Update fdk to v2.0.2

27 files changed, 771 insertions, 473 deletions

diff --git a/fdk-aac/libAACdec/include/aacdecoder_lib.h b/fdk-aac/libAACdec/include/aacdecoder_lib.h
index 2dfc65a..06272df 100644
--- a/fdk-aac/libAACdec/include/aacdecoder_lib.h
+++ b/fdk-aac/libAACdec/include/aacdecoder_lib.h
@@ -164,9 +164,6 @@ The contents of each file is described in detail in this document. All header
 files are provided for usage in specific C/C++ programs. The main AAC decoder
 library API functions are located in aacdecoder_lib.h header file.
 
-In binary releases the decoder core resides in statically linkable libraries,
-for example libAACdec.a.
-
 
 \section Calling_Sequence Calling Sequence
 
@@ -174,19 +171,7 @@ The following sequence is necessary for proper decoding of ISO/MPEG-2/4 AAC,
 HE-AAC v2, or MPEG-D USAC bitstreams. In the following description, input stream
 read and output write function details are left out, since they may be
 implemented in a variety of configurations depending on the user's specific
-requirements. The example implementation uses file-based input/output, and in
-such case one may call mpegFileRead_Open() to open an input file and to allocate
-memory for the required structures, and the corresponding mpegFileRead_Close()
-to close opened files and to de-allocate associated structures.
-mpegFileRead_Open() will attempt to detect the bitstream format and in case of
-MPEG-4 file format or Raw Packets file format (a proprietary Fraunhofer IIS file
-format suitable only for testing) it will read the Audio Specific Config data
-(ASC). An unsuccessful attempt to recognize the bitstream format requires the
-user to provide this information manually. For any other bitstream formats that
-are usually applicable in streaming applications, the decoder itself will try to
-synchronize and parse the given bitstream fragment using the FDK transport
-library. Hence, for streaming applications (without file access) this step is
-not necessary.
+requirements.
 
 
 -# Call aacDecoder_Open() to open and retrieve a handle to a new AAC decoder
@@ -205,19 +190,17 @@ do {
 working memory (a client-supplied input buffer "inBuffer" in framework). This
 buffer will be used to load AAC bitstream data to the decoder.  Only when all
 data in this buffer has been processed will the decoder signal an empty buffer.
-For file-based input, you may invoke mpegFileRead_Read() to acquire new
-bitstream data.
 -# Call aacDecoder_Fill() to fill the decoder's internal bitstream input buffer
 with the client-supplied bitstream input buffer. Note, if the data loaded in to
 the internal buffer is not sufficient to decode a frame,
 aacDecoder_DecodeFrame() will return ::AAC_DEC_NOT_ENOUGH_BITS until a
 sufficient amount of data is loaded in to the internal buffer. For streaming
 formats (ADTS, LOAS), it is acceptable to load more than one frame to the
-decoder. However, for RAW file format (Fraunhofer IIS proprietary format), only
-one frame may be loaded to the decoder per aacDecoder_DecodeFrame() call. For
-least amount of communication delay, fill and decode should be performed on a
-frame by frame basis. \code ErrorStatus = aacDecoder_Fill(aacDecoderInfo,
-inBuffer, bytesRead, bytesValid); \endcode
+decoder. However, for packed based formats, only one frame may be loaded to the
+decoder per aacDecoder_DecodeFrame() call. For least amount of communication
+delay, fill and decode should be performed on a frame by frame basis. \code
+    ErrorStatus = aacDecoder_Fill(aacDecoderInfo, inBuffer, bytesRead,
+bytesValid); \endcode
 -# Call aacDecoder_DecodeFrame(). This function decodes one frame and writes
 decoded PCM audio data to a client-supplied buffer. It is the client's
 responsibility to allocate a buffer which is large enough to hold the decoded
@@ -225,12 +208,9 @@ output data. \code ErrorStatus = aacDecoder_DecodeFrame(aacDecoderInfo,
 TimeData, OUT_BUF_SIZE, flags); \endcode If the bitstream configuration (number
 of channels, sample rate, frame size) is not known a priori, you may call
 aacDecoder_GetStreamInfo() to retrieve a structure that contains this
-information. You may use this data to initialize an audio output device. In the
-example program, if the number of channels or the sample rate has changed since
-program start or the previously decoded frame, the audio output device is then
-re-initialized. If WAVE file output is chosen, a new WAVE file for each new
-stream configuration is be created. \code p_si =
-aacDecoder_GetStreamInfo(aacDecoderInfo); \endcode
+information. You may use this data to initialize an audio output device. \code
+    p_si = aacDecoder_GetStreamInfo(aacDecoderInfo);
+\endcode
 -# Repeat steps 5 to 7 until no data is available to decode any more, or in case
 of error. \code } while (bytesRead[0] > 0 || doFlush || doBsFlush ||
 forceContinue); \endcode
@@ -239,7 +219,7 @@ structures. \code aacDecoder_Close(aacDecoderInfo); \endcode
 
 \image latex decode.png "Decode calling sequence" width=11cm
 
-\image latex change_source.png "Change data source sequence" width 5cm
+\image latex change_source.png "Change data source sequence" width=5cm
 
 \image latex conceal.png "Error concealment sequence" width=14cm
 
@@ -296,16 +276,14 @@ input buffer, and one to hold the decoded output PCM sample data. In resource
 limited applications, the output buffer may be reused as an external input
 buffer prior to the subsequence aacDecoder_Fill() function call.
 
-The external input buffer is set in the example program and its size is defined
-by ::IN_BUF_SIZE. You may freely choose different buffer sizes. To feed the data
-to the decoder-internal input buffer, use the function aacDecoder_Fill(). This
-function returns important information regarding the number of bytes in the
-external input buffer that have not yet been copied into the internal input
-buffer (variable bytesValid). Once the external buffer has been fully copied, it
-can be completely re-filled again. In case you wish to refill the buffer while
-there are unprocessed bytes (bytesValid is unequal 0), you should preserve the
-unconsumed data. However, we recommend to refill the buffer only when bytesValid
-returns 0.
+To feed the data to the decoder-internal input buffer, use the
+function aacDecoder_Fill(). This function returns important information
+regarding the number of bytes in the external input buffer that have not yet
+been copied into the internal input buffer (variable bytesValid). Once the
+external buffer has been fully copied, it can be completely re-filled again. In
+case you wish to refill the buffer while there are unprocessed bytes (bytesValid
+is unequal 0), you should preserve the unconsumed data. However, we recommend to
+refill the buffer only when bytesValid returns 0.
 
 The bytesValid parameter is an input and output parameter to the FDK decoder. As
 an input, it signals how many valid bytes are available in the external buffer.
@@ -340,10 +318,7 @@ explanation, please refer to ISO/IEC 13818-7:2005(E), chapter 8.5.3.2.
 In case a Program Config is included in the audio configuration, the channel
 mapping described within it will be adopted.
 
-In case of MPEG-D Surround the channel mapping will follow the same criteria
-described in ISO/IEC 13818-7:2005(E), but adding corresponding top channels (if
-available) to the channel types in order to avoid ambiguity. The examples below
-explain these aspects in detail.
+The examples below explain these aspects in detail.
 
 \section OutputFormatChange Changing the audio output format
 
@@ -459,8 +434,8 @@ Where N equals to CStreamInfo::frameSize .
 #include "genericStds.h"
 
 #define AACDECODER_LIB_VL0 3
-#define AACDECODER_LIB_VL1 1
-#define AACDECODER_LIB_VL2 2
+#define AACDECODER_LIB_VL1 2
+#define AACDECODER_LIB_VL2 0
 
 /**
  * \brief  AAC decoder error codes.
@@ -694,9 +669,7 @@ typedef enum {
                      2. If the parameter value is greater than that of
                  ::AAC_PCM_MAX_OUTPUT_CHANNELS both will be set to the same
                  value. \n
-                     3. This parameter does not affect MPEG Surround processing.
-                 \n
-                     4. This parameter will be ignored if the number of encoded
+                     3. This parameter will be ignored if the number of encoded
                  audio channels is greater than 8. */
   AAC_PCM_MAX_OUTPUT_CHANNELS =
       0x0012, /*!< Maximum number of PCM output channels. If lower than the
@@ -723,11 +696,7 @@ typedef enum {
                      2. If the parameter value is greater than zero but smaller
                  than ::AAC_PCM_MIN_OUTPUT_CHANNELS both will be set to same
                  value. \n
-                     3. The operating mode of the MPEG Surround module will be
-                 set accordingly. \n
-                     4. Setting this parameter with any value will disable the
-                 binaural processing of the MPEG Surround module
-                     5. This parameter will be ignored if the number of encoded
+                     3. This parameter will be ignored if the number of encoded
                  audio channels is greater than 8. */
   AAC_METADATA_PROFILE =
       0x0020, /*!< See ::AAC_MD_PROFILE for all available values. */
@@ -808,11 +777,11 @@ typedef enum {
                  sequences for fading in and out, if provided in the
                  bitstream.\n Enabled album mode makes use of dedicated album
                  loudness information, if provided in the bitstream.\n */
-  AAC_QMF_LOWPOWER = 0x0300, /*!< Quadrature Mirror Filter (QMF) Bank processing
-                                mode. \n -1: Use internal default. Implies MPEG
-                                Surround partially complex accordingly. \n 0:
-                                Use complex QMF data mode. \n 1: Use real (low
-                                power) QMF data mode. \n */
+  AAC_QMF_LOWPOWER =
+      0x0300, /*!< Quadrature Mirror Filter (QMF) Bank processing mode. \n
+                   -1: Use internal default. \n
+                    0: Use complex QMF data mode. \n
+                    1: Use real (low power) QMF data mode. \n */
   AAC_TPDEC_CLEAR_BUFFER =
       0x0603 /*!< Clear internal bit stream buffer of transport layers. The
                 decoder will start decoding at new data passed after this event
@@ -897,15 +866,25 @@ typedef struct {
                           1770. If no level has been found in the bitstream the
                           value is -1. */
   SCHAR
-  drcPresMode; /*!< DRC presentation mode. According to ETSI TS 101 154,
-                  this field indicates whether   light (MPEG-4 Dynamic Range
-                  Control tool) or heavy compression (DVB heavy
-                  compression)   dynamic range control shall take priority
-                  on the outputs.   For details, see ETSI TS 101 154, table
-                  C.33. Possible values are: \n   -1: No corresponding
-                  metadata found in the bitstream \n   0: DRC presentation
-                  mode not indicated \n   1: DRC presentation mode 1 \n   2:
-                  DRC presentation mode 2 \n   3: Reserved */
+  drcPresMode;        /*!< DRC presentation mode. According to ETSI TS 101 154,
+                         this field indicates whether   light (MPEG-4 Dynamic Range
+                         Control tool) or heavy compression (DVB heavy
+                         compression)   dynamic range control shall take priority
+                         on the outputs.   For details, see ETSI TS 101 154, table
+                         C.33. Possible values are: \n   -1: No corresponding
+                         metadata found in the bitstream \n   0: DRC presentation
+                         mode not indicated \n   1: DRC presentation mode 1 \n   2:
+                         DRC presentation mode 2 \n   3: Reserved */
+  INT outputLoudness; /*!< Audio output loudness in steps of -0.25 dB. Range: 0
+                         (0 dBFS) to 231 (-57.75 dBFS).\n  A value of -1
+                         indicates that no loudness metadata is present.\n  If
+                         loudness normalization is active, the value corresponds
+                         to the target loudness value set with
+                         ::AAC_DRC_REFERENCE_LEVEL.\n  If loudness normalization
+                         is not active, the output loudness value corresponds to
+                         the loudness metadata given in the bitstream.\n
+                           Loudness metadata can originate from MPEG-4 DRC or
+                         MPEG-D DRC. */
 
