Merge remote-tracking branch 'aosp/master'

34 files changed, 7707 insertions, 6540 deletions

diff --git a/Android.mk b/Android.mk
index f804089..9e56a2d 100644
--- a/Android.mk
+++ b/Android.mk
@@ -53,6 +53,11 @@ LOCAL_C_INCLUDES := \
         $(LOCAL_PATH)/libSBRdec/include \
         $(LOCAL_PATH)/libSBRenc/include
 
+# In tpdec_asc.cpp: address of array 'pPce->FrontElementIsCpe'
+# will always evaluate to 'true'.
+LOCAL_CLANG_CPPFLAGS += \
+        -Wno-pointer-bool-conversion
+
 LOCAL_MODULE:= libFraunhoferAAC
 
 include $(BUILD_STATIC_LIBRARY)
diff --git a/documentation/aacDecoder.pdf b/documentation/aacDecoder.pdf
index a8cee17..d388576 100644
--- a/documentation/aacDecoder.pdf
+++ b/documentation/aacDecoder.pdf
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 /Limits [(aacdecoder__lib_8h_afbc7e8b4f1613dfd8cb91b681eeb9d41) (aacdecoder__lib_8h_afbc7e8b4f1613dfd8cb91b681eeb9d41a42c5cbc358ada7261ba3d609c33ec3a5)]
 >> endobj
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 /Limits [(aacdecoder__lib_8h_afbc7e8b4f1613dfd8cb91b681eeb9d41a45662b23139ca5807ed6fe3b8940dc40) (aacdecoder__lib_8h_afbc7e8b4f1613dfd8cb91b681eeb9d41a5bc4e4af77d94fa08beb54def8f01f7f)]
 >> endobj
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 /Limits [(aacdecoder__lib_8h_afbc7e8b4f1613dfd8cb91b681eeb9d41a60690e0dbc450f62f83548fa484716ab) (aacdecoder__lib_8h_afbc7e8b4f1613dfd8cb91b681eeb9d41a7ea36c43c374f751f65be033057e3888)]
 >> endobj
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 /Limits [(aacdecoder__lib_8h_afbc7e8b4f1613dfd8cb91b681eeb9d41a856cdca9aa719c292b7c351cec03ae06) (aacdecoder__lib_8h_afbc7e8b4f1613dfd8cb91b681eeb9d41aa14848134610eea96115c59a6b944921)]
 >> endobj
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 /Limits [(aacdecoder__lib_8h_afbc7e8b4f1613dfd8cb91b681eeb9d41aa86450161814b428d08b620188302fbc) (aacdecoder__lib_8h_afbc7e8b4f1613dfd8cb91b681eeb9d41acb4252bc9a6eeee71738af951850379c)]
 >> endobj
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 >> endobj
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 /Limits [(aacdecoder__lib_8h_afbc7e8b4f1613dfd8cb91b681eeb9d41af16b1992a2a415168ab96228671ef337) (chapter*.1)]
 >> endobj
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->> endobj
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->> endobj
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->> endobj
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 >> endobj
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 trailer
-<< /Size 1404
-/Root 1402 0 R
-/Info 1403 0 R
-/ID [<F402B200711665781F0B684C4A0FC9D9> <F402B200711665781F0B684C4A0FC9D9>] >>
+<< /Size 1149
+/Root 1147 0 R
+/Info 1148 0 R
+/ID [<C984448BA326CC9A19372FC47EEB0753> <C984448BA326CC9A19372FC47EEB0753>] >>
 startxref
-527438
+482421
 %%EOF
diff --git a/libAACdec/include/aacdecoder_lib.h b/libAACdec/include/aacdecoder_lib.h
index 60efe8d..a0c0854 100644
--- a/libAACdec/include/aacdecoder_lib.h
+++ b/libAACdec/include/aacdecoder_lib.h
@@ -144,10 +144,9 @@ to allocate memory for the required structures, and the corresponding mpegFileRe
 files and to de-allocate associated structures. mpegFileRead_Open() tries to detect the bitstream format and
 in case of MPEG-4 file format or Raw Packets file format (a Fraunhofer IIS proprietary format) reads the Audio
 Specific Config data (ASC). An unsuccessful attempt to recognize the bitstream format requires the user to
-provide this information manually (see \ref CommandLineUsage). 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.
+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.
 
 -# Call aacDecoder_Open() to open and retrieve a handle to a new AAC decoder instance.
 \dontinclude main.cpp
@@ -430,21 +429,68 @@ typedef enum {
 typedef enum
 {
   AAC_PCM_OUTPUT_INTERLEAVED              = 0x0000,  /*!< PCM output mode (1: interleaved (default); 0: not interleaved). */
-  AAC_PCM_OUTPUT_CHANNELS                 = 0x0001,  /*!< Number of PCM output channels (if different from encoded audio channels, downmixing or
-                                                          upmixing is applied). \n
-                                                          -1: Disable up-/downmixing. The decoder output contains the same number of channels as the
-                                                              encoded bitstream. \n
-                                                           1: The decoder performs a mono matrix mix-down if the encoded audio channels are greater
-                                                              than one. Thus it ouputs always exact one channel. \n
-                                                           2: The decoder performs a stereo matrix mix-down if the encoded audio channels are greater
-                                                              than two. If the encoded audio channels are smaller than two the decoder duplicates the
-                                                              output. Thus it ouputs always exact two channels. \n */
-  AAC_PCM_DUAL_CHANNEL_OUTPUT_MODE        = 0x0002,  /*!< Defines how the decoder processes two channel signals:
-                                                          0: Leave both signals as they are (default).
-                                                          1: Create a dual mono output signal from channel 1.
-                                                          2: Create a dual mono output signal from channel 2.
+  AAC_PCM_DUAL_CHANNEL_OUTPUT_MODE        = 0x0002,  /*!< Defines how the decoder processes two channel signals: \n
+                                                          0: Leave both signals as they are (default). \n
+                                                          1: Create a dual mono output signal from channel 1. \n
+                                                          2: Create a dual mono output signal from channel 2. \n
                                                           3: Create a dual mono output signal by mixing both channels (L' = R' = 0.5*Ch1 + 0.5*Ch2). */
   AAC_PCM_OUTPUT_CHANNEL_MAPPING          = 0x0003,  /*!< Output buffer channel ordering. 0: MPEG PCE style order, 1: WAV file channel order (default). */
+  AAC_PCM_LIMITER_ENABLE                  = 0x0004,  /*!< Enable signal level limiting. \n
+                                                          -1: Auto-config. Enable limiter for all non-lowdelay configurations by default. \n
+                                                           0: Disable limiter in general. \n
+                                                           1: Enable limiter always.
+                                                          It is recommended to call the decoder with a AACDEC_CLRHIST flag to reset all states when
+                                                          the limiter switch is changed explicitly. */
+  AAC_PCM_LIMITER_ATTACK_TIME             = 0x0005,  /*!< Signal level limiting attack time in ms.
+                                                          Default confguration is 15 ms. Adjustable range from 1 ms to 15 ms. */
+  AAC_PCM_LIMITER_RELEAS_TIME             = 0x0006,  /*!< Signal level limiting release time in ms.
+                                                          Default configuration is 50 ms. Adjustable time must be larger than 0 ms. */
+  AAC_PCM_MIN_OUTPUT_CHANNELS             = 0x0011,  /*!< Minimum number of PCM output channels. If higher than the number of encoded audio channels,
+                                                          a simple channel extension is applied. \n
+                                                          -1, 0: Disable channel extenstion feature. The decoder output contains the same number of
+                                                                 channels as the encoded bitstream. \n
+                                                           1:    This value is currently needed only together with the mix-down feature. See
+                                                                 ::AAC_PCM_MAX_OUTPUT_CHANNELS and note 2 below. \n
+                                                           2:    Encoded mono signals will be duplicated to achieve a 2/0/0.0 channel output
+                                                                 configuration. \n
+                                                           6:    The decoder trys to reorder encoded signals with less than six channels to achieve
+                                                                 a 3/0/2.1 channel output signal. Missing channels will be filled with a zero signal.
+                                                                 If reordering is not possible the empty channels will simply be appended. Only
+                                                                 available if instance is configured to support multichannel output. \n
+                                                           8:    The decoder trys to reorder encoded signals with less than eight channels to
+                                                                 achieve a 3/0/4.1 channel output signal. Missing channels will be filled with a
+                                                                 zero signal. If reordering is not possible the empty channels will simply be
+                                                                 appended. Only available if instance is configured to support multichannel output.\n
+                                                          NOTE: \n
+                                                            1. The channel signalling (CStreamInfo::pChannelType and CStreamInfo::pChannelIndices)
+                                                               will not be modified. Added empty channels will be signalled with channel type
+                                                               AUDIO_CHANNEL_TYPE::ACT_NONE. \n
+                                                            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. */
+  AAC_PCM_MAX_OUTPUT_CHANNELS             = 0x0012,  /*!< Maximum number of PCM output channels. If lower than the number of encoded audio channels,
+                                                          downmixing is applied accordingly. If dedicated metadata is available in the stream it
+                                                          will be used to achieve better mixing results. \n
+                                                          -1, 0: Disable downmixing feature. The decoder output contains the same number of channels
+                                                                 as the encoded bitstream. \n
+                                                           1:    All encoded audio configurations with more than one channel will be mixed down to
+                                                                 one mono output signal. \n
+                                                           2:    The decoder performs a stereo mix-down if the number encoded audio channels is
+                                                                 greater than two. \n
+                                                           6:    If the number of encoded audio channels is greater than six the decoder performs a
+                                                                 mix-down to meet the target output configuration of 3/0/2.1 channels. Only
+                                                                 available if instance is configured to support multichannel output. \n
+                                                           8:    This value is currently needed only together with the channel extension feature.
+                                                                 See ::AAC_PCM_MIN_OUTPUT_CHANNELS and note 2 below. Only available if instance is
+                                                                 configured to support multichannel output. \n
+                                                          NOTE: \n
+                                                            1. Down-mixing of any seven or eight channel configuration not defined in ISO/IEC 14496-3
+                                                               PDAM 4 is not supported by this software version. \n
+                                                            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 param with any value will disable the binaural processing of the MPEG
+                                                               Surround module (::AAC_MPEGS_BINAURAL_ENABLE=0). */
 
   AAC_CONCEAL_METHOD                      = 0x0100,  /*!< Error concealment: Processing method. \n
                                                           0: Spectral muting. \n
@@ -485,18 +531,18 @@ typedef enum
  */
 typedef struct
 {
-  /* These three members are the only really relevant ones for the user.                                                           */
+  /* These five members are the only really relevant ones for the user.                                                            */
   INT               sampleRate;          /*!< The samplerate in Hz of the fully decoded PCM audio signal (after SBR processing).   */
   INT               frameSize;           /*!< The frame size of the decoded PCM audio signal. \n
                                               1024 or 960 for AAC-LC \n
                                               2048 or 1920 for HE-AAC (v2) \n
                                               512 or 480 for AAC-LD and AAC-ELD                                                    */
   INT               numChannels;         /*!< The number of output audio channels in the decoded and interleaved PCM audio signal. */
-  AUDIO_CHANNEL_TYPE *pChannelType;       /*!< Audio channel type of each output audio channel.           */
-  UCHAR             *pChannelIndices;     /*!< Audio channel index for each output audio channel.
+  AUDIO_CHANNEL_TYPE *pChannelType;      /*!< Audio channel type of each output audio channel.                                     */
+  UCHAR             *pChannelIndices;    /*!< Audio channel index for each output audio channel.
                                                See ISO/IEC 13818-7:2005(E), 8.5.3.2 Explicit channel mapping using a program_config_element() */
   /* Decoder internal members. */
-  INT               aacSampleRate;       /*!< sampling rate in Hz without SBR (from configuration info).                           */
+  INT               aacSampleRate;       /*!< Sampling rate in Hz without SBR (from configuration info).                           */
   INT               profile;             /*!< MPEG-2 profile (from file header) (-1: not applicable (e. g. MPEG-4)).               */
   AUDIO_OBJECT_TYPE aot;                 /*!< Audio Object Type (from ASC): is set to the appropriate value for MPEG-2 bitstreams (e. g. 2 for AAC-LC). */
   INT               channelConfig;       /*!< Channel configuration (0: PCE defined, 1: mono, 2: stereo, ...                       */
@@ -509,7 +555,9 @@ typedef struct
   AUDIO_OBJECT_TYPE extAot;              /*!< Extension Audio Object Type (from ASC)   */
   INT               extSamplingRate;     /*!< Extension sampling rate in Hz (from ASC) */
 
-  UINT              flags;               /*!< Copy if internal flags. Only to be written by the decoder, and only to be read externally. */
+  UINT              outputDelay;         /*!< The number of samples the output is additionally delayed by the decoder. */
+
+  UINT              flags;               /*!< Copy of internal flags. Only to be written by the decoder, and only to be read externally. */
 
   SCHAR             epConfig;            /*!< epConfig level (from ASC): only level 0 supported, -1 means no ER (e. g. AOT=2, MPEG-2 AAC, etc.)  */
 
@@ -522,10 +570,25 @@ typedef struct
   UINT              numTotalAccessUnits; /*!< This is the number of total access units that have passed through the decoder. */
   UINT              numBadAccessUnits;   /*!< This is the number of total access units that were considered with errors from numTotalBytes. */
 
+  /* Metadata */
+  SCHAR             drcProgRefLev;       /*!< DRC program reference level. Defines the reference level below full-scale.
+                                              It is quantized in steps of 0.25dB. The valid values range from 0 (0 dBFS) to 127 (-31.75 dBFS).
+                                              It is used to reflect the average loudness of the audio in LKFS accoring to ITU-R BS 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 */
+
 } CStreamInfo;
 
 
-typedef struct AAC_DECODER_INSTANCE *HANDLE_AACDECODER;
+typedef struct AAC_DECODER_INSTANCE *HANDLE_AACDECODER;  /*!< Pointer to a AAC decoder instance. */
 
 #ifdef __cplusplus
 extern "C"
@@ -634,11 +697,15 @@ aacDecoder_Fill ( HANDLE_AACDECODER  self,
                   const UINT         bufferSize[],
                   UINT              *bytesValid );
 
-#define AACDEC_CONCEAL  1 /*!< Flag for aacDecoder_DecodeFrame(): do not consider new input data. Do concealment. */
-#define AACDEC_FLUSH    2 /*!< Flag for aacDecoder_DecodeFrame(): Do not consider new input data. Flush filterbanks (output delayed audio). */
-#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. */
+#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. */
 
 /**
  * \brief            Decode one audio frame
diff --git a/libAACdec/src/aac_ram.cpp b/libAACdec/src/aac_ram.cpp
index a38f57c..1ff289b 100644
--- a/libAACdec/src/aac_ram.cpp
+++ b/libAACdec/src/aac_ram.cpp
@@ -108,15 +108,15 @@ C_ALLOC_MEM(AacDecoder, AAC_DECODER_INSTANCE, 1)
 /*! The structure CAacDecoderStaticChannelInfo contains the static sideinfo which is needed
     for the decoding of one aac channel. <br>
     Dimension: #AacDecoderChannels                                                      */
-C_ALLOC_MEM2(AacDecoderStaticChannelInfo, CAacDecoderStaticChannelInfo, 1, (6))
+C_ALLOC_MEM2(AacDecoderStaticChannelInfo, CAacDecoderStaticChannelInfo, 1, (8))
 
 /*! The structure CAacDecoderChannelInfo contains the dynamic sideinfo which is needed
     for the decoding of one aac channel. <br>
     Dimension: #AacDecoderChannels                                                      */
-C_ALLOC_MEM2(AacDecoderChannelInfo, CAacDecoderChannelInfo, 1, (6))
+C_AALLOC_MEM2(AacDecoderChannelInfo, CAacDecoderChannelInfo, 1, (8))
 
 /*! Overlap buffer */
-C_ALLOC_MEM2(OverlapBuffer, FIXP_DBL, OverlapBufferSize, (6))
+C_ALLOC_MEM2(OverlapBuffer, FIXP_DBL, OverlapBufferSize, (8))
 
 C_ALLOC_MEM(DrcInfo, CDrcInfo, 1)
 
@@ -128,7 +128,7 @@ C_ALLOC_MEM(DrcInfo, CDrcInfo, 1)
   Dynamic memory areas, might be reused in other algorithm sections,
   e.g. the sbr decoder
 */
-C_ALLOC_MEM_OVERLAY(WorkBufferCore2, FIXP_DBL, ((6)*1024), SECT_DATA_L2, WORKBUFFER2_TAG)
+C_ALLOC_MEM_OVERLAY(WorkBufferCore2, FIXP_DBL, ((8)*1024), SECT_DATA_L2, WORKBUFFER2_TAG)
 
 
 C_ALLOC_MEM_OVERLAY(WorkBufferCore1, CWorkBufferCore1, 1, SECT_DATA_L1, WORKBUFFER1_TAG)
diff --git a/libAACdec/src/aac_rom.cpp b/libAACdec/src/aac_rom.cpp
index fa04c42..607cb3b 100644
--- a/libAACdec/src/aac_rom.cpp
+++ b/libAACdec/src/aac_rom.cpp
@@ -1777,42 +1777,62 @@ const FIXP_TCC FDKaacDec_tnsCoeff4 [16] =
 };
 
 /* MPEG like mapping (no change). */
-const UCHAR channelMappingTablePassthrough[8][8] =
+const UCHAR channelMappingTablePassthrough[15][8] =
 {
+     { 0,  1,  2,  3,  4,  5,  6,  7},  /*  fallback  */
      { 0,  1,255,255,255,255,255,255},  /*  mono / PS */
      { 0,  1,255,255,255,255,255,255},  /*  stereo    */
      { 0,  1,  2,255,255,255,255,255},  /*  3ch       */
      { 0,  1,  2,  3,255,255,255,255},  /*  4ch       */
      { 0,  1,  2,  3,  4,255,255,255},  /*  5ch       */
      { 0,  1,  2,  3,  4,  5,255,255},  /*  5.1ch     */
-     { 0,  1,  2,  3,  4,  5,  6,  7},  /*  7ch       */
-     { 0,  1,  2,  3,  4,  5,  6,  7}   /*  7.1ch     */
+     { 0,  1,  2,  3,  4,  5,  6,  7},  /*  7.1 front */
+     { 0,  1,  2,  3,  4,  5,  6,  7},  /*  reserved  */
+     { 0,  1,  2,  3,  4,  5,  6,  7},  /*  reserved  */
+     { 0,  1,  2,  3,  4,  5,  6,  7},  /*  reserved  */
+     { 0,  1,  2,  3,  4,  5,  6,255},  /*  6.1ch     */
+     { 0,  1,  2,  3,  4,  5,  6,  7},  /*  7.1 rear  */
+     { 0,  1,  2,  3,  4,  5,  6,  7},  /*  reserved  */
+     { 0,  1,  2,  3,  4,  5,  6,  7}   /*  7.1 top   */
 };
 
 /* WAV file like mapping (from MPEG mapping). */
-const UCHAR channelMappingTableWAV[8][8] =
+const UCHAR channelMappingTableWAV[15][8] =
 {
+     { 0,  1,  2,  3,  4,  5,  6,  7},  /*  fallback  */
      { 0,  1,255,255,255,255,255,255},  /*  mono / PS */
      { 0,  1,255,255,255,255,255,255},  /*  stereo    */
      { 2,  0,  1,255,255,255,255,255},  /*  3ch       */
      { 2,  0,  1,  3,255,255,255,255},  /*  4ch       */
      { 2,  0,  1,  3,  4,255,255,255},  /*  5ch       */
      { 2,  0,  1,  4,  5,  3,255,255},  /*  5.1ch     */
-     { 0,  1,  2,  3,  4,  5,  6,  7},  /*  7ch       */
-     { 2,  0,  1,  6,  7,  4,  5,  3}   /*  7.1ch     */
+     { 2,  6,  7,  0,  1,  4,  5,  3},  /*  7.1 front */
+     { 0,  1,  2,  3,  4,  5,  6,  7},  /*  reserved  */
+     { 0,  1,  2,  3,  4,  5,  6,  7},  /*  reserved  */
+     { 0,  1,  2,  3,  4,  5,  6,  7},  /*  reserved  */
+     { 2,  0,  1,  4,  5,  6,  3,255},  /*  6.1ch     */
+     { 2,  0,  1,  6,  7,  4,  5,  3},  /*  7.1 rear  */
+     { 0,  1,  2,  3,  4,  5,  6,  7},  /*  reserved  */
+     { 2,  0,  1,  4,  5,  3,  6,  7}   /*  7.1 top   */
 };
 
 /* Lookup tables for elements in ER bitstream */
-const MP4_ELEMENT_ID elementsTab[8][7] =
+const MP4_ELEMENT_ID elementsTab[15][7] =
 {
-  {ID_SCE, ID_EXT, ID_END, ID_NONE, ID_NONE,ID_NONE,ID_NONE }, /* 1 channel  */
-  {ID_CPE, ID_EXT, ID_END, ID_NONE, ID_NONE,ID_NONE,ID_NONE }  /* 2 channels */
-  ,
-  {ID_SCE, ID_CPE, ID_EXT, ID_END, ID_NONE,ID_NONE,ID_NONE }, /* 3 channels */
-  {ID_SCE, ID_CPE, ID_SCE, ID_EXT, ID_END, ID_NONE,ID_NONE }, /* 4 channels */
-  {ID_SCE, ID_CPE, ID_CPE, ID_EXT, ID_END, ID_NONE,ID_NONE }, /* 5 channels */
-  {ID_SCE, ID_CPE, ID_CPE, ID_LFE, ID_EXT, ID_END, ID_NONE }, /* 6 channels */
-  {ID_SCE, ID_CPE, ID_CPE, ID_CPE, ID_LFE, ID_EXT, ID_END}    /* 8 channels */
+ /*  1 */  { ID_SCE,  ID_EXT,  ID_END,  ID_NONE, ID_NONE, ID_NONE, ID_NONE }, /* 1 channel  */
+ /*  2 */  { ID_CPE,  ID_EXT,  ID_END,  ID_NONE, ID_NONE, ID_NONE, ID_NONE }  /* 2 channels */
+ /*  3 */ ,{ ID_SCE,  ID_CPE,  ID_EXT,  ID_END,  ID_NONE, ID_NONE, ID_NONE }, /* 3 channels */
+ /*  4 */  { ID_SCE,  ID_CPE,  ID_SCE,  ID_EXT,  ID_END,  ID_NONE, ID_NONE }, /* 4 channels */
+ /*  5 */  { ID_SCE,  ID_CPE,  ID_CPE,  ID_EXT,  ID_END,  ID_NONE, ID_NONE }, /* 5 channels */
+ /*  6 */  { ID_SCE,  ID_CPE,  ID_CPE,  ID_LFE,  ID_EXT,  ID_END,  ID_NONE }  /* 6 channels */
+ /*  7 */ ,{ ID_SCE,  ID_CPE,  ID_CPE,  ID_CPE,  ID_LFE,  ID_EXT,  ID_END  },  /* 8 channels */
+ /*  8 */  { ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE },  /* reserved   */
+ /*  9 */  { ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE },  /* reserved   */
+ /* 10 */  { ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE },  /* reserved   */
+ /* 11 */  { ID_SCE,  ID_CPE,  ID_CPE,  ID_SCE,  ID_LFE,  ID_EXT,  ID_END  },  /* 7 channels */
+ /* 12 */  { ID_SCE,  ID_CPE,  ID_CPE,  ID_CPE,  ID_LFE,  ID_EXT,  ID_END  },  /* 8 channels */
+ /* 13 */  { ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE },  /* reserved   */
+ /* 14 */  { ID_SCE,  ID_CPE,  ID_CPE,  ID_LFE,  ID_CPE,  ID_EXT,  ID_END  }   /* 8 channels */
 };
 
 /*! Random sign bit used for concealment
diff --git a/libAACdec/src/aac_rom.h b/libAACdec/src/aac_rom.h
index 536d184..f314a2d 100644
--- a/libAACdec/src/aac_rom.h
+++ b/libAACdec/src/aac_rom.h
@@ -177,11 +177,12 @@ extern const USHORT randomSign[AAC_NF_NO_RANDOM_VAL/16];
 extern const FIXP_DBL pow2_div24minus1[47];
 extern const int offsetTab[2][16];
 
-/* Channel mapping indices for time domain I/O. First dimension is channel count-1. */
-extern const UCHAR channelMappingTablePassthrough[8][8];
-extern const UCHAR channelMappingTableWAV[8][8];
+/* Channel mapping indices for time domain I/O.
+   The first dimension is the channel configuration index. */
+extern const UCHAR channelMappingTablePassthrough[15][8];
+extern const UCHAR channelMappingTableWAV[15][8];
 
 /* Lookup tables for elements in ER bitstream */
-extern const MP4_ELEMENT_ID elementsTab[8][7];
+extern const MP4_ELEMENT_ID elementsTab[15][7];
 
 #endif /* #ifndef AAC_ROM_H */
diff --git a/libAACdec/src/aacdec_drc.cpp b/libAACdec/src/aacdec_drc.cpp
index 2666454..0c33a2b 100644
--- a/libAACdec/src/aacdec_drc.cpp
+++ b/libAACdec/src/aacdec_drc.cpp
@@ -130,7 +130,6 @@ void aacDecoder_drcInit (
   /* init control fields */
   self->enable = 0;
   self->numThreads = 0;
-  self->digitalNorm = 0;
 
   /* init params */
   pParams = &self->params;
@@ -139,12 +138,15 @@ void aacDecoder_drcInit (
   pParams->usrCut   = FL2FXCONST_DBL(0.0f);
   pParams->boost    = FL2FXCONST_DBL(0.0f);
   pParams->usrBoost = FL2FXCONST_DBL(0.0f);
-  pParams->targetRefLevel = AACDEC_DRC_DEFAULT_REF_LEVEL;
+  pParams->targetRefLevel = -1;
   pParams->expiryFrame = AACDEC_DRC_DFLT_EXPIRY_FRAMES;
+  pParams->applyDigitalNorm = 0;
   pParams->applyHeavyCompression = 0;
 
   /* initial program ref level = target ref level */
   self->progRefLevel = pParams->targetRefLevel;
+  self->progRefLevelPresent = 0;
+  self->presMode = -1;
 }
 
 
@@ -222,11 +224,12 @@ AAC_DECODER_ERROR aacDecoder_drcSetParam (
       return AAC_DEC_INVALID_HANDLE;
     }
     if (value < 0) {
-      self->digitalNorm = 0;
+      self->params.applyDigitalNorm = 0;
+      self->params.targetRefLevel = -1;
     }
     else {
       /* ref_level must be between 0 and MAX_REFERENCE_LEVEL, inclusive */
-      self->digitalNorm    = 1;
+      self->params.applyDigitalNorm = 1;
       if (self->params.targetRefLevel != (SCHAR)value) {
         self->params.targetRefLevel = (SCHAR)value;
         self->progRefLevel = (SCHAR)value;  /* Always set the program reference level equal to the
@@ -234,6 +237,16 @@ AAC_DECODER_ERROR aacDecoder_drcSetParam (
       }
     }
     break;
+  case APPLY_NORMALIZATION:
+    if (value < 0 || value > 1) {
+      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 < 0 || value > 1) {
       return AAC_DEC_SET_PARAM_FAIL;
@@ -278,7 +291,7 @@ AAC_DECODER_ERROR aacDecoder_drcSetParam (
   self->enable = ( (self->params.boost > (FIXP_DBL)0)
                 || (self->params.cut   > (FIXP_DBL)0)
                 || (self->params.applyHeavyCompression != 0)
-                || (self->digitalNorm == 1) );
+                || (self->params.targetRefLevel >= 0) );
 
 
   return ErrorStatus;
@@ -539,7 +552,7 @@ static int aacDecoder_drcReadCompression (
     UINT                  payloadPosition )
 {
   int  bitCnt = 0;
-  int  dmxLevelsPresent, compressionPresent;
+  int  dmxLevelsPresent, extensionPresent, compressionPresent;
   int  coarseGrainTcPresent, fineGrainTcPresent;
 
   /* Move to the beginning of the DRC payload field */
@@ -561,8 +574,9 @@ static int aacDecoder_drcReadCompression (
     return 0;
   }
   FDKreadBits(bs, 2);                          /* dolby_surround_mode */
-  FDKreadBits(bs, 2);                          /* presentation_mode */
-  if (FDKreadBits(bs, 2) != 0) {               /* reserved, set to 0 */
+  pDrcBs->presMode = FDKreadBits(bs, 2);       /* presentation_mode */
+  FDKreadBits(bs, 1);                          /* stereo_downmix_mode */
+  if (FDKreadBits(bs, 1) != 0) {               /* reserved, set to 0 */
     return 0;
   }
 
@@ -571,9 +585,7 @@ static int aacDecoder_drcReadCompression (
     return 0;
   }
   dmxLevelsPresent = FDKreadBits(bs, 1);       /* downmixing_levels_MPEG4_status */
-  if (FDKreadBits(bs, 1) != 0) {               /* reserved, set to 0 */
-    return 0;
-  }
+  extensionPresent = FDKreadBits(bs, 1);       /* ancillary_data_extension_status; */
   compressionPresent   = FDKreadBits(bs, 1);   /* audio_coding_mode_and_compression status */
   coarseGrainTcPresent = FDKreadBits(bs, 1);   /* coarse_grain_timecode_status */
   fineGrainTcPresent   = FDKreadBits(bs, 1);   /* fine_grain_timecode_status */
@@ -631,6 +643,19 @@ static int aacDecoder_drcReadCompression (
     bitCnt += 16;
   }
 
+  /* Read extension just to get the right amount of bits. */
+  if (extensionPresent) {
+    int  extBits = 8;
+
+    FDKreadBits(bs, 1);                     /* reserved, set to 0 */
+    if (FDKreadBits(bs, 1)) extBits += 8;   /* ext_downmixing_levels_status */
+    if (FDKreadBits(bs, 1)) extBits += 16;  /* ext_downmixing_global_gains_status */
+    if (FDKreadBits(bs, 1)) extBits += 8;   /* ext_downmixing_lfe_level_status */
+
+    FDKpushFor(bs, extBits - 4);            /* skip the extension payload remainder. */
+    bitCnt += extBits;
+  }
+
   return (bitCnt);
 }
 
@@ -780,9 +805,15 @@ static int aacDecoder_drcExtractAndMap (
      */
     if (pThreadBs->progRefLevel >= 0) {
       self->progRefLevel = pThreadBs->progRefLevel;
+      self->progRefLevelPresent = 1;
       self->prlExpiryCount = 0;  /* Got a new value -> Reset counter */
     }
 
+    if (drcPayloadType == DVB_DRC_ANC_DATA) {
+      /* Announce the presentation mode of this valid thread. */
+      self->presMode = pThreadBs->presMode;
+    }
+
     /* SCE, CPE and LFE */
     for (ch = 0; ch < validChannels; ch++) {
       int mapedChannel = channelMapping[ch];
@@ -802,6 +833,7 @@ static int aacDecoder_drcExtractAndMap (
   if ( (pParams->expiryFrame > 0)
     && (self->prlExpiryCount++ > pParams->expiryFrame) )
   { /* The program reference level is too old, so set it back to the target level. */
+    self->progRefLevelPresent = 0;
     self->progRefLevel = pParams->targetRefLevel;
     self->prlExpiryCount = 0;
   }
@@ -815,6 +847,7 @@ void aacDecoder_drcApply (
         void                   *pSbrDec,
         CAacDecoderChannelInfo *pAacDecoderChannelInfo,
         CDrcChannelData        *pDrcChData,
+        FIXP_DBL               *extGain,
         int  ch,   /* needed only for SBR */
         int  aacFrameSize,
         int  bSbrPresent )
@@ -826,8 +859,8 @@ void aacDecoder_drcApply (
   FIXP_DBL max_mantissa;
   INT max_exponent;
 
-  FIXP_DBL norm_mantissa = FL2FXCONST_DBL(0.0f);
-  INT  norm_exponent = 0;
+  FIXP_DBL norm_mantissa = FL2FXCONST_DBL(0.5f);
+  INT  norm_exponent = 1;
 
   FIXP_DBL fact_mantissa[MAX_DRC_BANDS];
   INT  fact_exponent[MAX_DRC_BANDS];
@@ -849,6 +882,15 @@ void aacDecoder_drcApply (
 
   if (!self->enable) {
     sbrDecoder_drcDisable( (HANDLE_SBRDECODER)pSbrDec, ch );
+    if (extGain != NULL) {
+      INT gainScale = (INT)*extGain;
+      /* The gain scaling must be passed to the function in the buffer pointed on by extGain. */
+      if (gainScale >= 0 && gainScale <= DFRACT_BITS) {
+        *extGain = scaleValue(norm_mantissa, norm_exponent-gainScale);
+      } else {
+        FDK_ASSERT(0);
+      }
+    }
     return;
   }
 
