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

15 files changed, 273 insertions, 217 deletions

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