PCM*)FDKcalloc(QMF_SYNTH_CHANNELS + 20 + 1, sizeof(INT_PCM));
+        (LONG*)FDKcalloc(QMF_SYNTH_CHANNELS + 20 + 1, sizeof(LONG));
     /* buffered time signal needs to be delayed by synthesis_size; max
      * synthesis_size = 20; */
     if (hQmfTran->inBuf_F == NULL) {
diff --git a/libSBRdec/src/hbe.h b/libSBRdec/src/hbe.h
index fdffe1e..3556783 100644
--- a/libSBRdec/src/hbe.h
+++ b/libSBRdec/src/hbe.h
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -132,6 +132,9 @@ typedef enum {
 } KEEP_STATES_SYNCED_MODE;
 
 struct hbeTransposer {
+  FIXP_DBL anaQmfStates[HBE_QMF_FILTER_STATE_ANA_SIZE];
+  FIXP_QSS synQmfStates[HBE_QMF_FILTER_STATE_SYN_SIZE];
+
   int xOverQmf[MAX_NUM_PATCHES_HBE];
 
   int maxStretch;
@@ -144,7 +147,7 @@ struct hbeTransposer {
   int stopBand;
   int bSbr41;
 
-  INT_PCM *inBuf_F;
+  LONG *inBuf_F;
   FIXP_DBL **qmfInBufReal_F;
   FIXP_DBL **qmfInBufImag_F;
 
@@ -156,9 +159,6 @@ struct hbeTransposer {
   FIXP_DBL const *synthesisQmfPreModCos_F;
   FIXP_DBL const *synthesisQmfPreModSin_F;
 
-  FIXP_QAS anaQmfStates[HBE_QMF_FILTER_STATE_ANA_SIZE];
-  FIXP_QSS synQmfStates[HBE_QMF_FILTER_STATE_SYN_SIZE];
-
   FIXP_DBL **qmfHBEBufReal_F;
   FIXP_DBL **qmfHBEBufImag_F;
 
diff --git a/libSBRdec/src/sbr_dec.cpp b/libSBRdec/src/sbr_dec.cpp
index 30611e7..b1fb0da 100644
--- a/libSBRdec/src/sbr_dec.cpp
+++ b/libSBRdec/src/sbr_dec.cpp
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -259,17 +259,18 @@ static void copyHarmonicSpectrum(int *xOverQmf, FIXP_DBL **qmfReal,
 
 void sbr_dec(
     HANDLE_SBR_DEC hSbrDec,             /*!< handle to Decoder channel */
-    INT_PCM *timeIn,                    /*!< pointer to input time signal */
-    INT_PCM *timeOut,                   /*!< pointer to output time signal */
+    LONG *timeIn,                       /*!< pointer to input time signal */
+    LONG *timeOut,                      /*!< pointer to output time signal */
     HANDLE_SBR_DEC hSbrDecRight,        /*!< handle to Decoder channel right */
-    INT_PCM *timeOutRight,              /*!< pointer to output time signal */
+    LONG *timeOutRight,                 /*!< pointer to output time signal */
     const int strideOut,                /*!< Time data traversal strideOut */
     HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */
     HANDLE_SBR_FRAME_DATA hFrameData,   /*!< Control data of current frame */
     HANDLE_SBR_PREV_FRAME_DATA
         hPrevFrameData,        /*!< Some control data of last frame */
     const int applyProcessing, /*!< Flag for SBR operation */
-    HANDLE_PS_DEC h_ps_d, const UINT flags, const int codecFrameSize) {
+    HANDLE_PS_DEC h_ps_d, const UINT flags, const int codecFrameSize,
+    const INT sbrInDataHeadroom) {
   int i, slot, reserve;
   int saveLbScale;
   int lastSlotOffs;
@@ -278,7 +279,7 @@ void sbr_dec(
   /* temporary pointer / variable for QMF;
      required as we want to use temporary buffer
      creating one frame delay for HBE in LP mode */
-  INT_PCM *pTimeInQmf = timeIn;
+  LONG *pTimeInQmf = timeIn;
 
   /* Number of QMF timeslots in the overlap buffer: */
   int ov_len = hSbrDec->LppTrans.pSettings->overlap;
@@ -341,8 +342,8 @@ void sbr_dec(
   } else {
     C_AALLOC_SCRATCH_START(qmfTemp, FIXP_DBL, 2 * (64));
     qmfAnalysisFiltering(&hSbrDec->qmfDomainInCh->fb, pReal, pImag,
-                         &hSbrDec->qmfDomainInCh->scaling, pTimeInQmf, 0, 1,
-                         qmfTemp);
+                         &hSbrDec->qmfDomainInCh->scaling, pTimeInQmf,
+                         0 + sbrInDataHeadroom, 1, qmfTemp);
 