@@ -864,7 +906,7 @@ void aacDecoder_drcApply (
   reduced DAC SNR (if signal is attenuated) or clipping (if signal is
   boosted) */
 
-  if (self->digitalNorm == 1)
+  if (pParams->targetRefLevel >= 0)
   {
     /* 0.5^((targetRefLevel - progRefLevel)/24) */
     norm_mantissa = fLdPow(
@@ -874,7 +916,18 @@ void aacDecoder_drcApply (
             3,
            &norm_exponent );
   }
-  else {
+  /* Always export the normalization gain (if possible). */
+  if (extGain != NULL) {
+    INT gainScale = (INT)*extGain;
+    /* The gain scaling must be passed to the function in the buffer pointed on by extGain. */
+    if (gainScale >= 0 && gainScale <= DFRACT_BITS) {
+      *extGain = scaleValue(norm_mantissa, norm_exponent-gainScale);
+    } else {
+      FDK_ASSERT(0);
+    }
+  }
+  if (self->params.applyDigitalNorm == 0) {
+    /* Reset normalization gain since this module must not apply it */
     norm_mantissa = FL2FXCONST_DBL(0.5f);
     norm_exponent = 1;
   }
@@ -1112,3 +1165,24 @@ int aacDecoder_drcEpilog (
   return err;
 }
 
+/*
+ * Export relevant metadata info from bitstream payload.
+ */
+void aacDecoder_drcGetInfo (
+        HANDLE_AAC_DRC  self,
+        SCHAR          *pPresMode,
+        SCHAR          *pProgRefLevel )
+{
+  if (self != NULL) {
+    if (pPresMode != NULL) {
+      *pPresMode = self->presMode;
+    }
+    if (pProgRefLevel != NULL) {
+      if (self->progRefLevelPresent) {
+        *pProgRefLevel = self->progRefLevel;
+      } else {
+        *pProgRefLevel = -1;
+      }
+    }
+  }
+}
diff --git a/libAACdec/src/aacdec_drc.h b/libAACdec/src/aacdec_drc.h
index 2ebae2c..c850aa5 100644
--- a/libAACdec/src/aacdec_drc.h
+++ b/libAACdec/src/aacdec_drc.h
@@ -98,7 +98,6 @@ amm-info@iis.fraunhofer.de
 #include "channel.h"
 #include "FDK_bitstream.h"
 
-#define AACDEC_DRC_DEFAULT_REF_LEVEL  ( 108 )   /* -27 dB below full scale (typical for movies) */
 #define AACDEC_DRC_DFLT_EXPIRY_FRAMES (  50 )   /* Default DRC data expiry time in AAC frames   */
 
 /**
@@ -111,6 +110,7 @@ typedef enum
   TARGET_REF_LEVEL,
   DRC_BS_DELAY,
   DRC_DATA_EXPIRY_FRAME,
+  APPLY_NORMALIZATION,
   APPLY_HEAVY_COMPRESSION
 
 } AACDEC_DRC_PARAM;
@@ -149,6 +149,8 @@ int aacDecoder_drcProlog (
  * \param pSbrDec pointer to SBR decoder instance
  * \param pAacDecoderChannelInfo AAC decoder channel instance to be processed
  * \param pDrcDat DRC channel data
+ * \param extGain Pointer to a FIXP_DBL where a externally applyable gain will be stored into (independently on whether it will be apply internally or not).
+ *                At function call the buffer must hold the scale (0 >= scale < DFRACT_BITS) to be applied on the gain value.
  * \param ch channel index
  * \param aacFrameSize AAC frame size
  * \param bSbrPresent flag indicating that SBR is present, in which case DRC is handed over to the SBR instance pSbrDec
@@ -158,6 +160,7 @@ void aacDecoder_drcApply (
         void                   *pSbrDec,
         CAacDecoderChannelInfo *pAacDecoderChannelInfo,
         CDrcChannelData        *pDrcDat,
+        FIXP_DBL               *extGain,
         int  ch,
         int  aacFrameSize,
         int  bSbrPresent );
@@ -170,5 +173,17 @@ int aacDecoder_drcEpilog (
         UCHAR  channelMapping[],
         int    validChannels );
 
+/**
+ * \brief Get metadata information found in bitstream.
+ * \param self DRC module instance handle.
+ * \param pPresMode Pointer to field where the presentation mode will be written to.
+ * \param pProgRefLevel Pointer to field where the program reference level will be written to.
+ * \return Nothing.
+ */
+void aacDecoder_drcGetInfo (
+        HANDLE_AAC_DRC  self,
+        SCHAR  *pPresMode,
+        SCHAR  *pProgRefLevel );
+
 
 #endif  /* AACDEC_DRC_H */
diff --git a/libAACdec/src/aacdec_drc_types.h b/libAACdec/src/aacdec_drc_types.h
index 1b5cd76..4c6d163 100644
--- a/libAACdec/src/aacdec_drc_types.h
+++ b/libAACdec/src/aacdec_drc_types.h
@@ -124,6 +124,7 @@ typedef struct
 {
   UINT   excludedChnsMask;
   SCHAR  progRefLevel;
+  SCHAR  presMode;     /* Presentation mode: 0 (not indicated), 1, 2, and 3 (reserved). */
   SCHAR  pceInstanceTag;
 
   CDrcChannelData channelData;
@@ -140,6 +141,7 @@ typedef struct
   UINT   expiryFrame;
   SCHAR  targetRefLevel;
   UCHAR  bsDelayEnable;
+  UCHAR  applyDigitalNorm;
   UCHAR  applyHeavyCompression;
 
 } CDrcParams;
@@ -155,9 +157,11 @@ typedef struct
   USHORT numPayloads;                        /* The number of DRC data payload elements found within frame */
   USHORT numThreads;                         /* The number of DRC data threads extracted from the found payload elements */
   SCHAR  progRefLevel;                       /* Program reference level for all channels */
+  UCHAR  progRefLevelPresent;                /* Program reference level found in bitstream */
 
   UINT   prlExpiryCount;                     /* Counter that can be used to monitor the life time of the program reference level. */
 
+  SCHAR  presMode;                            /* Presentation mode as defined in ETSI TS 101 154 */
   UCHAR  dvbAncDataAvailable;                  /* Flag that indicates whether DVB ancillary data is present or not */
   UINT   dvbAncDataPosition;                   /* Used to store the DVB ancillary data payload position in the bitstream (only one per frame) */
   UINT   drcPayloadPosition[MAX_DRC_THREADS];  /* Used to store the DRC payload positions in the bitstream */
diff --git a/libAACdec/src/aacdecoder.cpp b/libAACdec/src/aacdecoder.cpp
index 3a2a561..e19c501 100644
--- a/libAACdec/src/aacdecoder.cpp
+++ b/libAACdec/src/aacdecoder.cpp
@@ -338,17 +338,22 @@ AAC_DECODER_ERROR CAacDecoder_AncDataParse (
   \return  Error code
 */
 static AAC_DECODER_ERROR CDataStreamElement_Read (
+                                                  HANDLE_AACDECODER    self,
                                                   HANDLE_FDK_BITSTREAM bs,
-                                                  CAncData *ancData,
-                                                  HANDLE_AAC_DRC hDrcInfo,
-                                                  HANDLE_TRANSPORTDEC pTp,
                                                   UCHAR    *elementInstanceTag,
                                                   UINT      alignmentAnchor )
 {
+  HANDLE_TRANSPORTDEC  pTp;
+  CAncData *ancData;
   AAC_DECODER_ERROR error = AAC_DEC_OK;
-  UINT dataStart;
+  UINT dataStart, dseBits;
   int dataByteAlignFlag, count;
 
+  FDK_ASSERT(self != NULL);
+
+  ancData = &self->ancData;
+  pTp = self->hInput;
+
   int crcReg = transportDec_CrcStartReg(pTp, 0);
 
   /* Element Instance Tag */
@@ -361,6 +366,7 @@ static AAC_DECODER_ERROR CDataStreamElement_Read (
   if (count == 255) {
     count += FDKreadBits(bs,8); /* EscCount */
   }
+  dseBits = count*8;
 
   if (dataByteAlignFlag) {
     FDKbyteAlign(bs, alignmentAnchor);
@@ -372,19 +378,29 @@ static AAC_DECODER_ERROR CDataStreamElement_Read (
   transportDec_CrcEndReg(pTp, crcReg);
 
   {
-    INT readBits, dataBits = count<<3;
-
     /* Move to the beginning of the data junk */
     FDKpushBack(bs, dataStart-FDKgetValidBits(bs));
 
     /* Read Anc data if available */
-    readBits = aacDecoder_drcMarkPayload( hDrcInfo, bs, DVB_DRC_ANC_DATA );
+    aacDecoder_drcMarkPayload( self->hDrcInfo, bs, DVB_DRC_ANC_DATA );
+  }
+
+  {
+    PCMDMX_ERROR dmxErr = PCMDMX_OK;
+
+    /* Move to the beginning of the data junk */
+    FDKpushBack(bs, dataStart-FDKgetValidBits(bs));
 
-    if (readBits != dataBits) {
-      /* Move to the end again. */
-      FDKpushBiDirectional(bs, FDKgetValidBits(bs)-dataStart+dataBits);
+    /* Read DMX meta-data */
+    dmxErr = pcmDmx_Parse (
+                     self->hPcmUtils,
+                     bs,
+                     dseBits,
+                     0 /* not mpeg2 */ );
     }
-  }
+
+  /* Move to the very end of the element. */
+  FDKpushBiDirectional(bs, FDKgetValidBits(bs)-dataStart+dseBits);
 
   return error;
 }
@@ -701,6 +717,12 @@ void CStreamInfoInit(CStreamInfo *pStreamInfo)
   pStreamInfo->numChannels = 0;
   pStreamInfo->sampleRate = 0;
   pStreamInfo->frameSize = 0;
+
+  pStreamInfo->outputDelay = 0;
+
+  /* DRC */
+  pStreamInfo->drcProgRefLev = -1;                           /* set program reference level to not indicated */
+  pStreamInfo->drcPresMode = -1;                             /* default: presentation mode not indicated */
 }
 
 /*!
@@ -774,7 +796,7 @@ LINKSPEC_CPP void CAacDecoder_Close(HANDLE_AACDECODER self)
   if (self == NULL)
     return;
 
-  for (ch=0; ch<(6); ch++) {
+  for (ch=0; ch<(8); ch++) {
     if (self->pAacDecoderStaticChannelInfo[ch] != NULL) {
       if (self->pAacDecoderStaticChannelInfo[ch]->pOverlapBuffer != NULL) {
         FreeOverlapBuffer (&self->pAacDecoderStaticChannelInfo[ch]->pOverlapBuffer);
@@ -851,18 +873,19 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_Init(HANDLE_AACDECODER self, const CS
         /* valid number of channels -> copy program config element (PCE) from ASC */
         FDKmemcpy(&self->pce, &asc->m_progrConfigElement, sizeof(CProgramConfig));
         /* Built element table */
-        el = CProgramConfig_GetElementTable(&asc->m_progrConfigElement, self->elements, 7);
-        for (; el<7; el++) {
+        el = CProgramConfig_GetElementTable(&asc->m_progrConfigElement, self->elements, (8), &self->chMapIndex);
+        for (; el<(8); el++) {
           self->elements[el] = ID_NONE;
         }
       } else {
         return AAC_DEC_UNSUPPORTED_CHANNELCONFIG;
       }
     } else {
+      self->chMapIndex = 0;
       if (transportDec_GetFormat(self->hInput) == TT_MP4_ADTS) {
         /* set default max_channels for memory allocation because in implicit channel mapping mode
            we don't know the actual number of channels until we processed at least one raw_data_block(). */
-        ascChannels = (6);
+        ascChannels = (8);
       } else {
         return AAC_DEC_UNSUPPORTED_CHANNELCONFIG;
       }
@@ -874,26 +897,34 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_Init(HANDLE_AACDECODER self, const CS
   case 1: case 2: case 3: case 4: case 5: case 6:
     ascChannels = asc->m_channelConfiguration;
     break;
-  case 7:
+  case 11:
+    ascChannels = 7;
+    break;
+  case 7: case 12: case 14:
     ascChannels = 8;
     break;
   default:
     return AAC_DEC_UNSUPPORTED_CHANNELCONFIG;
   }
 
+  if (ascChannels > (8)) {
+    return AAC_DEC_UNSUPPORTED_CHANNELCONFIG;
+  }
+
   /* Initialize constant mappings for channel config 1-7 */
   if (asc->m_channelConfiguration > 0) {
     int el;
-    FDKmemcpy(self->elements, elementsTab[asc->m_channelConfiguration-1], sizeof(MP4_ELEMENT_ID)*FDKmin(7,7));
-    for (el=7; el<7; el++) {
+    FDKmemcpy(self->elements, elementsTab[asc->m_channelConfiguration-1], sizeof(MP4_ELEMENT_ID)*FDKmin(7,(8)));
+    for (el=7; el<(8); el++) {
       self->elements[el] = ID_NONE;
     }
     for (ch=0; ch<ascChannels; ch++) {
       self->chMapping[ch] = ch;
     }
-    for (; ch<(6); ch++) {
+    for (; ch<(8); ch++) {
       self->chMapping[ch] = 255;
     }
+    self->chMapIndex = asc->m_channelConfiguration;
   }
  #ifdef TP_PCE_ENABLE
   else {
@@ -909,9 +940,6 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_Init(HANDLE_AACDECODER self, const CS
 
   self->streamInfo.channelConfig = asc->m_channelConfiguration;
 
-  if (ascChannels > (6)) {
-    return AAC_DEC_UNSUPPORTED_CHANNELCONFIG;
-  }
   if (self->streamInfo.aot != asc->m_aot) {
     self->streamInfo.aot = asc->m_aot;
     ascChanged = 1;
@@ -1096,6 +1124,7 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
 
   MP4_ELEMENT_ID type = ID_NONE;            /* Current element type */
   INT aacChannels=0;                        /* Channel counter for channels found in the bitstream */
+  int chOutMapIdx;                          /* Output channel mapping index (see comment below) */
 
   INT auStartAnchor = (INT)FDKgetValidBits(bs);  /* AU start bit buffer position for AU byte alignment */
 
@@ -1119,12 +1148,12 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
 
     if (self->streamInfo.channelConfig == 0) {
       /* Init Channel/Element mapping table */
-      for (ch=0; ch<(6); ch++) {
+      for (ch=0; ch<(8); ch++) {
         self->chMapping[ch] = 255;
       }
       if (!CProgramConfig_IsValid(pce)) {
         int el;
-        for (el=0; el<7; el++) {
+        for (el=0; el<(8); el++) {
           self->elements[el] = ID_NONE;
         }
       }
@@ -1161,11 +1190,8 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
       CConcealment_InitChannelData(&self->pAacDecoderStaticChannelInfo[ch]->concealmentInfo,
                                    &self->concealCommonData,
                                     self->streamInfo.aacSamplesPerFrame );
-      /* Clear concealment buffers to get rid of the complete history */
-      FDKmemclear(self->pAacDecoderStaticChannelInfo[ch]->concealmentInfo.spectralCoefficient, 1024 * sizeof(FIXP_CNCL));
-      FDKmemclear(self->pAacDecoderStaticChannelInfo[ch]->concealmentInfo.specScale, 8 * sizeof(SHORT));
       /* Clear overlap-add buffers to avoid clicks. */
-      FDKmemclear(self->pAacDecoderStaticChannelInfo[ch]->IMdct.overlap.freq, OverlapBufferSize*sizeof(FIXP_DBL));
+      FDKmemclear(self->pAacDecoderStaticChannelInfo[ch]->pOverlapBuffer, OverlapBufferSize*sizeof(FIXP_DBL));
      }
   }
 
@@ -1378,10 +1404,8 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
         {
           UCHAR element_instance_tag;
 
-          CDataStreamElement_Read( bs,
-                                  &self->ancData,
-                                   self->hDrcInfo,
-                                   self->hInput,
+          CDataStreamElement_Read( self,
+                                   bs,
                                   &element_instance_tag,
                                    auStartAnchor );
 
@@ -1401,29 +1425,6 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
             //self->frameOK = 0;
           }
         }
-
-        {
-          UCHAR *pDvbAncData = NULL;
-          AAC_DECODER_ERROR ancErr;
-          int ancIndex;
-          int dvbAncDataSize = 0;
-
-          /* Ask how many anc data elements are in buffer */
-          ancIndex = self->ancData.nrElements - 1;
-          /* Get the last one (if available) */
-          ancErr   = CAacDecoder_AncDataGet( &self->ancData,
-                                              ancIndex,
-                                             &pDvbAncData,
-                                             &dvbAncDataSize );
-
-          if (ancErr == AAC_DEC_OK) {
-            pcmDmx_ReadDvbAncData (
-                  self->hPcmUtils,
-                  pDvbAncData,
-                  dvbAncDataSize,
-                  0 /* not mpeg2 */ );
-          }
-        }
         break;
 
 #ifdef TP_PCE_ENABLE
@@ -1442,9 +1443,9 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
           }
           else if ( result > 1 ) {
             /* Built element table */
-            int elIdx = CProgramConfig_GetElementTable(pce, self->elements, 7);
+            int elIdx = CProgramConfig_GetElementTable(pce, self->elements, (8), &self->chMapIndex);
             /* Reset the remaining tabs */
-            for ( ; elIdx<7; elIdx++) {
+            for ( ; elIdx<(8); elIdx++) {
               self->elements[elIdx] = ID_NONE;
             }
             /* Make new number of channel persistant */
@@ -1510,10 +1511,19 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
                 break;
               }
             }
-            if (err == SBRDEC_OK) {
+            switch (err) {
+            case SBRDEC_PARSE_ERROR:
+              /* Can not go on parsing because we do not
+                 know the length of the SBR extension data. */
+              FDKpushFor(bs, bitCnt);
+              bitCnt = 0;
+              break;
+            case SBRDEC_OK:
               self->sbrEnabled = 1;
-            } else {
+              break;
+            default:
               self->frameOK = 0;
+              break;
             }
           }
 
@@ -1603,13 +1613,17 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
     self->frameOK=0;
   }
 
-  /* store or restore the number of channels */
+  /* store or restore the number of channels and the corresponding info */
   if ( self->frameOK && !(flags &(AACDEC_CONCEAL|AACDEC_FLUSH)) ) {
-    self->concealChannels = aacChannels;  /* store */
+    self->aacChannelsPrev = aacChannels;  /* store */
+    FDKmemcpy(self->channelTypePrev, self->channelType, (8)*sizeof(AUDIO_CHANNEL_TYPE));  /* store */
+    FDKmemcpy(self->channelIndicesPrev, self->channelIndices, (8)*sizeof(UCHAR));         /* store */
     self->sbrEnabledPrev = self->sbrEnabled;
   } else {
     if (self->aacChannels > 0) {
-      aacChannels = self->concealChannels;  /* restore */
+      aacChannels = self->aacChannelsPrev;  /* restore */
+      FDKmemcpy(self->channelType, self->channelTypePrev, (8)*sizeof(AUDIO_CHANNEL_TYPE));  /* restore */
+      FDKmemcpy(self->channelIndices, self->channelIndicesPrev, (8)*sizeof(UCHAR));         /* restore */
       self->sbrEnabled = self->sbrEnabledPrev;
      }
   }
@@ -1632,12 +1646,31 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
     return ErrorStatus;
   }
 
+  /* Setup the output channel mapping. The table below shows the four possibilities:
+   *   # | chCfg | PCE | cChCfg | chOutMapIdx
+   *  ---+-------+-----+--------+------------------
+   *   1 |  > 0  |  no |    -   | chCfg
+   *   2 |   0   | yes |  > 0   | cChCfg
+   *   3 |   0   | yes |    0   | aacChannels || 0
+   *   4 |   0   |  no |    -   | aacChannels || 0
+   *  ---+-------+-----+--------+------------------
+   *  Where chCfg is the channel configuration index from ASC and cChCfg is a corresponding chCfg
+   *  derived from a given PCE. The variable aacChannels represents the number of channel found
+   *  during bitstream decoding. Due to the structure of the mapping table it can only be used for
+   *  mapping if its value is smaller than 7. Otherwise we use the fallback (0) which is a simple
+   *  pass-through. The possibility #4 should appear only with MPEG-2 (ADTS) streams. This is
+   *  mode is called "implicit channel mapping".
+   */
+  chOutMapIdx = ((self->chMapIndex==0) && (aacChannels<7)) ? aacChannels : self->chMapIndex;
+
   /*
     Inverse transform
   */
   {
     int stride, offset, c;
 
+    /* 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 );
     /* Extract DRC control data and map it to channels (without bitstream delay) */
     aacDecoder_drcProlog (
             self->hDrcInfo,
@@ -1663,13 +1696,18 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
       /* Setup offset and stride for time buffer traversal. */
       if (interleaved) {
         stride = aacChannels;
-        offset = self->channelOutputMapping[aacChannels-1][c];
+        offset = self->channelOutputMapping[chOutMapIdx][c];
       } else {
         stride = 1;
-        offset = self->channelOutputMapping[aacChannels-1][c] * self->streamInfo.aacSamplesPerFrame;
+        offset = self->channelOutputMapping[chOutMapIdx][c] * self->streamInfo.aacSamplesPerFrame;
       }
 
 
+      if ( flags&AACDEC_FLUSH ) {
+        /* Clear pAacDecoderChannelInfo->pSpectralCoefficient because with AACDEC_FLUSH set it contains undefined data. */
+        FDKmemclear(pAacDecoderChannelInfo->pSpectralCoefficient, sizeof(FIXP_DBL)*self->streamInfo.aacSamplesPerFrame);
+      }
+
       /*
         Conceal defective spectral data
       */
@@ -1688,12 +1726,15 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
         /* Reset DRC control data for this channel */
         aacDecoder_drcInitChannelData ( &self->pAacDecoderStaticChannelInfo[c]->drcData );
       }
+      /* The DRC module demands to be called with the gain field holding the gain scale. */
+      self->extGain[0] = (FIXP_DBL)TDL_GAIN_SCALING;
       /* DRC processing */
       aacDecoder_drcApply (
               self->hDrcInfo,
               self->hSbrDecoder,
               pAacDecoderChannelInfo,
              &self->pAacDecoderStaticChannelInfo[c]->drcData,
+              self->extGain,
               c,
               self->streamInfo.aacSamplesPerFrame,
               self->sbrEnabled
@@ -1711,6 +1752,7 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
                   (self->frameOK && !(flags&AACDEC_CONCEAL)),
                   self->aacCommonData.workBufferCore1->mdctOutTemp
                   );
+          self->extGainDelay = self->streamInfo.aacSamplesPerFrame;
           break;
         case AACDEC_RENDER_ELDFB:
           CBlock_FrequencyToTimeLowDelay(
@@ -1720,6 +1762,7 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
                   self->streamInfo.aacSamplesPerFrame,
                   stride
                   );
+          self->extGainDelay = (self->streamInfo.aacSamplesPerFrame*2 -  self->streamInfo.aacSamplesPerFrame/2 - 1)/2;
           break;
         default:
           ErrorStatus = AAC_DEC_UNKNOWN;
@@ -1743,11 +1786,20 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
           );
   }
 
+  /* Add additional concealment delay */
+  self->streamInfo.outputDelay += CConcealment_GetDelay(&self->concealCommonData) * self->streamInfo.aacSamplesPerFrame;
+
+  /* Map DRC data to StreamInfo structure */
+  aacDecoder_drcGetInfo (
+            self->hDrcInfo,
+           &self->streamInfo.drcPresMode,
+           &self->streamInfo.drcProgRefLev
+          );
 
   /* Reorder channel type information tables.  */
   {
-    AUDIO_CHANNEL_TYPE types[(6)];
-    UCHAR idx[(6)];
+    AUDIO_CHANNEL_TYPE types[(8)];
+    UCHAR idx[(8)];
     int c;
 
     FDK_ASSERT(sizeof(self->channelType) == sizeof(types));
@@ -1757,8 +1809,8 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
     FDKmemcpy(idx, self->channelIndices, sizeof(idx));
 
     for (c=0; c<aacChannels; c++) {
-      self->channelType[self->channelOutputMapping[aacChannels-1][c]] = types[c];
-      self->channelIndices[self->channelOutputMapping[aacChannels-1][c]] = idx[c];
+      self->channelType[self->channelOutputMapping[chOutMapIdx][c]] = types[c];
+      self->channelIndices[self->channelOutputMapping[chOutMapIdx][c]] = idx[c];
     }
   }
 
diff --git a/libAACdec/src/aacdecoder.h b/libAACdec/src/aacdecoder.h
index 16351e6..3541773 100644
--- a/libAACdec/src/aacdecoder.h
+++ b/libAACdec/src/aacdecoder.h
@@ -111,6 +111,7 @@ amm-info@iis.fraunhofer.de
 #include "aacdec_drc.h"
 
  #include "pcmutils_lib.h"
+ #include "limiter.h"
 
 
 /* Capabilities flags */
@@ -176,27 +177,31 @@ struct AAC_DECODER_INSTANCE {
 
   UINT                  flags;                       /*!< Flags for internal decoder use. DO NOT USE self::streaminfo::flags ! */
 
-  MP4_ELEMENT_ID        elements[7]; /*!< Table where the element Id's are listed          */
-  UCHAR                 elTags[7];   /*!< Table where the elements id Tags are listed      */
-  UCHAR                 chMapping[(6)];   /*!< Table of MPEG canonical order to bitstream channel order mapping. */
+  MP4_ELEMENT_ID        elements[(8)]; /*!< Table where the element Id's are listed          */
+  UCHAR                 elTags[(8)];   /*!< Table where the elements id Tags are listed      */
+  UCHAR                 chMapping[(8)];   /*!< Table of MPEG canonical order to bitstream channel order mapping. */
 
-  AUDIO_CHANNEL_TYPE    channelType[(6)];    /*!< Audio channel type of each output audio channel (from 0 upto numChannels).           */
-  UCHAR                 channelIndices[(6)]; /*!< Audio channel index for each output audio channel (from 0 upto numChannels).         */
+  AUDIO_CHANNEL_TYPE    channelType[(8)];    /*!< Audio channel type of each output audio channel (from 0 upto numChannels).           */
+  UCHAR                 channelIndices[(8)]; /*!< Audio channel index for each output audio channel (from 0 upto numChannels).         */
                                                              /* See ISO/IEC 13818-7:2005(E), 8.5.3.2 Explicit channel mapping using a program_config_element() */
 
 
   const UCHAR         (*channelOutputMapping)[8];    /*!< Table for MPEG canonical order to output channel order mapping. */
-
+  UCHAR                 chMapIndex;                  /*!< Index to access one line of the channelOutputMapping table. This is required
+                                                          because not all 8 channel configurations have the same output mapping. */
 
   CProgramConfig                pce;
   CStreamInfo                   streamInfo;         /*!< pointer to StreamInfo data (read from the bitstream) */
-  CAacDecoderChannelInfo       *pAacDecoderChannelInfo[(6)];       /*!< Temporal channel memory */
-  CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo[(6)]; /*!< Persistent channel memory */
+  CAacDecoderChannelInfo       *pAacDecoderChannelInfo[(8)];       /*!< Temporal channel memory */
+  CAacDecoderStaticChannelInfo *pAacDecoderStaticChannelInfo[(8)]; /*!< Persistent channel memory */
 
   CAacDecoderCommonData         aacCommonData;             /*!< Temporal shared data for all channels hooked into pAacDecoderChannelInfo */
 
   CConcealParams                concealCommonData;
-  INT   concealChannels;
+
+  INT                   aacChannelsPrev;                          /*!< The amount of AAC core channels of the last successful decode call.         */
+  AUDIO_CHANNEL_TYPE    channelTypePrev[(8)];     /*!< Array holding the channelType values of the last successful decode call.    */
+  UCHAR                 channelIndicesPrev[(8)];  /*!< Array holding the channelIndices values of the last successful decode call. */
 
 
   HANDLE_SBRDECODER   hSbrDecoder;                   /*!< SBR decoder handle.                        */
@@ -214,6 +219,12 @@ struct AAC_DECODER_INSTANCE {
   CAncData      ancData;                             /*!< structure to handle ancillary data         */
 
   HANDLE_PCM_DOWNMIX  hPcmUtils;                     /*!< privat data for the PCM utils.             */
+  TDLimiterPtr hLimiter;                             /*!< Handle of time domain limiter.             */
+  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. */
 
 };
 
diff --git a/libAACdec/src/aacdecoder_lib.cpp b/libAACdec/src/aacdecoder_lib.cpp
index ec8f41e..82f85ab 100644
--- a/libAACdec/src/aacdecoder_lib.cpp
+++ b/libAACdec/src/aacdecoder_lib.cpp
@@ -110,7 +110,7 @@ amm-info@iis.fraunhofer.de
 /* Decoder library info */
 #define AACDECODER_LIB_VL0 2
 #define AACDECODER_LIB_VL1 5
-#define AACDECODER_LIB_VL2 5
+#define AACDECODER_LIB_VL2 10
 #define AACDECODER_LIB_TITLE "AAC Decoder Lib"
 #define AACDECODER_LIB_BUILD_DATE __DATE__
 #define AACDECODER_LIB_BUILD_TIME __TIME__
@@ -397,12 +397,14 @@ aacDecoder_SetParam ( const HANDLE_AACDECODER  self,   /*!< Handle of the decode
   CConcealParams  *pConcealData = NULL;
   HANDLE_AAC_DRC hDrcInfo = NULL;
   HANDLE_PCM_DOWNMIX hPcmDmx = NULL;
+  TDLimiterPtr hPcmTdl = NULL;
 
   /* check decoder handle */
   if (self != NULL) {
     pConcealData = &self->concealCommonData;
     hDrcInfo = self->hDrcInfo;
     hPcmDmx = self->hPcmUtils;
+    hPcmTdl = self->hLimiter;
   } else {
     errorStatus = AAC_DEC_INVALID_HANDLE;
   }
@@ -420,8 +422,8 @@ aacDecoder_SetParam ( const HANDLE_AACDECODER  self,   /*!< Handle of the decode
     self->outputInterleaved = value;
     break;
 
-  case AAC_PCM_OUTPUT_CHANNELS:
-    if (value < -1 || value > (6)) {
+  case AAC_PCM_MIN_OUTPUT_CHANNELS:
+    if (value < -1 || value > (8)) {
       return AAC_DEC_SET_PARAM_FAIL;
     }
     {
@@ -429,7 +431,30 @@ aacDecoder_SetParam ( const HANDLE_AACDECODER  self,   /*!< Handle of the decode
 
       err = pcmDmx_SetParam (
               hPcmDmx,
-              NUMBER_OF_OUTPUT_CHANNELS,
+              MIN_NUMBER_OF_OUTPUT_CHANNELS,
+              value );
+
+      switch (err) {
+      case PCMDMX_OK:
+        break;
+      case PCMDMX_INVALID_HANDLE:
+        return AAC_DEC_INVALID_HANDLE;
+      default:
+        return AAC_DEC_SET_PARAM_FAIL;
+      }
+    }
+    break;
+
+  case AAC_PCM_MAX_OUTPUT_CHANNELS:
+    if (value < -1 || value > (8)) {
+      return AAC_DEC_SET_PARAM_FAIL;
+    }
+    {
+      PCMDMX_ERROR err;
+
+      err = pcmDmx_SetParam (
+              hPcmDmx,
+              MAX_NUMBER_OF_OUTPUT_CHANNELS,
               value );
 
       switch (err) {
@@ -449,7 +474,7 @@ aacDecoder_SetParam ( const HANDLE_AACDECODER  self,   /*!< Handle of the decode
 
       err = pcmDmx_SetParam (
               hPcmDmx,
-              DUAL_CHANNEL_DOWNMIX_MODE,
+              DMX_DUAL_CHANNEL_MODE,
               value );
 
       switch (err) {
@@ -463,6 +488,47 @@ aacDecoder_SetParam ( const HANDLE_AACDECODER  self,   /*!< Handle of the decode
     }
     break;
 