 } CStreamInfo;
 
@@ -1033,21 +1012,24 @@ LINKSPEC_H AAC_DECODER_ERROR aacDecoder_Fill(HANDLE_AACDECODER self,
                                              const UINT bufferSize[],
                                              UINT *bytesValid);
 
-#define AACDEC_CONCEAL                                                        \
-  1 /*!< Flag for aacDecoder_DecodeFrame(): Trigger the built-in error        \
-       concealment module to generate a substitute signal for one lost frame. \
-       New input data will not be considered. */
-#define AACDEC_FLUSH                                                         \
-  2 /*!< Flag for aacDecoder_DecodeFrame(): Flush all filterbanks to get all \
-       delayed audio without having new input data. Thus new input data will \
-       not be considered.*/
-#define AACDEC_INTR                                                         \
-  4 /*!< Flag for aacDecoder_DecodeFrame(): Signal an input bit stream data \
-       discontinuity. Resync any internals as necessary. */
-#define AACDEC_CLRHIST                                                        \
-  8 /*!< Flag for aacDecoder_DecodeFrame(): Clear all signal delay lines and  \
-       history buffers. CAUTION: This can cause discontinuities in the output \
-       signal. */
+/** Flag for aacDecoder_DecodeFrame(): Trigger the built-in error concealment
+ * module to generate a substitute signal for one lost frame. New input data
+ * will not be considered.
+ */
+#define AACDEC_CONCEAL 1
+/** Flag for aacDecoder_DecodeFrame(): Flush all filterbanks to get all delayed
+ * audio without having new input data. Thus new input data will not be
+ * considered.
+ */
+#define AACDEC_FLUSH 2
+/** Flag for aacDecoder_DecodeFrame(): Signal an input bit stream data
+ * discontinuity. Resync any internals as necessary.
+ */
+#define AACDEC_INTR 4
+/** Flag for aacDecoder_DecodeFrame(): Clear all signal delay lines and history
+ * buffers. CAUTION: This can cause discontinuities in the output signal.
+ */
+#define AACDEC_CLRHIST 8
 
 /**
  * \brief               Decode one audio frame
diff --git a/fdk-aac/libAACdec/src/FDK_delay.cpp b/fdk-aac/libAACdec/src/FDK_delay.cpp
index 0ab1a66..0cc869c 100644
--- a/fdk-aac/libAACdec/src/FDK_delay.cpp
+++ b/fdk-aac/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/fdk-aac/libAACdec/src/FDK_delay.h b/fdk-aac/libAACdec/src/FDK_delay.h
index f89c3a2..6317d9d 100644
--- a/fdk-aac/libAACdec/src/FDK_delay.h
+++ b/fdk-aac/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/fdk-aac/libAACdec/src/aac_ram.cpp b/fdk-aac/libAACdec/src/aac_ram.cpp
index e13167d..aa8f6a6 100644
--- a/fdk-aac/libAACdec/src/aac_ram.cpp
+++ b/fdk-aac/libAACdec/src/aac_ram.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
@@ -105,12 +105,7 @@ amm-info@iis.fraunhofer.de
 
 #define WORKBUFFER1_TAG 0
 #define WORKBUFFER2_TAG 1
-
-#define WORKBUFFER3_TAG 4
-#define WORKBUFFER4_TAG 5
-
 #define WORKBUFFER5_TAG 6
-
 #define WORKBUFFER6_TAG 7
 
 /*! The structure AAC_DECODER_INSTANCE is the top level structure holding all
@@ -169,9 +164,6 @@ C_ALLOC_MEM2(TimeDataFlush, INT_PCM, TIME_DATA_FLUSH_SIZE, (8))
 C_ALLOC_MEM_OVERLAY(WorkBufferCore2, FIXP_DBL, ((8) * 1024), SECT_DATA_L2,
                     WORKBUFFER2_TAG)
 
-C_ALLOC_MEM_OVERLAY(WorkBufferCore3, FIXP_DBL, WB_SECTION_SIZE, SECT_DATA_L2,
-                    WORKBUFFER3_TAG)
-C_AALLOC_MEM(WorkBufferCore4, FIXP_DBL, WB_SECTION_SIZE)
 C_ALLOC_MEM_OVERLAY(WorkBufferCore6, SCHAR,
                     fMax((INT)(sizeof(FIXP_DBL) * WB_SECTION_SIZE),
                          (INT)sizeof(CAacDecoderCommonData)),
diff --git a/fdk-aac/libAACdec/src/aac_ram.h b/fdk-aac/libAACdec/src/aac_ram.h
index a861e25..b9b95b7 100644
--- a/fdk-aac/libAACdec/src/aac_ram.h
+++ b/fdk-aac/libAACdec/src/aac_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
@@ -136,12 +136,7 @@ H_ALLOC_MEM(TimeDataFlush, INT_PCM)
 
 H_ALLOC_MEM_OVERLAY(WorkBufferCore1, CWorkBufferCore1)
 H_ALLOC_MEM_OVERLAY(WorkBufferCore2, FIXP_DBL)
-
-H_ALLOC_MEM_OVERLAY(WorkBufferCore3, FIXP_DBL)
-H_ALLOC_MEM(WorkBufferCore4, FIXP_DBL)
-
 H_ALLOC_MEM_OVERLAY(WorkBufferCore5, PCM_DEC)
-
 H_ALLOC_MEM_OVERLAY(WorkBufferCore6, SCHAR)
 
 #endif /* #ifndef AAC_RAM_H */
diff --git a/fdk-aac/libAACdec/src/aac_rom.h b/fdk-aac/libAACdec/src/aac_rom.h
index ffaf951..7a1597c 100644
--- a/fdk-aac/libAACdec/src/aac_rom.h
+++ b/fdk-aac/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/fdk-aac/libAACdec/src/aacdec_drc.cpp b/fdk-aac/libAACdec/src/aacdec_drc.cpp
index 4129d0f..b6f5b49 100644
--- a/fdk-aac/libAACdec/src/aacdec_drc.cpp
+++ b/fdk-aac/libAACdec/src/aacdec_drc.cpp
@@ -150,6 +150,20 @@ static INT convert_drcParam(FIXP_DBL param_dbl) {
 }
 
 /*!
+\brief Reset DRC information
+
+\self Handle of DRC info
+
+\return none
+*/
+void aacDecoder_drcReset(HANDLE_AAC_DRC self) {
+  self->applyExtGain = 0;
+  self->additionalGainPrev = AACDEC_DRC_GAIN_INIT_VALUE;
+  self->additionalGainFilterState = AACDEC_DRC_GAIN_INIT_VALUE;
+  self->additionalGainFilterState1 = AACDEC_DRC_GAIN_INIT_VALUE;
+}
+
+/*!
   \brief Initialize DRC information
 
   \self Handle of DRC info
@@ -176,7 +190,6 @@ void aacDecoder_drcInit(HANDLE_AAC_DRC self) {
   pParams->usrBoost = FL2FXCONST_DBL(0.0f);
   pParams->targetRefLevel = 96;
   pParams->expiryFrame = AACDEC_DRC_DFLT_EXPIRY_FRAMES;
-  pParams->applyDigitalNorm = ON;
   pParams->applyHeavyCompression = OFF;
   pParams->usrApplyHeavyCompression = OFF;
 
@@ -192,6 +205,8 @@ void aacDecoder_drcInit(HANDLE_AAC_DRC self) {
   self->progRefLevelPresent = 0;
   self->presMode = -1;
   self->uniDrcPrecedence = 0;
+
+  aacDecoder_drcReset(self);
 }
 
 /*!
@@ -258,11 +273,8 @@ AAC_DECODER_ERROR aacDecoder_drcSetParam(HANDLE_AAC_DRC self,
         return AAC_DEC_INVALID_HANDLE;
       }
       if (value < 0) {
-        self->params.applyDigitalNorm = OFF;
         self->params.targetRefLevel = -1;
       } else {
-        /* ref_level must be between 0 and MAX_REFERENCE_LEVEL, inclusive */
-        self->params.applyDigitalNorm = ON;
         if (self->params.targetRefLevel != (SCHAR)value) {
           self->params.targetRefLevel = (SCHAR)value;
           self->progRefLevel = (SCHAR)value; /* Always set the program reference
@@ -273,16 +285,6 @@ AAC_DECODER_ERROR aacDecoder_drcSetParam(HANDLE_AAC_DRC self,
         self->update = 1;
       }
       break;
-    case APPLY_NORMALIZATION:
-      if ((value != OFF) && (value != ON)) {
-        return AAC_DEC_SET_PARAM_FAIL;
-      }
-      if (self == NULL) {
-        return AAC_DEC_INVALID_HANDLE;
-      }
-      /* Store new parameter value */
-      self->params.applyDigitalNorm = (UCHAR)value;
-      break;
     case APPLY_HEAVY_COMPRESSION:
       if ((value != OFF) && (value != ON)) {
         return AAC_DEC_SET_PARAM_FAIL;
@@ -910,11 +912,9 @@ void aacDecoder_drcApply(HANDLE_AAC_DRC self, void *pSbrDec,
       FDK_ASSERT(0);
     }
   }
-  if (self->params.applyDigitalNorm == OFF) {
-    /* Reset normalization gain since this module must not apply it */
-    norm_mantissa = FL2FXCONST_DBL(0.5f);
-    norm_exponent = 1;
-  }
+  /* Reset normalization gain since this module must not apply it */
+  norm_mantissa = FL2FXCONST_DBL(0.5f);
+  norm_exponent = 1;
 