     C_AALLOC_SCRATCH_END(qmfTemp, FIXP_DBL, 2 * (64));
   }
@@ -658,7 +659,7 @@ void sbr_dec(
 
   if (!(flags & SBRDEC_PS_DECODED)) {
     if (!(flags & SBRDEC_SKIP_QMF_SYN)) {
-      int outScalefactor = 0;
+      int outScalefactor = -(8);
 
       if (h_ps_d != NULL) {
         h_ps_d->procFrameBased = 1; /* we here do frame based processing */
@@ -743,6 +744,7 @@ void sbr_dec(
       */
       FDK_ASSERT(hSbrDec->qmfDomainInCh->pGlobalConf->nBandsSynthesis <=
                  QMF_MAX_SYNTHESIS_BANDS);
+      qmfChangeOutScalefactor(synQmfRight, -(8));
       FDKmemcpy(synQmfRight->FilterStates, synQmf->FilterStates,
                 9 * hSbrDec->qmfDomainInCh->pGlobalConf->nBandsSynthesis *
                     sizeof(FIXP_QSS));
@@ -814,7 +816,8 @@ void sbr_dec(
                   : scaleFactorLowBand_no_ov,
               scaleFactorHighBand, synQmf->lsb, synQmf->usb);
 
-          outScalefactorL = outScalefactorR = 1; /* psDiffScale! (MPEG-PS) */
+          outScalefactorL = outScalefactorR =
+              1 + sbrInDataHeadroom; /* psDiffScale! (MPEG-PS) */
         }
 
         sbrDecoder_drcApplySlot(/* right channel */
@@ -831,6 +834,9 @@ void sbr_dec(
         outScalefactorL += maxShift;
 
         if (!(flags & SBRDEC_SKIP_QMF_SYN)) {
+          qmfChangeOutScalefactor(synQmf, -(8));
+          qmfChangeOutScalefactor(synQmfRight, -(8));
+
           qmfSynthesisFilteringSlot(
               synQmfRight, rQmfReal, /* QMF real buffer */
               rQmfImag,              /* QMF imag buffer */
diff --git a/libSBRdec/src/sbr_dec.h b/libSBRdec/src/sbr_dec.h
index 156da03..eb9c394 100644
--- a/libSBRdec/src/sbr_dec.h
+++ b/libSBRdec/src/sbr_dec.h
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -176,17 +176,18 @@ typedef SBR_CHANNEL *HANDLE_SBR_CHANNEL;
 
 void sbr_dec(
     HANDLE_SBR_DEC hSbrDec,             /*!< handle to Decoder channel */
-    INT_PCM *timeIn,                    /*!< pointer to input time signal */
-    INT_PCM *timeOut,                   /*!< pointer to output time signal */
+    LONG *timeIn,                       /*!< pointer to input time signal */
+    LONG *timeOut,                      /*!< pointer to output time signal */
     HANDLE_SBR_DEC hSbrDecRight,        /*!< handle to Decoder channel right */
-    INT_PCM *timeOutRight,              /*!< pointer to output time signal */
+    LONG *timeOutRight,                 /*!< pointer to output time signal */
     INT strideOut,                      /*!< Time data traversal strideOut */
     HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */
     HANDLE_SBR_FRAME_DATA hFrameData,   /*!< Control data of current frame */
     HANDLE_SBR_PREV_FRAME_DATA
         hPrevFrameData,        /*!< Some control data of last frame */
     const int applyProcessing, /*!< Flag for SBR operation */
-    HANDLE_PS_DEC h_ps_d, const UINT flags, const int codecFrameSize);
+    HANDLE_PS_DEC h_ps_d, const UINT flags, const int codecFrameSize,
+    const INT sbrInDataHeadroom);
 
 SBR_ERROR
 createSbrDec(SBR_CHANNEL *hSbrChannel, HANDLE_SBR_HEADER_DATA hHeaderData,
diff --git a/libSBRdec/src/sbr_ram.h b/libSBRdec/src/sbr_ram.h
index e00f8b5..452f835 100644
--- a/libSBRdec/src/sbr_ram.h
+++ b/libSBRdec/src/sbr_ram.h
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -170,6 +170,9 @@ struct SBR_DECODER_INSTANCE {
                        flushed consecutively. */
 
   UINT flags;
+
+  INT sbrInDataHeadroom; /* Headroom of the SBR input time signal to prevent
+                            clipping */
 };
 
 H_ALLOC_MEM(Ram_SbrDecElement, SBR_DECODER_ELEMENT)
diff --git a/libSBRdec/src/sbrdecoder.cpp b/libSBRdec/src/sbrdecoder.cpp
index 52403f6..6e4aad4 100644
--- a/libSBRdec/src/sbrdecoder.cpp
+++ b/libSBRdec/src/sbrdecoder.cpp
@@ -155,7 +155,7 @@ amm-info@iis.fraunhofer.de
 