+
+  case AAC_PCM_LIMITER_ENABLE:
+    if (value < -1 || value > 1) {
+      return AAC_DEC_SET_PARAM_FAIL;
+    }
+    if (self == NULL) {
+      return AAC_DEC_INVALID_HANDLE;
+    }
+    self->limiterEnableUser = value;
+    break;
+
+  case AAC_PCM_LIMITER_ATTACK_TIME:
+    if (value <= 0) {  /* module function converts value to unsigned */
+      return AAC_DEC_SET_PARAM_FAIL;
+    }
+    switch (setLimiterAttack(hPcmTdl, value)) {
+    case TDLIMIT_OK:
+      break;
+    case TDLIMIT_INVALID_HANDLE:
+      return AAC_DEC_INVALID_HANDLE;
+    case TDLIMIT_INVALID_PARAMETER:
+    default:
+      return AAC_DEC_SET_PARAM_FAIL;
+    }
+    break;
+
+  case AAC_PCM_LIMITER_RELEAS_TIME:
+    if (value <= 0) {  /* module function converts value to unsigned */
+      return AAC_DEC_SET_PARAM_FAIL;
+    }
+    switch (setLimiterRelease(hPcmTdl, value)) {
+    case TDLIMIT_OK:
+      break;
+    case TDLIMIT_INVALID_HANDLE:
+      return AAC_DEC_INVALID_HANDLE;
+    case TDLIMIT_INVALID_PARAMETER:
+    default:
+      return AAC_DEC_SET_PARAM_FAIL;
+    }
+    break;
+
   case AAC_PCM_OUTPUT_CHANNEL_MAPPING:
     switch (value) {
       case 0:
@@ -609,6 +675,14 @@ LINKSPEC_CPP HANDLE_AACDECODER aacDecoder_Open(TRANSPORT_TYPE transportFmt, UINT
     goto bail;
   }
 
+  aacDec->hLimiter = createLimiter(TDL_ATTACK_DEFAULT_MS, TDL_RELEASE_DEFAULT_MS, SAMPLE_MAX, (8), 96000);
+  if (NULL == aacDec->hLimiter) {
+    err = -1;
+    goto bail;
+  }
+  aacDec->limiterEnableUser = (UCHAR)-1;
+  aacDec->limiterEnableCurr = 0;
+
 
 
   /* Assure that all modules have same delay */
@@ -768,6 +842,7 @@ LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_DecodeFrame(
     /* Signal bit stream interruption to other modules if required. */
     if ( fTpInterruption || (flags & (AACDEC_INTR|AACDEC_CLRHIST)) )
     {
+      sbrDecoder_SetParam(self->hSbrDecoder, SBR_CLEAR_HISTORY, (flags&AACDEC_CLRHIST));
       aacDecoder_SignalInterruption(self);
       if ( ! (flags & AACDEC_INTR) ) {
         ErrorStatus = AAC_DEC_TRANSPORT_SYNC_ERROR;
@@ -783,6 +858,19 @@ LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_DecodeFrame(
       self->streamInfo.numBadBytes = 0;
       self->streamInfo.numTotalBytes = 0;
     }
+    /* Reset the output delay field. The modules will add their figures one after another. */
+    self->streamInfo.outputDelay = 0;
+
+    if (self->limiterEnableUser==(UCHAR)-1) {
+      /* Enbale limiter for all non-lowdelay AOT's. */
+      self->limiterEnableCurr = ( self->flags & (AC_LD|AC_ELD) ) ? 0 : 1;
+    }
+    else {
+      /* 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));
 
 
     ErrorStatus = CAacDecoder_DecodeFrame(self,
@@ -825,11 +913,15 @@ LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_DecodeFrame(
     if (self->sbrEnabled)
     {
       SBR_ERROR sbrError = SBRDEC_OK;
+      int chOutMapIdx = ((self->chMapIndex==0) && (self->streamInfo.numChannels<7)) ? self->streamInfo.numChannels : self->chMapIndex;
 
       /* set params */
       sbrDecoder_SetParam ( self->hSbrDecoder,
                             SBR_SYSTEM_BITSTREAM_DELAY,
                             self->sbrParams.bsDelay);
+      sbrDecoder_SetParam ( self->hSbrDecoder,
+                            SBR_FLUSH_DATA,
+                            (flags & AACDEC_FLUSH) );
 
       if ( self->streamInfo.aot == AOT_ER_AAC_ELD ) {
         /* Configure QMF */
@@ -838,7 +930,16 @@ LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_DecodeFrame(
                               (self->flags & AC_LD_MPS) ? 1 : 0 );
       }
 
+      {
+        PCMDMX_ERROR dmxErr;
+        INT  maxOutCh = 0;
 
+        dmxErr = pcmDmx_GetParam(self->hPcmUtils, MAX_NUMBER_OF_OUTPUT_CHANNELS, &maxOutCh);
+        if ( (dmxErr == PCMDMX_OK) && (maxOutCh == 1) ) {
+          /* Disable PS processing if we have to create a mono output signal. */
+          self->psPossible = 0;
+        }
+      }
 
 
       /* apply SBR processing */
@@ -846,23 +947,29 @@ LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_DecodeFrame(
                                     pTimeData,
                                    &self->streamInfo.numChannels,
                                    &self->streamInfo.sampleRate,
-                                    self->channelOutputMapping[self->streamInfo.numChannels-1],
+                                    self->channelOutputMapping[chOutMapIdx],
                                     interleaved,
                                     self->frameOK,
                                    &self->psPossible);
 
 
      if (sbrError == SBRDEC_OK) {
+       #define UPS_SCALE  2  /* Maximum upsampling factor is 4 (CELP+SBR) */
+       FIXP_DBL  upsampleFactor = FL2FXCONST_DBL(1.0f/(1<<UPS_SCALE));
 
        /* Update data in streaminfo structure. Assume that the SBR upsampling factor is either 1 or 2 */
        self->flags |= AC_SBR_PRESENT;
        if (self->streamInfo.aacSampleRate != self->streamInfo.sampleRate) {
          if (self->streamInfo.frameSize == 768) {
-           self->streamInfo.frameSize =  (self->streamInfo.aacSamplesPerFrame * 8) / 3;
+           upsampleFactor = FL2FXCONST_DBL(8.0f/(3<<UPS_SCALE));
          } else {
-           self->streamInfo.frameSize =  self->streamInfo.aacSamplesPerFrame << 1;
+           upsampleFactor = FL2FXCONST_DBL(2.0f/(1<<UPS_SCALE));
          }
        }
+       /* Apply upsampling factor to both the core frame length and the core delay */
+       self->streamInfo.frameSize    =       (INT)fMult((FIXP_DBL)self->streamInfo.aacSamplesPerFrame<<UPS_SCALE, upsampleFactor);
+       self->streamInfo.outputDelay  = (UINT)(INT)fMult((FIXP_DBL)self->streamInfo.outputDelay<<UPS_SCALE, upsampleFactor);
+       self->streamInfo.outputDelay += sbrDecoder_GetDelay( self->hSbrDecoder );
 
        if (self->psPossible) {
          self->flags |= AC_PS_PRESENT;
@@ -870,19 +977,20 @@ LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_DecodeFrame(
          self->channelType[1] = ACT_FRONT;
          self->channelIndices[0] = 0;
          self->channelIndices[1] = 1;
-       } else {
-         self->flags &= ~AC_PS_PRESENT;
        }
      }
    }
 
 
+    {
+    INT pcmLimiterScale = 0;
+    PCMDMX_ERROR dmxErr = PCMDMX_OK;
     if ( flags & (AACDEC_INTR | AACDEC_CLRHIST) ) {
       /* delete data from the past (e.g. mixdown coeficients) */
       pcmDmx_Reset( self->hPcmUtils, PCMDMX_RESET_BS_DATA );
     }
     /* do PCM post processing */
-    pcmDmx_ApplyFrame (
+    dmxErr = pcmDmx_ApplyFrame (
             self->hPcmUtils,
             pTimeData,
             self->streamInfo.frameSize,
@@ -890,9 +998,39 @@ LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_DecodeFrame(
             interleaved,
             self->channelType,
             self->channelIndices,
-            self->channelOutputMapping
+            self->channelOutputMapping,
+            (self->limiterEnableCurr) ? &pcmLimiterScale : NULL
       );
+    if (dmxErr == PCMDMX_INVALID_MODE) {
+      /* Announce the framework that the current combination of channel configuration and downmix
+       * settings are not know to produce a predictable behavior and thus maybe produce strange output. */
+      ErrorStatus = AAC_DEC_DECODE_FRAME_ERROR;
+    }
 
+    if ( flags & AACDEC_CLRHIST ) {
+      /* Delete the delayed signal. */
+      resetLimiter(self->hLimiter);
+    }
+    if (self->limiterEnableCurr)
+    {
+      /* Set actual signal parameters */
+      setLimiterNChannels(self->hLimiter, self->streamInfo.numChannels);
+      setLimiterSampleRate(self->hLimiter, self->streamInfo.sampleRate);
+
+      applyLimiter(
+              self->hLimiter,
+              pTimeData,
+              self->extGain,
+             &pcmLimiterScale,
+              1,
+              self->extGainDelay,
+              self->streamInfo.frameSize
+              );
+
+      /* Announce the additional limiter output delay */
+      self->streamInfo.outputDelay += getLimiterDelay(self->hLimiter);
+    }
+    }
 
 
     /* Signal interruption to take effect in next frame. */
@@ -917,6 +1055,9 @@ LINKSPEC_CPP void aacDecoder_Close ( HANDLE_AACDECODER self )
     return;
 
 
+  if (self->hLimiter != NULL) {
+    destroyLimiter(self->hLimiter);
+  }
 
   if (self->hPcmUtils != NULL) {
     pcmDmx_Close( &self->hPcmUtils );
diff --git a/libAACdec/src/conceal.cpp b/libAACdec/src/conceal.cpp
index c26051c..1c313ef 100644
--- a/libAACdec/src/conceal.cpp
+++ b/libAACdec/src/conceal.cpp
@@ -762,7 +762,6 @@ int
   CConcealment_UpdateState( hConcealmentInfo,
                             frameOk );
 
-  if ( !frameOk )
   {
     /* Create data for signal rendering according to the selected concealment method and decoder operating mode. */
 
@@ -775,11 +774,13 @@ int
       {
       default:
       case ConcealMethodMute:
-        /* Mute spectral data in case of errors */
-        FDKmemclear(pAacDecoderChannelInfo->pSpectralCoefficient, samplesPerFrame * sizeof(FIXP_DBL));
-        /* Set last window shape */
-        pAacDecoderChannelInfo->icsInfo.WindowShape = hConcealmentInfo->windowShape;
-        appliedProcessing = 1;
+        if (!frameOk) {
+          /* Mute spectral data in case of errors */
+          FDKmemclear(pAacDecoderChannelInfo->pSpectralCoefficient, samplesPerFrame * sizeof(FIXP_DBL));
+          /* Set last window shape */
+          pAacDecoderChannelInfo->icsInfo.WindowShape = hConcealmentInfo->windowShape;
+          appliedProcessing = 1;
+        }
         break;
 
       case ConcealMethodNoise:
@@ -801,7 +802,7 @@ int
                                    pSamplingRateInfo,
                                    samplesPerFrame,
                                    0,  /* don't use tonal improvement */
-                                   0);
+                                   frameOk);
         break;
 
       }
diff --git a/libFDK/include/FDK_archdef.h b/libFDK/include/FDK_archdef.h
index cffb943..a831727 100644
--- a/libFDK/include/FDK_archdef.h
+++ b/libFDK/include/FDK_archdef.h
@@ -215,7 +215,7 @@ amm-info@iis.fraunhofer.de
 
 #else
 
-  #error  >>>> Please set architecture characterization defines for your platform (FDK_HIGH_PERFORMANCE)! <<<<
+  #warning  >>>> Please set architecture characterization defines for your platform (FDK_HIGH_PERFORMANCE)! <<<<
 
 #endif /* Architecture switches */
 
diff --git a/libFDK/include/cplx_mul.h b/libFDK/include/cplx_mul.h
index 093ffa6..ce5c9f7 100644
--- a/libFDK/include/cplx_mul.h
+++ b/libFDK/include/cplx_mul.h
@@ -96,7 +96,7 @@ amm-info@iis.fraunhofer.de
 #if defined(__CC_ARM) || defined(__arm__) || defined(_M_ARM)	/* cppp replaced: elif */
 #include "arm/cplx_mul.h"
 
-#elif defined(__GNUC__) && defined(__mips__)	/* cppp replaced: elif */
+#elif defined(__GNUC__) && defined(__mips__) && __mips_isa_rev < 6
 #include "mips/cplx_mul.h"
 
 #endif /* #if defined all cores: bfin, arm, etc. */
diff --git a/libFDK/include/mips/cplx_mul.h b/libFDK/include/mips/cplx_mul.h
index e05d2b6..43cdbd0 100644
--- a/libFDK/include/mips/cplx_mul.h
+++ b/libFDK/include/mips/cplx_mul.h
@@ -89,7 +89,7 @@ amm-info@iis.fraunhofer.de
 ******************************************************************************/
 
 
-#if defined(__GNUC__) && defined(__mips__)
+#if defined(__GNUC__) && defined(__mips__) && __mips_isa_rev < 6
 
 
 //#define FUNCTION_cplxMultDiv2_32x16
diff --git a/libMpegTPDec/include/tp_data.h b/libMpegTPDec/include/tp_data.h
index 5269858..c6e89b5 100644
--- a/libMpegTPDec/include/tp_data.h
+++ b/libMpegTPDec/include/tp_data.h
@@ -146,12 +146,15 @@ typedef struct
 
   UCHAR FrontElementIsCpe[PC_FSB_CHANNELS_MAX];
   UCHAR FrontElementTagSelect[PC_FSB_CHANNELS_MAX];
+  UCHAR FrontElementHeightInfo[PC_FSB_CHANNELS_MAX];
 
   UCHAR SideElementIsCpe[PC_FSB_CHANNELS_MAX];
   UCHAR SideElementTagSelect[PC_FSB_CHANNELS_MAX];
+  UCHAR SideElementHeightInfo[PC_FSB_CHANNELS_MAX];
 
   UCHAR BackElementIsCpe[PC_FSB_CHANNELS_MAX];
   UCHAR BackElementTagSelect[PC_FSB_CHANNELS_MAX];
+  UCHAR BackElementHeightInfo[PC_FSB_CHANNELS_MAX];
 
   UCHAR LfeElementTagSelect[PC_LFE_CHANNELS_MAX];
 
@@ -324,16 +327,23 @@ int getSamplingRateIndex( UINT samplingRate )
  */
 static inline int getNumberOfTotalChannels(int channelConfig)
 {
-  if (channelConfig > 0 && channelConfig < 8)
-    return (channelConfig == 7)?8:channelConfig;
-  else
+  switch (channelConfig) {
+  case 1: case 2: case 3:
+  case 4: case 5: case 6:
+    return channelConfig;
+  case 7: case 12: case 14:
+    return 8;
+  case 11:
+    return 7;
+  default:
     return 0;
+  }
 }
 
 static inline
 int getNumberOfEffectiveChannels(const int channelConfig)
-{
-  const int n[] = {0,1,2,3,4,5,5,7};
+{        /* index: 0,1,2,3,4,5,6,7,8,9,10,11,12,13,14,15 */
+  const int n[] = {0,1,2,3,4,5,5,7,0,0, 0, 6, 7, 0, 7, 0};
   return n[channelConfig];
 }
 
diff --git a/libMpegTPDec/include/tpdec_lib.h b/libMpegTPDec/include/tpdec_lib.h
index fb4c41d..2ad397d 100644
--- a/libMpegTPDec/include/tpdec_lib.h
+++ b/libMpegTPDec/include/tpdec_lib.h
@@ -151,6 +151,7 @@ typedef enum {
 #define PC_ASSOCDATA_MAX    8
 #define PC_CCEL_MAX         16 /* CC elements */
 #define PC_COMMENTLENGTH    256
+#define PC_NUM_HEIGHT_LAYER 3
 
 
 /*!
@@ -239,14 +240,20 @@ int CProgramConfig_LookupElement(
         );
 
 /**
- * \brief        Get table of elements in canonical order.
- * \param pPce   A valid program config structure.
- * \param table  An array where the element IDs are stored.
- * \return       Total element count including all SCE, CPE and LFE.
+ * \brief             Get table of elements in canonical order from a
+ *                    give program config field.
+ * \param pPce        A valid program config structure.
+ * \param table       An array where the element IDs are stored.
+ * \param elListSize  The length of the table array.
+ * \param pChMapIdx   Pointer to a field receiving the corresponding
+ *                    implicit channel configuration index of the given
+ *                    PCE. If none can be found it receives the value 0.
+ * \return            Total element count including all SCE, CPE and LFE.
  */
 int CProgramConfig_GetElementTable( const CProgramConfig *pPce,
                                     MP4_ELEMENT_ID  table[],
-                                    const INT elListSize );
+                                    const INT elListSize,
+                                    UCHAR *pChMapIdx );
 
 /**
  * \brief       Initialize a given AudioSpecificConfig structure.
diff --git a/libMpegTPDec/src/tpdec_asc.cpp b/libMpegTPDec/src/tpdec_asc.cpp
index 4d7cc02..82adec1 100644
--- a/libMpegTPDec/src/tpdec_asc.cpp
+++ b/libMpegTPDec/src/tpdec_asc.cpp
@@ -90,6 +90,9 @@ amm-info@iis.fraunhofer.de
 
 #include "tpdec_lib.h"
 #include "tp_data.h"
+#ifdef TP_PCE_ENABLE
+#include "FDK_crc.h"
+#endif
 
 
 void CProgramConfig_Reset(CProgramConfig *pPce)
@@ -111,13 +114,75 @@ int  CProgramConfig_IsValid ( const CProgramConfig *pPce )
 }
 
 #ifdef TP_PCE_ENABLE
+#define PCE_HEIGHT_EXT_SYNC  ( 0xAC )
+
+/*
+ * Read the extension for height info.
+ * return 0 if successfull or -1 if the CRC failed.
+ */
+static
+int CProgramConfig_ReadHeightExt(
+                                  CProgramConfig *pPce,
+                                  HANDLE_FDK_BITSTREAM bs,
+                                  int * const bytesAvailable,
+                                  const UINT alignmentAnchor
+                                )
+{
+  int err = 0;
+  FDK_CRCINFO crcInfo;    /* CRC state info */
+  INT crcReg;
+  FDKcrcInit(&crcInfo, 0x07, 0xFF, 8);
+  crcReg = FDKcrcStartReg(&crcInfo, bs, 0);
+  UINT startAnchor = FDKgetValidBits(bs);
+
+  FDK_ASSERT(pPce != NULL);
+  FDK_ASSERT(bs != NULL);
+  FDK_ASSERT(bytesAvailable != NULL);
+
+  if ( (startAnchor >= 24) && (*bytesAvailable >= 3)
+    && (FDKreadBits(bs,8) == PCE_HEIGHT_EXT_SYNC) )
+  {
+    int i;
+
+    for (i=0; i < pPce->NumFrontChannelElements; i++)
+    {
+      pPce->FrontElementHeightInfo[i] = (UCHAR) FDKreadBits(bs,2);
+    }
+    for (i=0; i < pPce->NumSideChannelElements; i++)
+    {
+      pPce->SideElementHeightInfo[i] = (UCHAR) FDKreadBits(bs,2);
+    }
+    for (i=0; i < pPce->NumBackChannelElements; i++)
+    {
+      pPce->BackElementHeightInfo[i] = (UCHAR) FDKreadBits(bs,2);
+    }
+    FDKbyteAlign(bs, alignmentAnchor);
+
+    FDKcrcEndReg(&crcInfo, bs, crcReg);
+    if ((USHORT)FDKreadBits(bs,8) != FDKcrcGetCRC(&crcInfo)) {
+      /* CRC failed */
+      err = -1;
+    }
+  }
+  else {
+    /* No valid extension data found -> restore the initial bitbuffer state */
+    FDKpushBack(bs, startAnchor - FDKgetValidBits(bs));
+  }
+
+  /* Always report the bytes read. */
+  *bytesAvailable -= (startAnchor - FDKgetValidBits(bs)) >> 3;
+
+  return (err);
+}
+
 void CProgramConfig_Read(
                           CProgramConfig *pPce,
                           HANDLE_FDK_BITSTREAM bs,
                           UINT alignmentAnchor
                         )
 {
-  int i;
+  int i, err = 0;
+  int commentBytes;
 
   pPce->NumEffectiveChannels = 0;
   pPce->NumChannels = 0;
@@ -190,8 +255,12 @@ void CProgramConfig_Read(
   FDKbyteAlign(bs, alignmentAnchor);
 
   pPce->CommentFieldBytes = (UCHAR) FDKreadBits(bs,8);
+  commentBytes = pPce->CommentFieldBytes;
+
+  /* Search for height info extension and read it if available */
+  err = CProgramConfig_ReadHeightExt( pPce, bs, &commentBytes, alignmentAnchor );
 
-  for (i=0; i < pPce->CommentFieldBytes; i++)
+  for (i=0; i < commentBytes; i++)
   {
     UCHAR text;
 
@@ -203,7 +272,7 @@ void CProgramConfig_Read(
     }
   }
 
-  pPce->isValid = 1;
+  pPce->isValid = (err) ? 0 : 1;
 }
 
 /*
@@ -235,6 +304,10 @@ int CProgramConfig_Compare ( const CProgramConfig * const pPce1,
       } else {
         int el, numCh1 = 0, numCh2 = 0;
         for (el = 0; el < pPce1->NumFrontChannelElements; el += 1) {
+          if (pPce1->FrontElementHeightInfo[el] != pPce2->FrontElementHeightInfo[el]) {
+            result = 2; /* different height info */
+            break;
+          }
           numCh1 += pPce1->FrontElementIsCpe[el] ? 2 : 1;
           numCh2 += pPce2->FrontElementIsCpe[el] ? 2 : 1;
         }
@@ -248,6 +321,10 @@ int CProgramConfig_Compare ( const CProgramConfig * const pPce1,
       } else {
         int el, numCh1 = 0, numCh2 = 0;
         for (el = 0; el < pPce1->NumSideChannelElements; el += 1) {
+          if (pPce1->SideElementHeightInfo[el] != pPce2->SideElementHeightInfo[el]) {
+            result = 2; /* different height info */
+            break;
+          }
           numCh1 += pPce1->SideElementIsCpe[el] ? 2 : 1;
           numCh2 += pPce2->SideElementIsCpe[el] ? 2 : 1;
         }
@@ -261,6 +338,10 @@ int CProgramConfig_Compare ( const CProgramConfig * const pPce1,
       } else {
         int el, numCh1 = 0, numCh2 = 0;
         for (el = 0; el < pPce1->NumBackChannelElements; el += 1) {
+          if (pPce1->BackElementHeightInfo[el] != pPce2->BackElementHeightInfo[el]) {
+            result = 2; /* different height info */
+            break;
+          }
           numCh1 += pPce1->BackElementIsCpe[el] ? 2 : 1;
           numCh2 += pPce2->BackElementIsCpe[el] ? 2 : 1;
         }
@@ -290,6 +371,44 @@ void CProgramConfig_GetDefault( CProgramConfig *pPce,
 
   switch (channelConfig) {
   /* - - - - - - - - - - - - - - - - - - - - - - - - - - - - */
+  case 32: /* 7.1 side channel configuration as defined in FDK_audio.h */
+    pPce->NumFrontChannelElements  = 2;
+    pPce->FrontElementIsCpe[0]     = 0;
+    pPce->FrontElementIsCpe[1]     = 1;
+    pPce->NumSideChannelElements   = 1;
+    pPce->SideElementIsCpe[0]      = 1;
+    pPce->NumBackChannelElements   = 1;
+    pPce->BackElementIsCpe[0]      = 1;
+    pPce->NumLfeChannelElements    = 1;
+    pPce->NumChannels              = 8;
+    pPce->NumEffectiveChannels     = 7;
+    pPce->isValid                  = 1;
+    break;
+  /* - - - - - - - - - - - - - - - - - - - - - - - - - - - - */
+  case 12:  /* 3/0/4.1ch surround back */
+    pPce->BackElementIsCpe[1]      = 1;
+    pPce->NumChannels             += 1;
+    pPce->NumEffectiveChannels    += 1;
+  case 11:  /* 3/0/3.1ch */
+    pPce->NumFrontChannelElements += 2;
+    pPce->FrontElementIsCpe[0]     = 0;
+    pPce->FrontElementIsCpe[1]     = 1;
+    pPce->NumBackChannelElements  += 2;
+    pPce->BackElementIsCpe[0]      = 1;
+    pPce->BackElementIsCpe[1]     += 0;
+    pPce->NumLfeChannelElements   += 1;
+    pPce->NumChannels             += 7;
+    pPce->NumEffectiveChannels    += 6;
+    pPce->isValid                  = 1;
+    break;
+  /* - - - - - - - - - - - - - - - - - - - - - - - - - - - - */
+  case 14:  /* 2/0/0-3/0/2-0.1ch front height */
+    pPce->FrontElementHeightInfo[2] = 1;      /* Top speaker */
+  case 7:   /* 5/0/2.1ch front */
+    pPce->NumFrontChannelElements += 1;
+    pPce->FrontElementIsCpe[2]     = 1;
+    pPce->NumChannels             += 2;
+    pPce->NumEffectiveChannels    += 2;
   case 6:   /* 3/0/2.1ch */
     pPce->NumLfeChannelElements   += 1;
     pPce->NumChannels             += 1;
@@ -348,10 +467,11 @@ void CProgramConfig_GetDefault( CProgramConfig *pPce,
 
 /**
  * \brief get implicit audio channel type for given channelConfig and MPEG ordered channel index
- * \param channelConfig MPEG channelConfiguration from 1 upto 7
+ * \param channelConfig MPEG channelConfiguration from 1 upto 14
  * \param index MPEG channel order index
  * \return audio channel type.
  */
+static
 void getImplicitAudioChannelTypeAndIndex(
         AUDIO_CHANNEL_TYPE *chType,
         UCHAR *chIndex,
@@ -364,9 +484,9 @@ void getImplicitAudioChannelTypeAndIndex(
     *chIndex = index;
   } else {
     switch (channelConfig) {
-      case MODE_1_2_1:
-      case MODE_1_2_2:
-      case MODE_1_2_2_1:
+      case 4:  /* SCE, CPE, SCE */
+      case 5:  /* SCE, CPE, CPE */
+      case 6:  /* SCE, CPE, CPE, LFE */
         switch (index) {
           case 3:
           case 4:
@@ -379,12 +499,12 @@ void getImplicitAudioChannelTypeAndIndex(
             break;
         }
         break;
-      case MODE_1_2_2_2_1:
+      case 7:  /* SCE,CPE,CPE,CPE,LFE */
         switch (index) {
           case 3:
           case 4:
-            *chType = ACT_SIDE;
-            *chIndex = index - 3;
+            *chType = ACT_FRONT;
+            *chIndex = index;
             break;
           case 5:
           case 6:
@@ -397,6 +517,42 @@ void getImplicitAudioChannelTypeAndIndex(
             break;
         }
         break;
+      case 11:  /* SCE,CPE,CPE,SCE,LFE */
+        if (index < 6) {
+          *chType = ACT_BACK;
+          *chIndex = index - 3;
+        } else {
+          *chType = ACT_LFE;
+          *chIndex = 0;
+        }
+        break;
+      case 12:  /* SCE,CPE,CPE,CPE,LFE */
+        if (index < 7) {
+          *chType = ACT_BACK;
+          *chIndex = index - 3;
+        } else {
+          *chType = ACT_LFE;
+          *chIndex = 0;
+        }
+        break;
+      case 14:  /* SCE,CPE,CPE,LFE,CPE */
+        switch (index) {
+          case 3:
+          case 4:
+            *chType = ACT_BACK;
+            *chIndex = index - 3;
+            break;
+          case 5:
+            *chType = ACT_LFE;
+            *chIndex = 0;
+            break;
+          case 6:
+          case 7:
+            *chType = ACT_FRONT_TOP;
+            *chIndex = index - 6;  /* handle the top layer independently */
+            break;
+        }
+        break;
       default:
         *chType = ACT_NONE;
         break;
@@ -467,7 +623,24 @@ int CProgramConfig_LookupElement(
     else {
       /* Accept the additional channel(s), only if the tag is in the lists */
       int isCpe = 0, i;
-      int cc = 0, fc = 0, sc = 0, bc = 0, lc = 0, ec = 0; /* Channel and element counters */
+      /* Element counter */
+      int ec[PC_NUM_HEIGHT_LAYER] = {0};
+      /* Channel counters */
+      int cc[PC_NUM_HEIGHT_LAYER] = {0};
+      int fc[PC_NUM_HEIGHT_LAYER] = {0};
+      int sc[PC_NUM_HEIGHT_LAYER] = {0};
+      int bc[PC_NUM_HEIGHT_LAYER] = {0};
+      int lc = 0;;
+
+      /* General MPEG (PCE) composition rules:
+         - Over all:
+             <normal height channels><top height channels><bottom height channels>
+         - Within each height layer:
+             <front channels><side channels><back channels>
+         - Exception:
+             The LFE channels have no height info and thus they are arranged at the very
+             end of the normal height layer channels.
+       */
 