   /* calc scale factors */
   for (band = 0; band < numBands; band++) {
@@ -1353,3 +1353,152 @@ void aacDecoder_drcGetInfo(HANDLE_AAC_DRC self, SCHAR *pPresMode,
     }
   }
 }
+
+/**
+ * \brief  Apply DRC Level Normalization.
+ *
+ *         This function prepares/applies the gain values for the DRC Level
+ * Normalization and returns the exponent of the time data. The following two
+ * cases are handled:
+ *
+ *         - Limiter enabled:
+ *           The input data must be interleaved.
+ *           One gain per sample is written to the buffer pGainPerSample.
+ *           If necessary the time data is rescaled.
+ *
+ *         - Limiter disabled:
+ *           The input data can be interleaved or deinterleaved.
+ *           The gain values are applied to the time data.
+ *           If necessary the time data is rescaled.
+ *
+ * \param hDrcInfo                     [i/o] handle to drc data structure.
+ * \param samplesIn                    [i/o] pointer to time data.
+ * \param pGain                        [i  ] pointer to gain to be applied to
+ * the time data.
+ * \param pGainPerSample               [o  ] pointer to the gain per sample to
+ * be applied to the time data in the limiter.
+ * \param gain_scale                   [i  ] exponent to be applied to the time
+ * data.
+ * \param gain_delay                   [i  ] delay[samples] with which the gains
+ * in pGain shall be applied (gain_delay <= nSamples).
+ * \param nSamples                     [i  ] number of samples per frame.
+ * \param channels                     [i  ] number of channels.
+ * \param stride                       [i  ] channel stride of time data.
+ * \param limiterEnabled               [i  ] 1 if limiter is enabled, otherwise
+ * 0.
+ *
+ * \return exponent of time data
+ */
+INT applyDrcLevelNormalization(HANDLE_AAC_DRC hDrcInfo, PCM_DEC *samplesIn,
+                               FIXP_DBL *pGain, FIXP_DBL *pGainPerSample,
+                               const INT gain_scale, const UINT gain_delay,
+                               const UINT nSamples, const UINT channels,
+                               const UINT stride, const UINT limiterEnabled) {
+  UINT i;
+  INT additionalGain_scaling;
+  FIXP_DBL additionalGain;
+
+  FDK_ASSERT(gain_delay <= nSamples);
+
+  FIXP_DBL additionalGainSmoothState = hDrcInfo->additionalGainFilterState;
+  FIXP_DBL additionalGainSmoothState1 = hDrcInfo->additionalGainFilterState1;
+
+  if (!gain_delay) {
+    additionalGain = pGain[0];
+
+    /* Apply the additional scaling gain_scale[0] that has no delay and no
+     * smoothing */
+    additionalGain_scaling =
+        fMin(gain_scale, CntLeadingZeros(additionalGain) - 1);
+    additionalGain = scaleValue(additionalGain, additionalGain_scaling);
+
+    /* if it's not possible to fully apply gain_scale to additionalGain, apply
+     * it to the input signal */
+    additionalGain_scaling -= gain_scale;
+
+    if (additionalGain_scaling) {
+      scaleValuesSaturate(samplesIn, channels * nSamples,
+                          -additionalGain_scaling);
+    }
+
+    if (limiterEnabled) {
+      FDK_ASSERT(pGainPerSample != NULL);
+
+      for (i = 0; i < nSamples; i++) {
+        pGainPerSample[i] = additionalGain;
+      }
+    } else {
+      for (i = 0; i < channels * nSamples; i++) {
+        samplesIn[i] = FIXP_DBL2PCM_DEC(fMult(samplesIn[i], additionalGain));
+      }
+    }
+  } else {
+    UINT inc;
+    FIXP_DBL additionalGainUnfiltered;
+
+    inc = (stride == 1) ? channels : 1;
+
+    for (i = 0; i < nSamples; i++) {
+      if (i < gain_delay) {
+        additionalGainUnfiltered = hDrcInfo->additionalGainPrev;
+      } else {
+        additionalGainUnfiltered = pGain[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 gain_scale[0] that has no delay and no
+       * smoothing */
+      additionalGain_scaling =
+          fMin(gain_scale, CntLeadingZeros(additionalGain) - 1);
+      additionalGain = scaleValue(additionalGain, additionalGain_scaling);
+
+      /* if it's not possible to fully apply gain_scale[0] to additionalGain,
+       * apply it to the input signal */
+      additionalGain_scaling -= gain_scale;
+
+      if (limiterEnabled) {
+        FDK_ASSERT(stride == 1);
+        FDK_ASSERT(pGainPerSample != NULL);
+
+        if (additionalGain_scaling) {
+          scaleValuesSaturate(samplesIn, channels, -additionalGain_scaling);
+        }
+
+        pGainPerSample[i] = additionalGain;
+      } else {
+        if (additionalGain_scaling) {
+          for (UINT k = 0; k < channels; k++) {
+            scaleValuesSaturate(&samplesIn[k * stride], 1,
+                                -additionalGain_scaling);
+          }
+        }
+
+        for (UINT k = 0; k < channels; k++) {
+          samplesIn[k * stride] =
+              FIXP_DBL2PCM_DEC(fMult(samplesIn[k * stride], additionalGain));
+        }
+      }
+
+      samplesIn += inc;
+    }
+  }
+
+  hDrcInfo->additionalGainPrev = pGain[0];
+  hDrcInfo->additionalGainFilterState = additionalGainSmoothState;
+  hDrcInfo->additionalGainFilterState1 = additionalGainSmoothState1;
+
+  return (AACDEC_DRC_GAIN_SCALING);
+}
diff --git a/fdk-aac/libAACdec/src/aacdec_drc.h b/fdk-aac/libAACdec/src/aacdec_drc.h
index 924ec6f..76a44d6 100644
--- a/fdk-aac/libAACdec/src/aacdec_drc.h
+++ b/fdk-aac/libAACdec/src/aacdec_drc.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
@@ -109,6 +109,11 @@ amm-info@iis.fraunhofer.de
 #include "channel.h"
 #include "FDK_bitstream.h"
 
+#define AACDEC_DRC_GAIN_SCALING (11) /* Scaling of DRC gains */
+#define AACDEC_DRC_GAIN_INIT_VALUE \
+  (FL2FXCONST_DBL(                 \
+      1.0f / (1 << AACDEC_DRC_GAIN_SCALING))) /* Init value for DRC gains */
+
 #define AACDEC_DRC_DFLT_EXPIRY_FRAMES \
   (0) /* Default DRC data expiry time in AAC frames   */
 
@@ -125,7 +130,6 @@ typedef enum {
   TARGET_REF_LEVEL,
   DRC_BS_DELAY,
   DRC_DATA_EXPIRY_FRAME,
-  APPLY_NORMALIZATION,
   APPLY_HEAVY_COMPRESSION,
   DEFAULT_PRESENTATION_MODE,
   ENCODER_TARGET_LEVEL,
@@ -136,6 +140,8 @@ typedef enum {
 /**
  * \brief DRC module interface functions
  */
+void aacDecoder_drcReset(HANDLE_AAC_DRC self);
+
 void aacDecoder_drcInit(HANDLE_AAC_DRC self);
 
 void aacDecoder_drcInitChannelData(CDrcChannelData *pDrcChannel);
@@ -189,4 +195,45 @@ int aacDecoder_drcEpilog(
 void aacDecoder_drcGetInfo(HANDLE_AAC_DRC self, SCHAR *pPresMode,
                            SCHAR *pProgRefLevel);
 
+/**
+ * \brief  Apply DRC Level Normalization.
+ *
+ *         This function prepares/applies the gain values for the DRC Level
+ * Normalization and returns the exponent of the time data. The following two
+ * cases are handled:
+ *
+ *         - Limiter enabled:
+ *           The input data must be interleaved.
+ *           One gain per sample is written to the buffer pGainPerSample.
+ *           If necessary the time data is rescaled.
+ *
+ *         - Limiter disabled:
+ *           The input data can be interleaved or deinterleaved.
+ *           The gain values are applied to the time data.
+ *           If necessary the time data is rescaled.
+ *
+ * \param hDrcInfo                     [i/o] handle to drc data structure.
+ * \param samplesIn                    [i/o] pointer to time data.
+ * \param pGain                        [i  ] pointer to gain to be applied to
+ * the time data.
+ * \param pGainPerSample               [o  ] pointer to the gain per sample to
+ * be applied to the time data in the limiter.
+ * \param gain_scale                   [i  ] exponent to be applied to the time
+ * data.
+ * \param gain_delay                   [i  ] delay[samples] with which the gains
+ * in pGain shall be applied (gain_delay <= nSamples).
+ * \param nSamples                     [i  ] number of samples per frame.
+ * \param channels                     [i  ] number of channels.
+ * \param stride                       [i  ] channel stride of time data.
+ * \param limiterEnabled               [i  ] 1 if limiter is enabled, otherwise
+ * 0.
+ *
+ * \return exponent of time data
+ */
+INT applyDrcLevelNormalization(HANDLE_AAC_DRC hDrcInfo, PCM_DEC *samplesIn,
+                               FIXP_DBL *pGain, FIXP_DBL *pGainPerSample,
+                               const INT gain_scale, const UINT gain_delay,
+                               const UINT nSamples, const UINT channels,
+                               const UINT stride, const UINT limiterEnabled);
+
 #endif /* AACDEC_DRC_H */
diff --git a/fdk-aac/libAACdec/src/aacdec_drc_types.h b/fdk-aac/libAACdec/src/aacdec_drc_types.h
index 76c35d0..d4393f7 100644
--- a/fdk-aac/libAACdec/src/aacdec_drc_types.h
+++ b/fdk-aac/libAACdec/src/aacdec_drc_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
@@ -168,7 +168,6 @@ typedef struct {
 