 /* Decoder library info */
 #define SBRDECODER_LIB_VL0 3
-#define SBRDECODER_LIB_VL1 0
+#define SBRDECODER_LIB_VL1 1
 #define SBRDECODER_LIB_VL2 0
 #define SBRDECODER_LIB_TITLE "SBR Decoder"
 #ifdef __ANDROID__
@@ -1570,10 +1570,10 @@ bail:
  * \return SBRDEC_OK if successfull, else error code
  */
 static SBR_ERROR sbrDecoder_DecodeElement(
-    HANDLE_SBRDECODER self, QDOM_PCM *input, INT_PCM *timeData,
-    const int timeDataSize, const FDK_channelMapDescr *const mapDescr,
-    const int mapIdx, int channelIndex, const int elementIndex,
-    const int numInChannels, int *numOutChannels, const int psPossible) {
+    HANDLE_SBRDECODER self, LONG *input, LONG *timeData, const int timeDataSize,
+    const FDK_channelMapDescr *const mapDescr, const int mapIdx,
+    int channelIndex, const int elementIndex, const int numInChannels,
+    int *numOutChannels, const int psPossible) {
   SBR_DECODER_ELEMENT *hSbrElement = self->pSbrElement[elementIndex];
   HANDLE_SBR_CHANNEL *pSbrChannel =
       self->pSbrElement[elementIndex]->pSbrChannel;
@@ -1743,7 +1743,7 @@ static SBR_ERROR sbrDecoder_DecodeElement(
             timeData + offset1, strideOut, hSbrHeader, hFrameDataLeft,
             &pSbrChannel[0]->prevFrameData,
             (hSbrHeader->syncState == SBR_ACTIVE), h_ps_d, self->flags,
-            codecFrameSize);
+            codecFrameSize, self->sbrInDataHeadroom);
 
     if (stereo) {
       /* Process right channel */
@@ -1751,7 +1751,7 @@ static SBR_ERROR sbrDecoder_DecodeElement(
               timeData + offset1, NULL, NULL, strideOut, hSbrHeader,
               hFrameDataRight, &pSbrChannel[1]->prevFrameData,
               (hSbrHeader->syncState == SBR_ACTIVE), NULL, self->flags,
-              codecFrameSize);
+              codecFrameSize, self->sbrInDataHeadroom);
     }
 
     C_ALLOC_SCRATCH_END(pPsScratch, struct PS_DEC_COEFFICIENTS, 1)
@@ -1771,14 +1771,14 @@ static SBR_ERROR sbrDecoder_DecodeElement(
       int copyFrameSize =
           codecFrameSize * self->pQmfDomain->QmfDomainOut->fb.no_channels;
       copyFrameSize /= self->pQmfDomain->QmfDomainIn->fb.no_channels;
-      INT_PCM *ptr;
+      LONG *ptr;
       INT i;
       FDK_ASSERT(strideOut == 2);
 
       ptr = timeData;
       for (i = copyFrameSize >> 1; i--;) {
-        INT_PCM tmp; /* This temporal variable is required because some
-                        compilers can't do *ptr++ = *ptr++ correctly. */
+        LONG tmp; /* This temporal variable is required because some compilers
+                     can't do *ptr++ = *ptr++ correctly. */
         tmp = *ptr++;
         *ptr++ = tmp;
         tmp = *ptr++;
@@ -1791,12 +1791,13 @@ static SBR_ERROR sbrDecoder_DecodeElement(
   return errorStatus;
 }
 
-SBR_ERROR sbrDecoder_Apply(HANDLE_SBRDECODER self, INT_PCM *input,
-                           INT_PCM *timeData, const int timeDataSize,
-                           int *numChannels, int *sampleRate,
+SBR_ERROR sbrDecoder_Apply(HANDLE_SBRDECODER self, LONG *input, LONG *timeData,
+                           const int timeDataSize, int *numChannels,
+                           int *sampleRate,
                            const FDK_channelMapDescr *const mapDescr,
                            const int mapIdx, const int coreDecodedOk,
-                           UCHAR *psDecoded) {
+                           UCHAR *psDecoded, const INT inDataHeadroom,
+                           INT *outDataHeadroom) {
   SBR_ERROR errorStatus = SBRDEC_OK;
 
   int psPossible;
@@ -1833,6 +1834,9 @@ SBR_ERROR sbrDecoder_Apply(HANDLE_SBRDECODER self, INT_PCM *input,
     psPossible = 0;
   }
 
+  self->sbrInDataHeadroom = inDataHeadroom;
+  *outDataHeadroom = (INT)(8);
+
   /* Make sure that even if no SBR data was found/parsed *psDecoded is returned
    * 1 if psPossible was 0. */
   if (psPossible == 0) {