       switch (elType)
       {
@@ -476,87 +649,206 @@ int CProgramConfig_LookupElement(
       case ID_SCE:
         /* search in front channels */
         for (i = 0; i < pPce->NumFrontChannelElements; i++) {
+          int heightLayer = pPce->FrontElementHeightInfo[i];
           if (isCpe == pPce->FrontElementIsCpe[i] && pPce->FrontElementTagSelect[i] == tag) {
-            chMapping[cc] = channelIdx;
-            chType[cc] = ACT_FRONT;
-            chIndex[cc] = fc;
+            int h, elIdx = ec[heightLayer], chIdx = cc[heightLayer];
+            AUDIO_CHANNEL_TYPE aChType = (AUDIO_CHANNEL_TYPE)((heightLayer<<4) | ACT_FRONT);
+            for (h = heightLayer-1; h >= 0; h-=1) {
+              int el;
+              /* Count front channels/elements */
+              for (el = 0; el < pPce->NumFrontChannelElements; el+=1) {
+                if (pPce->FrontElementHeightInfo[el] == h) {
+                  elIdx += 1;
+                  chIdx += (pPce->FrontElementIsCpe[el]) ? 2 : 1;
+                }
+              }
+              /* Count side channels/elements */
+              for (el = 0; el < pPce->NumSideChannelElements; el+=1) {
+                if (pPce->SideElementHeightInfo[el] == h) {
+                  elIdx += 1;
+                  chIdx += (pPce->SideElementIsCpe[el]) ? 2 : 1;
+                }
+              }
+              /* Count back channels/elements */
+              for (el = 0; el < pPce->NumBackChannelElements; el+=1) {
+                if (pPce->BackElementHeightInfo[el] == h) {
+                  elIdx += 1;
+                  chIdx += (pPce->BackElementIsCpe[el]) ? 2 : 1;
+                }
+              }
+              if (h == 0) {  /* normal height */
+                elIdx += pPce->NumLfeChannelElements;
+                chIdx += pPce->NumLfeChannelElements;
+              }
+            }
+            chMapping[chIdx] = channelIdx;
+            chType[chIdx] = aChType;
+            chIndex[chIdx] = fc[heightLayer];
             if (isCpe) {
-              chMapping[cc+1] = channelIdx+1;
-              chType[cc+1] = ACT_FRONT;
-              chIndex[cc+1] = fc+1;
+              chMapping[chIdx+1] = channelIdx+1;
+              chType[chIdx+1] = aChType;
+              chIndex[chIdx+1] = fc[heightLayer]+1;
             }
-            *elMapping = ec;
+            *elMapping = elIdx;
             return 1;
           }
-          ec++;
+          ec[heightLayer] += 1;
           if (pPce->FrontElementIsCpe[i]) {
-            cc+=2; fc+=2;
+            cc[heightLayer] += 2;
+            fc[heightLayer] += 2;
           } else {
-            cc++; fc++;
+            cc[heightLayer] += 1;
+            fc[heightLayer] += 1;
           }
         }
         /* search in side channels */
         for (i = 0; i < pPce->NumSideChannelElements; i++) {
+          int heightLayer = pPce->SideElementHeightInfo[i];
           if (isCpe == pPce->SideElementIsCpe[i] && pPce->SideElementTagSelect[i] == tag) {
-            chMapping[cc] = channelIdx;
-            chType[cc] = ACT_SIDE;
-            chIndex[cc] = sc;
+            int h, elIdx = ec[heightLayer], chIdx = cc[heightLayer];
+            AUDIO_CHANNEL_TYPE aChType = (AUDIO_CHANNEL_TYPE)((heightLayer<<4) | ACT_SIDE);
+            for (h = heightLayer-1; h >= 0; h-=1) {
+              int el;
+              /* Count front channels/elements */
+              for (el = 0; el < pPce->NumFrontChannelElements; el+=1) {
+                if (pPce->FrontElementHeightInfo[el] == h) {
+                  elIdx += 1;
+                  chIdx += (pPce->FrontElementIsCpe[el]) ? 2 : 1;
+                }
+              }
+              /* Count side channels/elements */
+              for (el = 0; el < pPce->NumSideChannelElements; el+=1) {
+                if (pPce->SideElementHeightInfo[el] == h) {
+                  elIdx += 1;
+                  chIdx += (pPce->SideElementIsCpe[el]) ? 2 : 1;
+                }
+              }
+              /* Count back channels/elements */
+              for (el = 0; el < pPce->NumBackChannelElements; el+=1) {
+                if (pPce->BackElementHeightInfo[el] == h) {
+                  elIdx += 1;
+                  chIdx += (pPce->BackElementIsCpe[el]) ? 2 : 1;
+                }
+              }
+              if (h == 0) {  /* LFE channels belong to the normal height layer */
+                elIdx += pPce->NumLfeChannelElements;
+                chIdx += pPce->NumLfeChannelElements;
+              }
+            }
+            chMapping[chIdx] = channelIdx;
+            chType[chIdx] = aChType;
+            chIndex[chIdx] = sc[heightLayer];
             if (isCpe) {
-              chMapping[cc+1] = channelIdx+1;
-              chType[cc+1] = ACT_SIDE;
-              chIndex[cc+1] = sc+1;
+              chMapping[chIdx+1] = channelIdx+1;
+              chType[chIdx+1] = aChType;
+              chIndex[chIdx+1] = sc[heightLayer]+1;
             }
-            *elMapping = ec;
+            *elMapping = elIdx;
             return 1;
           }
-          ec++;
+          ec[heightLayer] += 1;
           if (pPce->SideElementIsCpe[i]) {
-            cc+=2; sc+=2;
+            cc[heightLayer] += 2;
+            sc[heightLayer] += 2;
           } else {
-            cc++; sc++;
+            cc[heightLayer] += 1;
+            sc[heightLayer] += 1;
           }
         }
         /* search in back channels */
         for (i = 0; i < pPce->NumBackChannelElements; i++) {
+          int heightLayer = pPce->BackElementHeightInfo[i];
           if (isCpe == pPce->BackElementIsCpe[i] && pPce->BackElementTagSelect[i] == tag) {
-            chMapping[cc] = channelIdx;
-            chType[cc] = ACT_BACK;
-            chIndex[cc] = bc;
+            int h, elIdx = ec[heightLayer], chIdx = cc[heightLayer];
+            AUDIO_CHANNEL_TYPE aChType = (AUDIO_CHANNEL_TYPE)((heightLayer<<4) | ACT_BACK);
+            for (h = heightLayer-1; h >= 0; h-=1) {
+              int el;
+              /* Count front channels/elements */
+              for (el = 0; el < pPce->NumFrontChannelElements; el+=1) {
+                if (pPce->FrontElementHeightInfo[el] == h) {
+                  elIdx += 1;
+                  chIdx += (pPce->FrontElementIsCpe[el]) ? 2 : 1;
+                }
+              }
+              /* Count side channels/elements */
+              for (el = 0; el < pPce->NumSideChannelElements; el+=1) {
+                if (pPce->SideElementHeightInfo[el] == h) {
+                  elIdx += 1;
+                  chIdx += (pPce->SideElementIsCpe[el]) ? 2 : 1;
+                }
+              }
+              /* Count back channels/elements */
+              for (el = 0; el < pPce->NumBackChannelElements; el+=1) {
+                if (pPce->BackElementHeightInfo[el] == h) {
+                  elIdx += 1;
+                  chIdx += (pPce->BackElementIsCpe[el]) ? 2 : 1;
+                }
+              }
+              if (h == 0) {  /* normal height */
+                elIdx += pPce->NumLfeChannelElements;
+                chIdx += pPce->NumLfeChannelElements;
+              }
+            }
+            chMapping[chIdx] = channelIdx;
+            chType[chIdx] = aChType;
+            chIndex[chIdx] = bc[heightLayer];
             if (isCpe) {
-              chMapping[cc+1] = channelIdx+1;
-              chType[cc+1] = ACT_BACK;
-              chIndex[cc+1] = bc+1;
+              chMapping[chIdx+1] = channelIdx+1;
+              chType[chIdx+1] = aChType;
+              chIndex[chIdx+1] = bc[heightLayer]+1;
             }
-            *elMapping = ec;
+            *elMapping = elIdx;
             return 1;
           }
-          ec++;
+          ec[heightLayer] += 1;
           if (pPce->BackElementIsCpe[i]) {
-            cc+=2; bc+=2;
+            cc[heightLayer] += 2;
+            bc[heightLayer] += 2;
           } else {
-            cc++; bc++;
+            cc[heightLayer] += 1;
+            bc[heightLayer] += 1;
           }
         }
         break;
 
       case ID_LFE:
-        /* Initialize channel counter and element counter */
-        cc = pPce->NumEffectiveChannels;
-        ec = pPce->NumFrontChannelElements+ pPce->NumSideChannelElements + pPce->NumBackChannelElements;
+      { /* Unfortunately we have to go through all normal height
+           layer elements to get the position of the LFE channels.
+           Start with counting the front channels/elements at normal height */
+        for (i = 0; i < pPce->NumFrontChannelElements; i+=1) {
+          int heightLayer = pPce->FrontElementHeightInfo[i];
+          ec[heightLayer] += 1;
+          cc[heightLayer] += (pPce->FrontElementIsCpe[i]) ? 2 : 1;
+        }
+        /* Count side channels/elements at normal height */
+        for (i = 0; i < pPce->NumSideChannelElements; i+=1) {
+          int heightLayer = pPce->SideElementHeightInfo[i];
+          ec[heightLayer] += 1;
+          cc[heightLayer] += (pPce->SideElementIsCpe[i]) ? 2 : 1;
+        }
+        /* Count back channels/elements at normal height */
+        for (i = 0; i < pPce->NumBackChannelElements; i+=1) {
+          int heightLayer = pPce->BackElementHeightInfo[i];
+          ec[heightLayer] += 1;
+          cc[heightLayer] += (pPce->BackElementIsCpe[i]) ? 2 : 1;
+        }
+
         /* search in lfe channels */
         for (i = 0; i < pPce->NumLfeChannelElements; i++) {
+          int elIdx = ec[0];  /* LFE channels belong to the normal height layer */
+          int chIdx = cc[0];
           if ( pPce->LfeElementTagSelect[i] == tag ) {
-            chMapping[cc] = channelIdx;
-            *elMapping = ec;
-            chType[cc] = ACT_LFE;
-            chIndex[cc] = lc;
+            chMapping[chIdx] = channelIdx;
+            *elMapping = elIdx;
+            chType[chIdx] = ACT_LFE;
+            chIndex[chIdx] = lc;
             return 1;
           }
-          ec++;
-          cc++;
-          lc++;
+          ec[0] += 1;
+          cc[0] += 1;
+          lc += 1;
         }
-        break;
+      } break;
 
       /* Non audio elements */
       case ID_CCE:
@@ -590,13 +882,19 @@ int CProgramConfig_LookupElement(
 int CProgramConfig_GetElementTable(
         const CProgramConfig *pPce,
         MP4_ELEMENT_ID  elList[],
-        const INT elListSize
+        const INT elListSize,
+        UCHAR *pChMapIdx
        )
 {
   int i, el = 0;
 
-  if ( elListSize 
-    < pPce->NumFrontChannelElements + pPce->NumSideChannelElements + pPce->NumBackChannelElements + pPce->NumLfeChannelElements 
+  FDK_ASSERT(elList != NULL);
+  FDK_ASSERT(pChMapIdx != NULL);
+
+  *pChMapIdx = 0;
+
+  if ( elListSize
+    < pPce->NumFrontChannelElements + pPce->NumSideChannelElements + pPce->NumBackChannelElements + pPce->NumLfeChannelElements
     )
   {
     return 0;
@@ -623,6 +921,47 @@ int CProgramConfig_GetElementTable(
   }
 
 
+  /* Find an corresponding channel configuration if possible */
+  switch (pPce->NumChannels) {
+  case 1: case 2: case 3: case 4: case 5: case 6:
+    /* One and two channels have no alternatives. The other ones are mapped directly to the
+       corresponding channel config. Because of legacy reasons or for lack of alternative mappings. */
+    *pChMapIdx = pPce->NumChannels;
+    break;
+  case 7:
+    {
+      C_ALLOC_SCRATCH_START(tmpPce, CProgramConfig, 1);
+      /* Create a PCE for the config to test ... */
+      CProgramConfig_GetDefault(tmpPce, 11);
+      /* ... and compare it with the given one. */
+      *pChMapIdx = (!(CProgramConfig_Compare(pPce, tmpPce)&0xE)) ? 11 : 0;
+      /* If compare result is 0 or 1 we can be sure that it is channel config 11. */
+      C_ALLOC_SCRATCH_END(tmpPce, CProgramConfig, 1);
+    }
+    break;
+  case 8:
+    { /* Try the four possible 7.1ch configurations. One after the other. */
+      UCHAR testCfg[4] = { 32, 14, 12, 7};
+      C_ALLOC_SCRATCH_START(tmpPce, CProgramConfig, 1);
+      for (i=0; i<4; i+=1) {
+        /* Create a PCE for the config to test ... */
+        CProgramConfig_GetDefault(tmpPce, testCfg[i]);
+        /* ... and compare it with the given one. */
+        if (!(CProgramConfig_Compare(pPce, tmpPce)&0xE)) {
+          /* If the compare result is 0 or 1 than the two channel configurations match. */
+          /* Explicit mapping of 7.1 side channel configuration to 7.1 rear channel mapping. */
+          *pChMapIdx = (testCfg[i]==32) ? 12 : testCfg[i];
+        }
+      }
+      C_ALLOC_SCRATCH_END(tmpPce, CProgramConfig, 1);
+    }
+    break;
+  default:
+    /* The PCE does not match any predefined channel configuration. */
+    *pChMapIdx = 0;
+    break;
+  }
+
   return el;
 }
 #endif
@@ -714,7 +1053,7 @@ TRANSPORTDEC_ERROR GaSpecificConfig_Parse( CSGaSpecificConfig    *self,
 #ifdef TP_ELD_ENABLE
 
 static INT ld_sbr_header( const CSAudioSpecificConfig *asc,
-                           HANDLE_FDK_BITSTREAM hBs, 
+                           HANDLE_FDK_BITSTREAM hBs,
                            CSTpCallBacks *cb )
 {
   const int channelConfiguration = asc->m_channelConfiguration;
@@ -728,6 +1067,8 @@ static INT ld_sbr_header( const CSAudioSpecificConfig *asc,
   }
 
   switch ( channelConfiguration ) {
+    case 14:
+    case 12:
     case 7:
       error |= cb->cbSbr(cb->cbSbrData, hBs, asc->m_samplingFrequency, asc->m_extensionSamplingFrequency, asc->m_samplesPerFrame, AOT_ER_AAC_ELD, ID_CPE, i++);
     case 6:
@@ -737,6 +1078,8 @@ static INT ld_sbr_header( const CSAudioSpecificConfig *asc,
       error |= cb->cbSbr(cb->cbSbrData, hBs, asc->m_samplingFrequency, asc->m_extensionSamplingFrequency, asc->m_samplesPerFrame, AOT_ER_AAC_ELD, ID_CPE, i++);
       break;
 
+    case 11:
+      error |= cb->cbSbr(cb->cbSbrData, hBs, asc->m_samplingFrequency, asc->m_extensionSamplingFrequency, asc->m_samplesPerFrame, AOT_ER_AAC_ELD, ID_CPE, i++);
     case 4:
       error |= cb->cbSbr(cb->cbSbrData, hBs, asc->m_samplingFrequency, asc->m_extensionSamplingFrequency, asc->m_samplesPerFrame, AOT_ER_AAC_ELD, ID_CPE, i++);
       error |= cb->cbSbr(cb->cbSbrData, hBs, asc->m_samplingFrequency, asc->m_extensionSamplingFrequency, asc->m_samplesPerFrame, AOT_ER_AAC_ELD, ID_SCE, i++);
@@ -802,24 +1145,6 @@ TRANSPORTDEC_ERROR EldSpecificConfig_Parse(
     }
 
     switch (eldExtType) {
-      case ELDEXT_LDSAC:
-        esc->m_useLdQmfTimeAlign = 1;
-        if (cb->cbSsc != NULL) {
-          ErrorStatus = (TRANSPORTDEC_ERROR)cb->cbSsc(
-                  cb->cbSscData,
-                  hBs,
-                  asc->m_aot,
-                  asc->m_samplingFrequency,
-                  1,  /* muxMode */
-                  len
-                  );
-        } else {
-          ErrorStatus = TRANSPORTDEC_UNSUPPORTED_FORMAT;
-        }
-        if (ErrorStatus != TRANSPORTDEC_OK) {
-          goto bail;
-        }
-        break;
       default:
         for(cnt=0; cnt<len; cnt++) {
           FDKreadBits(hBs, 8 );
diff --git a/libMpegTPDec/src/version b/libMpegTPDec/src/version
index 2803347..cda01c8 100644
--- a/libMpegTPDec/src/version
+++ b/libMpegTPDec/src/version
@@ -2,7 +2,7 @@
 /* library info */
 #define TP_LIB_VL0 2
 #define TP_LIB_VL1 3
-#define TP_LIB_VL2 3
+#define TP_LIB_VL2 4
 #define TP_LIB_TITLE "MPEG Transport"
 #define TP_LIB_BUILD_DATE __DATE__
 #define TP_LIB_BUILD_TIME __TIME__
diff --git a/libMpegTPEnc/include/tp_data.h b/libMpegTPEnc/include/tp_data.h
index 5269858..c6e89b5 100644
--- a/libMpegTPEnc/include/tp_data.h
+++ b/libMpegTPEnc/include/tp_data.h
@@ -146,12 +146,15 @@ typedef struct
 
   UCHAR FrontElementIsCpe[PC_FSB_CHANNELS_MAX];
   UCHAR FrontElementTagSelect[PC_FSB_CHANNELS_MAX];
+  UCHAR FrontElementHeightInfo[PC_FSB_CHANNELS_MAX];
 
   UCHAR SideElementIsCpe[PC_FSB_CHANNELS_MAX];
   UCHAR SideElementTagSelect[PC_FSB_CHANNELS_MAX];
+  UCHAR SideElementHeightInfo[PC_FSB_CHANNELS_MAX];
 
   UCHAR BackElementIsCpe[PC_FSB_CHANNELS_MAX];
   UCHAR BackElementTagSelect[PC_FSB_CHANNELS_MAX];
+  UCHAR BackElementHeightInfo[PC_FSB_CHANNELS_MAX];
 
   UCHAR LfeElementTagSelect[PC_LFE_CHANNELS_MAX];
 
@@ -324,16 +327,23 @@ int getSamplingRateIndex( UINT samplingRate )
  */
 static inline int getNumberOfTotalChannels(int channelConfig)
 {
-  if (channelConfig > 0 && channelConfig < 8)
-    return (channelConfig == 7)?8:channelConfig;
-  else
+  switch (channelConfig) {
+  case 1: case 2: case 3:
+  case 4: case 5: case 6:
+    return channelConfig;
+  case 7: case 12: case 14:
+    return 8;
+  case 11:
+    return 7;
+  default:
     return 0;
+  }
 }
 
 static inline
 int getNumberOfEffectiveChannels(const int channelConfig)
-{
-  const int n[] = {0,1,2,3,4,5,5,7};
+{        /* index: 0,1,2,3,4,5,6,7,8,9,10,11,12,13,14,15 */
+  const int n[] = {0,1,2,3,4,5,5,7,0,0, 0, 6, 7, 0, 7, 0};
   return n[channelConfig];
 }
 
diff --git a/libMpegTPEnc/src/version b/libMpegTPEnc/src/version
index 2803347..cda01c8 100644
--- a/libMpegTPEnc/src/version
+++ b/libMpegTPEnc/src/version
@@ -2,7 +2,7 @@
 /* library info */
 #define TP_LIB_VL0 2
 #define TP_LIB_VL1 3
-#define TP_LIB_VL2 3
+#define TP_LIB_VL2 4
 #define TP_LIB_TITLE "MPEG Transport"
 #define TP_LIB_BUILD_DATE __DATE__
 #define TP_LIB_BUILD_TIME __TIME__
diff --git a/libPCMutils/include/limiter.h b/libPCMutils/include/limiter.h
new file mode 100644
index 0000000..0d3d701
--- /dev/null
+++ b/libPCMutils/include/limiter.h
@@ -0,0 +1,233 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+� Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur F�rderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/************************  FDK PCM postprocessor module  *********************
+
+   Author(s):   Matthias Neusinger
+   Description: Hard limiter for clipping prevention
+
+*******************************************************************************/
+
+#ifndef _LIMITER_H_
+#define _LIMITER_H_
+
+
+#include "common_fix.h"
+
+#define TDL_ATTACK_DEFAULT_MS      (15)              /* default attack  time in ms */
+#define TDL_RELEASE_DEFAULT_MS     (50)              /* default release time in ms */
+
+#define TDL_GAIN_SCALING           (15)              /* scaling of gain value. */
+
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+
+typedef enum {
+  TDLIMIT_OK = 0,
+
+  __error_codes_start = -100,
+
+  TDLIMIT_INVALID_HANDLE,
+  TDLIMIT_INVALID_PARAMETER,
+
+  __error_codes_end
+} TDLIMITER_ERROR;
+
+struct TDLimiter;
+typedef struct TDLimiter* TDLimiterPtr;
+
+/******************************************************************************
+* createLimiter                                                               *
+* maxAttackMs:   maximum and initial attack/lookahead time in milliseconds    *
+* releaseMs:     release time in milliseconds (90% time constant)             *
+* threshold:     limiting threshold                                           *
+* maxChannels:   maximum and initial number of channels                       *
+* maxSampleRate: maximum and initial sampling rate in Hz                      *
+* returns:       limiter handle                                               *
+******************************************************************************/
+TDLimiterPtr createLimiter(unsigned int  maxAttackMs,
+                           unsigned int  releaseMs,
+                           INT_PCM       threshold,
+                           unsigned int  maxChannels,
+                           unsigned int  maxSampleRate);
+
+
+/******************************************************************************
+* resetLimiter                                                                *
+* limiter: limiter handle                                                     *
+* returns: error code                                                         *
+******************************************************************************/
+TDLIMITER_ERROR resetLimiter(TDLimiterPtr limiter);
+
+
+/******************************************************************************
+* destroyLimiter                                                              *
+* limiter: limiter handle                                                     *
+* returns: error code                                                         *
+******************************************************************************/
+TDLIMITER_ERROR destroyLimiter(TDLimiterPtr limiter);
+
+/******************************************************************************
+* applyLimiter                                                                *
+* limiter:    limiter handle                                                  *
+* pGain :     pointer to gains to be applied to the signal before limiting,   *
+*             which are downscaled by TDL_GAIN_SCALING bit.                   *
+*             These gains are delayed by gain_delay, and smoothed.            *
+*             Smoothing is done by a butterworth lowpass filter with a cutoff *
+*             frequency which is fixed with respect to the sampling rate.     *
+*             It is a substitute for the smoothing due to windowing and       *
+*             overlap/add, if a gain is applied in frequency domain.          *
+* gain_scale: pointer to scaling exponents to be applied to the signal before *
+*             limiting, without delay and without smoothing                   *
+* gain_size:  number of elements in pGain, currently restricted to 1          *
+* gain_delay: delay [samples] with which the gains in pGain shall be applied  *
+*             gain_delay <= nSamples                                          *
+* samples:    input/output buffer containing interleaved samples              *
+*             precision of output will be DFRACT_BITS-TDL_GAIN_SCALING bits   *
+* nSamples:   number of samples per channel                                   *
+* returns:    error code                                                      *
+******************************************************************************/
+TDLIMITER_ERROR applyLimiter(TDLimiterPtr limiter,
+                 INT_PCM*    samples,
+                 FIXP_DBL*    pGain,
+                 const INT*   gain_scale,
+                 const UINT   gain_size,
+                 const UINT   gain_delay,
+                 const UINT   nSamples);
+
+/******************************************************************************
+* getLimiterDelay                                                             *
+* limiter: limiter handle                                                     *
+* returns: exact delay caused by the limiter in samples                       *
+******************************************************************************/
+unsigned int getLimiterDelay(TDLimiterPtr limiter);
+
+/******************************************************************************
+* setLimiterNChannels                                                         *
+* limiter:   limiter handle                                                   *
+* nChannels: number of channels ( <= maxChannels specified on create)         *
+* returns:   error code                                                       *
+******************************************************************************/
+TDLIMITER_ERROR setLimiterNChannels(TDLimiterPtr limiter, unsigned int nChannels);
+
+/******************************************************************************
+* setLimiterSampleRate                                                        *
+* limiter:    limiter handle                                                  *
+* sampleRate: sampling rate in Hz ( <= maxSampleRate specified on create)     *
+* returns:    error code                                                      *
+******************************************************************************/
+TDLIMITER_ERROR setLimiterSampleRate(TDLimiterPtr limiter, unsigned int sampleRate);
+
+/******************************************************************************
+* setLimiterAttack                                                            *
+* limiter:    limiter handle                                                  *
+* attackMs:   attack time in ms ( <= maxAttackMs specified on create)         *
+* returns:    error code                                                      *
+******************************************************************************/
+TDLIMITER_ERROR setLimiterAttack(TDLimiterPtr limiter, unsigned int attackMs);
+
+/******************************************************************************
+* setLimiterRelease                                                           *
+* limiter:    limiter handle                                                  *
+* releaseMs:  release time in ms                                              *
+* returns:    error code                                                      *
+******************************************************************************/
+TDLIMITER_ERROR setLimiterRelease(TDLimiterPtr limiter, unsigned int releaseMs);
+
+/******************************************************************************
+* setLimiterThreshold                                                         *
+* limiter:    limiter handle                                                  *
+* threshold:  limiter threshold                                               *
+* returns:    error code                                                      *
+******************************************************************************/
+TDLIMITER_ERROR setLimiterThreshold(TDLimiterPtr limiter, INT_PCM threshold);
+
+#ifdef __cplusplus
+}
+#endif
+
+
+#endif  //#ifndef _LIMITER_H_
diff --git a/libPCMutils/include/pcmutils_lib.h b/libPCMutils/include/pcmutils_lib.h
index 5ba74be..e7e6a41 100644
--- a/libPCMutils/include/pcmutils_lib.h
+++ b/libPCMutils/include/pcmutils_lib.h
@@ -95,69 +95,88 @@ amm-info@iis.fraunhofer.de
 #include "machine_type.h"
 #include "common_fix.h"
 #include "FDK_audio.h"
+#include "FDK_bitstream.h"
 
-/* ------------------------ *
- *     MODULE SETTINGS:     *
- * ------------------------ */
-/* #define PCM_UPMIX_ENABLE */       /*!< Generally enable up mixing.                           */
-#define PCM_DOWNMIX_ENABLE           /*!< Generally enable down mixing.                         */
-#define DVB_MIXDOWN_ENABLE           /*!< Enable this to support DVB ancillary data for encoder
-                                          assisted downmixing of MPEG-4 AAC and
-                                          MPEG-1/2 layer 2 streams. PCM_DOWNMIX_ENABLE has to
-                                          be enabled, too!                                      */
-#define MPEG_PCE_MIXDOWN_ENABLE      /*!< Enable this to support MPEG matrix mixdown with a
-                                          coefficient carried in the PCE. PCM_DOWNMIX_ENABLE
-                                          has to be enabled, too!                               */
-/* #define ARIB_MIXDOWN_ENABLE */    /*!< Enable modifications to the MPEG PCE mixdown method
-                                          to fulfill ARIB standard. MPEG_PCE_MIXDOWN_ENABLE has
-                                          to be set.                                            */
 
 /* ------------------------ *
  *     ERROR CODES:         *
  * ------------------------ */
 typedef enum
 {
-  PCMDMX_OK              = 0x0,    /*!< No error happened.                                     */
-  PCMDMX_OUT_OF_MEMORY   = 0x2,    /*!< Not enough memory to set up an instance of the module. */
-  PCMDMX_UNKNOWN         = 0x5,    /*!< Error condition is of unknown reason, or from a third
-                                          party module.                                        */
-  PCMDMX_INVALID_HANDLE,           /*!< The given instance handle is not valid.                */
-  PCMDMX_INVALID_ARGUMENT,         /*!< One of the parameters handed over is invalid.          */
-  PCMDMX_INVALID_CH_CONFIG,        /*!< The given channel configuration is not supported and
-                                          thus no processing was performed.                    */
-  PCMDMX_INVALID_MODE,             /*!< The set configuration/mode is not applicable.          */
-  PCMDMX_UNKNOWN_PARAM,            /*!< The handed parameter is not known/supported.           */
-  PCMDMX_UNABLE_TO_SET_PARAM,      /*!< Unable to set the specific parameter. Most probably
-                                          the value ist out of range.                          */
-  PCMDMX_CORRUPT_ANC_DATA          /*!< The read ancillary data was corrupt.                   */
+  PCMDMX_OK              = 0x0,   /*!< No error happened.                                        */
+
+  pcm_dmx_fatal_error_start,
+  PCMDMX_OUT_OF_MEMORY   = 0x2,   /*!< Not enough memory to set up an instance of the module.    */
+  PCMDMX_UNKNOWN         = 0x5,   /*!< Error condition is of unknown reason, or from a third
+                                       party module.                                             */
+  pcm_dmx_fatal_error_end,
+
+  PCMDMX_INVALID_HANDLE,          /*!< The given instance handle is not valid.                   */
+  PCMDMX_INVALID_ARGUMENT,        /*!< One of the parameters handed over is invalid.             */
+  PCMDMX_INVALID_CH_CONFIG,       /*!< The given channel configuration is not supported and thus
+                                       no processing was performed.                              */
+  PCMDMX_INVALID_MODE,            /*!< The set configuration/mode is not applicable.             */
+  PCMDMX_UNKNOWN_PARAM,           /*!< The handed parameter is not known/supported.              */
+  PCMDMX_UNABLE_TO_SET_PARAM,     /*!< Unable to set the specific parameter. Most probably the
+                                       value ist out of range.                                   */
+  PCMDMX_CORRUPT_ANC_DATA         /*!< The read ancillary data was corrupt.                      */
 
 } PCMDMX_ERROR;
 
+/** Macro to identify fatal errors. */
+#define PCMDMX_IS_FATAL_ERROR(err)   ( (((err)>=pcm_dmx_fatal_error_start)   && ((err)<=pcm_dmx_fatal_error_end))   ? 1 : 0)
 
 /* ------------------------ *
  *     RUNTIME PARAMS:      *
  * ------------------------ */
 typedef enum
 {
-  DMX_BS_DATA_EXPIRY_FRAME,          /*!< The number of frames without new metadata that have to
-                                            go by before the bitstream data expires. The value 0
-                                            disables expiry.                                     */
-  DMX_BS_DATA_DELAY,                 /*!< The number of delay frames of the output samples
-                                            compared to the bitstream data.                      */
-  NUMBER_OF_OUTPUT_CHANNELS ,        /*!< The number of output channels (equals the number of
-                                            channels processed by the audio output setup).       */
-  DUAL_CHANNEL_DOWNMIX_MODE          /*!< Downmix mode for two channel audio data.               */
-
+  DMX_BS_DATA_EXPIRY_FRAME,       /*!< The number of frames without new metadata that have to go
+                                       by before the bitstream data expires. The value 0 disables
+                                       expiry.                                                   */
+  DMX_BS_DATA_DELAY,              /*!< The number of delay frames of the output samples compared
+                                       to the bitstream data.                                    */
+  MIN_NUMBER_OF_OUTPUT_CHANNELS,  /*!< The minimum number of output channels. For all input
+                                       configurations that have less than the given channels the
+                                       module will modify the output automatically to obtain the
+                                       given number of output channels. Mono signals will be
+                                       duplicated. If more than two output channels are desired
+                                       the module just adds empty channels. The parameter value
+                                       must be either -1, 0, 1, 2, 6 or 8. If the value is
+                                       greater than zero and exceeds the value of parameter
+                                       MAX_NUMBER_OF_OUTPUT_CHANNELS the latter will be set to
+                                       the same value. Both values -1 and 0 disable the feature. */
+  MAX_NUMBER_OF_OUTPUT_CHANNELS,  /*!< The maximum number of output channels. For all input
+                                       configurations that have more than the given channels the
+                                       module will apply a mixdown automatically to obtain the
+                                       given number of output channels. The value must be either
+                                       -1, 0, 1, 2, 6 or 8. If it is greater than zero and lower
+                                       or equal than the value of MIN_NUMBER_OF_OUTPUT_CHANNELS
+                                       parameter the latter will be set to the same value.
+                                       The values -1 and 0 disable the feature.                  */
+  DMX_DUAL_CHANNEL_MODE,          /*!< Downmix mode for two channel audio data.                  */
+  DMX_PSEUDO_SURROUND_MODE        /*!< Defines how module handles pseudo surround compatible
+                                       signals. See PSEUDO_SURROUND_MODE type for details.       */
 } PCMDMX_PARAM;
 
-
+/* Parameter value types */
 typedef enum
 {
-  STEREO_MODE = 0x0,           /*!< Leave stereo signals as they are.                            */
-  CH1_MODE    = 0x1,           /*!< Create a dual mono output signal from channel 1.             */
-  CH2_MODE    = 0x2,           /*!< Create a dual mono output signal from channel 2.             */
-  MIXED_MODE  = 0x3            /*!< Create a dual mono output signal by mixing the two channels. */
+  NEVER_DO_PS_DMX = -1,           /*!< Never create a pseudo surround compatible downmix.        */
+  AUTO_PS_DMX     =  0,           /*!< Create a pseudo surround compatible downmix only if
+                                       signalled in bitstreams meta data. (Default)              */
+  FORCE_PS_DMX    =  1            /*!< Always create a pseudo surround compatible downmix.
+                                       CAUTION: This can lead to excessive signal cancellations
+                                       and signal level differences for non-compatible signals.  */
+} PSEUDO_SURROUND_MODE;
 
+typedef enum
+{
+  STEREO_MODE = 0x0,              /*!< Leave stereo signals as they are.                         */
+  CH1_MODE    = 0x1,              /*!< Create a dual mono output signal from channel 1.          */
+  CH2_MODE    = 0x2,              /*!< Create a dual mono output signal from channel 2.          */
+  MIXED_MODE  = 0x3               /*!< Create a dual mono output signal by mixing the two
+                                       channels.                                                 */
 } DUAL_CHANNEL_MODE;
 
 
@@ -178,7 +197,7 @@ extern "C"
 
 /** Open and initialize an instance of the PCM downmix module
  * @param [out] Pointer to a buffer receiving the handle of the new instance.
- * @returns Returns an error code.
+ * @returns     Returns an error code.
  **/
 PCMDMX_ERROR pcmDmx_Open (
     HANDLE_PCM_DOWNMIX *pSelf
@@ -188,20 +207,46 @@ PCMDMX_ERROR pcmDmx_Open (
  * @param [in] Handle of PCM downmix instance.
  * @param [in] Parameter to be set.
  * @param [in] Parameter value.
- * @returns Returns an error code.
+ * @returns    Returns an error code.
  **/
 PCMDMX_ERROR pcmDmx_SetParam (
     HANDLE_PCM_DOWNMIX  self,
-    PCMDMX_PARAM        param,
-    UINT                value
+    const PCMDMX_PARAM  param,
+    const INT           value
+  );
+
+/** Get one parameter value of one PCM downmix module instance.
+ * @param [in] Handle of PCM downmix module instance.
+ * @param [in] Parameter to be set.
+ * @param [out] Pointer to buffer receiving the parameter value.
+ * @returns Returns an error code.
+ **/
+PCMDMX_ERROR pcmDmx_GetParam (
+    HANDLE_PCM_DOWNMIX  self,
+    const PCMDMX_PARAM  param,
+    INT * const         pValue
   );
 