   UINT expiryFrame;
   UCHAR bsDelayEnable;
-  UCHAR applyDigitalNorm;
 
   AACDEC_DRC_PARAMETER_HANDLING defaultPresentationMode;
   UCHAR encoderTargetLevel;
@@ -213,6 +212,13 @@ typedef struct {
   uniDrcPrecedence; /* Flag for signalling that uniDrc is active and takes
                        precedence over legacy DRC */
 
+  UCHAR applyExtGain; /* Flag is 1 if extGain has to be applied, otherwise 0. */
+
+  FIXP_DBL additionalGainPrev; /* Gain of previous frame to be applied to the
+                                  time data */
+  FIXP_DBL additionalGainFilterState;  /* Filter state for the gain smoothing */
+  FIXP_DBL additionalGainFilterState1; /* Filter state for the gain smoothing */
+
 } CDrcInfo;
 
 typedef CDrcInfo *HANDLE_AAC_DRC;
diff --git a/fdk-aac/libAACdec/src/aacdec_hcr.cpp b/fdk-aac/libAACdec/src/aacdec_hcr.cpp
index 26fdd97..a7e9cce 100644
--- a/fdk-aac/libAACdec/src/aacdec_hcr.cpp
+++ b/fdk-aac/libAACdec/src/aacdec_hcr.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
@@ -137,7 +137,7 @@ static void DeriveNumberOfExtendedSortedSectionsInSets(
 static INT DecodeEscapeSequence(HANDLE_FDK_BITSTREAM bs, const INT bsAnchor,
                                 INT quantSpecCoef, INT *pLeftStartOfSegment,
                                 SCHAR *pRemainingBitsInSegment,
-                                int *pNumDecodedBits);
+                                int *pNumDecodedBits, UINT *errorWord);
 
 static int DecodePCW_Sign(HANDLE_FDK_BITSTREAM bs, const INT bsAnchor,
                           UINT codebookDim, const SCHAR *pQuantVal,
@@ -1179,8 +1179,8 @@ static void DecodePCWs(HANDLE_FDK_BITSTREAM bs, H_HCR_INFO pHcr) {
                   bs, pHcr->decInOut.bitstreamAnchor,
                   pQuantizedSpectralCoefficients
                       [quantizedSpectralCoefficientsIdx],
-                  pLeftStartOfSegment, pRemainingBitsInSegment,
-                  &numDecodedBits);
+                  pLeftStartOfSegment, pRemainingBitsInSegment, &numDecodedBits,
+                  &pHcr->decInOut.errorLog);
         }
         quantizedSpectralCoefficientsIdx++;
         if (quantizedSpectralCoefficientsIdx >= 1024) {
@@ -1195,8 +1195,8 @@ static void DecodePCWs(HANDLE_FDK_BITSTREAM bs, H_HCR_INFO pHcr) {
                   bs, pHcr->decInOut.bitstreamAnchor,
                   pQuantizedSpectralCoefficients
                       [quantizedSpectralCoefficientsIdx],
-                  pLeftStartOfSegment, pRemainingBitsInSegment,
-                  &numDecodedBits);
+                  pLeftStartOfSegment, pRemainingBitsInSegment, &numDecodedBits,
+                  &pHcr->decInOut.errorLog);
         }
         quantizedSpectralCoefficientsIdx++;
         if (quantizedSpectralCoefficientsIdx >= 1024) {
@@ -1386,7 +1386,7 @@ value == 16, a escapeSequence is decoded in two steps:
 static INT DecodeEscapeSequence(HANDLE_FDK_BITSTREAM bs, const INT bsAnchor,
                                 INT quantSpecCoef, INT *pLeftStartOfSegment,
                                 SCHAR *pRemainingBitsInSegment,
-                                int *pNumDecodedBits) {
+                                int *pNumDecodedBits, UINT *errorWord) {
   UINT i;
   INT sign;
   UINT escapeOnesCounter = 0;
@@ -1400,6 +1400,9 @@ static INT DecodeEscapeSequence(HANDLE_FDK_BITSTREAM bs, const INT bsAnchor,
                                        FROM_LEFT_TO_RIGHT);
     *pRemainingBitsInSegment -= 1;
     *pNumDecodedBits += 1;
+    if (*pRemainingBitsInSegment < 0) {
+      return Q_VALUE_INVALID;
+    }
 
     if (carryBit != 0) {
       escapeOnesCounter += 1;
@@ -1416,6 +1419,9 @@ static INT DecodeEscapeSequence(HANDLE_FDK_BITSTREAM bs, const INT bsAnchor,
                                        FROM_LEFT_TO_RIGHT);
     *pRemainingBitsInSegment -= 1;
     *pNumDecodedBits += 1;
+    if (*pRemainingBitsInSegment < 0) {
+      return Q_VALUE_INVALID;
+    }
 
     escape_word <<= 1;
     escape_word = escape_word | carryBit;
@@ -1423,11 +1429,12 @@ static INT DecodeEscapeSequence(HANDLE_FDK_BITSTREAM bs, const INT bsAnchor,
 
   sign = (quantSpecCoef >= 0) ? 1 : -1;
 
-  if (escapeOnesCounter > 30)
-    escapeOnesCounter = 30;
-
-  quantSpecCoef = sign * (((INT)1 << escapeOnesCounter) + escape_word);
-
+  if (escapeOnesCounter < 13) {
+    quantSpecCoef = sign * (((INT)1 << escapeOnesCounter) + escape_word);
+  } else {
+    *errorWord |= TOO_MANY_PCW_BODY_SIGN_ESC_BITS_DECODED;
+    quantSpecCoef = Q_VALUE_INVALID;
+  }
   return quantSpecCoef;
 }
 
diff --git a/fdk-aac/libAACdec/src/aacdec_hcrs.cpp b/fdk-aac/libAACdec/src/aacdec_hcrs.cpp
index d2bc867..44b32a5 100644
--- a/fdk-aac/libAACdec/src/aacdec_hcrs.cpp
+++ b/fdk-aac/libAACdec/src/aacdec_hcrs.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
@@ -1324,6 +1324,10 @@ UINT Hcr_State_BODY_SIGN_ESC__ESC_PREFIX(HANDLE_FDK_BITSTREAM bs, void *ptr) {
     /* count ones and store sum in escapePrefixUp */
     if (carryBit == 1) {
       escapePrefixUp += 1; /* update conter for ones */
+      if (escapePrefixUp > 8) {
+        pHcr->decInOut.errorLog |= STATE_ERROR_BODY_SIGN_ESC__ESC_PREFIX;
+        return BODY_SIGN_ESC__ESC_PREFIX;
+      }
 
       /* store updated counter in sideinfo of current codeword */
       pEscapeSequenceInfo[codewordOffset] &=
diff --git a/fdk-aac/libAACdec/src/aacdecoder.cpp b/fdk-aac/libAACdec/src/aacdecoder.cpp
index 7617937..965631b 100644
--- a/fdk-aac/libAACdec/src/aacdecoder.cpp
+++ b/fdk-aac/libAACdec/src/aacdecoder.cpp
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -1225,6 +1225,8 @@ static void CStreamInfoInit(CStreamInfo *pStreamInfo) {
   pStreamInfo->drcProgRefLev =
       -1; /* set program reference level to not indicated */
   pStreamInfo->drcPresMode = -1; /* default: presentation mode not indicated */
+
+  pStreamInfo->outputLoudness = -1; /* default: no loudness metadata present */
 }
 
 /*!
@@ -1279,6 +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;
@@ -1303,7 +1306,8 @@ static void CAacDecoder_DeInit(HANDLE_AACDECODER self,
                                const int subStreamIndex) {
   int ch;
   int aacChannelOffset = 0, aacChannels = (8);
-  int numElements = (((8)) + (8)), elementOffset = 0;
+  int numElements = (3 * ((8) * 2) + (((8) * 2)) / 2 + 4 * (1) + 1),
+      elementOffset = 0;
 
   if (self == NULL) return;
 
@@ -1453,6 +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);
@@ -1490,6 +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)
@@ -1847,6 +1857,12 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc,
           self->streamInfo.extSamplingRate / self->downscaleFactor;
     }
   }
+  if ((asc->m_aot == AOT_AAC_LC) && (asc->m_sbrPresentFlag == 1) &&
+      (asc->m_extensionSamplingFrequency > (2 * asc->m_samplingFrequency))) {
+    return AAC_DEC_UNSUPPORTED_SAMPLINGRATE; /* Core decoder supports at most a
+                                                1:2 upsampling for HE-AAC and
+                                                HE-AACv2 */
+  }
 
   /* --------- vcb11 ------------ */
   self->flags[streamIndex] |= (asc->m_vcb11Flag) ? AC_ER_VCB11 : 0;
@@ -1928,6 +1944,9 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc,
           self->samplingRateInfo[0].samplingRate / self->downscaleFactor;
       self->streamInfo.aacSamplesPerFrame =
           asc->m_samplesPerFrame / self->downscaleFactor;
+      if (self->streamInfo.aacSampleRate <= 0) {
+        return AAC_DEC_UNSUPPORTED_SAMPLINGRATE;
+      }
     }
   }
 
@@ -2362,6 +2381,13 @@ CAacDecoder_Init(HANDLE_AACDECODER self, const CSAudioSpecificConfig *asc,
       goto bail;
   }
 
+  if (*configChanged) {
+    if (asc->m_aot == AOT_USAC) {
+      self->hDrcInfo->enable = 0;
+      self->hDrcInfo->progRefLevelPresent = 0;
+    }
+  }
+
   if (asc->m_aot == AOT_USAC) {
     pcmLimiter_SetAttack(self->hLimiter, (5));
     pcmLimiter_SetThreshold(self->hLimiter, FL2FXCONST_DBL(0.89125094f));
@@ -2375,7 +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;
 
@@ -3151,11 +3177,6 @@ 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);
-
     /* deactivate legacy DRC in case uniDrc is active, i.e. uniDrc payload is
      * present and one of DRC or Loudness Normalization is switched on */
     aacDecoder_drcSetParam(
@@ -3168,9 +3189,15 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
         self->hDrcInfo, bs, self->pAacDecoderStaticChannelInfo,
         pce->ElementInstanceTag, drcChMap, aacChannels);
     if (mapped > 0) {
-      /* If at least one DRC thread has been mapped to a channel threre was DRC
-       * data in the bitstream. */
-      self->flags[streamIndex] |= AC_DRC_PRESENT;
+      if (!(self->flags[streamIndex] & (AC_USAC | AC_RSV603DA))) {
+        /* If at least one DRC thread has been mapped to a channel there was DRC
+         * data in the bitstream. */
+        self->flags[streamIndex] |= AC_DRC_PRESENT;
+      } else {
+        self->hDrcInfo->enable = 0;
+        self->hDrcInfo->progRefLevelPresent = 0;
+        ErrorStatus = AAC_DEC_UNSUPPORTED_FORMAT;
+      }
     }
 
     /* Create a reverse mapping table */
@@ -3300,9 +3327,11 @@ 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,
@@ -3318,7 +3347,7 @@ 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:
@@ -3330,7 +3359,7 @@ 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;
@@ -3351,7 +3380,7 @@ 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;
@@ -3363,7 +3392,8 @@ 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 */
@@ -3385,22 +3415,17 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
       if ((aacChannels == 2) && bsPseudoLr) {
         int i, offset2;
         const FIXP_SGL invSqrt2 = FL2FXCONST_SGL(0.707106781186547f);
-        FIXP_PCM *pTD = pTimeData;
+        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] = FX_DBL2FX_PCM(fAddSaturate(L + R, (FIXP_DBL)0x8000));
-          pTD[offset2] = FX_DBL2FX_PCM(fAddSaturate(L - R, (FIXP_DBL)0x8000));
-#else
-          pTD[0] = FX_DBL2FX_PCM(L + R);
-          pTD[offset2] = FX_DBL2FX_PCM(L - R);
-#endif
+          pTD[0] = L + R;
+          pTD[offset2] = L - R;
           pTD++;
         }
       }
@@ -3411,9 +3436,15 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
         self->hDrcInfo, bs, self->pAacDecoderStaticChannelInfo,
         pce->ElementInstanceTag, drcChMap, aacChannels);
     if (mapped > 0) {
-      /* If at least one DRC thread has been mapped to a channel threre was DRC
-       * data in the bitstream. */
-      self->flags[streamIndex] |= AC_DRC_PRESENT;
+      if (!(self->flags[streamIndex] & (AC_USAC | AC_RSV603DA))) {
+        /* If at least one DRC thread has been mapped to a channel there was DRC
+         * data in the bitstream. */
+        self->flags[streamIndex] |= AC_DRC_PRESENT;
+      } else {
+        self->hDrcInfo->enable = 0;
+        self->hDrcInfo->progRefLevelPresent = 0;
+        ErrorStatus = AAC_DEC_UNSUPPORTED_FORMAT;
+      }
     }
   }
 
diff --git a/fdk-aac/libAACdec/src/aacdecoder.h b/fdk-aac/libAACdec/src/aacdecoder.h
index 20f4c45..bd1f38f 100644
--- a/fdk-aac/libAACdec/src/aacdecoder.h
+++ b/fdk-aac/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/fdk-aac/libAACdec/src/aacdecoder_lib.cpp b/fdk-aac/libAACdec/src/aacdecoder_lib.cpp
index 8685a30..bcbd46c 100644
--- a/fdk-aac/libAACdec/src/aacdecoder_lib.cpp
+++ b/fdk-aac/libAACdec/src/aacdecoder_lib.cpp
@@ -119,10 +119,10 @@ amm-info@iis.fraunhofer.de
 
 /* Decoder library info */
 #define AACDECODER_LIB_VL0 3
-#define AACDECODER_LIB_VL1 1
-#define AACDECODER_LIB_VL2 2
+#define AACDECODER_LIB_VL1 2
+#define AACDECODER_LIB_VL2 0
 #define AACDECODER_LIB_TITLE "AAC Decoder Lib"
-#ifdef __ANDROID__
+#ifdef SUPPRESS_BUILD_DATE_INFO
 #define AACDECODER_LIB_BUILD_DATE ""
 #define AACDECODER_LIB_BUILD_TIME ""
 #else
@@ -1133,35 +1133,31 @@ 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;
   }
@@ -1271,9 +1267,9 @@ aacDecoder_DecodeFrame(HANDLE_AACDECODER self, INT_PCM *pTimeData_extern,
     }
 
     /* Signal bit stream interruption to other modules if required. */
-    if (fTpInterruption || (flags & AACDEC_INTR)) {
+    if (fTpInterruption || ((flags & AACDEC_INTR) && (accessUnit == 0))) {
       aacDecoder_SignalInterruption(self);
-      if (!(flags & AACDEC_INTR)) {
+      if (!((flags & AACDEC_INTR) && (accessUnit == 0))) {
         ErrorStatus = AAC_DEC_TRANSPORT_SYNC_ERROR;
         goto bail;
       }
@@ -1317,19 +1313,23 @@ 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 */
@@ -1354,7 +1354,7 @@ 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;
@@ -1368,10 +1368,10 @@ 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
@@ -1418,8 +1418,6 @@ 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:
@@ -1476,18 +1474,18 @@ 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
@@ -1566,10 +1564,11 @@ 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) {
@@ -1592,8 +1591,8 @@ 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;
@@ -1614,11 +1613,11 @@ 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
@@ -1646,17 +1645,15 @@ 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);
+          }
         }
       }
 
@@ -1711,22 +1708,21 @@ 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 */
@@ -1761,16 +1757,51 @@ 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);
           }
         }
       }
+      if (FDK_drcDec_GetParam(self->hUniDrcDecoder, DRC_DEC_IS_ACTIVE)) {
+        /* return output loudness information for MPEG-D DRC */
+        LONG outputLoudness =
+            FDK_drcDec_GetParam(self->hUniDrcDecoder, DRC_DEC_OUTPUT_LOUDNESS);
+        if (outputLoudness == DRC_DEC_LOUDNESS_NOT_PRESENT) {
+          /* no valid MPEG-D DRC loudness value contained */
+          self->streamInfo.outputLoudness = -1;
+        } else {
+          if (outputLoudness > 0) {
+            /* positive output loudness values (very unusual) are limited to 0
+             * dB */
+            self->streamInfo.outputLoudness = 0;
+          } else {
+            self->streamInfo.outputLoudness =
+                -(INT)outputLoudness >>
+                22; /* negate and scale from e = 7 to e = (31-2) */
+          }
+        }
+      } else {
+        /* return output loudness information for MPEG-4 DRC */
+        if (self->streamInfo.drcProgRefLev <
+            0) { /* no MPEG-4 DRC loudness metadata contained */
+          self->streamInfo.outputLoudness = -1;
+        } else {
+          if (self->defaultTargetLoudness <
+              0) { /* loudness normalization is off */
+            self->streamInfo.outputLoudness = self->streamInfo.drcProgRefLev;
+          } else {
+            self->streamInfo.outputLoudness = self->defaultTargetLoudness;
+          }
+        }
+      }
 
       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)) {
+        if ((flags & AACDEC_INTR) && (accessUnit == 0)) {
           /* delete data from the past (e.g. mixdown coeficients) */
           pcmDmx_Reset(self->hPcmUtils, PCMDMX_RESET_BS_DATA);
         }
@@ -1781,17 +1812,12 @@ 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;
@@ -1803,13 +1829,35 @@ 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;
@@ -1818,44 +1866,72 @@ 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,
@@ -1951,20 +2027,8 @@ 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/fdk-aac/libAACdec/src/block.cpp b/fdk-aac/libAACdec/src/block.cpp
index b3d09a6..0bca577 100644
--- a/fdk-aac/libAACdec/src/block.cpp
+++ b/fdk-aac/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/fdk-aac/libAACdec/src/block.h b/fdk-aac/libAACdec/src/block.h
index f0f56cd..f5118a2 100644
--- a/fdk-aac/libAACdec/src/block.h
+++ b/fdk-aac/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/fdk-aac/libAACdec/src/conceal.cpp b/fdk-aac/libAACdec/src/conceal.cpp
index 5895cb8..0939bb5 100644
--- a/fdk-aac/libAACdec/src/conceal.cpp
+++ b/fdk-aac/libAACdec/src/conceal.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
@@ -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;
@@ -1228,7 +1230,6 @@ static void CConcealment_InterpolateBuffer(FIXP_DBL *spectrum,
   int sfb, line = 0;
   int fac_shift;
   int fac_mod;
-  FIXP_DBL accu;
 
   for (sfb = 0; sfb < sfbCnt; sfb++) {
     fac_shift =
@@ -1236,15 +1237,11 @@ static void CConcealment_InterpolateBuffer(FIXP_DBL *spectrum,
     fac_mod = fac_shift & 3;
     fac_shift = (fac_shift >> 2) + 1;
     fac_shift += *pSpecScalePrv - fixMax(*pSpecScalePrv, *pSpecScaleAct);
+    fac_shift = fMax(fMin(fac_shift, DFRACT_BITS - 1), -(DFRACT_BITS - 1));
 
     for (; line < pSfbOffset[sfb + 1]; line++) {
-      accu = fMult(*(spectrum + line), facMod4Table[fac_mod]);
-      if (fac_shift < 0) {
-        accu >>= -fac_shift;
-      } else {
-        accu <<= fac_shift;
-      }
-      *(spectrum + line) = accu;
+      FIXP_DBL accu = fMult(*(spectrum + line), facMod4Table[fac_mod]);
+      *(spectrum + line) = scaleValue(accu, fac_shift);
     }
   }
   *pSpecScaleOut = fixMax(*pSpecScalePrv, *pSpecScaleAct);
@@ -1618,7 +1615,7 @@ static void CConcealment_ApplyRandomSign(int randomPhase, FIXP_DBL *spec,
     }
 
     if (packedSign & 0x1) {
-      spec[i] = -spec[i];
+      spec[i] = -fMax(spec[i], (FIXP_DBL)(MINVAL_DBL + 1));
     }
     packedSign >>= 1;
 