-/** Read the ancillary data transported in DSEs of DVB streams with MPEG-4 content
+/** Read downmix meta-data directly from a given bitstream.
+ * @param [in] Handle of PCM downmix instance.
+ * @param [in] Handle of FDK bitstream buffer.
+ * @param [in] Length of ancillary data in bits.
+ * @param [in] Flag indicating wheter the ancillary data is from a MPEG-1/2 or an MPEG-4 stream.
+ * @returns    Returns an error code.
+ **/
+PCMDMX_ERROR pcmDmx_Parse (
+    HANDLE_PCM_DOWNMIX  self,
+    HANDLE_FDK_BITSTREAM  hBitStream,
+    UINT   ancDataBits,
+    int    isMpeg2
+  );
+
+/** Read downmix meta-data from a given data buffer.
  * @param [in] Handle of PCM downmix instance.
  * @param [in] Pointer to ancillary data buffer.
- * @param [in] Size of ancillary data.
+ * @param [in] Size of ancillary data in bytes.
  * @param [in] Flag indicating wheter the ancillary data is from a MPEG-1/2 or an MPEG-4 stream.
- * @returns Returns an error code.
+ * @returns    Returns an error code.
  **/
 PCMDMX_ERROR pcmDmx_ReadDvbAncData (
     HANDLE_PCM_DOWNMIX  self,
@@ -211,12 +256,11 @@ PCMDMX_ERROR pcmDmx_ReadDvbAncData (
   );
 
 /** Set the matrix mixdown information extracted from the PCE of an AAC bitstream.
- *  Note: Call only if matrix_mixdown_idx_present is true.
  * @param [in] Handle of PCM downmix instance.
  * @param [in] Matrix mixdown index present flag extracted from PCE.
  * @param [in] The 2 bit matrix mixdown index extracted from PCE.
  * @param [in] The pseudo surround enable flag extracted from PCE.
- * @returns Returns an error code.
+ * @returns    Returns an error code.
  **/
 PCMDMX_ERROR pcmDmx_SetMatrixMixdownFromPce (
     HANDLE_PCM_DOWNMIX  self,
@@ -235,34 +279,42 @@ PCMDMX_ERROR pcmDmx_Reset (
     UINT                flags
   );
 
-/** Apply down or up mixing.
+/** Create a mixdown, bypass or extend the output signal depending on the modules settings and the
+ *  respective given input configuration.
  *
  * \param [in]    Handle of PCM downmix module instance.
  * \param [inout] Pointer to time buffer with decoded PCM samples.
- * \param [in]    Pointer where the amount of output samples is returned into.
- * \param [inout] Pointer where the amount of output channels is returned into.
- * \param [in]    Flag which indicates if output time data are writtern interleaved or as subsequent blocks.
- * \param [inout] Array were the corresponding channel type for each output audio channel is stored into.
- * \param [inout] Array were the corresponding channel type index for each output audio channel is stored into.
- * \param [in]    Array containing the output channel mapping to be used (From MPEG PCE ordering to whatever is required).
- *
- * @returns Returns an error code.
+ * \param [in]    The I/O block size which is the number of samples per channel.
+ * \param [inout] Pointer to buffer that holds the number of input channels and where the
+ *                amount of output channels is written to.
+ * \param [in]    Flag which indicates if output time data is writtern interleaved or as
+ *                subsequent blocks.
+ * \param [inout] Array were the corresponding channel type for each output audio channel is
+ *                stored into.
+ * \param [inout] Array were the corresponding channel type index for each output audio channel
+ *                is stored into.
+ * \param [in]    Array containing the output channel mapping to be used (from MPEG PCE ordering
+ *                to whatever is required).
+ * \param [out]   Pointer on a field receiving the scale factor that has to be applied on all
+ *                samples afterwards. If the handed pointer is NULL the final scaling is done
+ *                internally.
+ * @returns       Returns an error code.
  **/
 PCMDMX_ERROR pcmDmx_ApplyFrame (
     HANDLE_PCM_DOWNMIX      self,
     INT_PCM                *pPcmBuf,
     UINT                    frameSize,
     INT                    *nChannels,
-
     int                     fInterleaved,
     AUDIO_CHANNEL_TYPE      channelType[],
     UCHAR                   channelIndices[],
-    const UCHAR             channelMapping[][8]
+    const UCHAR             channelMapping[][8],
+    INT                    *pDmxOutScale
   );
 
 /** Close an instance of the PCM downmix module.
  * @param [inout] Pointer to a buffer containing the handle of the instance.
- * @returns Returns an error code.
+ * @returns       Returns an error code.
  **/
 PCMDMX_ERROR pcmDmx_Close (
     HANDLE_PCM_DOWNMIX *pSelf
@@ -270,7 +322,7 @@ PCMDMX_ERROR pcmDmx_Close (
 
 /** Get library info for this module.
  * @param [out] Pointer to an allocated LIB_INFO structure.
- * @returns Returns an error code.
+ * @returns     Returns an error code.
  */
 PCMDMX_ERROR pcmDmx_GetLibInfo( LIB_INFO *info );
 
diff --git a/libPCMutils/src/limiter.cpp b/libPCMutils/src/limiter.cpp
new file mode 100644
index 0000000..af724f0
--- /dev/null
+++ b/libPCMutils/src/limiter.cpp
@@ -0,0 +1,498 @@
+
+/* -----------------------------------------------------------------------------------------------------------
+Software License for The Fraunhofer FDK AAC Codec Library for Android
+
+� Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur F�rderung der angewandten Forschung e.V.
+  All rights reserved.
+
+ 1.    INTRODUCTION
+The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software that implements
+the MPEG Advanced Audio Coding ("AAC") encoding and decoding scheme for digital audio.
+This FDK AAC Codec software is intended to be used on a wide variety of Android devices.
+
+AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient general perceptual
+audio codecs. AAC-ELD is considered the best-performing full-bandwidth communications codec by
+independent studies and is widely deployed. AAC has been standardized by ISO and IEC as part
+of the MPEG specifications.
+
+Patent licenses for necessary patent claims for the FDK AAC Codec (including those of Fraunhofer)
+may be obtained through Via Licensing (www.vialicensing.com) or through the respective patent owners
+individually for the purpose of encoding or decoding bit streams in products that are compliant with
+the ISO/IEC MPEG audio standards. Please note that most manufacturers of Android devices already license
+these patent claims through Via Licensing or directly from the patent owners, and therefore FDK AAC Codec
+software may already be covered under those patent licenses when it is used for those licensed purposes only.
+
+Commercially-licensed AAC software libraries, including floating-point versions with enhanced sound quality,
+are also available from Fraunhofer. Users are encouraged to check the Fraunhofer website for additional
+applications information and documentation.
+
+2.    COPYRIGHT LICENSE
+
+Redistribution and use in source and binary forms, with or without modification, are permitted without
+payment of copyright license fees provided that you satisfy the following conditions:
+
+You must retain the complete text of this software license in redistributions of the FDK AAC Codec or
+your modifications thereto in source code form.
+
+You must retain the complete text of this software license in the documentation and/or other materials
+provided with redistributions of the FDK AAC Codec or your modifications thereto in binary form.
+You must make available free of charge copies of the complete source code of the FDK AAC Codec and your
+modifications thereto to recipients of copies in binary form.
+
+The name of Fraunhofer may not be used to endorse or promote products derived from this library without
+prior written permission.
+
+You may not charge copyright license fees for anyone to use, copy or distribute the FDK AAC Codec
+software or your modifications thereto.
+
+Your modified versions of the FDK AAC Codec must carry prominent notices stating that you changed the software
+and the date of any change. For modified versions of the FDK AAC Codec, the term
+"Fraunhofer FDK AAC Codec Library for Android" must be replaced by the term
+"Third-Party Modified Version of the Fraunhofer FDK AAC Codec Library for Android."
+
+3.    NO PATENT LICENSE
+
+NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without limitation the patents of Fraunhofer,
+ARE GRANTED BY THIS SOFTWARE LICENSE. Fraunhofer provides no warranty of patent non-infringement with
+respect to this software.
+
+You may use this FDK AAC Codec software or modifications thereto only for purposes that are authorized
+by appropriate patent licenses.
+
+4.    DISCLAIMER
+
+This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright holders and contributors
+"AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES, including but not limited to the implied warranties
+of merchantability and fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
+CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary, or consequential damages,
+including but not limited to procurement of substitute goods or services; loss of use, data, or profits,
+or business interruption, however caused and on any theory of liability, whether in contract, strict
+liability, or tort (including negligence), arising in any way out of the use of this software, even if
+advised of the possibility of such damage.
+
+5.    CONTACT INFORMATION
+
+Fraunhofer Institute for Integrated Circuits IIS
+Attention: Audio and Multimedia Departments - FDK AAC LL
+Am Wolfsmantel 33
+91058 Erlangen, Germany
+
+www.iis.fraunhofer.de/amm
+amm-info@iis.fraunhofer.de
+----------------------------------------------------------------------------------------------------------- */
+
+/************************  FDK PCM postprocessor module  *********************
+
+   Author(s):   Matthias Neusinger
+   Description: Hard limiter for clipping prevention
+
+*******************************************************************************/
+
+#include "limiter.h"
+
+
+struct TDLimiter {
+  unsigned int  attack;
+  FIXP_DBL      attackConst, releaseConst;
+  unsigned int  attackMs, releaseMs, maxAttackMs;
+  FIXP_PCM      threshold;
+  unsigned int  channels, maxChannels;
+  unsigned int  sampleRate, maxSampleRate;
+  FIXP_DBL      cor, max;
+  FIXP_DBL*     maxBuf;
+  FIXP_DBL*     delayBuf;
+  unsigned int  maxBufIdx, delayBufIdx;
+  FIXP_DBL      smoothState0;
+  FIXP_DBL      minGain;
+
+  FIXP_DBL      additionalGainPrev;
+  FIXP_DBL      additionalGainFilterState;
+  FIXP_DBL      additionalGainFilterState1;
+};
+
+/* create limiter */
+TDLimiterPtr createLimiter(
+                           unsigned int  maxAttackMs,
+                           unsigned int  releaseMs,
+                           INT_PCM       threshold,
+                           unsigned int  maxChannels,
+                           unsigned int  maxSampleRate
+                           )
+{
+  TDLimiterPtr limiter = NULL;
+  unsigned int attack, release;
+  FIXP_DBL attackConst, releaseConst, exponent;
+  INT e_ans;
+
+  /* calc attack and release time in samples */
+  attack = (unsigned int)(maxAttackMs * maxSampleRate / 1000);
+  release = (unsigned int)(releaseMs * maxSampleRate / 1000);
+
+  /* alloc limiter struct */
+  limiter = (TDLimiterPtr)FDKcalloc(1, sizeof(struct TDLimiter));
+  if (!limiter) return NULL;
+
+  /* alloc max and delay buffers */
+  limiter->maxBuf   = (FIXP_DBL*)FDKcalloc(attack + 1, sizeof(FIXP_DBL));
+  limiter->delayBuf = (FIXP_DBL*)FDKcalloc(attack * maxChannels, sizeof(FIXP_DBL));
+
+  if (!limiter->maxBuf || !limiter->delayBuf) {
+    destroyLimiter(limiter);
+    return NULL;
+  }
+
+  /* attackConst = pow(0.1, 1.0 / (attack + 1)) */
+  exponent = invFixp(attack+1);
+  attackConst = fPow(FL2FXCONST_DBL(0.1f), 0, exponent, 0, &e_ans);
+  attackConst = scaleValue(attackConst, e_ans);
+
+  /* releaseConst  = (float)pow(0.1, 1.0 / (release + 1)) */
+  exponent = invFixp(release + 1);
+  releaseConst = fPow(FL2FXCONST_DBL(0.1f), 0, exponent, 0, &e_ans);
+  releaseConst = scaleValue(releaseConst, e_ans);
+
+  /* init parameters */
+  limiter->attackMs      = maxAttackMs;
+  limiter->maxAttackMs   = maxAttackMs;
+  limiter->releaseMs     = releaseMs;
+  limiter->attack        = attack;
+  limiter->attackConst   = attackConst;
+  limiter->releaseConst  = releaseConst;
+  limiter->threshold     = (FIXP_PCM)threshold;
+  limiter->channels      = maxChannels;
+  limiter->maxChannels   = maxChannels;
+  limiter->sampleRate    = maxSampleRate;
+  limiter->maxSampleRate = maxSampleRate;
+
+  resetLimiter(limiter);
+
+  return limiter;
+}
+
+
+/* reset limiter */
+TDLIMITER_ERROR resetLimiter(TDLimiterPtr limiter)
+{
+  if (limiter != NULL) {
+
+    limiter->maxBufIdx = 0;
+    limiter->delayBufIdx = 0;
+    limiter->max = (FIXP_DBL)0;
+    limiter->cor = FL2FXCONST_DBL(1.0f/(1<<1));
+    limiter->smoothState0 = FL2FXCONST_DBL(1.0f/(1<<1));
+    limiter->minGain = FL2FXCONST_DBL(1.0f/(1<<1));
+
+    limiter->additionalGainPrev = FL2FXCONST_DBL(1.0f/(1<<TDL_GAIN_SCALING));
+    limiter->additionalGainFilterState = FL2FXCONST_DBL(1.0f/(1<<TDL_GAIN_SCALING));
+    limiter->additionalGainFilterState1 = FL2FXCONST_DBL(1.0f/(1<<TDL_GAIN_SCALING));
+
+    FDKmemset(limiter->maxBuf,   0, (limiter->attack + 1) * sizeof(FIXP_DBL) );
+    FDKmemset(limiter->delayBuf, 0, limiter->attack * limiter->channels * sizeof(FIXP_DBL) );
+  }
+  else {
+    return TDLIMIT_INVALID_HANDLE;
+  }
+
+  return TDLIMIT_OK;
+}
+
+
+/* destroy limiter */
+TDLIMITER_ERROR destroyLimiter(TDLimiterPtr limiter)
+{
+  if (limiter != NULL) {
+    FDKfree(limiter->maxBuf);
+    FDKfree(limiter->delayBuf);
+
+    FDKfree(limiter);
+  }
+  else {
+    return TDLIMIT_INVALID_HANDLE;
+  }
+  return TDLIMIT_OK;
+}
+
+/* apply limiter */
+TDLIMITER_ERROR applyLimiter(TDLimiterPtr limiter,
+                 INT_PCM*    samples,
+                 FIXP_DBL*    pGain,
+                 const INT*   gain_scale,
+                 const UINT   gain_size,
+                 const UINT   gain_delay,
+                 const UINT   nSamples)
+{
+  unsigned int i, j;
+  FIXP_PCM tmp1, tmp2;
+  FIXP_DBL tmp, old, gain, additionalGain, additionalGainUnfiltered;
+  FIXP_DBL minGain = FL2FXCONST_DBL(1.0f/(1<<1));
+
+  FDK_ASSERT(gain_size == 1);
+  FDK_ASSERT(gain_delay <= nSamples);
+
+  if ( limiter == NULL ) return TDLIMIT_INVALID_HANDLE;
+
+  {
+    unsigned int channels       = limiter->channels;
+    unsigned int attack         = limiter->attack;
+    FIXP_DBL     attackConst    = limiter->attackConst;
+    FIXP_DBL     releaseConst   = limiter->releaseConst;
+    FIXP_DBL     threshold      = FX_PCM2FX_DBL(limiter->threshold)>>TDL_GAIN_SCALING;
+
+    FIXP_DBL     max            = limiter->max;
+    FIXP_DBL*    maxBuf         = limiter->maxBuf;
+    unsigned int maxBufIdx      = limiter->maxBufIdx;
+    FIXP_DBL     cor            = limiter->cor;
+    FIXP_DBL*    delayBuf       = limiter->delayBuf;
+    unsigned int delayBufIdx    = limiter->delayBufIdx;
+
+    FIXP_DBL     smoothState0   = limiter->smoothState0;
+    FIXP_DBL     additionalGainSmoothState = limiter->additionalGainFilterState;
+    FIXP_DBL     additionalGainSmoothState1 = limiter->additionalGainFilterState1;
+
+    for (i = 0; i < nSamples; i++) {
+
+      if (i < gain_delay) {
+        additionalGainUnfiltered = limiter->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[] = { FL2FXCONST_SGL(1.000000), FL2FXCONST_SGL(-0.96907) };
+      /* [b,a] = butter(1, 0.001) */
+      //static const FIXP_SGL b[] = { FL2FXCONST_SGL(0.0015683*2.0), FL2FXCONST_SGL( 0.0015683*2.0) };
+      //static const FIXP_SGL a[] = { FL2FXCONST_SGL(1.0000000), FL2FXCONST_SGL(-0.99686) };
+      additionalGain = - fMult(additionalGainSmoothState, a[1]) + fMultDiv2( additionalGainUnfiltered, b[0]) + fMultDiv2(additionalGainSmoothState1, b[1]);
+      additionalGainSmoothState1 = additionalGainUnfiltered;
+      additionalGainSmoothState = additionalGain;
+
+      /* Apply the additional scaling that has no delay and no smoothing */
+      if (gain_scale[0] > 0) {
+        additionalGain <<= gain_scale[0];
+      } else {
+        additionalGain >>= gain_scale[0];
+      }
+
+      /* get maximum absolute sample value of all channels, including the additional gain. */
+      tmp1 = (FIXP_PCM)0;
+      for (j = 0; j < channels; j++) {
+          tmp2 = (FIXP_PCM)samples[i * channels + j];
+          if (tmp2 == (FIXP_PCM)SAMPLE_MIN) /* protect fAbs from -1.0 value */
+              tmp2 = (FIXP_PCM)(SAMPLE_MIN+1);
+        tmp1 = fMax(tmp1, fAbs(tmp2));
+      }
+      tmp = SATURATE_LEFT_SHIFT(fMultDiv2(tmp1, additionalGain), 1, DFRACT_BITS);
+
+      /* set threshold as lower border to save calculations in running maximum algorithm */
+      tmp = fMax(tmp, threshold);
+
+      /* running maximum */
+      old = maxBuf[maxBufIdx];
+      maxBuf[maxBufIdx] = tmp;
+
+      if (tmp >= max) {
+        /* new sample is greater than old maximum, so it is the new maximum */
+        max = tmp;
+      }
+      else if (old < max) {
+        /* maximum does not change, as the sample, which has left the window was
+           not the maximum */
+      }
+      else {
+        /* the old maximum has left the window, we have to search the complete
+           buffer for the new max */
+        max = maxBuf[0];
+        for (j = 1; j <= attack; j++) {
+          if (maxBuf[j] > max) max = maxBuf[j];
+        }
+      }
+      maxBufIdx++;
+      if (maxBufIdx >= attack+1) maxBufIdx = 0;
+
+      /* calc gain */
+      /* gain is downscaled by one, so that gain = 1.0 can be represented */
+      if (max > threshold) {
+        gain = fDivNorm(threshold, max)>>1;
+      }
+      else {
+        gain = FL2FXCONST_DBL(1.0f/(1<<1));
+      }
+
+      /* gain smoothing, method: TDL_EXPONENTIAL */
+      /* first order IIR filter with attack correction to avoid overshoots */
+
+      /* correct the 'aiming' value of the exponential attack to avoid the remaining overshoot */
+      if (gain < smoothState0) {
+        cor = fMin(cor, fMultDiv2((gain - fMultDiv2(FL2FXCONST_SGL(0.1f*(1<<1)),smoothState0)), FL2FXCONST_SGL(1.11111111f/(1<<1)))<<2);
+      }
+      else {
+        cor = gain;
+      }
+
+      /* smoothing filter */
+      if (cor < smoothState0) {
+        smoothState0 = fMult(attackConst,(smoothState0 - cor)) + cor;  /* attack */
+        smoothState0 = fMax(smoothState0, gain); /* avoid overshooting target */
+      }
+      else {
+        /* sign inversion twice to round towards +infinity,
+           so that gain can converge to 1.0 again,
+           for bit-identical output when limiter is not active */
+        smoothState0 = -fMult(releaseConst,-(smoothState0 - cor)) + cor; /* release */
+      }
+
+      gain = smoothState0;
+
+      /* lookahead delay, apply gain */
+      for (j = 0; j < channels; j++) {
+
+        tmp = delayBuf[delayBufIdx * channels + j];
+        delayBuf[delayBufIdx * channels + j] = fMult((FIXP_PCM)samples[i * channels + j], additionalGain);
+
+        /* Apply gain to delayed signal */
+        if (gain < FL2FXCONST_DBL(1.0f/(1<<1)))
+          tmp = fMult(tmp,gain<<1);
+
+        samples[i * channels + j] = FX_DBL2FX_PCM((FIXP_DBL)SATURATE_LEFT_SHIFT(tmp,TDL_GAIN_SCALING,DFRACT_BITS));
+      }
+      delayBufIdx++;
+      if (delayBufIdx >= attack) delayBufIdx = 0;
+
+      /* save minimum gain factor */
+      if (gain < minGain) minGain = gain;
+    }
+
+
+    limiter->max = max;
+    limiter->maxBufIdx = maxBufIdx;
+    limiter->cor = cor;
+    limiter->delayBufIdx = delayBufIdx;
+
+    limiter->smoothState0 = smoothState0;
+    limiter->additionalGainFilterState = additionalGainSmoothState;
+    limiter->additionalGainFilterState1 = additionalGainSmoothState1;
+
+    limiter->minGain = minGain;
+
+    limiter->additionalGainPrev = pGain[0];
+
+    return TDLIMIT_OK;
+  }
+}
+
+/* get delay in samples */
+unsigned int getLimiterDelay(TDLimiterPtr limiter)
+{
+  FDK_ASSERT(limiter != NULL);
+  return limiter->attack;
+}
+
+/* set number of channels */
+TDLIMITER_ERROR setLimiterNChannels(TDLimiterPtr limiter, unsigned int nChannels)
+{
+  if ( limiter == NULL ) return TDLIMIT_INVALID_HANDLE;
+
+  if (nChannels > limiter->maxChannels) return TDLIMIT_INVALID_PARAMETER;
+
+  limiter->channels = nChannels;
+  //resetLimiter(limiter);
+
+  return TDLIMIT_OK;
+}
+
+/* set sampling rate */
+TDLIMITER_ERROR setLimiterSampleRate(TDLimiterPtr limiter, unsigned int sampleRate)
+{
+  unsigned int attack, release;
+  FIXP_DBL attackConst, releaseConst, exponent;
+  INT e_ans;
+
+  if ( limiter == NULL ) return TDLIMIT_INVALID_HANDLE;
+
+  if (sampleRate > limiter->maxSampleRate) return TDLIMIT_INVALID_PARAMETER;
+
+  /* update attack and release time in samples */
+  attack = (unsigned int)(limiter->attackMs * sampleRate / 1000);
+  release = (unsigned int)(limiter->releaseMs * sampleRate / 1000);
+
+  /* attackConst = pow(0.1, 1.0 / (attack + 1)) */
+  exponent = invFixp(attack+1);
+  attackConst = fPow(FL2FXCONST_DBL(0.1f), 0, exponent, 0, &e_ans);
+  attackConst = scaleValue(attackConst, e_ans);
+
+  /* releaseConst  = (float)pow(0.1, 1.0 / (release + 1)) */
+  exponent = invFixp(release + 1);
+  releaseConst = fPow(FL2FXCONST_DBL(0.1f), 0, exponent, 0, &e_ans);
+  releaseConst = scaleValue(releaseConst, e_ans);
+
+  limiter->attack        = attack;
+  limiter->attackConst   = attackConst;
+  limiter->releaseConst  = releaseConst;
+  limiter->sampleRate    = sampleRate;
+
+  /* reset */
+  //resetLimiter(limiter);
+
+  return TDLIMIT_OK;
+}
+
+/* set attack time */
+TDLIMITER_ERROR setLimiterAttack(TDLimiterPtr limiter, unsigned int attackMs)
+{
+  unsigned int attack;
+  FIXP_DBL attackConst, exponent;
+  INT e_ans;
+
+  if ( limiter == NULL ) return TDLIMIT_INVALID_HANDLE;
+
+  if (attackMs > limiter->maxAttackMs) return TDLIMIT_INVALID_PARAMETER;
+
+  /* calculate attack time in samples */
+  attack = (unsigned int)(attackMs * limiter->sampleRate / 1000);
+
+  /* attackConst = pow(0.1, 1.0 / (attack + 1)) */
+  exponent = invFixp(attack+1);
+  attackConst = fPow(FL2FXCONST_DBL(0.1f), 0, exponent, 0, &e_ans);
+  attackConst = scaleValue(attackConst, e_ans);
+
+  limiter->attack        = attack;
+  limiter->attackConst   = attackConst;
+  limiter->attackMs      = attackMs;
+
+  return TDLIMIT_OK;
+}
+
+/* set release time */
+TDLIMITER_ERROR setLimiterRelease(TDLimiterPtr limiter, unsigned int releaseMs)
+{
+  unsigned int release;
+  FIXP_DBL releaseConst, exponent;
+  INT e_ans;
+
+  if ( limiter == NULL ) return TDLIMIT_INVALID_HANDLE;
+
+  /* calculate  release time in samples */
+  release = (unsigned int)(releaseMs * limiter->sampleRate / 1000);
+
+  /* releaseConst  = (float)pow(0.1, 1.0 / (release + 1)) */
+  exponent = invFixp(release + 1);
+  releaseConst = fPow(FL2FXCONST_DBL(0.1f), 0, exponent, 0, &e_ans);
+  releaseConst = scaleValue(releaseConst, e_ans);
+
+  limiter->releaseConst  = releaseConst;
+  limiter->releaseMs     = releaseMs;
+
+  return TDLIMIT_OK;
+}
+
+/* set limiter threshold */
+TDLIMITER_ERROR setLimiterThreshold(TDLimiterPtr limiter, INT_PCM threshold)
+{
+  if ( limiter == NULL ) return TDLIMIT_INVALID_HANDLE;
+
+  limiter->threshold = (FIXP_PCM)threshold;
+
+  return TDLIMIT_OK;
+}
diff --git a/libPCMutils/src/pcmutils_lib.cpp b/libPCMutils/src/pcmutils_lib.cpp
index a84a050..32d8437 100644
--- a/libPCMutils/src/pcmutils_lib.cpp
+++ b/libPCMutils/src/pcmutils_lib.cpp
@@ -84,8 +84,8 @@ amm-info@iis.fraunhofer.de
 /****************************  FDK PCM utils module  **************************
 
    Author(s):   Christian Griebel
-   Description: Defines functions to interface with the PCM post processing
-                module.
+   Description: Defines functions that perform downmixing or a simple channel
+                expansion in the PCM time domain.
 
 *******************************************************************************/
 
@@ -93,21 +93,66 @@ amm-info@iis.fraunhofer.de
 
 #include "genericStds.h"
 #include "fixpoint_math.h"
+#include "FDK_core.h"
+
+
+/* ------------------------ *
+ *  GLOBAL SETTINGS (GFR):  *
+ * ------------------------ */
+#define DSE_METADATA_ENABLE          /*!< Enable this to support MPEG/ETSI DVB ancillary data for
+                                          encoder assisted downmixing of MPEG-4 AAC and
+                                          MPEG-1/2 layer 2 streams.                             */
+#define PCE_METADATA_ENABLE          /*!< Enable this to support MPEG matrix mixdown with a
+                                          coefficient carried in the PCE.                       */
+
+#define PCM_DMX_MAX_IN_CHANNELS          ( 8 )   /* Neither the maximum number of input nor the maximum number of output channels ... */
+#define PCM_DMX_MAX_OUT_CHANNELS         ( 8 )   /* ... must exceed the maximum number of channels that the framework can handle. */
+
+/* ------------------------ *
+ *    SPECIFIC SETTINGS:    *
+ * ------------------------ */
+#define PCM_CHANNEL_EXTENSION_ENABLE             /*!< Allow module to duplicate mono signals or add zero channels to achieve the
+                                                      desired number of output channels. */
+
+#define PCM_DMX_DFLT_MAX_OUT_CHANNELS    ( 6 )   /*!< The maximum number of output channels. If the value is greater than 0 the module
+                                                      will automatically create a mixdown for all input signals with more channels
+                                                      than specified. */
+#define PCM_DMX_DFLT_MIN_OUT_CHANNELS    ( 0 )   /*!< The minimum number of output channels. If the value is greater than 0 the module
+                                                      will do channel extension automatically for all input signals with less channels
+                                                      than specified. */
+#define PCM_DMX_MAX_DELAY_FRAMES         ( 1 )   /*!< The maximum delay frames to align the bitstreams payload with the PCM output. */
+#define PCM_DMX_DFLT_EXPIRY_FRAME        ( 50 )  /*!< If value is greater than 0 the mixdown coefficients will expire by default after the
+                                                      given number of frames. The value 50 corresponds to at least 500ms (FL 960 @ 96kHz) */
+/* #define PCMDMX_DEBUG */
+
+/* Derived setting:
+ *   No need to edit beyond this line. */
+#if defined(DSE_METADATA_ENABLE) || defined(PCE_METADATA_ENABLE) || defined(ARIB_MIXDOWN_ENABLE)
+ #define PCM_DOWNMIX_ENABLE                      /*!< Generally enable down mixing.                         */
+#endif
+#if (PCM_DMX_MAX_IN_CHANNELS > 2) || (PCM_DMX_MAX_OUT_CHANNELS > 2)
+ #define PCM_DMX_MAX_CHANNELS            ( 8 )
+ #define PCM_DMX_MAX_CHANNEL_GROUPS      ( 4 )
+ #define PCM_DMX_MAX_CHANNELS_PER_GROUP  PCM_DMX_MAX_CHANNELS   /* All channels can be in one group */
+#else
+ #define PCM_DMX_MAX_CHANNELS            ( 3 )   /* Need to add 1 because there are three channel positions in first channel group. */
+ #define PCM_DMX_MAX_CHANNEL_GROUPS      ( 1 )   /* Only front channels supported. */
+ #define PCM_DMX_MAX_CHANNELS_PER_GROUP  ( 2 )   /* The maximum over all channel groups */
+#endif
+#if (PCM_DMX_MAX_IN_CHANNELS > PCM_DMX_MAX_OUT_CHANNELS)
+ #define PCM_DMX_MAX_IO_CHANNELS  PCM_DMX_MAX_IN_CHANNELS
+#else
+ #define PCM_DMX_MAX_IO_CHANNELS  PCM_DMX_MAX_OUT_CHANNELS
+#endif
 
 /* Decoder library info */
 #define PCMDMX_LIB_VL0 2
 #define PCMDMX_LIB_VL1 4
-#define PCMDMX_LIB_VL2 0
+#define PCMDMX_LIB_VL2 2
 #define PCMDMX_LIB_TITLE "PCM Downmix Lib"
 #define PCMDMX_LIB_BUILD_DATE __DATE__
 #define PCMDMX_LIB_BUILD_TIME __TIME__
 
-/* Library settings */
-#define PCM_DMX_MAX_DELAY_FRAMES        ( 1 )
-#define PCM_DMX_MAX_CHANNELS            ( 8 )
-#define PCM_DMX_MAX_CHANNEL_GROUPS      ( 4 )
-#define PCM_DMX_MAX_CHANNELS_PER_GROUP  ( 3 )   /* The maximum over all groups */
-#define PCMDMX_DFLT_EXPIRY_FRAME        ( 50 )  /* At least 500ms (FL 960 @ 96kHz) */
 
 /* Fixed and unique channel group indices.
  * The last group index has to be smaller than PCM_DMX_MAX_CHANNEL_GROUPS. */
@@ -122,22 +167,64 @@ amm-info@iis.fraunhofer.de
 #define CENTER_FRONT_CHANNEL    ( 0 )     /* C  */
 #define LEFT_FRONT_CHANNEL      ( 1 )     /* L  */
 #define RIGHT_FRONT_CHANNEL     ( 2 )     /* R  */
-#define LEFT_OUTSIDE_CHANNEL    ( 3 )     /* Lo */
-#define RIGHT_OUTSIDE_CHANNEL   ( 4 )     /* Ro */
-#define LEFT_REAR_CHANNEL       ( 5 )     /* Lr  aka left back channel  */
-#define RIGHT_REAR_CHANNEL      ( 6 )     /* Rr  aka right back channel */
-#define LOW_FREQUENCY_CHANNEL   ( 7 )     /* Lf */
+#define LEFT_REAR_CHANNEL       ( 3 )     /* Lr (aka left back channel) or center back channel */
+#define RIGHT_REAR_CHANNEL      ( 4 )     /* Rr (aka right back channel) */
+#define LOW_FREQUENCY_CHANNEL   ( 5 )     /* Lf */
+#define LEFT_MULTIPRPS_CHANNEL  ( 6 )     /* Left multipurpose channel */
+#define RIGHT_MULTIPRPS_CHANNEL ( 7 )     /* Right multipurpose channel */
 
 /* More constants */
-#define ANC_DATA_SYNC_BYTE      ( 0xBC )  /* ancillary data sync byte. */
-#define ATTENUATION_FACTOR_1    ( FL2FXCONST_SGL(0.70710678f) )
+#define ONE_CHANNEL             ( 1 )
 #define TWO_CHANNEL             ( 2 )
+#define SIX_CHANNEL             ( 6 )
+#define EIGHT_CHANNEL           ( 8 )
+
+#define PCMDMX_A_IDX_DEFAULT    ( 2 )
+#define PCMDMX_B_IDX_DEFAULT    ( 2 )
+#define PCMDMX_LFE_IDX_DEFAULT  ( 15 )
+#define PCMDMX_GAIN_5_DEFAULT   ( 0 )
+#define PCMDMX_GAIN_2_DEFAULT   ( 0 )
+
+#define PCMDMX_MAX_HEADROOM     ( 3 )     /* Defines the maximum PCM scaling headroom that can be done by a
+                                             postprocessing step. This value must be greater or equal to 0. */
+
+#define FALSE  0
+#define TRUE   1
+#define IN     0
+#define OUT    1
+
+/* Type definitions: */
+#ifndef DMX_HIGH_PRECISION_ENABLE
+ #define FIXP_DMX          FIXP_SGL
+ #define FX_DMX2FX_DBL(x)  FX_SGL2FX_DBL((FIXP_SGL)(x))
+ #define FX_DBL2FX_DMX(x)  FX_DBL2FX_SGL(x)
+ #define FL2FXCONST_DMX(x) FL2FXCONST_SGL(x)
+ #define MAXVAL_DMX        MAXVAL_SGL
+ #define FX_DMX2SHRT(x)    ((SHORT)(x))
+ #define FX_DMX2FL(x)      FX_DBL2FL(FX_DMX2FX_DBL(x))
+#else
+ #define FIXP_DMX          FIXP_DBL
+ #define FX_DMX2FX_DBL(x)  ((FIXP_DBL)(x))
+ #define FX_DBL2FX_DMX(x)  ((FIXP_DBL)(x)
+ #define FL2FXCONST_DMX(x) FL2FXCONST_DBL(x)
+ #define MAXVAL_DMX        MAXVAL_DBL
+ #define FX_DMX2SHRT(x)    ((SHORT)((x)>>FRACT_BITS))
+ #define FX_DMX2FL(x)      FX_DBL2FL(x)
+#endif
 
-/* Sanity checks on library setting: */
+/* The number of channels positions for each group in the internal representation.
+ * See the channel labels above. */
+static const UCHAR maxChInGrp[PCM_DMX_MAX_CHANNEL_GROUPS] = {
+#if (PCM_DMX_MAX_CHANNELS > 3)
+  3, 0, 2, 1
+#else
+  PCM_DMX_MAX_CHANNELS_PER_GROUP
+#endif
+};
 
 /* List of packed channel modes */
 typedef enum
-{ /* CH_MODE_<numFrontCh>_<numOutsideCh>_<numRearCh>_<numLfCh> */
+{ /* CH_MODE_<numFrontCh>_<numSideCh>_<numBackCh>_<numLfCh> */
   CH_MODE_UNDEFINED = 0x0000,
   /* 1 channel */
   CH_MODE_1_0_0_0   = 0x0001,   /* chCfg 1 */
@@ -151,53 +238,93 @@ typedef enum
   CH_MODE_3_0_1_0   = 0x0103,   /* chCfg 4 */
   CH_MODE_2_0_2_0   = 0x0202,
   CH_MODE_2_0_1_1   = 0x1102,
+  CH_MODE_4_0_0_0   = 0x0004,
   /* 5 channels */
   CH_MODE_3_0_2_0   = 0x0203,   /* chCfg 5 */
   CH_MODE_2_0_2_1   = 0x1202,
   CH_MODE_3_0_1_1   = 0x1103,
   CH_MODE_3_2_0_0   = 0x0023,
+  CH_MODE_5_0_0_0   = 0x0005,
   /* 6 channels */
   CH_MODE_3_0_2_1   = 0x1203,   /* chCfg 6 */
+  CH_MODE_3_2_0_1   = 0x1023,
   CH_MODE_3_2_1_0   = 0x0123,
+  CH_MODE_5_0_1_0   = 0x0105,
+  CH_MODE_6_0_0_0   = 0x0006,
   /* 7 channels */
   CH_MODE_2_2_2_1   = 0x1222,
+  CH_MODE_3_0_3_1   = 0x1303,   /* chCfg 11 */
   CH_MODE_3_2_1_1   = 0x1123,
   CH_MODE_3_2_2_0   = 0x0223,
+  CH_MODE_3_0_2_2   = 0x2203,
+  CH_MODE_5_0_2_0   = 0x0205,
+  CH_MODE_5_0_1_1   = 0x1105,
+  CH_MODE_7_0_0_0   = 0x0007,
   /* 8 channels */
-  CH_MODE_3_2_2_1   = 0x1222,   /* chCfg 7 */
+  CH_MODE_3_2_2_1   = 0x1223,
+  CH_MODE_3_0_4_1   = 0x1403,   /* chCfg 12 */
+  CH_MODE_5_0_2_1   = 0x1205,   /* chCfg 7 + 14 */
+  CH_MODE_5_2_1_0   = 0x0125,
   CH_MODE_3_2_1_2   = 0x2123,
-  CH_MODE_2_2_2_2   = 0x2222
+  CH_MODE_2_2_2_2   = 0x2222,
+  CH_MODE_3_0_3_2   = 0x2303,
+  CH_MODE_8_0_0_0   = 0x0008
 
 } PCM_DMX_CHANNEL_MODE;
 
 
 /* These are the channel configurations linked to
    the number of output channels give by the user: */
-static const PCM_DMX_CHANNEL_MODE outChModeTable[PCM_DMX_MAX_CHANNELS] =
+static const PCM_DMX_CHANNEL_MODE outChModeTable[PCM_DMX_MAX_CHANNELS+1] =
 {
+  CH_MODE_UNDEFINED,
   CH_MODE_1_0_0_0,  /* 1 channel  */
   CH_MODE_2_0_0_0,  /* 2 channels */
-  CH_MODE_3_0_0_0,  /* 3 channels */
-  CH_MODE_3_0_1_0,  /* 4 channels */
+  CH_MODE_3_0_0_0   /* 3 channels */
+#if (PCM_DMX_MAX_CHANNELS > 3)
+ ,CH_MODE_3_0_1_0,  /* 4 channels */
   CH_MODE_3_0_2_0,  /* 5 channels */
   CH_MODE_3_0_2_1,  /* 6 channels */
-  CH_MODE_3_2_2_0,  /* 7 channels */
-  CH_MODE_3_2_2_1   /* 8 channels */
+  CH_MODE_3_0_3_1,  /* 7 channels */
+  CH_MODE_3_0_4_1   /* 8 channels */
+#endif
 };
 
-static const FIXP_SGL dvbDownmixFactors[8] =
+static const FIXP_DMX abMixLvlValueTab[8] =
 {
-  FL2FXCONST_SGL(1.0f),
-  FL2FXCONST_SGL(0.841f),
-  FL2FXCONST_SGL(0.707f),
-  FL2FXCONST_SGL(0.596f),
-  FL2FXCONST_SGL(0.500f),
-  FL2FXCONST_SGL(0.422f),
-  FL2FXCONST_SGL(0.355f),
-  FL2FXCONST_SGL(0.0f)
+  FL2FXCONST_DMX(0.500f),   /* scaled by 1 */
+  FL2FXCONST_DMX(0.841f),
+  FL2FXCONST_DMX(0.707f),
+  FL2FXCONST_DMX(0.596f),
+  FL2FXCONST_DMX(0.500f),
+  FL2FXCONST_DMX(0.422f),
+  FL2FXCONST_DMX(0.355f),
+  FL2FXCONST_DMX(0.0f)
+};
+
+static const FIXP_DMX lfeMixLvlValueTab[16] =
+{ /*             value,        scale */
+  FL2FXCONST_DMX(0.7905f),  /*     2 */
+  FL2FXCONST_DMX(0.5000f),  /*     2 */
+  FL2FXCONST_DMX(0.8395f),  /*     1 */
+  FL2FXCONST_DMX(0.7065f),  /*     1 */
+  FL2FXCONST_DMX(0.5945f),  /*     1 */
+  FL2FXCONST_DMX(0.500f),   /*     1 */
+  FL2FXCONST_DMX(0.841f),   /*     0 */
+  FL2FXCONST_DMX(0.707f),   /*     0 */
+  FL2FXCONST_DMX(0.596f),   /*     0 */
+  FL2FXCONST_DMX(0.500f),   /*     0 */
+  FL2FXCONST_DMX(0.316f),   /*     0 */
+  FL2FXCONST_DMX(0.178f),   /*     0 */
+  FL2FXCONST_DMX(0.100f),   /*     0 */
+  FL2FXCONST_DMX(0.032f),   /*     0 */
+  FL2FXCONST_DMX(0.010f),   /*     0 */
+  FL2FXCONST_DMX(0.000f)    /*     0 */
 };
 
 
+
+#ifdef PCE_METADATA_ENABLE
   /* MPEG matrix mixdown:
       Set 1:  L' = (1 + 2^-0.5 + A )^-1 * [L + C * 2^-0.5 + A * Ls];
               R' = (1 + 2^-0.5 + A )^-1 * [R + C * 2^-0.5 + A * Rs];
@@ -207,111 +334,176 @@ static const FIXP_SGL dvbDownmixFactors[8] =
 
       M = (3 + 2A)^-1 * [L + C + R + A*(Ls + Rs)];
   */
-  static const FIXP_SGL mpegMixDownIdx2Coef[4] =
+  static const FIXP_DMX mpegMixDownIdx2Coef[4] =
   {
-    FL2FXCONST_SGL(0.70710678f),
-    FL2FXCONST_SGL(0.5f),
-    FL2FXCONST_SGL(0.35355339f),
-    FL2FXCONST_SGL(0.0f)
+    FL2FXCONST_DMX(0.70710678f),
+    FL2FXCONST_DMX(0.5f),
+    FL2FXCONST_DMX(0.35355339f),
+    FL2FXCONST_DMX(0.0f)
   };
 
-  static const FIXP_SGL mpegMixDownIdx2PreFact[4] =
-  {
-    FL2FXCONST_SGL(0.4142135623730950f),
-    FL2FXCONST_SGL(0.4530818393219728f),
-    FL2FXCONST_SGL(0.4852813742385703f),
-    FL2FXCONST_SGL(0.5857864376269050f)
+  static const FIXP_SGL mpegMixDownIdx2PreFact[3][4] =
+  { {  /* Set 1: */
+    FL2FXCONST_DMX(0.4142135623730950f),
+    FL2FXCONST_DMX(0.4530818393219728f),
+    FL2FXCONST_DMX(0.4852813742385703f),
+    FL2FXCONST_DMX(0.5857864376269050f)
+  },{  /* Set 2: */
+    FL2FXCONST_DMX(0.3203772410170407f),
+    FL2FXCONST_DMX(0.3693980625181293f),
+    FL2FXCONST_DMX(0.4142135623730950f),
+    FL2FXCONST_DMX(0.5857864376269050f)
+  },{  /* Mono DMX set: */
+    FL2FXCONST_DMX(0.2265409196609864f),
+    FL2FXCONST_DMX(0.25f),
+    FL2FXCONST_DMX(0.2697521433898179f),
+    FL2FXCONST_DMX(0.3333333333333333f) }
   };
+#endif  /* PCE_METADATA_ENABLE */
 
-  typedef struct
-  {
-    USHORT  matrixMixdownIdx;       /*!< MPEG mixdown index extracted from PCE.            */
-    USHORT  pseudoSurroundEnable;   /*!< Pseudo surround enable flag extracted from PCE.   */
-    USHORT  mixdownAvailable;       /*!< Will be set to 1 if we found a valid coefficient. */
-
-  } MPEG_MIXDOWN_INFO;
 
+#define TYPE_NONE      ( 0x0 )
+#define TYPE_DSE_DATA  ( 0x1 )
+#define TYPE_PCE_DATA  ( 0x2 )
 
 typedef struct
 {
-  FIXP_SGL  centerMixLevelValue;    /*!< DVB mixdown level for the center channel extracted from anc data.  */
-  FIXP_SGL  surroundMixLevelValue;  /*!< DVB mixdown level for back channels extracted from anc data.       */
-
-  UCHAR     mixLevelsAvail;         /*!< Will be set to 1 if we found a valid coefficient.                  */
+  UINT   typeFlags;
+  /* From DSE */
+  UCHAR  cLevIdx;
+  UCHAR  sLevIdx;
+  UCHAR  dmixIdxA;
+  UCHAR  dmixIdxB;
+  UCHAR  dmixIdxLfe;
+  UCHAR  dmxGainIdx2;
+  UCHAR  dmxGainIdx5;
+#ifdef PCE_METADATA_ENABLE
+  /* From PCE */
+  UCHAR  matrixMixdownIdx;
+#endif
+  /* Attributes: */
+  SCHAR  pseudoSurround;               /*!< If set to 1 the signal is pseudo surround compatible. The value 0 tells
+                                            that it is not. If the value is -1 the information is not available.  */
+  UINT   expiryCount;                  /*!< Counter to monitor the life time of a meta data set. */
+
+} DMX_BS_META_DATA;
+
+/* Default metadata */
+static const DMX_BS_META_DATA  dfltMetaData = {
+  0, 2, 2, 2, 2, 15, 0, 0,
+#ifdef PCE_METADATA_ENABLE
+  0,
+#endif
+  -1, 0
+};
 
-} DVB_MIXDOWN_LEVELS;
+/* Dynamic (user) params:
+     See the definition of PCMDMX_PARAM for details on the specific fields. */
+typedef struct
+{
+  UINT   expiryFrame;                   /*!< Linked to DMX_BS_DATA_EXPIRY_FRAME       */
+  DUAL_CHANNEL_MODE dualChannelMode;    /*!< Linked to DMX_DUAL_CHANNEL_MODE          */
+  PSEUDO_SURROUND_MODE pseudoSurrMode;  /*!< Linked to DMX_PSEUDO_SURROUND_MODE       */
+  SHORT  numOutChannelsMin;             /*!< Linked to MIN_NUMBER_OF_OUTPUT_CHANNELS  */
+  SHORT  numOutChannelsMax;             /*!< Linked to MAX_NUMBER_OF_OUTPUT_CHANNELS  */
+  UCHAR  frameDelay;                    /*!< Linked to DMX_BS_DATA_DELAY              */
 
+} PCM_DMX_USER_PARAMS;
 
 /* Modules main data structure: */
 struct PCM_DMX_INSTANCE
 {
-  DVB_MIXDOWN_LEVELS  dvbMixDownLevels[PCM_DMX_MAX_DELAY_FRAMES+1];
-  MPEG_MIXDOWN_INFO   mpegMixDownInfo[PCM_DMX_MAX_DELAY_FRAMES+1];
-  DUAL_CHANNEL_MODE dualChannelMode;
-  UINT expiryFrame;
-  UINT expiryCount;
-  SHORT numOutputChannels;
-  UCHAR applyProcessing;
-  UCHAR frameDelay;
+  /* Metadata */
+  DMX_BS_META_DATA     bsMetaData[PCM_DMX_MAX_DELAY_FRAMES+1];
+  PCM_DMX_USER_PARAMS  userParams;
+
+  UCHAR  applyProcessing;              /*!< Flag to en-/disable modules processing.
+                                            The max channel limiting is done independently. */
 };
 
 /* Memory allocation macro */
 C_ALLOC_MEM_STATIC(PcmDmxInstance, struct PCM_DMX_INSTANCE, 1)
 
 
-/** Evaluate a given channel configuration and extract a packed channel mode and generate a channel offset table
+/** Evaluate a given channel configuration and extract a packed channel mode. In addition the
+ *  function generates a channel offset table for the mapping to the internal representation.
  *  This function is the inverse to the getChannelDescription() routine.
  * @param [in] The total number of channels of the given configuration.
  * @param [in] Array holding the corresponding channel types for each channel.
  * @param [in] Array holding the corresponding channel type indices for each channel.
  * @param [out] Array where the buffer offsets for each channel are stored into.
- * @returns Returns the packed channel mode.
+ * @param [out] The generated packed channel mode that represents the given input configuration.
+ * @returns Returns an error code.
  **/
 static
-PCM_DMX_CHANNEL_MODE getChannelMode (
+PCMDMX_ERROR getChannelMode (
         const INT                numChannels,                           /* in */
         const AUDIO_CHANNEL_TYPE channelType[],                         /* in */
         const UCHAR              channelIndices[],                      /* in */
-        UCHAR                    offsetTable[PCM_DMX_MAX_CHANNELS]      /* out */
+        UCHAR                    offsetTable[PCM_DMX_MAX_CHANNELS],     /* out */
+        PCM_DMX_CHANNEL_MODE    *chMode                                 /* out */
       )
 {
-  UINT  chMode = CH_MODE_UNDEFINED;
   UCHAR chIdx[PCM_DMX_MAX_CHANNEL_GROUPS][PCM_DMX_MAX_CHANNELS_PER_GROUP];
-  UCHAR numChInGrp[PCM_DMX_MAX_CHANNEL_GROUPS];
-  int   ch, grpIdx, err = 0;
+  UCHAR numChInGrp[PCM_DMX_MAX_CHANNEL_GROUPS]; /* Total num of channels per group of the input config */
+  UCHAR numChFree[PCM_DMX_MAX_CHANNEL_GROUPS];  /* Number of free slots per group in the internal repr. */
+  UCHAR hardToPlace[PCM_DMX_MAX_CHANNELS];      /* List of channels not matching the internal repr. */
+  UCHAR h2pSortIdx[PCM_DMX_MAX_CHANNELS];
+  PCMDMX_ERROR err = PCMDMX_OK;
+  int   ch, grpIdx;
+  int   numChToPlace = 0;
 
   FDK_ASSERT(channelType != NULL);
   FDK_ASSERT(channelIndices != NULL);
   FDK_ASSERT(offsetTable != NULL);
+  FDK_ASSERT(chMode != NULL);
 
   /* For details see ISO/IEC 13818-7:2005(E), 8.5.3 Channel configuration */
   FDKmemclear(numChInGrp, PCM_DMX_MAX_CHANNEL_GROUPS*sizeof(UCHAR));
   FDKmemset(offsetTable, 255, PCM_DMX_MAX_CHANNELS*sizeof(UCHAR));
   FDKmemset(chIdx, 255, PCM_DMX_MAX_CHANNEL_GROUPS*PCM_DMX_MAX_CHANNELS_PER_GROUP*sizeof(UCHAR));
+  FDKmemset(hardToPlace, 255, PCM_DMX_MAX_CHANNELS*sizeof(UCHAR));
+  FDKmemset(h2pSortIdx, 255, PCM_DMX_MAX_CHANNELS*sizeof(UCHAR));
+  /* Get the restrictions of the internal representation */
+  FDKmemcpy(numChFree, maxChInGrp, PCM_DMX_MAX_CHANNEL_GROUPS*sizeof(UCHAR));
+
+  *chMode = CH_MODE_UNDEFINED;
 
   /* Categorize channels */
   for (ch = 0; ch < numChannels; ch += 1) {
-    int i = 0, j, chGrpIdx = channelIndices[ch];
+    UCHAR chGrpIdx = channelIndices[ch];
+    int i = 0, j;
 
     switch (channelType[ch]) {
-    case ACT_FRONT:
     case ACT_FRONT_TOP:
+      chGrpIdx += numChInGrp[CH_GROUP_FRONT];  /* Append after normal plain */
+    case ACT_FRONT:
       grpIdx = CH_GROUP_FRONT;
       break;
-    case ACT_SIDE:
+#if (PCM_DMX_MAX_CHANNEL_GROUPS > 1)
     case ACT_SIDE_TOP:
+      chGrpIdx += numChInGrp[CH_GROUP_SIDE];   /* Append after normal plain */
+    case ACT_SIDE:
       grpIdx = CH_GROUP_SIDE;
       break;
-    case ACT_BACK:
     case ACT_BACK_TOP:
+      chGrpIdx += numChInGrp[CH_GROUP_REAR];   /* Append after normal plain */
+    case ACT_BACK:
       grpIdx = CH_GROUP_REAR;
       break;
     case ACT_LFE:
       grpIdx = CH_GROUP_LFE;
       break;
+#endif
     default:
-      err = -1;
-      continue;
+      /* Found a channel that can not be categorized! Most likely due to corrupt input signalling.
+         The rescue strategy is to append it to the front channels (=> ignore index).
+         This could cause strange behaviour so return an error to signal it. */
+      err = PCMDMX_INVALID_MODE;
+      grpIdx = CH_GROUP_FRONT;
+      chGrpIdx = numChannels + numChToPlace;
+      numChToPlace += 1;
+      break;
     }
 
     if (numChInGrp[grpIdx] < PCM_DMX_MAX_CHANNELS_PER_GROUP) {
@@ -327,76 +519,152 @@ PCM_DMX_CHANNEL_MODE getChannelMode (
     }
   }
 
-  /* Compose channel offset table */
+#if (PCM_DMX_MAX_CHANNEL_GROUPS > 1)
+  FDK_ASSERT( (numChInGrp[CH_GROUP_FRONT]+numChInGrp[CH_GROUP_SIDE]
+              +numChInGrp[CH_GROUP_REAR]+numChInGrp[CH_GROUP_LFE]) == numChannels);
+#else
+  FDK_ASSERT( numChInGrp[CH_GROUP_FRONT] == numChannels );
+#endif
+
+  /* Compose channel offset table:
+   * Map all channels to the internal representation. */
+  numChToPlace = 0;
 
   /* Non-symmetric channels */
   if (numChInGrp[CH_GROUP_FRONT] & 0x1) {
     /* Odd number of front channels -> we have a center channel.
        In MPEG-4 the center has the index 0. */
     offsetTable[CENTER_FRONT_CHANNEL] = chIdx[CH_GROUP_FRONT][0];
+    numChFree[CH_GROUP_FRONT] -= 1;
   }
 
   for (grpIdx = 0; grpIdx < PCM_DMX_MAX_CHANNEL_GROUPS; grpIdx += 1) {
-    int chMapPos, maxChannels = 0;
-    ch = 0;
+    int chMapPos = 0;
+    ch = 0;  /* Index of channel within the specific group */
 
     switch (grpIdx) {
     case CH_GROUP_FRONT:
       chMapPos = LEFT_FRONT_CHANNEL;
-      maxChannels = 3;
       ch = numChInGrp[grpIdx] & 0x1;
       break;
+#if (PCM_DMX_MAX_CHANNEL_GROUPS > 1)
     case CH_GROUP_SIDE:
-      chMapPos = LEFT_OUTSIDE_CHANNEL;
-      maxChannels = 2;
       break;
     case CH_GROUP_REAR:
       chMapPos = LEFT_REAR_CHANNEL;
-      maxChannels = 2;
       break;
     case CH_GROUP_LFE:
       chMapPos = LOW_FREQUENCY_CHANNEL;
-      maxChannels = 1;
       break;
+#endif
     default:
-      err = -1;
+      FDK_ASSERT(0);
       continue;
     }
 
+    /* Map all channels of the group */
     for ( ; ch < numChInGrp[grpIdx]; ch += 1) {
-      if (ch < maxChannels) {
+      if (numChFree[grpIdx] > 0) {
         offsetTable[chMapPos] = chIdx[grpIdx][ch];
         chMapPos += 1;
+        numChFree[grpIdx] -= 1;
       } else {
-        err = -1;
+        /* Add to the list of hardship cases considering a MPEG-like sorting order: */
+        int pos, sortIdx = grpIdx*PCM_DMX_MAX_CHANNELS_PER_GROUP + channelIndices[chIdx[grpIdx][ch]];
+        for (pos = numChToPlace; pos > 0; pos -= 1) {
+          if (h2pSortIdx[pos-1] > sortIdx) {
+            hardToPlace[pos] = hardToPlace[pos-1];
+            h2pSortIdx[pos] = h2pSortIdx[pos-1];
+          } else {
+            /* Insert channel at the current index/position */
+            break;
+          }
+        }
+        hardToPlace[pos] = chIdx[grpIdx][ch];
+        h2pSortIdx[pos] = sortIdx;
+        numChToPlace += 1;
       }
     }
   }
 
-  if (err == 0) {
-    /* Compose the channel mode */
-    chMode = (numChInGrp[CH_GROUP_LFE]   & 0xF) << 12
-           | (numChInGrp[CH_GROUP_REAR]  & 0xF) <<  8
-           | (numChInGrp[CH_GROUP_SIDE]  & 0xF) <<  4
-           | (numChInGrp[CH_GROUP_FRONT] & 0xF);
+  { /* Assign the hardship cases */
+    int chMapPos = 0;
+    int mappingHeat = 0;
+    for (ch = 0; ch < numChToPlace; ch+=1) {
+      int chAssigned = 0;
+
+      /* Just assigning the channels to the next best slot can lead to undesired results (especially for x/x/1.x
+         configurations). Thus use the MPEG-like sorting index to find the best fitting slot for each channel.
+         If this is not possible the sorting index will be ignored (mappingHeat >= 2). */
+      for ( ; chMapPos < PCM_DMX_MAX_CHANNELS; chMapPos+=1) {
+        if (offsetTable[chMapPos] == 255) {
+          int prvSortIdx = 0;
+          int nxtSortIdx = (CH_GROUP_LFE+1)*PCM_DMX_MAX_CHANNELS_PER_GROUP;
+
+          if (mappingHeat < 2) {
+            if (chMapPos < LEFT_REAR_CHANNEL) {
+              /* Got front channel slot */
+              prvSortIdx = CH_GROUP_FRONT*PCM_DMX_MAX_CHANNELS_PER_GROUP + chMapPos - CENTER_FRONT_CHANNEL;
+              nxtSortIdx = CH_GROUP_SIDE *PCM_DMX_MAX_CHANNELS_PER_GROUP;
+            }
+            else if (chMapPos < LOW_FREQUENCY_CHANNEL) {
+              /* Got back channel slot */
+              prvSortIdx = CH_GROUP_REAR*PCM_DMX_MAX_CHANNELS_PER_GROUP + chMapPos - LEFT_REAR_CHANNEL;
+              nxtSortIdx = CH_GROUP_LFE *PCM_DMX_MAX_CHANNELS_PER_GROUP;
+            }
+            else if (chMapPos < LEFT_MULTIPRPS_CHANNEL) {
+              /* Got lfe channel slot */
+              prvSortIdx =  CH_GROUP_LFE   *PCM_DMX_MAX_CHANNELS_PER_GROUP + chMapPos - LOW_FREQUENCY_CHANNEL;
+              nxtSortIdx = (CH_GROUP_LFE+1)*PCM_DMX_MAX_CHANNELS_PER_GROUP;
+            }
+          }
+
+          /* Assign the channel only if its sort index is within the range */
+          if ( (h2pSortIdx[ch] >= prvSortIdx)
+            && (h2pSortIdx[ch] <  nxtSortIdx) ) {
+            offsetTable[chMapPos++] = hardToPlace[ch];
+            chAssigned = 1;
+            break;
+          }
+        }
+      }
+      if (chAssigned == 0) {
+        chMapPos = 0;
+        ch -= 1;
+        mappingHeat += 1;
+        continue;
+      }
+    }
   }
 
-  return (PCM_DMX_CHANNEL_MODE)chMode;
+  /* Compose the channel mode */
+  *chMode = (PCM_DMX_CHANNEL_MODE)( (numChInGrp[CH_GROUP_FRONT] & 0xF)
+#if (PCM_DMX_MAX_CHANNEL_GROUPS > 1)
+                                  | (numChInGrp[CH_GROUP_SIDE]  & 0xF) <<  4
+                                  | (numChInGrp[CH_GROUP_REAR]  & 0xF) <<  8
+                                  | (numChInGrp[CH_GROUP_LFE]   & 0xF) << 12
+#endif
+                                  );
+
+  return err;
 }
 
 
 /** Generate a channel offset table and complete channel description for a given (packed) channel mode.
- *  This function is the inverse to the getChannelMode() routine.
- * @param [in] The total number of channels of the given configuration.
+ *  This function is the inverse to the getChannelMode() routine but does not support weird channel
+ *  configurations. All channels have to be in the normal height layer and there must not be more
+ *  channels in each group than given by maxChInGrp.
+ * @param [in] The packed channel mode of the configuration to be processed.
  * @param [in] Array containing the channel mapping to be used (From MPEG PCE ordering to whatever is required).
  * @param [out] Array where corresponding channel types for each channels are stored into.
  * @param [out] Array where corresponding channel type indices for each output channel are stored into.
  * @param [out] Array where the buffer offsets for each channel are stored into.
  * @returns None.
  **/
+static
 void getChannelDescription (
         const PCM_DMX_CHANNEL_MODE  chMode,                                 /* in */
-        const UCHAR                 channelMapping[][PCM_DMX_MAX_CHANNELS], /* in */
+        const UCHAR                 channelMapping[][8],                    /* in */
         AUDIO_CHANNEL_TYPE          channelType[],                          /* out */
         UCHAR                       channelIndices[],                       /* out */
         UCHAR                       offsetTable[PCM_DMX_MAX_CHANNELS]       /* out */
@@ -412,15 +680,17 @@ void getChannelDescription (
   FDK_ASSERT(offsetTable != NULL);
 
   /* Init output arrays */
-  FDKmemclear(channelType,    PCM_DMX_MAX_CHANNELS*sizeof(AUDIO_CHANNEL_TYPE));
-  FDKmemclear(channelIndices, PCM_DMX_MAX_CHANNELS*sizeof(UCHAR));
+  FDKmemclear(channelType,    PCM_DMX_MAX_IO_CHANNELS*sizeof(AUDIO_CHANNEL_TYPE));
+  FDKmemclear(channelIndices, PCM_DMX_MAX_IO_CHANNELS*sizeof(UCHAR));
   FDKmemset(offsetTable, 255, PCM_DMX_MAX_CHANNELS*sizeof(UCHAR));
 
   /* Extract the number of channels per group */
   numChInGrp[CH_GROUP_FRONT] =  chMode        & 0xF;
+#if (PCM_DMX_MAX_CHANNEL_GROUPS > 1)
   numChInGrp[CH_GROUP_SIDE]  = (chMode >>  4) & 0xF;
   numChInGrp[CH_GROUP_REAR]  = (chMode >>  8) & 0xF;
   numChInGrp[CH_GROUP_LFE]   = (chMode >> 12) & 0xF;
+#endif
 
   /* Summerize to get the total number of channels */
   for (grpIdx = 0; grpIdx < PCM_DMX_MAX_CHANNEL_GROUPS; grpIdx += 1) {
@@ -428,7 +698,29 @@ void getChannelDescription (
   }
 
   /* Get the appropriate channel map */
-  pChannelMap = channelMapping[numChannels-1];
+  switch (chMode) {
+  case CH_MODE_1_0_0_0:
+  case CH_MODE_2_0_0_0:
+  case CH_MODE_3_0_0_0:
+  case CH_MODE_3_0_1_0:
+  case CH_MODE_3_0_2_0:
+  case CH_MODE_3_0_2_1:
+    pChannelMap = channelMapping[numChannels];
+    break;
+  case CH_MODE_3_0_3_1:
+    pChannelMap = channelMapping[11];
+    break;
+  case CH_MODE_3_0_4_1:
+    pChannelMap = channelMapping[12];
+    break;
+  case CH_MODE_5_0_2_1:
+    pChannelMap = channelMapping[7];
+    break;
+  default:
+    /* fallback */
+    pChannelMap = channelMapping[0];
+    break;
+  }
 
   /* Compose channel offset table */
 
@@ -436,15 +728,17 @@ void getChannelDescription (
   if (numChInGrp[CH_GROUP_FRONT] & 0x1) {
     /* Odd number of front channels -> we have a center channel.
        In MPEG-4 the center has the index 0. */
-    offsetTable[CENTER_FRONT_CHANNEL] = pChannelMap[0];
-    channelType[0] = ACT_FRONT;
+    int mappedIdx = pChannelMap[ch];
+    offsetTable[CENTER_FRONT_CHANNEL] = mappedIdx;
+    channelType[mappedIdx]    = ACT_FRONT;
+    channelIndices[mappedIdx] = 0;
     ch += 1;
   }
 
   for (grpIdx = 0; grpIdx < PCM_DMX_MAX_CHANNEL_GROUPS; grpIdx += 1) {
-    AUDIO_CHANNEL_TYPE type;
-    int chMapPos, maxChannels = 0;
-    int chIdx = 0;
+    AUDIO_CHANNEL_TYPE type = ACT_NONE;
+    int chMapPos = 0, maxChannels = 0;
+    int chIdx = 0;  /* Index of channel within the specific group */
 
     switch (grpIdx) {
     case CH_GROUP_FRONT:
@@ -453,10 +747,11 @@ void getChannelDescription (
       maxChannels = 3;
       chIdx = numChInGrp[grpIdx] & 0x1;
       break;
+#if (PCM_DMX_MAX_CHANNEL_GROUPS > 1)
     case CH_GROUP_SIDE:
+      /* Always map side channels to the multipurpose group. */
       type = ACT_SIDE;
-      chMapPos = LEFT_OUTSIDE_CHANNEL;
-      maxChannels = 2;
+      chMapPos = LEFT_MULTIPRPS_CHANNEL;
       break;
     case CH_GROUP_REAR:
       type = ACT_BACK;
@@ -468,20 +763,646 @@ void getChannelDescription (
       chMapPos = LOW_FREQUENCY_CHANNEL;
       maxChannels = 1;
       break;
+#endif
     default:
       break;
     }
 