@@ -1849,7 +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
   */
@@ -1962,7 +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;
@@ -1974,12 +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;
@@ -1992,7 +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));
   }
 }
 
@@ -2055,9 +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;
@@ -2075,18 +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/fdk-aac/libAACdec/src/conceal.h b/fdk-aac/libAACdec/src/conceal.h
index e01a796..0c002a5 100644
--- a/fdk-aac/libAACdec/src/conceal.h
+++ b/fdk-aac/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/fdk-aac/libAACdec/src/conceal_types.h b/fdk-aac/libAACdec/src/conceal_types.h
index d90374e..36e7dec 100644
--- a/fdk-aac/libAACdec/src/conceal_types.h
+++ b/fdk-aac/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/fdk-aac/libAACdec/src/ldfiltbank.cpp b/fdk-aac/libAACdec/src/ldfiltbank.cpp
index c7d2928..13e61a5 100644
--- a/fdk-aac/libAACdec/src/ldfiltbank.cpp
+++ b/fdk-aac/libAACdec/src/ldfiltbank.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
@@ -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
@@ -131,24 +134,26 @@ static void multE2_DinvF_fdk(FIXP_PCM *output, FIXP_DBL *x, const FIXP_WTB *fb,
     FIXP_DBL z0, z2, tmp;
 
     z2 = x[N / 2 + i];
-    z0 = z2 + (fMultDiv2(z[N / 2 + i], fb[2 * N + i]) >> (-WTS2 - 1));
+    z0 = fAddSaturate(z2,
+                      (fMultDiv2(z[N / 2 + i], fb[2 * N + i]) >> (-WTS2 - 1)));
 
-    z[N / 2 + i] = x[N / 2 - 1 - i] +
-                   (fMultDiv2(z[N + i], fb[2 * N + N / 2 + i]) >> (-WTS2 - 1));
+    z[N / 2 + i] = fAddSaturate(
+        x[N / 2 - 1 - i],
+        (fMultDiv2(z[N + i], fb[2 * N + N / 2 + i]) >> (-WTS2 - 1)));
 
     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;
@@ -159,32 +164,34 @@ static void multE2_DinvF_fdk(FIXP_PCM *output, FIXP_DBL *x, const FIXP_WTB *fb,
     FIXP_DBL z0, z2, tmp0, tmp1;
 
     z2 = x[N / 2 + i];
-    z0 = z2 + (fMultDiv2(z[N / 2 + i], fb[2 * N + i]) >> (-WTS2 - 1));
+    z0 = fAddSaturate(z2,
+                      (fMultDiv2(z[N / 2 + i], fb[2 * N + i]) >> (-WTS2 - 1)));
 
-    z[N / 2 + i] = x[N / 2 - 1 - i] +
-                   (fMultDiv2(z[N + i], fb[2 * N + N / 2 + i]) >> (-WTS2 - 1));
+    z[N / 2 + i] = fAddSaturate(
+        x[N / 2 - 1 - i],
+        (fMultDiv2(z[N + i], fb[2 * N + N / 2 + i]) >> (-WTS2 - 1)));
 
     tmp0 = (fMultDiv2(z[N / 2 + i], fb[N / 2 - 1 - i]) +
             fMultDiv2(z[i], fb[N / 2 + i]));
     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;
@@ -194,22 +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/fdk-aac/libAACdec/src/ldfiltbank.h b/fdk-aac/libAACdec/src/ldfiltbank.h
index b63da6b..02971d0 100644
--- a/fdk-aac/libAACdec/src/ldfiltbank.h
+++ b/fdk-aac/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/fdk-aac/libAACdec/src/stereo.cpp b/fdk-aac/libAACdec/src/stereo.cpp
index eed826b..47f1a31 100644
--- a/fdk-aac/libAACdec/src/stereo.cpp
+++ b/fdk-aac/libAACdec/src/stereo.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
@@ -807,19 +807,17 @@ void CJointStereo_ApplyMS(
                   for (int i = 0; i < windowLen; i++) {
                     dmx_re_prev[i] =
                         ((staticSpectralCoeffsL[index_offset + i] >>
-                          srLeftChan) +
+                          fMin(DFRACT_BITS - 1, srLeftChan + 1)) +
                          (staticSpectralCoeffsR[index_offset + i] >>
-                          srRightChan)) >>
-                        1;
+                          fMin(DFRACT_BITS - 1, srRightChan + 1)));
                   }
                 } else {
                   for (int i = 0; i < windowLen; i++) {
                     dmx_re_prev[i] =
                         ((staticSpectralCoeffsL[index_offset + i] >>
-                          srLeftChan) -
+                          fMin(DFRACT_BITS - 1, srLeftChan + 1)) -
                          (staticSpectralCoeffsR[index_offset + i] >>
-                          srRightChan)) >>
-                        1;
+                          fMin(DFRACT_BITS - 1, srRightChan + 1)));
                   }
                 }
               }
@@ -854,12 +852,13 @@ void CJointStereo_ApplyMS(
           if (window == 0) {
             if (dmx_re_prev_e < frameMaxScale) {
               if (mainband_flag == 0) {
-                scaleValues(dmx_re_prev, store_dmx_re_prev, windowLen,
-                            -(frameMaxScale - dmx_re_prev_e));
+                scaleValues(
+                    dmx_re_prev, store_dmx_re_prev, windowLen,
+                    -fMin(DFRACT_BITS - 1, (frameMaxScale - dmx_re_prev_e)));
               } else {
-                for (int i = 0; i < windowLen; i++) {
-                  dmx_re_prev[i] >>= (frameMaxScale - dmx_re_prev_e);
-                }
+                scaleValues(
+                    dmx_re_prev, windowLen,
+                    -fMin(DFRACT_BITS - 1, (frameMaxScale - dmx_re_prev_e)));
               }
             } else {
               if (mainband_flag == 0) {
@@ -873,10 +872,9 @@ void CJointStereo_ApplyMS(
             FDK_ASSERT(pAacDecoderChannelInfo[L]->icsInfo.WindowSequence ==
                        BLOCK_SHORT);
             if (specScaleL[window - 1] < frameMaxScale) {
-              for (int i = 0; i < windowLen; i++) {
-                dmx_re[windowLen * (window - 1) + i] >>=
-                    (frameMaxScale - specScaleL[window - 1]);
-              }
+              scaleValues(&dmx_re[windowLen * (window - 1)], windowLen,
+                          -fMin(DFRACT_BITS - 1,
+                                (frameMaxScale - specScaleL[window - 1])));
             } else {
               specScaleL[window] = specScaleL[window - 1];
               specScaleR[window] = specScaleR[window - 1];
@@ -991,7 +989,7 @@ void CJointStereo_ApplyMS(
         }   /* if ( pJointStereoData->complex_coef == 1 ) */
 
         /* 4. upmix process */
-        INT pred_dir = cplxPredictionData->pred_dir ? -1 : 1;
+        LONG pred_dir = cplxPredictionData->pred_dir ? -1 : 1;
         /* 0.1 in Q-3.34 */
         const FIXP_DBL pointOne = 0x66666666; /* 0.8 */
         /* Shift value for the downmix */
@@ -1041,34 +1039,24 @@ void CJointStereo_ApplyMS(
                 the downmix. "dmx_re" and "specL" are two different pointers
                 pointing to separate arrays, which may or may not contain the
                 same data (with different scaling).
-              */
-
-              /* help1: alpha_re[i] * dmx_re[i] */
-              FIXP_DBL help1 = fMultDiv2(alpha_re_tmp, *p2dmxRe++);
 
-              /* tmp: dmx_im[i] */
-              FIXP_DBL tmp = (*p2dmxIm++) << shift_dmx;
-
-              /* help2: alpha_im[i] * dmx_im[i] */
-              FIXP_DBL help2 = fMultDiv2(alpha_im_tmp, tmp);
-
-              /* help3: alpha_re[i] * dmx_re[i] + alpha_im[i] * dmx_im[i] */
-              FIXP_DBL help3 = help1 + help2;
+                specL[i] =   + (specL[i] + side);
+                specR[i] = -/+ (specL[i] - side);
+              */
+              FIXP_DBL side, left, right;
 
-              /* side (= help4) = specR[i] - (dmx_re[i] * specL[i] + alpha_im[i]
-               * * dmx_im[i]) */
-              FIXP_DBL help4 = *p2CoeffR - scaleValue(help3, help3_shift);
+              side = fMultAddDiv2(fMultDiv2(alpha_re_tmp, *p2dmxRe++),
+                                  alpha_im_tmp, (*p2dmxIm++) << shift_dmx);
+              side = ((*p2CoeffR) >> 2) -
+                     (FIXP_DBL)SATURATE_SHIFT(side, -(help3_shift - 2),
+                                              DFRACT_BITS - 2);
 
-              /* We calculate the left and right output by using the helper
-               * function */
-              /* specR[i] = -/+ (specL[i] - side); */
-              *p2CoeffR =
-                  (FIXP_DBL)((LONG)(*p2CoeffL - help4) * (LONG)pred_dir);
-              p2CoeffR++;
+              left = ((*p2CoeffL) >> 2) + side;
+              right = ((*p2CoeffL) >> 2) - side;
+              right = (FIXP_DBL)((LONG)right * pred_dir);
 