-    for ( ; (chIdx < numChInGrp[grpIdx]) && (chIdx < maxChannels); chIdx += 1) {
-      offsetTable[chMapPos] = pChannelMap[ch];
-      channelType[ch]    = type;
-      channelIndices[ch] = chIdx;
+    /* Map all channels in this group */
+    for ( ; chIdx < numChInGrp[grpIdx]; chIdx += 1) {
+      int mappedIdx = pChannelMap[ch];
+      if (chIdx == maxChannels) {
+        /* No space left in this channel group!
+           Use the multipurpose group instead: */
+        chMapPos = LEFT_MULTIPRPS_CHANNEL;
+      }
+      offsetTable[chMapPos]     = mappedIdx;
+      channelType[mappedIdx]    = type;
+      channelIndices[mappedIdx] = chIdx;
       chMapPos += 1;
       ch += 1;
     }
   }
 }
 
+/** Private helper function for downmix matrix manipulation that initializes
+ *  one row in a given downmix matrix (corresponding to one output channel).
+ * @param [inout] Pointer to fixed-point parts of the downmix matrix.
+ * @param [inout] Pointer to scale factor matrix associated to the downmix factors.
+ * @param [in]    Index of channel (row) to be initialized.
+ * @returns       Nothing to return.
+ **/
+static
+void dmxInitChannel(
+        FIXP_DMX            mixFactors[PCM_DMX_MAX_CHANNELS][PCM_DMX_MAX_CHANNELS],
+        INT                 mixScales[PCM_DMX_MAX_CHANNELS][PCM_DMX_MAX_CHANNELS],
+        const unsigned int  outCh
+       )
+{
+  unsigned int inCh;
+  for (inCh=0; inCh < PCM_DMX_MAX_CHANNELS; inCh+=1) {
+    if (inCh == outCh) {
+      mixFactors[outCh][inCh] = FL2FXCONST_DMX(0.5f);
+      mixScales[outCh][inCh]  = 1;
+    } else {
+      mixFactors[outCh][inCh] = FL2FXCONST_DMX(0.0f);
+      mixScales[outCh][inCh]  = 0;
+    }
+  }
+}
+
+/** Private helper function for downmix matrix manipulation that does a reset
+ *  of one row in a given downmix matrix (corresponding to one output channel).
+ * @param [inout] Pointer to fixed-point parts of the downmix matrix.
+ * @param [inout] Pointer to scale factor matrix associated to the downmix factors.
+ * @param [in]    Index of channel (row) to be cleared/reset.
+ * @returns       Nothing to return.
+ **/
+static
+void dmxClearChannel(
+        FIXP_DMX            mixFactors[PCM_DMX_MAX_CHANNELS][PCM_DMX_MAX_CHANNELS],
+        INT                 mixScales[PCM_DMX_MAX_CHANNELS][PCM_DMX_MAX_CHANNELS],
+        const unsigned int  outCh
+       )
+{
+  FDKmemclear(&mixFactors[outCh], PCM_DMX_MAX_CHANNELS*sizeof(FIXP_DMX));
+  FDKmemclear(&mixScales[outCh],  PCM_DMX_MAX_CHANNELS*sizeof(INT));
+}
+
+/** Private helper function for downmix matrix manipulation that applies a source channel (row)
+ *  scaled by a given mix factor to a destination channel (row) in a given downmix matrix.
+ *  Existing mix factors of the destination channel (row) will get overwritten.
+ * @param [inout] Pointer to fixed-point parts of the downmix matrix.
+ * @param [inout] Pointer to scale factor matrix associated to the downmix factors.
+ * @param [in]    Index of source channel (row).
+ * @param [in]    Index of destination channel (row).
+ * @param [in]    Fixed-point part of mix factor to be applied.
+ * @param [in]    Scale factor of mix factor to be applied.
+ * @returns       Nothing to return.
+ **/
+static
+void dmxSetChannel(
+        FIXP_DMX            mixFactors[PCM_DMX_MAX_CHANNELS][PCM_DMX_MAX_CHANNELS],
+        INT                 mixScales[PCM_DMX_MAX_CHANNELS][PCM_DMX_MAX_CHANNELS],
+        const unsigned int  dstCh,
+        const unsigned int  srcCh,
+        const FIXP_DMX      factor,
+        const INT           scale
+       )
+{
+  int ch;
+  for (ch=0; ch < PCM_DMX_MAX_CHANNELS; ch+=1) {
+    if (mixFactors[srcCh][ch] != (FIXP_DMX)0) {
+      mixFactors[dstCh][ch] = FX_DBL2FX_DMX(fMult(mixFactors[srcCh][ch], factor));
+      mixScales[dstCh][ch]  = mixScales[srcCh][ch] + scale;
+    }
+  }
+}
+
+/** Private helper function for downmix matrix manipulation that adds a source channel (row)
+ *  scaled by a given mix factor to a destination channel (row) in a given downmix matrix.
+ * @param [inout] Pointer to fixed-point parts of the downmix matrix.
+ * @param [inout] Pointer to scale factor matrix associated to the downmix factors.
+ * @param [in]    Index of source channel (row).
+ * @param [in]    Index of destination channel (row).
+ * @param [in]    Fixed-point part of mix factor to be applied.
+ * @param [in]    Scale factor of mix factor to be applied.
+ * @returns       Nothing to return.
+ **/
+static
+void dmxAddChannel(
+        FIXP_DMX            mixFactors[PCM_DMX_MAX_CHANNELS][PCM_DMX_MAX_CHANNELS],
+        INT                 mixScales[PCM_DMX_MAX_CHANNELS][PCM_DMX_MAX_CHANNELS],
+        const unsigned int  dstCh,
+        const unsigned int  srcCh,
+        const FIXP_DMX      factor,
+        const INT           scale
+       )
+{
+  int ch;
+  for (ch=0; ch < PCM_DMX_MAX_CHANNELS; ch+=1) {
+    FIXP_DBL addFact = fMult(mixFactors[srcCh][ch], factor);
+    if (addFact != (FIXP_DMX)0) {
+      INT newScale = mixScales[srcCh][ch] + scale;
+      if (mixFactors[dstCh][ch] != (FIXP_DMX)0) {
+        if (newScale > mixScales[dstCh][ch]) {
+          mixFactors[dstCh][ch] >>= newScale - mixScales[dstCh][ch];
+        } else {
+          addFact >>= mixScales[dstCh][ch] - newScale;
+          newScale  = mixScales[dstCh][ch];
+        }
+      }
+      mixFactors[dstCh][ch] += FX_DBL2FX_DMX(addFact);
+      mixScales[dstCh][ch]   = newScale;
+    }
+  }
+}
+
+
+/** Private function that creates a downmix factor matrix depending on the input and output
+ *  configuration, the user parameters as well as the given metadata. This function is the modules
+ *  brain and hold all downmix algorithms.
+ * @param [in]  Flag that indicates if inChMode holds a real (packed) channel mode or has been
+                converted to a MPEG-4 channel configuration index.
+ * @param [in]  Dependent on the inModeIsCfg flag this field hands in a (packed) channel mode or
+                the corresponding MPEG-4 channel configuration index.of the input configuration.
+ * @param [in]  The (packed) channel mode of the output configuration.
+ * @param [in]  Pointer to structure holding all current user parameter.
+ * @param [in]  Pointer to field holding all current meta data.
+ * @param [out] Pointer to fixed-point parts of the downmix matrix. Normalized to one scale factor.
+ * @param [out] The common scale factor of the downmix matrix.
+ * @returns     An error code.
+ **/
+static
+PCMDMX_ERROR getMixFactors (
+        const UCHAR                 inModeIsCfg,
+        PCM_DMX_CHANNEL_MODE        inChMode,
+        const PCM_DMX_CHANNEL_MODE  outChMode,
+        const PCM_DMX_USER_PARAMS  *pParams,
+        const DMX_BS_META_DATA     *pMetaData,
+        FIXP_DMX                    mixFactors[PCM_DMX_MAX_CHANNELS][PCM_DMX_MAX_CHANNELS],
+        INT                        *pOutScale
+      )
+{
+  PCMDMX_ERROR err = PCMDMX_OK;
+  INT  mixScales[PCM_DMX_MAX_CHANNELS][PCM_DMX_MAX_CHANNELS];
+  INT  maxScale = 0;
+  int  numInChannel, numOutChannel;
+  unsigned int  outCh, inCh, inChCfg = 0;
+  unsigned int  valid[PCM_DMX_MAX_CHANNELS] = { 0 };
+
+  FDK_ASSERT(pMetaData  != NULL);
+  FDK_ASSERT(mixFactors != NULL);
+  /* Check on a supported output configuration */
+  FDK_ASSERT( (outChMode == CH_MODE_1_0_0_0)
+           || (outChMode == CH_MODE_2_0_0_0)
+           || (outChMode == CH_MODE_3_0_2_1) );
+
+  if (inModeIsCfg) {
+    /* Workaround for the ambiguity of the internal channel modes.
+       Convert channel config to channel mode: */
+    inChCfg  = (unsigned int)inChMode;
+    switch (inChCfg) {
+    case 1: case 2: case 3:
+#if (PCM_DMX_MAX_CHANNELS > 3)
+    case 4: case 5: case 6:
+#endif
+      inChMode = outChModeTable[inChCfg];
+      break;
+    case 11:
+      inChMode = CH_MODE_3_0_3_1;
+      break;
+    case 12:
+      inChMode = CH_MODE_3_0_4_1;
+      break;
+    case 7: case 14:
+      inChMode = CH_MODE_5_0_2_1;
+      break;
+    default:
+      FDK_ASSERT(0);
+    }
+  }
+
+  /* Extract the total number of input channels */
+  numInChannel  =  (inChMode&0xF)
+                + ((inChMode>> 4)&0xF)
+                + ((inChMode>> 8)&0xF)
+                + ((inChMode>>12)&0xF);
+  /* Extract the total number of output channels */
+  numOutChannel =  (outChMode&0xF)
+                + ((outChMode>> 4)&0xF)
+                + ((outChMode>> 8)&0xF)
+                + ((outChMode>>12)&0xF);
+
+  /* MPEG ammendment 4 aka ETSI metadata and fallback mode: */
+
+
+  /* Create identity DMX matrix: */
+  for (outCh=0; outCh < PCM_DMX_MAX_CHANNELS; outCh+=1) {
+    dmxInitChannel( mixFactors, mixScales, outCh );
+  }
+  if (((inChMode>>12)&0xF) == 0) {
+    /* Clear empty or wrongly mapped input channel */
+    dmxClearChannel( mixFactors, mixScales, LOW_FREQUENCY_CHANNEL );
+  }
+
+  /* FIRST STAGE: */
+  if (numInChannel > SIX_CHANNEL)
+  { /* Always use MPEG equations either with meta data or with default values. */
+    FIXP_DMX  dMixFactA, dMixFactB;
+    INT       dMixScaleA, dMixScaleB;
+    int       isValidCfg = TRUE;
+
+    /* Get factors from meta data */
+    dMixFactA  = abMixLvlValueTab[pMetaData->dmixIdxA];
+    dMixScaleA = (pMetaData->dmixIdxA==0) ? 1 : 0;
+    dMixFactB  = abMixLvlValueTab[pMetaData->dmixIdxB];
+    dMixScaleB = (pMetaData->dmixIdxB==0) ? 1 : 0;
+
+    /* Check if input is in the list of supported configurations */
+    switch (inChMode) {
+    case CH_MODE_3_0_3_1:   /* chCfg 11 */
+      /* 6.1ch:  C' = C;  L' = L;  R' = R;  LFE' = LFE;
+                 Ls' = Ls*dmix_a_idx + Cs*dmix_b_idx;
+                 Rs' = Rs*dmix_a_idx + Cs*dmix_b_idx; */
+      dmxClearChannel( mixFactors, mixScales, RIGHT_MULTIPRPS_CHANNEL );  /* clear empty input channel */
+      dmxSetChannel( mixFactors, mixScales, LEFT_REAR_CHANNEL,  LEFT_REAR_CHANNEL,      dMixFactA, dMixScaleA );
+      dmxSetChannel( mixFactors, mixScales, LEFT_REAR_CHANNEL,  LEFT_MULTIPRPS_CHANNEL, dMixFactB, dMixScaleB );
+      dmxSetChannel( mixFactors, mixScales, RIGHT_REAR_CHANNEL, RIGHT_REAR_CHANNEL,     dMixFactA, dMixScaleA );
+      dmxSetChannel( mixFactors, mixScales, RIGHT_REAR_CHANNEL, LEFT_MULTIPRPS_CHANNEL, dMixFactB, dMixScaleB );
+      break;
+    case CH_MODE_3_2_1_0:
+    case CH_MODE_3_2_1_1:   /* chCfg 11 but with side channels */
+      /* 6.1ch:  C' = C;  L' = L;  R' = R;  LFE' = LFE;
+                 Ls' = Ls*dmix_a_idx + Cs*dmix_b_idx;
+                 Rs' = Rs*dmix_a_idx + Cs*dmix_b_idx; */
+      dmxClearChannel( mixFactors, mixScales, RIGHT_REAR_CHANNEL );  /* clear empty input channel */
+      dmxSetChannel( mixFactors, mixScales, RIGHT_REAR_CHANNEL, LEFT_REAR_CHANNEL,       dMixFactB, dMixScaleB );
+      dmxSetChannel( mixFactors, mixScales, RIGHT_REAR_CHANNEL, RIGHT_MULTIPRPS_CHANNEL, dMixFactA, dMixScaleA );
+      dmxSetChannel( mixFactors, mixScales, LEFT_REAR_CHANNEL,  LEFT_REAR_CHANNEL,       dMixFactB, dMixScaleB );
+      dmxSetChannel( mixFactors, mixScales, LEFT_REAR_CHANNEL,  LEFT_MULTIPRPS_CHANNEL,  dMixFactA, dMixScaleA );
+      isValidCfg = FALSE;
+      err = PCMDMX_INVALID_MODE;
+      break;
+    case CH_MODE_5_2_1_0:
+    case CH_MODE_5_0_1_0:
+    case CH_MODE_5_0_1_1:
+      /*         Ls' = Cs*dmix_a_idx;
+                 Rs' = Cs*dmix_a_idx; */
+      dmxClearChannel( mixFactors, mixScales, RIGHT_REAR_CHANNEL );  /* clear empty input channel */
+      dmxSetChannel( mixFactors, mixScales, RIGHT_REAR_CHANNEL, LEFT_REAR_CHANNEL, dMixFactA, dMixScaleA );
+      dmxSetChannel( mixFactors, mixScales, LEFT_REAR_CHANNEL,  LEFT_REAR_CHANNEL, dMixFactA, dMixScaleA );
+      isValidCfg = FALSE;
+      err = PCMDMX_INVALID_MODE;
+      break;
+    case CH_MODE_3_0_4_1:   /* chCfg 12 */
+      /* 7.1ch Surround Back:  C' = C;  L' = L;  R' = R;  LFE' = LFE;
+                               Ls' = Ls*dmix_a_idx + Lsr*dmix_b_idx;
+                               Rs' = Rs*dmix_a_idx + Rsr*dmix_b_idx; */
+      dmxSetChannel( mixFactors, mixScales, LEFT_REAR_CHANNEL,  LEFT_REAR_CHANNEL,       dMixFactA, dMixScaleA );
+      dmxSetChannel( mixFactors, mixScales, LEFT_REAR_CHANNEL,  LEFT_MULTIPRPS_CHANNEL,  dMixFactB, dMixScaleB );
+      dmxSetChannel( mixFactors, mixScales, RIGHT_REAR_CHANNEL, RIGHT_REAR_CHANNEL,      dMixFactA, dMixScaleA );
+      dmxSetChannel( mixFactors, mixScales, RIGHT_REAR_CHANNEL, RIGHT_MULTIPRPS_CHANNEL, dMixFactB, dMixScaleB );
+      break;
+    case CH_MODE_5_0_2_1:   /* chCfg 7 || 14 */
+      if (inChCfg == 14) {
+        /* 7.1ch Front Height:  C' = C;  Ls' = Ls;  Rs' = Rs;  LFE' = LFE;
+                                L' = L*dmix_a_idx + Lv*dmix_b_idx;
+                                R' = R*dmix_a_idx + Rv*dmix_b_idx; */
+        dmxSetChannel( mixFactors, mixScales, LEFT_FRONT_CHANNEL,  LEFT_FRONT_CHANNEL,      dMixFactA, dMixScaleA );
+        dmxSetChannel( mixFactors, mixScales, LEFT_FRONT_CHANNEL,  LEFT_MULTIPRPS_CHANNEL,  dMixFactB, dMixScaleB );
+        dmxSetChannel( mixFactors, mixScales, RIGHT_FRONT_CHANNEL, RIGHT_FRONT_CHANNEL,     dMixFactA, dMixScaleA );
+        dmxSetChannel( mixFactors, mixScales, RIGHT_FRONT_CHANNEL, RIGHT_MULTIPRPS_CHANNEL, dMixFactB, dMixScaleB );
+      } else {
+        /* 7.1ch Front:  Ls' = Ls;  Rs' = Rs;  LFE' = LFE;
+                         C' = C + (Lc+Rc)*dmix_a_idx;
+                         L' = L + Lc*dmix_b_idx;
+                         R' = R + Rc*dmix_b_idx;
+                         CAUTION: L+R are not at (MPEG) index 1+2. */
+        dmxSetChannel( mixFactors, mixScales, CENTER_FRONT_CHANNEL, LEFT_FRONT_CHANNEL,      dMixFactA, dMixScaleA );
+        dmxSetChannel( mixFactors, mixScales, CENTER_FRONT_CHANNEL, RIGHT_FRONT_CHANNEL,     dMixFactA, dMixScaleA );
+        dmxSetChannel( mixFactors, mixScales, LEFT_FRONT_CHANNEL,   LEFT_FRONT_CHANNEL,      dMixFactB, dMixScaleB );
+        dmxSetChannel( mixFactors, mixScales, LEFT_FRONT_CHANNEL,   LEFT_MULTIPRPS_CHANNEL,  FL2FXCONST_DMX(0.5f), 1 );
+        dmxSetChannel( mixFactors, mixScales, RIGHT_FRONT_CHANNEL,  RIGHT_FRONT_CHANNEL,     dMixFactB, dMixScaleB );
+        dmxSetChannel( mixFactors, mixScales, RIGHT_FRONT_CHANNEL,  RIGHT_MULTIPRPS_CHANNEL, FL2FXCONST_DMX(0.5f), 1 );
+      }
+      break;
+    default:
+      /* Nothing to do. Just use the identity matrix. */
+      isValidCfg = FALSE;
+      err = PCMDMX_INVALID_MODE;
+      break;
+    }
+
+    /* Add additional DMX gain */
+    if ( (isValidCfg == TRUE)
+      && (pMetaData->dmxGainIdx5 != 0))
+    { /* Apply DMX gain 5 */
+      FIXP_DMX dmxGain;
+      INT      dmxScale;
+      INT      sign = (pMetaData->dmxGainIdx5 & 0x40) ? -1 : 1;
+      INT      val  = pMetaData->dmxGainIdx5 & 0x3F;
+
+      /* 10^(dmx_gain_5/80) */
+      dmxGain = FX_DBL2FX_DMX( fLdPow(
+                                    FL2FXCONST_DBL(0.830482023721841f), 2,  /* log2(10) */
+                                    (FIXP_DBL)(sign*val*(LONG)FL2FXCONST_DBL(0.0125f)), 0,
+                                   &dmxScale )
+                               );
+      /* Currently only positive scale factors supported! */
+      if (dmxScale < 0) {
+        dmxGain >>= -dmxScale;
+        dmxScale  =  0;
+      }
+
+      dmxSetChannel( mixFactors, mixScales, CENTER_FRONT_CHANNEL,  CENTER_FRONT_CHANNEL,  dmxGain, dmxScale );
+      dmxSetChannel( mixFactors, mixScales, LEFT_FRONT_CHANNEL,    LEFT_FRONT_CHANNEL,    dmxGain, dmxScale );
+      dmxSetChannel( mixFactors, mixScales, RIGHT_FRONT_CHANNEL,   RIGHT_FRONT_CHANNEL,   dmxGain, dmxScale );
+      dmxSetChannel( mixFactors, mixScales, LEFT_REAR_CHANNEL,     LEFT_REAR_CHANNEL,     dmxGain, dmxScale );
+      dmxSetChannel( mixFactors, mixScales, RIGHT_REAR_CHANNEL,    RIGHT_REAR_CHANNEL,    dmxGain, dmxScale );
+      dmxSetChannel( mixFactors, mixScales, LOW_FREQUENCY_CHANNEL, LOW_FREQUENCY_CHANNEL, dmxGain, dmxScale );
+    }
+
+    /* Mark the output channels */
+    valid[CENTER_FRONT_CHANNEL]  = 1;
+    valid[LEFT_FRONT_CHANNEL]    = 1;
+    valid[RIGHT_FRONT_CHANNEL]   = 1;
+    valid[LEFT_REAR_CHANNEL]     = 1;
+    valid[RIGHT_REAR_CHANNEL]    = 1;
+    valid[LOW_FREQUENCY_CHANNEL] = 1;
+
+    /* Update channel mode for the next stage */
+    inChMode = CH_MODE_3_0_2_1;
+  }
+
+  /* SECOND STAGE: */
+  if (numOutChannel <= TWO_CHANNEL) {
+    /* Create DMX matrix according to input configuration */
+    switch (inChMode) {
+    case CH_MODE_2_0_0_0:   /* chCfg 2 */
+      /* Apply the dual channel mode. */
+      switch (pParams->dualChannelMode) {
+      case CH1_MODE:  /* L' = 0.707 * Ch1;
+                         R' = 0.707 * Ch1; */
+        dmxSetChannel( mixFactors, mixScales, LEFT_FRONT_CHANNEL,  LEFT_FRONT_CHANNEL,  FL2FXCONST_DMX(0.707f), 0 );
+        dmxSetChannel( mixFactors, mixScales, RIGHT_FRONT_CHANNEL, LEFT_FRONT_CHANNEL,  FL2FXCONST_DMX(0.707f), 0 );
+        break;
+      case CH2_MODE:  /* L' = 0.707 * Ch2;
+                         R' = 0.707 * Ch2; */
+        dmxSetChannel( mixFactors, mixScales, LEFT_FRONT_CHANNEL,  RIGHT_FRONT_CHANNEL, FL2FXCONST_DMX(0.707f), 0 );
+        dmxSetChannel( mixFactors, mixScales, RIGHT_FRONT_CHANNEL, RIGHT_FRONT_CHANNEL, FL2FXCONST_DMX(0.707f), 0 );
+        break;
+      case MIXED_MODE:  /* L' = 0.5*Ch1 + 0.5*Ch2;
+                           R' = 0.5*Ch1 + 0.5*Ch2; */
+        dmxSetChannel( mixFactors, mixScales, LEFT_FRONT_CHANNEL,  LEFT_FRONT_CHANNEL,  FL2FXCONST_DMX(0.5f), 0 );
+        dmxAddChannel( mixFactors, mixScales, LEFT_FRONT_CHANNEL,  RIGHT_FRONT_CHANNEL, FL2FXCONST_DMX(0.5f), 0 );
+        dmxSetChannel( mixFactors, mixScales, RIGHT_FRONT_CHANNEL, LEFT_FRONT_CHANNEL,  FL2FXCONST_DMX(0.5f), 0 );
+        dmxAddChannel( mixFactors, mixScales, RIGHT_FRONT_CHANNEL, RIGHT_FRONT_CHANNEL, FL2FXCONST_DMX(0.5f), 0 );
+        break;
+      default:
+      case STEREO_MODE:
+        /* Nothing to do */
+        break;
+      }
+      break;
+    case CH_MODE_2_0_1_0:
+      /* L' = L + 0.707*S;
+         R' = R + 0.707*S; */
+      dmxAddChannel( mixFactors, mixScales, LEFT_FRONT_CHANNEL,  LEFT_REAR_CHANNEL,   FL2FXCONST_DMX(0.707f), 0 );
+      dmxAddChannel( mixFactors, mixScales, RIGHT_FRONT_CHANNEL, LEFT_REAR_CHANNEL,   FL2FXCONST_DMX(0.707f), 0 );
+      break;
+    case CH_MODE_3_0_0_0:   /* chCfg 3 */
+      /* L' = L + 0.707*C;
+         R' = R + 0.707*C; */
+      dmxAddChannel( mixFactors, mixScales, LEFT_FRONT_CHANNEL,  CENTER_FRONT_CHANNEL, FL2FXCONST_DMX(0.707f), 0 );
+      dmxAddChannel( mixFactors, mixScales, RIGHT_FRONT_CHANNEL, CENTER_FRONT_CHANNEL, FL2FXCONST_DMX(0.707f), 0 );
+      break;
+    case CH_MODE_3_0_1_0:   /* chCfg 4 */
+      /* L' = L + 0.707*C + 0.707*S;
+         R' = R + 0.707*C + 0.707*S; */
+      dmxAddChannel( mixFactors, mixScales, LEFT_FRONT_CHANNEL,  CENTER_FRONT_CHANNEL, FL2FXCONST_DMX(0.707f), 0 );
+      dmxAddChannel( mixFactors, mixScales, LEFT_FRONT_CHANNEL,  LEFT_REAR_CHANNEL,    FL2FXCONST_DMX(0.707f), 0 );
+      dmxAddChannel( mixFactors, mixScales, RIGHT_FRONT_CHANNEL, CENTER_FRONT_CHANNEL, FL2FXCONST_DMX(0.707f), 0 );
+      dmxAddChannel( mixFactors, mixScales, RIGHT_FRONT_CHANNEL, LEFT_REAR_CHANNEL,    FL2FXCONST_DMX(0.707f), 0 );
+      break;
+    case CH_MODE_3_0_2_0:   /* chCfg 5 */
+    case CH_MODE_3_0_2_1:   /* chCfg 6 */
+      /* MPEG + ITU + DLB
+         But because the default downmix equations and coefficients are equal we stick to MPEG. */
+      if (  (pMetaData->typeFlags & TYPE_DSE_DATA)
+        || !(pMetaData->typeFlags & TYPE_PCE_DATA) )
+      {
+        FIXP_DMX  cMixLvl, sMixLvl, lMixLvl;
+        INT       cMixScale, sMixScale, lMixScale;
+
+        /* Get factors from meta data */
+        cMixLvl   = abMixLvlValueTab[pMetaData->cLevIdx];
+        cMixScale = (pMetaData->cLevIdx==0) ? 1 : 0;
+        sMixLvl   = abMixLvlValueTab[pMetaData->sLevIdx];
+        sMixScale = (pMetaData->sLevIdx==0) ? 1 : 0;
+        lMixLvl   = lfeMixLvlValueTab[pMetaData->dmixIdxLfe];
+        if (pMetaData->dmixIdxLfe <= 1) {
+          lMixScale = 2;
+        } else if (pMetaData->dmixIdxLfe <= 5) {
+          lMixScale = 1;
+        } else {
+          lMixScale = 0;
+        }
+        /* Setup the DMX matrix */
+        if ( (pParams->pseudoSurrMode == FORCE_PS_DMX)
+          || ((pParams->pseudoSurrMode == AUTO_PS_DMX) && (pMetaData->pseudoSurround==1)))
+        { /* L' = L + C*clev - (Ls+Rs)*slev + LFE*lflev;
+             R' = R + C*clev + (Ls+Rs)*slev + LFE*lflev; */
+          dmxAddChannel( mixFactors, mixScales, LEFT_FRONT_CHANNEL,  CENTER_FRONT_CHANNEL,  cMixLvl, cMixScale );
+          dmxAddChannel( mixFactors, mixScales, LEFT_FRONT_CHANNEL,  LEFT_REAR_CHANNEL,    -sMixLvl, sMixScale );
+          dmxAddChannel( mixFactors, mixScales, LEFT_FRONT_CHANNEL,  RIGHT_REAR_CHANNEL,   -sMixLvl, sMixScale );
+          dmxAddChannel( mixFactors, mixScales, LEFT_FRONT_CHANNEL,  LOW_FREQUENCY_CHANNEL, lMixLvl, lMixScale );
+          dmxAddChannel( mixFactors, mixScales, RIGHT_FRONT_CHANNEL, CENTER_FRONT_CHANNEL,  cMixLvl, cMixScale );
+          dmxAddChannel( mixFactors, mixScales, RIGHT_FRONT_CHANNEL, LEFT_REAR_CHANNEL,     sMixLvl, sMixScale );
+          dmxAddChannel( mixFactors, mixScales, RIGHT_FRONT_CHANNEL, RIGHT_REAR_CHANNEL,    sMixLvl, sMixScale );
+          dmxAddChannel( mixFactors, mixScales, RIGHT_FRONT_CHANNEL, LOW_FREQUENCY_CHANNEL, lMixLvl, lMixScale );
+        }
+        else
+        { /* L' = L + C*clev + Ls*slev + LFE*llev;
+             R' = R + C*clev + Rs*slev + LFE*llev; */
+          dmxAddChannel( mixFactors, mixScales, LEFT_FRONT_CHANNEL,  CENTER_FRONT_CHANNEL,  cMixLvl, cMixScale );
+          dmxAddChannel( mixFactors, mixScales, LEFT_FRONT_CHANNEL,  LEFT_REAR_CHANNEL,     sMixLvl, sMixScale );
+          dmxAddChannel( mixFactors, mixScales, LEFT_FRONT_CHANNEL,  LOW_FREQUENCY_CHANNEL, lMixLvl, lMixScale );
+          dmxAddChannel( mixFactors, mixScales, RIGHT_FRONT_CHANNEL, CENTER_FRONT_CHANNEL,  cMixLvl, cMixScale );
+          dmxAddChannel( mixFactors, mixScales, RIGHT_FRONT_CHANNEL, RIGHT_REAR_CHANNEL,    sMixLvl, sMixScale );
+          dmxAddChannel( mixFactors, mixScales, RIGHT_FRONT_CHANNEL, LOW_FREQUENCY_CHANNEL, lMixLvl, lMixScale );
+        }
+
+        /* Add additional DMX gain */
+        if ( pMetaData->dmxGainIdx2 != 0 )
+        { /* Apply DMX gain 2 */
+          FIXP_DMX dmxGain;
+          INT      dmxScale;
+          INT      sign = (pMetaData->dmxGainIdx2 & 0x40) ? -1 : 1;
+          INT      val  = pMetaData->dmxGainIdx2 & 0x3F;
+
+          /* 10^(dmx_gain_2/80) */
+          dmxGain = FX_DBL2FX_DMX( fLdPow(
+                                        FL2FXCONST_DBL(0.830482023721841f), 2,  /* log2(10) */
+                                        (FIXP_DBL)(sign*val*(LONG)FL2FXCONST_DBL(0.0125f)), 0,
+                                       &dmxScale )
+                                   );
+          /* Currently only positive scale factors supported! */
+          if (dmxScale < 0) {
+            dmxGain >>= -dmxScale;
+            dmxScale  =  0;
+          }
+
+          dmxSetChannel( mixFactors, mixScales, LEFT_FRONT_CHANNEL,  LEFT_FRONT_CHANNEL,  dmxGain, dmxScale );
+          dmxSetChannel( mixFactors, mixScales, RIGHT_FRONT_CHANNEL, RIGHT_FRONT_CHANNEL, dmxGain, dmxScale );
+        }
+      }
+#ifdef PCE_METADATA_ENABLE
+      else {
+        FIXP_DMX  flev, clev, slevLL, slevLR, slevRL, slevRR;
+        FIXP_DMX  mtrxMixDwnCoef = mpegMixDownIdx2Coef[pMetaData->matrixMixdownIdx];
+
+        if ( (pParams->pseudoSurrMode == FORCE_PS_DMX)
+          || ((pParams->pseudoSurrMode == AUTO_PS_DMX) && (pMetaData->pseudoSurround==1)))
+        { /* 3/2 input: L' = (1.707+2*A)^-1 * [L+0.707*C-A*Ls-A*Rs];
+                        R' = (1.707+2*A)^-1 * [R+0.707*C+A*Ls+A*Rs]; */
+          flev = mpegMixDownIdx2PreFact[1][pMetaData->matrixMixdownIdx];
+          slevRR = slevRL = FX_DBL2FX_DMX(fMult(flev, mtrxMixDwnCoef));
+          slevLL = slevLR = -slevRL;
+        }
+        else {
+          /* 3/2 input: L' = (1.707+A)^-1 * [L+0.707*C+A*Ls];
+                        R' = (1.707+A)^-1 * [R+0.707*C+A*Rs]; */
+          flev = mpegMixDownIdx2PreFact[0][pMetaData->matrixMixdownIdx];
+          slevRR = slevLL = FX_DBL2FX_DMX(fMult(flev, mtrxMixDwnCoef));
+          slevLR = slevRL = (FIXP_SGL)0;
+        }
+        /* common factor */
+        clev  = FX_DBL2FX_DMX(fMult(flev, mpegMixDownIdx2Coef[0] /* 0.707 */));
+
+        dmxSetChannel( mixFactors, mixScales, LEFT_FRONT_CHANNEL,  LEFT_FRONT_CHANNEL,   flev,   0 );
+        dmxSetChannel( mixFactors, mixScales, LEFT_FRONT_CHANNEL,  CENTER_FRONT_CHANNEL, clev,   0 );
+        dmxSetChannel( mixFactors, mixScales, LEFT_FRONT_CHANNEL,  LEFT_REAR_CHANNEL,    slevLL, 0 );
+        dmxSetChannel( mixFactors, mixScales, LEFT_FRONT_CHANNEL,  RIGHT_REAR_CHANNEL,   slevLR, 0 );
+
+        dmxSetChannel( mixFactors, mixScales, RIGHT_FRONT_CHANNEL, RIGHT_FRONT_CHANNEL,  flev,   0 );
+        dmxSetChannel( mixFactors, mixScales, RIGHT_FRONT_CHANNEL, CENTER_FRONT_CHANNEL, clev,   0 );
+        dmxSetChannel( mixFactors, mixScales, RIGHT_FRONT_CHANNEL, LEFT_REAR_CHANNEL,    slevRL, 0 );
+        dmxSetChannel( mixFactors, mixScales, RIGHT_FRONT_CHANNEL, RIGHT_REAR_CHANNEL,   slevRR, 0 );
+      }
+#endif  /* PCE_METADATA_ENABLE */
+      break;
+    default:
+      /* This configuration does not fit to any known downmix equation! */
+      err = PCMDMX_INVALID_MODE;
+      break;
+    }
+    /* Mark the output channels */
+    FDKmemclear(valid, PCM_DMX_MAX_CHANNELS*sizeof(unsigned int));
+    valid[LEFT_FRONT_CHANNEL]  = 1;
+    valid[RIGHT_FRONT_CHANNEL] = 1;
+    /* Update channel mode for the next stage */
+    inChMode = CH_MODE_2_0_0_0;
+  }
+
+  if (numOutChannel == ONE_CHANNEL) {
+    FIXP_DMX monoMixLevel;
+    INT      monoMixScale;
+
+#ifdef PCE_METADATA_ENABLE
+    if (  (pMetaData->typeFlags & TYPE_PCE_DATA)
+      && !(pMetaData->typeFlags & TYPE_DSE_DATA) )
+    { /* C' = (3+2*A)^-1 * [C+L+R+A*Ls+A+Rs]; */
+      monoMixLevel = mpegMixDownIdx2PreFact[2][pMetaData->matrixMixdownIdx];
+      monoMixScale = 0;
+
+      dmxClearChannel( mixFactors, mixScales, CENTER_FRONT_CHANNEL );
+      mixFactors[CENTER_FRONT_CHANNEL][CENTER_FRONT_CHANNEL] = monoMixLevel;
+      mixFactors[CENTER_FRONT_CHANNEL][LEFT_FRONT_CHANNEL]   = monoMixLevel;
+      mixFactors[CENTER_FRONT_CHANNEL][RIGHT_FRONT_CHANNEL]  = monoMixLevel;
+      monoMixLevel = FX_DBL2FX_DMX(fMult(monoMixLevel, mpegMixDownIdx2Coef[pMetaData->matrixMixdownIdx]));
+      mixFactors[CENTER_FRONT_CHANNEL][LEFT_REAR_CHANNEL]    = monoMixLevel;
+      mixFactors[CENTER_FRONT_CHANNEL][RIGHT_REAR_CHANNEL]   = monoMixLevel;
+    }
+    else
+#endif
+    { /* C' = L + R; [default] */
+      monoMixLevel = FL2FXCONST_DMX(0.5f);
+      monoMixScale = 1;
+      dmxClearChannel( mixFactors, mixScales, CENTER_FRONT_CHANNEL );  /* C is not in the mix */
+      dmxSetChannel( mixFactors, mixScales, CENTER_FRONT_CHANNEL, LEFT_FRONT_CHANNEL,  monoMixLevel, monoMixScale );
+      dmxAddChannel( mixFactors, mixScales, CENTER_FRONT_CHANNEL, RIGHT_FRONT_CHANNEL, monoMixLevel, monoMixScale );
+    }
+
+    /* Mark the output channel */
+    FDKmemclear(valid, PCM_DMX_MAX_CHANNELS*sizeof(unsigned int));
+    valid[CENTER_FRONT_CHANNEL] = 1;
+  }
+
+#define MAX_SEARCH_START_VAL  ( -7 )
+
+  {
+    LONG chSum[PCM_DMX_MAX_CHANNELS];
+    INT  chSumMax = MAX_SEARCH_START_VAL;
+
+    /* Determine the current maximum scale factor */
+    for (outCh=0; outCh < PCM_DMX_MAX_CHANNELS; outCh+=1) {
+      if (valid[outCh]!=0) {
+        for (inCh=0; inCh < PCM_DMX_MAX_CHANNELS; inCh+=1) {
+          if (mixScales[outCh][inCh] > maxScale)
+          { /* Store the new maximum */
+            maxScale = mixScales[outCh][inCh];
+          }
+        }
+      }
+    }
+
+    /* Individualy analyse output chanal levels */
+    for (outCh=0; outCh < PCM_DMX_MAX_CHANNELS; outCh+=1) {
+      chSum[outCh] = MAX_SEARCH_START_VAL;
+      if (valid[outCh]!=0) {
+        int  ovrflwProtScale = 0;
+
+        /* Accumulate all factors for each output channel */
+        chSum[outCh] = 0;
+        for (inCh=0; inCh < PCM_DMX_MAX_CHANNELS; inCh+=1) {
+          SHORT addFact = FX_DMX2SHRT(mixFactors[outCh][inCh]);
+          if ( mixScales[outCh][inCh] <= maxScale ) {
+            addFact >>= maxScale - mixScales[outCh][inCh];
+          } else {
+            addFact <<= mixScales[outCh][inCh] - maxScale;
+          }
+          chSum[outCh] += addFact;
+        }
+        if (chSum[outCh] > (LONG)MAXVAL_SGL) {
+          while (chSum[outCh] > (LONG)MAXVAL_SGL) {
+            ovrflwProtScale += 1;
+            chSum[outCh] >>= 1;
+          }
+        } else if (chSum[outCh] > 0) {
+          while ((chSum[outCh]<<1) <= (LONG)MAXVAL_SGL) {
+            ovrflwProtScale -= 1;
+            chSum[outCh] <<= 1;
+          }
+        }
+        /* Store the differential scaling in the same array */
+        chSum[outCh] = ovrflwProtScale;
+      }
+    }
+
+    for (outCh=0; outCh < PCM_DMX_MAX_CHANNELS; outCh+=1) {
+      if ( (valid[outCh] != 0)
+        && (chSum[outCh] > chSumMax) )
+      { /* Store the new maximum */
+        chSumMax = chSum[outCh];
+      }
+    }
+    maxScale = FDKmax(maxScale+chSumMax, 0);
+
+    /* Normalize all factors */
+    for (outCh=0; outCh < PCM_DMX_MAX_CHANNELS; outCh+=1) {
+      if (valid[outCh]!=0) {
+        for (inCh=0; inCh < PCM_DMX_MAX_CHANNELS; inCh+=1) {
+          if (mixFactors[outCh][inCh] != (FIXP_DMX)0) {
+            if ( mixScales[outCh][inCh] <= maxScale ) {
+              mixFactors[outCh][inCh] >>= maxScale - mixScales[outCh][inCh];
+            } else {
+              mixFactors[outCh][inCh] <<= mixScales[outCh][inCh] - maxScale;
+            }
+            mixScales[outCh][inCh] = maxScale;
+          }
+        }
+      }
+    }
+  }
+
+
+  /* return the scale factor */
+  *pOutScale = maxScale;
+
+  return (err);
+}
+
 