-              /* specL[i] = specL[i] + side; */
-              *p2CoeffL = *p2CoeffL + help4;
-              p2CoeffL++;
+              *p2CoeffL++ = SATURATE_LEFT_SHIFT_ALT(left, 2, DFRACT_BITS);
+              *p2CoeffR++ = SATURATE_LEFT_SHIFT_ALT(right, 2, DFRACT_BITS);
             }
           }
 
diff --git a/fdk-aac/libAACdec/src/usacdec_acelp.cpp b/fdk-aac/libAACdec/src/usacdec_acelp.cpp
index a606459..fbe3188 100644
--- a/fdk-aac/libAACdec/src/usacdec_acelp.cpp
+++ b/fdk-aac/libAACdec/src/usacdec_acelp.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
@@ -131,7 +131,7 @@ void E_UTIL_preemph(const FIXP_DBL *in, FIXP_DBL *out, INT L) {
   int i;
 
   for (i = 0; i < L; i++) {
-    out[i] = in[i] - fMult(PREEMPH_FAC, in[i - 1]);
+    out[i] = fAddSaturate(in[i], -fMult(PREEMPH_FAC, in[i - 1]));
   }
 
   return;
@@ -465,7 +465,9 @@ void BuildAdaptiveExcitation(
 /* Note: code[L_SUBFR] and exc2[L_SUBFR] share the same memory!
          If exc2[i] is written, code[i] will be destroyed!
 */
-#define SF (SF_CODE + SF_GAIN_C + 1 - SF_EXC)
+#define SF_HEADROOM (1)
+#define SF (SF_CODE + SF_GAIN_C + 1 - SF_EXC - SF_HEADROOM)
+#define SF_GAIN_P2 (SF_GAIN_P - SF_HEADROOM)
 
   int i;
   FIXP_DBL tmp, cpe, code_smooth_prev, code_smooth;
@@ -477,8 +479,8 @@ void BuildAdaptiveExcitation(
   cpe = (period_fac >> (2 - SF_PFAC)) + FL2FXCONST_DBL(0.25f);
 
   /* u'(n) */
-  tmp = fMultDiv2(*exc, gain_pit) << (SF_GAIN_P + 1); /* v(0)*g_p */
-  *exc++ = tmp + (fMultDiv2(code[0], gain_code) << SF);
+  tmp = fMultDiv2(*exc, gain_pit) << (SF_GAIN_P2 + 1); /* v(0)*g_p */
+  *exc++ = (tmp + (fMultDiv2(code[0], gain_code) << SF)) << SF_HEADROOM;
 
   /* u(n) */
   code_smooth_prev = fMultDiv2(*code++, gain_code_smoothed)
@@ -487,15 +489,15 @@ void BuildAdaptiveExcitation(
   code_smooth = fMultDiv2(code_i, gain_code_smoothed) << SF; /* c(1) * g_sc */
   tmp += code_smooth_prev; /* tmp = v(0)*g_p + c(0)*g_sc */
   cpe_code_smooth = fMultDiv2(cpe, code_smooth);
-  *exc2++ = tmp - cpe_code_smooth;
+  *exc2++ = (tmp - cpe_code_smooth) << SF_HEADROOM;
   cpe_code_smooth_prev = fMultDiv2(cpe, code_smooth_prev);
 
   i = L_SUBFR - 2;
   do /* ARM926: 22 cycles per iteration */
   {
     /* u'(n) */
-    tmp = fMultDiv2(*exc, gain_pit) << (SF_GAIN_P + 1);
-    *exc++ = tmp + (fMultDiv2(code_i, gain_code) << SF);
+    tmp = fMultDiv2(*exc, gain_pit) << (SF_GAIN_P2 + 1);
+    *exc++ = (tmp + (fMultDiv2(code_i, gain_code) << SF)) << SF_HEADROOM;
     /* u(n) */
     tmp += code_smooth; /* += g_sc * c(i) */
     tmp -= cpe_code_smooth_prev;
@@ -503,16 +505,17 @@ void BuildAdaptiveExcitation(
     code_i = *code++;
     code_smooth = fMultDiv2(code_i, gain_code_smoothed) << SF;
     cpe_code_smooth = fMultDiv2(cpe, code_smooth);
-    *exc2++ = tmp - cpe_code_smooth; /* tmp - c_pe * g_sc * c(i+1) */
+    *exc2++ = (tmp - cpe_code_smooth)
+              << SF_HEADROOM; /* tmp - c_pe * g_sc * c(i+1) */
   } while (--i != 0);
 
   /* u'(n) */
-  tmp = fMultDiv2(*exc, gain_pit) << (SF_GAIN_P + 1);
-  *exc = tmp + (fMultDiv2(code_i, gain_code) << SF);
+  tmp = fMultDiv2(*exc, gain_pit) << (SF_GAIN_P2 + 1);
+  *exc = (tmp + (fMultDiv2(code_i, gain_code) << SF)) << SF_HEADROOM;
   /* u(n) */
   tmp += code_smooth;
   tmp -= cpe_code_smooth_prev;
-  *exc2++ = tmp;
+  *exc2++ = tmp << SF_HEADROOM;
 
   return;
 }
diff --git a/fdk-aac/libAACdec/src/usacdec_fac.cpp b/fdk-aac/libAACdec/src/usacdec_fac.cpp
index 0d3d844..b246171 100644
--- a/fdk-aac/libAACdec/src/usacdec_fac.cpp
+++ b/fdk-aac/libAACdec/src/usacdec_fac.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
@@ -344,7 +344,7 @@ INT CLpd_FAC_Mdct2Acelp(H_MDCT hMdct, FIXP_DBL *output, FIXP_DBL *pFac,
       /* Overlap Add */
       x0 = -fMult(*pOvl--, pWindow[i].v.re);
 
-      *pOut0 += IMDCT_SCALE_DBL(x0);
+      *pOut0 = fAddSaturate(*pOut0, IMDCT_SCALE_DBL(x0));
       pOut0++;
     }
   } else {
@@ -354,7 +354,7 @@ INT CLpd_FAC_Mdct2Acelp(H_MDCT hMdct, FIXP_DBL *output, FIXP_DBL *pFac,
       /* Overlap Add */
       x0 = fMult(*pOvl--, pWindow[i].v.re);
 
-      *pOut0 += IMDCT_SCALE_DBL(x0);
+      *pOut0 = fAddSaturate(*pOut0, IMDCT_SCALE_DBL(x0));
       pOut0++;
     }
   }
@@ -362,7 +362,7 @@ INT CLpd_FAC_Mdct2Acelp(H_MDCT hMdct, FIXP_DBL *output, FIXP_DBL *pFac,
       0) { /* this should only happen for ACELP -> TCX20 -> ACELP transition */
     FIXP_DBL *pOut = pOut0 - fl / 2; /* fl/2 == fac_length */
     for (i = 0; i < fl / 2; i++) {
-      pOut[i] += IMDCT_SCALE_DBL(hMdct->pFacZir[i]);
+      pOut[i] = fAddSaturate(pOut[i], IMDCT_SCALE_DBL(hMdct->pFacZir[i]));
     }
     hMdct->pFacZir = NULL;
   }
@@ -493,9 +493,7 @@ INT CLpd_FAC_Acelp2Mdct(H_MDCT hMdct, FIXP_DBL *output, FIXP_DBL *_pSpec,
           /* Div2 is compensated by table scaling */
           x = fMultDiv2(pTmp2[i], FacWindowZir[w]);
           x += fMultDiv2(pTmp1[-i - 1], FacWindowSynth[w]);
-          x += pFAC_and_FAC_ZIR[i];
-          pOut1[i] = x;
-
+          pOut1[i] = fAddSaturate(x, pFAC_and_FAC_ZIR[i]);
           w++;
         }
       }
@@ -552,7 +550,7 @@ INT CLpd_FAC_Acelp2Mdct(H_MDCT hMdct, FIXP_DBL *output, FIXP_DBL *_pSpec,
     FDK_ASSERT((pOut1 >= hMdct->overlap.time &&
                 pOut1 < hMdct->overlap.time + hMdct->ov_size) ||
                (pOut1 >= output && pOut1 < output + 1024));
-    *pOut1 += IMDCT_SCALE_DBL(-x1);
+    *pOut1 = fAddSaturate(*pOut1, IMDCT_SCALE_DBL(-x1));
     pOut1--;
   }
 
@@ -578,7 +576,7 @@ INT CLpd_FAC_Acelp2Mdct(H_MDCT hMdct, FIXP_DBL *output, FIXP_DBL *_pSpec,
     FIXP_DBL x = -(*pCurr--);
     /* 5) (item 4) Synthesis filter Zir component, FAC ZIR (another one). */
     if (i < f_len) {
-      x += *pF++;
+      x = fAddSaturate(x, *pF++);
     }
 
     FDK_ASSERT((pOut1 >= hMdct->overlap.time &&
@@ -668,9 +666,9 @@ INT CLpd_FAC_Acelp2Mdct(H_MDCT hMdct, FIXP_DBL *output, FIXP_DBL *_pSpec,
       for (i = 0; i < fl / 2; i++) {
         FIXP_DBL x0, x1;
 
-        cplxMult(&x1, &x0, *pCurr++, -*pOvl--, pWindow_prev[i]);
-        *pOut0 = IMDCT_SCALE_DBL(x0);
-        *pOut1 = IMDCT_SCALE_DBL(-x1);
+        cplxMultDiv2(&x1, &x0, *pCurr++, -*pOvl--, pWindow_prev[i]);
+        *pOut0 = IMDCT_SCALE_DBL_LSH1(x0);
+        *pOut1 = IMDCT_SCALE_DBL_LSH1(-x1);
         pOut0++;
         pOut1--;
       }
@@ -680,9 +678,9 @@ INT CLpd_FAC_Acelp2Mdct(H_MDCT hMdct, FIXP_DBL *output, FIXP_DBL *_pSpec,
         for (i = 0; i < fl / 2; i++) {
           FIXP_DBL x0, x1;
 
-          cplxMult(&x1, &x0, *pCurr++, -*pOvl--, pWindow_prev[i]);
-          *pOut0 = IMDCT_SCALE_DBL(x0);
-          *pOut1 = IMDCT_SCALE_DBL(x1);
+          cplxMultDiv2(&x1, &x0, *pCurr++, -*pOvl--, pWindow_prev[i]);
+          *pOut0 = IMDCT_SCALE_DBL_LSH1(x0);
+          *pOut1 = IMDCT_SCALE_DBL_LSH1(x1);
           pOut0++;
           pOut1--;
         }
@@ -691,9 +689,9 @@ INT CLpd_FAC_Acelp2Mdct(H_MDCT hMdct, FIXP_DBL *output, FIXP_DBL *_pSpec,
         for (i = 0; i < fl / 2; i++) {
           FIXP_DBL x0, x1;
 
-          cplxMult(&x1, &x0, *pCurr++, *pOvl--, pWindow_prev[i]);
-          *pOut0 = IMDCT_SCALE_DBL(x0);
-          *pOut1 = IMDCT_SCALE_DBL(x1);
+          cplxMultDiv2(&x1, &x0, *pCurr++, *pOvl--, pWindow_prev[i]);
+          *pOut0 = IMDCT_SCALE_DBL_LSH1(x0);
+          *pOut1 = IMDCT_SCALE_DBL_LSH1(x1);
           pOut0++;
           pOut1--;
         }
@@ -705,7 +703,7 @@ INT CLpd_FAC_Acelp2Mdct(H_MDCT hMdct, FIXP_DBL *output, FIXP_DBL *_pSpec,
       FIXP_DBL *pOut = pOut0 - fl / 2;
       FDK_ASSERT(fl / 2 <= 128);
       for (i = 0; i < fl / 2; i++) {
-        pOut[i] += IMDCT_SCALE_DBL(hMdct->pFacZir[i]);
+        pOut[i] = fAddSaturate(pOut[i], IMDCT_SCALE_DBL(hMdct->pFacZir[i]));
       }
       hMdct->pFacZir = NULL;
     }
diff --git a/fdk-aac/libAACdec/src/usacdec_lpc.cpp b/fdk-aac/libAACdec/src/usacdec_lpc.cpp
index 271463f..88601b7 100644
--- a/fdk-aac/libAACdec/src/usacdec_lpc.cpp
+++ b/fdk-aac/libAACdec/src/usacdec_lpc.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
@@ -231,7 +231,7 @@ void nearest_neighbor_2D8(FIXP_ZF x[8], int y[8]) {
 void RE8_PPV(FIXP_ZF x[], SHORT y[], int r) {
   int i, y0[8], y1[8];
   FIXP_ZF x1[8], tmp;
-  FIXP_DBL e;
+  INT64 e;
 
   /* find the nearest neighbor y0 of x in 2D8 */
   nearest_neighbor_2D8(x, y0);
@@ -245,16 +245,16 @@ void RE8_PPV(FIXP_ZF x[], SHORT y[], int r) {
   }
 
   /* compute e0=||x-y0||^2 and e1=||x-y1||^2 */
-  e = (FIXP_DBL)0;
+  e = 0;
   for (i = 0; i < 8; i++) {
     tmp = x[i] - INT2ZF(y0[i], 0);
-    e += fPow2Div2(
+    e += (INT64)fPow2Div2(
         tmp << r); /* shift left to ensure that no fract part bits get lost. */
     tmp = x[i] - INT2ZF(y1[i], 0);
-    e -= fPow2Div2(tmp << r);
+    e -= (INT64)fPow2Div2(tmp << r);
   }
   /* select best candidate y0 or y1 to minimize distortion */
-  if (e < (FIXP_DBL)0) {
+  if (e < 0) {
     for (i = 0; i < 8; i++) {
       y[i] = y0[i];
     }
@@ -565,7 +565,8 @@ static void lsf_weight_2st(FIXP_LPC *lsfq, FIXP_DBL *xq, int nk_mode) {
   /* add non-weighted residual LSF vector to LSF1st */
   for (i = 0; i < M_LP_FILTER_ORDER; i++) {
     w = (LONG)fMultDiv2(factor, sqrtFixp(fMult(d[i], d[i + 1])));
-    lsfq[i] = fAddSaturate(lsfq[i], FX_DBL2FX_LPC((FIXP_DBL)(w * (LONG)xq[i])));
+    lsfq[i] = fAddSaturate(lsfq[i],
+                           FX_DBL2FX_LPC((FIXP_DBL)((INT64)w * (LONG)xq[i])));
   }
 
   return;
@@ -1138,9 +1139,12 @@ static void get_lsppol(FIXP_LPC lsp[], FIXP_DBL f[], int n, int flag) {
   for (i = 2; i <= n; i++) {
     plsp += 2;
     b = -FX_LPC2FX_DBL(*plsp);
-    f[i] = ((fMultDiv2(b, f[i - 1]) << 1) + (f[i - 2])) << 1;
+    f[i] = SATURATE_LEFT_SHIFT((fMultDiv2(b, f[i - 1]) + (f[i - 2] >> 1)), 2,
+                               DFRACT_BITS);
     for (j = i - 1; j > 1; j--) {
-      f[j] = f[j] + (fMultDiv2(b, f[j - 1]) << 2) + f[j - 2];
+      f[j] = SATURATE_LEFT_SHIFT(
+          ((f[j] >> 2) + fMultDiv2(b, f[j - 1]) + (f[j - 2] >> 2)), 2,
+          DFRACT_BITS);
     }
     f[1] = f[1] + (b >> (SF_F - 1));
   }
@@ -1167,6 +1171,9 @@ void E_LPC_f_lsp_a_conversion(FIXP_LPC *lsp, FIXP_LPC *a, INT *a_exp) {
   /*-----------------------------------------------------*
    *  Multiply F1(z) by (1+z^-1) and F2(z) by (1-z^-1)   *
    *-----------------------------------------------------*/
+  scaleValues(f1, NC + 1, -2);
+  scaleValues(f2, NC + 1, -2);
+
   for (i = NC; i > 0; i--) {
     f1[i] += f1[i - 1];
     f2[i] -= f2[i - 1];
@@ -1175,13 +1182,8 @@ void E_LPC_f_lsp_a_conversion(FIXP_LPC *lsp, FIXP_LPC *a, INT *a_exp) {
   FIXP_DBL aDBL[M_LP_FILTER_ORDER];
 
   for (i = 1, k = M_LP_FILTER_ORDER - 1; i <= NC; i++, k--) {
-    FIXP_DBL tmp1, tmp2;
-
-    tmp1 = f1[i] >> 1;
-    tmp2 = f2[i] >> 1;
-
-    aDBL[i - 1] = (tmp1 + tmp2);
-    aDBL[k] = (tmp1 - tmp2);
+    aDBL[i - 1] = f1[i] + f2[i];
+    aDBL[k] = f1[i] - f2[i];
   }
 
   int headroom_a = getScalefactor(aDBL, M_LP_FILTER_ORDER);
@@ -1190,5 +1192,5 @@ void E_LPC_f_lsp_a_conversion(FIXP_LPC *lsp, FIXP_LPC *a, INT *a_exp) {
     a[i] = FX_DBL2FX_LPC(aDBL[i] << headroom_a);
   }
 
-  *a_exp = 8 - headroom_a;
+  *a_exp = SF_F + (2 - 1) - headroom_a;
 }
diff --git a/fdk-aac/libAACdec/src/usacdec_lpd.cpp b/fdk-aac/libAACdec/src/usacdec_lpd.cpp
index e0a2631..fbf6fab 100644
--- a/fdk-aac/libAACdec/src/usacdec_lpd.cpp
+++ b/fdk-aac/libAACdec/src/usacdec_lpd.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
@@ -122,17 +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 += fMultDiv2((noise[i - j] + noise[i + j]), filt[j]);
+      tmp += fMult((noise[i - j] >> 1) + (noise[i + j] >> 1), filt[j]);
     }
-    syn_out[i] = (FIXP_PCM)(IMDCT_SCALE(syn[i] - tmp));
+    syn_out[i] = (PCM_DEC)(
+        IMDCT_SCALE((syn[i] >> 1) - (tmp >> 1), aacOutDataHeadroom - 1));
   }
 }
 
@@ -142,8 +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;
@@ -335,17 +341,22 @@ void bass_pf_1sf_delay(
 
       {
         for (i = 0; i < lg; i++) {
-          /* scaled with SF_SYNTH + gain_sf + 1 */
+          /* scaled with SF_SYNTH + gain_sf + 1; composition of scalefactor 2:
+           * one additional shift of syn values + fMult => fMultDiv2 */
           noise_in[i] =
-              (fMult(gainSGL, syn[i + i_subfr] - (syn[i + i_subfr - T] >> 1) -
-                                  (syn[i + i_subfr + T] >> 1))) >>
-              s1;
+              scaleValue(fMultDiv2(gainSGL, (syn[i + i_subfr] >> 1) -
+                                                (syn[i + i_subfr - T] >> 2) -
+                                                (syn[i + i_subfr + T] >> 2)),
+                         2 - s1);
         }
 
         for (i = lg; i < L_SUBFR; i++) {
-          /* scaled with SF_SYNTH + gain_sf + 1 */
+          /* scaled with SF_SYNTH + gain_sf + 1; composition of scalefactor 2:
+           * one additional shift of syn values + fMult => fMultDiv2 */
           noise_in[i] =
-              (fMult(gainSGL, syn[i + i_subfr] - syn[i + i_subfr - T])) >> s1;
+              scaleValue(fMultDiv2(gainSGL, (syn[i + i_subfr] >> 1) -
+                                                (syn[i + i_subfr - T] >> 1)),
+                         2 - s1);
         }
       }
     } else {
@@ -364,7 +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);
     }
   }
 
@@ -377,9 +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;
@@ -1222,7 +1233,7 @@ AAC_DECODER_ERROR CLpdChannelStream_Read(
       (INT)(samplingRate * PIT_MIN_12k8 + (FSCALE_DENOM / 2)) / FSCALE_DENOM -
       (INT)PIT_MIN_12k8;
 
-  if ((samplingRate < 6000) || (samplingRate > 24000)) {
+  if ((samplingRate < FAC_FSCALE_MIN) || (samplingRate > FAC_FSCALE_MAX)) {
     error = AAC_DEC_PARSE_ERROR;
     goto bail;
   }
@@ -1546,9 +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;
@@ -2011,7 +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/fdk-aac/libAACdec/src/usacdec_lpd.h b/fdk-aac/libAACdec/src/usacdec_lpd.h
index 3e7938d..448dc55 100644
--- a/fdk-aac/libAACdec/src/usacdec_lpd.h
+++ b/fdk-aac/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