 /** Open and initialize an instance of the PCM downmix module
  * @param [out] Pointer to a buffer receiving the handle of the new instance.
@@ -526,24 +1447,24 @@ PCMDMX_ERROR pcmDmx_Reset (
   if (self == NULL) { return (PCMDMX_INVALID_HANDLE); }
 
   if (flags & PCMDMX_RESET_PARAMS) {
-    self->dualChannelMode   = STEREO_MODE;
-    self->numOutputChannels = 0;
-    self->applyProcessing   = 0;
-    self->frameDelay        = 0;
-    self->expiryFrame       = PCMDMX_DFLT_EXPIRY_FRAME;
+    PCM_DMX_USER_PARAMS *pParams = &self->userParams;
+
+    pParams->dualChannelMode   = STEREO_MODE;
+    pParams->pseudoSurrMode    = NEVER_DO_PS_DMX;
+    pParams->numOutChannelsMax = PCM_DMX_DFLT_MAX_OUT_CHANNELS;
+    pParams->numOutChannelsMin = PCM_DMX_DFLT_MIN_OUT_CHANNELS;
+    pParams->frameDelay        = 0;
+    pParams->expiryFrame       = PCM_DMX_DFLT_EXPIRY_FRAME;
+
+    self->applyProcessing      = 0;
   }
 
   if (flags & PCMDMX_RESET_BS_DATA) {
     int slot;
+    /* Init all slots with a default set */
     for (slot = 0; slot <= PCM_DMX_MAX_DELAY_FRAMES; slot += 1) {
-      self->dvbMixDownLevels[slot].centerMixLevelValue    = dvbDownmixFactors[2]; /* 0.707 */
-      self->dvbMixDownLevels[slot].surroundMixLevelValue  = dvbDownmixFactors[0]; /* 1.000 */
-      self->dvbMixDownLevels[slot].mixLevelsAvail = 0;
-
-      self->mpegMixDownInfo[slot].mixdownAvailable = 0;
+      FDKmemcpy(&self->bsMetaData[slot], &dfltMetaData, sizeof(DMX_BS_META_DATA));
     }
-    /* Reset expiry counter */
-    self->expiryCount = 0;
   }
 
   return (PCMDMX_OK);
@@ -558,8 +1479,8 @@ PCMDMX_ERROR pcmDmx_Reset (
  **/
 PCMDMX_ERROR pcmDmx_SetParam (
     HANDLE_PCM_DOWNMIX  self,
-    PCMDMX_PARAM        param,
-    UINT                value
+    const PCMDMX_PARAM  param,
+    const INT           value
   )
 {
   switch (param)
@@ -567,39 +1488,70 @@ PCMDMX_ERROR pcmDmx_SetParam (
   case DMX_BS_DATA_EXPIRY_FRAME:
     if (self == NULL)
       return (PCMDMX_INVALID_HANDLE);
-    self->expiryFrame = value;
+    self->userParams.expiryFrame = (value > 0) ? (UINT)value : 0;
     break;
 
   case DMX_BS_DATA_DELAY:
-    if (value > PCM_DMX_MAX_DELAY_FRAMES) {
+    if ( (value > PCM_DMX_MAX_DELAY_FRAMES)
+      || (value < 0) ) {
       return (PCMDMX_UNABLE_TO_SET_PARAM);
     }
     if (self == NULL) {
       return (PCMDMX_INVALID_HANDLE);
     }
-    self->frameDelay = value;
+    self->userParams.frameDelay = (UCHAR)value;
     break;
 
-  case NUMBER_OF_OUTPUT_CHANNELS:
-    switch ((int)value) {  /* supported output channels */
-    case -1: case 0: case 1: case 2:
-    case 6: case 8:
+  case MIN_NUMBER_OF_OUTPUT_CHANNELS:
+    switch (value) {  /* supported output channels */
+    case -1: case 0: case ONE_CHANNEL: case TWO_CHANNEL:
+#if (PCM_DMX_MAX_OUT_CHANNELS >= 6)
+    case SIX_CHANNEL:
+#endif
+#if (PCM_DMX_MAX_OUT_CHANNELS >= 8)
+    case EIGHT_CHANNEL:
+#endif
       break;
     default:
       return (PCMDMX_UNABLE_TO_SET_PARAM);
     }
     if (self == NULL)
       return (PCMDMX_INVALID_HANDLE);
-    if ((int)value > 0) {
-      self->numOutputChannels = (int)value;
-      self->applyProcessing = 1;
-    } else {
-      self->numOutputChannels = 0;
-      self->applyProcessing = 0;
+    /* Store the new value */
+    self->userParams.numOutChannelsMin = (value > 0) ? value : -1;
+    if ( (value > 0)
+      && (self->userParams.numOutChannelsMax > 0)
+      && (value > self->userParams.numOutChannelsMax) )
+    { /* MIN > MAX would be an invalid state. Thus set MAX = MIN in this case. */
+      self->userParams.numOutChannelsMax = self->userParams.numOutChannelsMin;
     }
     break;
 
-  case DUAL_CHANNEL_DOWNMIX_MODE:
+  case MAX_NUMBER_OF_OUTPUT_CHANNELS:
+    switch (value) {  /* supported output channels */
+    case -1: case 0: case ONE_CHANNEL: case TWO_CHANNEL:
+#if (PCM_DMX_MAX_OUT_CHANNELS >= 6)
+    case SIX_CHANNEL:
+#endif
+#if (PCM_DMX_MAX_OUT_CHANNELS >= 8)
+    case EIGHT_CHANNEL:
+#endif
+      break;
+    default:
+      return (PCMDMX_UNABLE_TO_SET_PARAM);
+    }
+    if (self == NULL)
+      return (PCMDMX_INVALID_HANDLE);
+    /* Store the new value */
+    self->userParams.numOutChannelsMax = (value > 0) ? value : -1;
+    if ( (value > 0)
+      && (value < self->userParams.numOutChannelsMin) )
+    { /* MAX < MIN would be an invalid state. Thus set MIN = MAX in this case. */
+      self->userParams.numOutChannelsMin = self->userParams.numOutChannelsMax;
+    }
+    break;
+
+  case DMX_DUAL_CHANNEL_MODE:
     switch ((DUAL_CHANNEL_MODE)value) {
     case STEREO_MODE:
     case CH1_MODE:
@@ -611,8 +1563,22 @@ PCMDMX_ERROR pcmDmx_SetParam (
     }
     if (self == NULL)
       return (PCMDMX_INVALID_HANDLE);
-    self->dualChannelMode = (DUAL_CHANNEL_MODE)value;
-    self->applyProcessing = 1;
+    self->userParams.dualChannelMode = (DUAL_CHANNEL_MODE)value;
+    self->applyProcessing = 1;  /* Force processing */
+    break;
+
+  case DMX_PSEUDO_SURROUND_MODE:
+    switch ((PSEUDO_SURROUND_MODE)value) {
+    case NEVER_DO_PS_DMX:
+    case AUTO_PS_DMX:
+    case FORCE_PS_DMX:
+      break;
+    default:
+      return (PCMDMX_UNABLE_TO_SET_PARAM);
+    }
+    if (self == NULL)
+      return (PCMDMX_INVALID_HANDLE);
+    self->userParams.pseudoSurrMode = (PSEUDO_SURROUND_MODE)value;
     break;
 
   default:
@@ -622,87 +1588,239 @@ PCMDMX_ERROR pcmDmx_SetParam (
   return (PCMDMX_OK);
 }
 
-
-/** Read the ancillary data transported in DSEs of DVB streams with MPEG-4 content
+/** Get one parameter value of one PCM downmix module instance.
  * @param [in] Handle of PCM downmix module instance.
- * @param [in] Pointer to ancillary data buffer.
- * @param [in] Size of ancillary data.
- * @param [in] Flag indicating wheter the DVB ancillary data is from an MPEG-1/2 or an MPEG-4 stream.
+ * @param [in] Parameter to be set.
+ * @param [out] Pointer to buffer receiving the parameter value.
  * @returns Returns an error code.
  **/
-PCMDMX_ERROR pcmDmx_ReadDvbAncData (
+PCMDMX_ERROR pcmDmx_GetParam (
     HANDLE_PCM_DOWNMIX  self,
-    UCHAR *pAncDataBuf,
-    UINT   ancDataBytes,
+    const PCMDMX_PARAM  param,
+    INT * const         pValue
+  )
+{
+  PCM_DMX_USER_PARAMS *pUsrParams;
+
+  if ( (self == NULL)
+    || (pValue == NULL) ) {
+    return (PCMDMX_INVALID_HANDLE);
+  }
+  pUsrParams = &self->userParams;
+
+  switch (param)
+  {
+  case DMX_BS_DATA_EXPIRY_FRAME:
+    *pValue = (INT)pUsrParams->expiryFrame;
+    break;
+  case DMX_BS_DATA_DELAY:
+    *pValue = (INT)pUsrParams->frameDelay;
+    break;
+  case MIN_NUMBER_OF_OUTPUT_CHANNELS:
+    *pValue = (INT)pUsrParams->numOutChannelsMin;
+    break;
+  case MAX_NUMBER_OF_OUTPUT_CHANNELS:
+    *pValue = (INT)pUsrParams->numOutChannelsMax;
+    break;
+  case DMX_DUAL_CHANNEL_MODE:
+    *pValue = (INT)pUsrParams->dualChannelMode;
+    break;
+  case DMX_PSEUDO_SURROUND_MODE:
+    *pValue = (INT)pUsrParams->pseudoSurrMode;
+    break;
+  default:
+    return (PCMDMX_UNKNOWN_PARAM);
+  }
+
+  return (PCMDMX_OK);
+}
+
+
+#ifdef DSE_METADATA_ENABLE
+
+#define MAX_DSE_ANC_BYTES       ( 16 )    /* 15 bytes */
+#define ANC_DATA_SYNC_BYTE      ( 0xBC )  /* ancillary data sync byte. */
+
+/*
+ * Read DMX meta-data from a data stream element.
+ */
+PCMDMX_ERROR pcmDmx_Parse (
+    HANDLE_PCM_DOWNMIX  self,
+    HANDLE_FDK_BITSTREAM  hBs,
+    UINT  ancDataBits,
     int    isMpeg2
   )
 {
-  DVB_MIXDOWN_LEVELS *pDownmixLevels = &self->dvbMixDownLevels[0];
+  PCMDMX_ERROR errorStatus = PCMDMX_OK;
+  DMX_BS_META_DATA *pBsMetaData = &self->bsMetaData[0];
 
-  int   offset = (isMpeg2) ? 2 : 0;
-  UCHAR ancDataStatus;
+  int   skip4Dmx = 0, skip4Ext = 0;
+  int   dmxLvlAvail = 0, extDataAvail = 0;
+  int   foundNewData = 0;
+  UINT  minAncBits = ((isMpeg2) ? 5 : 3)*8;
 
-  if (self == NULL) { return (PCMDMX_INVALID_HANDLE); }
+  if ( (self == NULL)
+    || (hBs  == NULL) ) { return (PCMDMX_INVALID_HANDLE); }
+
+  ancDataBits = FDKgetValidBits(hBs);
 
   /* sanity checks */
-  if (pAncDataBuf == NULL || ancDataBytes < (UCHAR)(3+offset)) {
+  if ( (ancDataBits < minAncBits)
+    || (ancDataBits > FDKgetValidBits(hBs)) ) {
     return (PCMDMX_CORRUPT_ANC_DATA);
   }
 
+  pBsMetaData = &self->bsMetaData[0];
+
+  if (isMpeg2) {
+    /* skip DVD ancillary data */
+    FDKpushFor(hBs, 16);
+  }
+
   /* check sync word */
-  if (pAncDataBuf[offset] != ANC_DATA_SYNC_BYTE) {
+  if (FDKreadBits(hBs,8) != ANC_DATA_SYNC_BYTE) {
     return (PCMDMX_CORRUPT_ANC_DATA);
   }
 
-  offset += 2;
-  ancDataStatus = pAncDataBuf[offset++];
+  /* skip MPEG audio type and Dolby surround mode */
+  FDKpushFor(hBs, 4);
 
   if (isMpeg2) {
-    /* skip advanced_dynamic_range_control */
-    if (ancDataStatus & 0x80) offset += 3;
-    /* skip dialog_normalization */
-    if (ancDataStatus & 0x40) offset += 1;
-    /* skip reproduction_level */
-    if (ancDataStatus & 0x20) offset += 1;
+    /* int numAncBytes = */ FDKreadBits(hBs, 4);
+    /* advanced dynamic range control */
+    if (FDKreadBit(hBs)) skip4Dmx += 24;
+    /* dialog normalization */
+    if (FDKreadBit(hBs)) skip4Dmx += 8;
+    /* reproduction_level */
+    if (FDKreadBit(hBs)) skip4Dmx += 8;
+  } else {
+    FDKpushFor(hBs, 2);   /* drc presentation mode */
+    pBsMetaData->pseudoSurround = FDKreadBit(hBs);
+    FDKpushFor(hBs, 4);   /* reserved bits */
   }
-  else {
-    /* check reserved bits */
-    if (ancDataStatus & 0xE8) { return (PCMDMX_CORRUPT_ANC_DATA); }
+
+  /* downmixing levels MPEGx status */
+  dmxLvlAvail  = FDKreadBit(hBs);
+
+  if (isMpeg2) {
+    /* scale factor CRC status */
+    if (FDKreadBit(hBs)) skip4Ext += 16;
+  } else {
+    /* ancillary data extension status */
+    extDataAvail = FDKreadBit(hBs);
   }
 
+  /* audio coding and compression status */
+  if (FDKreadBit(hBs)) skip4Ext += 16;
+  /* coarse grain timecode status */
+  if (FDKreadBit(hBs)) skip4Ext += 16;
+  /* fine grain timecode status */
+  if (FDKreadBit(hBs)) skip4Ext += 16;
+
+  /* skip the useless data to get to the DMX levels */
+  FDKpushFor(hBs, skip4Dmx);
+
   /* downmix_levels_MPEGX */
-  if (ancDataStatus & 0x10)
+  if (dmxLvlAvail)
   {
-    int   foundNewData = 0;
-    UCHAR downmixData = pAncDataBuf[offset++];
-
-    if (downmixData & 0x80) {  /* center_mix_level_on */
-      pDownmixLevels->centerMixLevelValue =
-        dvbDownmixFactors[(downmixData >> 4) & 0x07];
+    if (FDKreadBit(hBs)) {  /* center_mix_level_on */
+      pBsMetaData->cLevIdx = FDKreadBits(hBs, 3);
       foundNewData = 1;
     } else {
-      pDownmixLevels->centerMixLevelValue = dvbDownmixFactors[0];
-      if (downmixData & 0x70) { return (PCMDMX_CORRUPT_ANC_DATA); }
+      FDKreadBits(hBs, 3);
     }
-
-    if (downmixData & 0x08) {  /* surround_mix_level_on */
-      pDownmixLevels->surroundMixLevelValue =
-        dvbDownmixFactors[downmixData & 0x07];
+    if (FDKreadBit(hBs)) {  /* surround_mix_level_on */
+      pBsMetaData->sLevIdx = FDKreadBits(hBs, 3);
       foundNewData = 1;
     } else {
-      pDownmixLevels->surroundMixLevelValue = dvbDownmixFactors[0];
-      if (downmixData & 0x07) { return (PCMDMX_CORRUPT_ANC_DATA); }
+      FDKreadBits(hBs, 3);
+    }
+  }
+
+  /* skip the useless data to get to the ancillary data extension */
+  FDKpushFor(hBs, skip4Ext);
+
+  /* anc data extension (MPEG-4 only) */
+  if (extDataAvail) {
+    int extDmxLvlSt, extDmxGainSt, extDmxLfeSt;
+
+    FDKreadBit(hBs);        /* reserved bit */
+    extDmxLvlSt  = FDKreadBit(hBs);
+    extDmxGainSt = FDKreadBit(hBs);
+    extDmxLfeSt  = FDKreadBit(hBs);
+    FDKreadBits(hBs, 4);    /* reserved bits */
+
+    if (extDmxLvlSt) {
+      pBsMetaData->dmixIdxA = FDKreadBits(hBs, 3);
+      pBsMetaData->dmixIdxB = FDKreadBits(hBs, 3);
+      FDKreadBits(hBs, 2);  /* reserved bits */
+      foundNewData = 1;
+    }
+    if (extDmxGainSt) {
+      pBsMetaData->dmxGainIdx5 = FDKreadBits(hBs, 7);
+      FDKreadBit(hBs);      /* reserved bit */
+      pBsMetaData->dmxGainIdx2 = FDKreadBits(hBs, 7);
+      FDKreadBit(hBs);      /* reserved bit */
+      foundNewData = 1;
+    }
+    if (extDmxLfeSt) {
+      pBsMetaData->dmixIdxLfe = FDKreadBits(hBs, 4);
+      FDKreadBits(hBs, 4);  /* reserved bits */
+      foundNewData = 1;
     }
+  }
 
-    pDownmixLevels->mixLevelsAvail = foundNewData;
+  /* final sanity check on the amount of read data */
+  if ((INT)FDKgetValidBits(hBs) < 0) {
+    errorStatus = PCMDMX_CORRUPT_ANC_DATA;
   }
 
-  /* Reset expiry counter */
-  self->expiryCount = 0;
+  if ( (errorStatus  == PCMDMX_OK)
+    && (foundNewData == 1) ) {
+    /* announce new data */
+    pBsMetaData->typeFlags |= TYPE_DSE_DATA;
+    /* reset expiry counter */
+    pBsMetaData->expiryCount = 0;
+  }
 
-  return (PCMDMX_OK);
+  return (errorStatus);
+}
+
+/*
+ * Read DMX meta-data from a data stream element.
+ */
+PCMDMX_ERROR pcmDmx_ReadDvbAncData (
+    HANDLE_PCM_DOWNMIX  self,
+    UCHAR *pAncDataBuf,
+    UINT   ancDataBytes,
+    int    isMpeg2
+  )
+{
+  FDK_BITSTREAM bs;
+  HANDLE_FDK_BITSTREAM hBs = &bs;
+  PCMDMX_ERROR errorStatus = PCMDMX_OK;
+
+  if (self == NULL) { return (PCMDMX_INVALID_HANDLE); }
+
+  /* sanity checks */
+  if ( (pAncDataBuf == NULL)
+    || (ancDataBytes == 0) ) {
+    return (PCMDMX_CORRUPT_ANC_DATA);
+  }
+
+  FDKinitBitStream (hBs, pAncDataBuf, MAX_DSE_ANC_BYTES, ancDataBytes*8, BS_READER);
+
+  errorStatus = pcmDmx_Parse (
+                        self,
+                        hBs,
+                        ancDataBytes*8,
+                        isMpeg2 );
+
+  return (errorStatus);
 }
+#endif  /* DSE_METADATA_ENABLE */
 
+#ifdef PCE_METADATA_ENABLE
 /** Set the matrix mixdown information extracted from the PCE of an AAC bitstream.
  *  Note: Call only if matrix_mixdown_idx_present is true.
  * @param [in] Handle of PCM downmix module instance.
@@ -717,38 +1835,36 @@ PCMDMX_ERROR pcmDmx_SetMatrixMixdownFromPce (
     int                 pseudoSurroundEnable
   )
 {
-  MPEG_MIXDOWN_INFO *pMpegMixDownInfo;
+  DMX_BS_META_DATA *pBsMetaData = &self->bsMetaData[0];
 
   if (self == NULL) {
     return (PCMDMX_INVALID_HANDLE);
   }
 
-  pMpegMixDownInfo = &self->mpegMixDownInfo[0];
-
   if (matrixMixdownPresent) {
-    pMpegMixDownInfo->matrixMixdownIdx     = matrixMixdownIdx & 0x03;
-    pMpegMixDownInfo->pseudoSurroundEnable = pseudoSurroundEnable;
+    pBsMetaData->pseudoSurround = pseudoSurroundEnable;
+    pBsMetaData->matrixMixdownIdx = matrixMixdownIdx & 0x03;
+    pBsMetaData->typeFlags |= TYPE_PCE_DATA;
+    /* Reset expiry counter */
+    pBsMetaData->expiryCount = 0;
   }
 
-  pMpegMixDownInfo->mixdownAvailable = matrixMixdownPresent;
-  /* Reset expiry counter */
-  self->expiryCount = 0;
-
   return (PCMDMX_OK);
 }
+#endif  /* PCE_METADATA_ENABLE */
 
 
 /** Apply down or up mixing.
  * @param [in]    Handle of PCM downmix module instance.
- * @param [inout] Pointer to time buffer. Depending on interface configuration, the content of pTimeData is ignored,
- *                and the internal QMF buffer will be used as input data source. Otherwise, the MPEG Surround processing is
- *                applied to the timesignal pTimeData. For both variants, the resulting MPEG Surround signal is written into pTimeData.
+ * @param [inout] Pointer to buffer that hold the time domain signal.
  * @param [in]    Pointer where the amount of output samples is returned into.
  * @param [inout] Pointer where the amount of output channels is returned into.
  * @param [in]    Flag which indicates if output time data are writtern interleaved or as subsequent blocks.
  * @param [inout] Array where the corresponding channel type for each output audio channel is stored into.
  * @param [inout] Array where the corresponding channel type index for each output audio channel is stored into.
- * @param [in]    Array containing the output channel mapping to be used (From MPEG PCE ordering to whatever is required).
+ * @param [in]    Array containing the out channel mapping to be used (From MPEG PCE ordering to whatever is required).
+ * @param [out]   Pointer on a field receiving the scale factor that has to be applied on all samples afterwards.
+ *                If the handed pointer is NULL scaling is done internally.
  * @returns Returns an error code.
  **/
 PCMDMX_ERROR pcmDmx_ApplyFrame (
@@ -756,97 +1872,173 @@ PCMDMX_ERROR pcmDmx_ApplyFrame (
         INT_PCM                *pPcmBuf,
         UINT                    frameSize,
         INT                    *nChannels,
-
         int                     fInterleaved,
         AUDIO_CHANNEL_TYPE      channelType[],
         UCHAR                   channelIndices[],
-        const UCHAR             channelMapping[][8]
+        const UCHAR             channelMapping[][8],
+        INT                    *pDmxOutScale
   )
 {
+  PCM_DMX_USER_PARAMS  *pParam = NULL;
   PCMDMX_ERROR  errorStatus = PCMDMX_OK;
   DUAL_CHANNEL_MODE  dualChannelMode;
   PCM_DMX_CHANNEL_MODE  inChMode;
-  int   numOutChannels;
-  int   numInChannels = *nChannels;
-  int   slot;
+  PCM_DMX_CHANNEL_MODE  outChMode;
+  INT   devNull;  /* Just a dummy to avoid a lot of branches in the code */
+  int   numOutChannels, numInChannels;
+  int   inStride, outStride, offset;
+  int   dmxMaxScale, dmxScale;
+  int   ch, slot;
   UCHAR inOffsetTable[PCM_DMX_MAX_CHANNELS];
 
-  MPEG_MIXDOWN_INFO   mpegMixDownInfo;
-  DVB_MIXDOWN_LEVELS  dvbMixDownLevels;
+  DMX_BS_META_DATA  bsMetaData;
 
-  if (self == NULL) { return (PCMDMX_INVALID_HANDLE); }
-
-  if ( (self->expiryFrame > 0)
-    && (++self->expiryCount > self->expiryFrame) )
-  { /* The metadata read from bitstream is too old. */
-    errorStatus = pcmDmx_Reset(self, PCMDMX_RESET_BS_DATA);
+  if ( (self           == NULL)
+    || (nChannels      == NULL)
+    || (channelType    == NULL)
+    || (channelIndices == NULL)
+    || (channelMapping == NULL) ) {
+    return (PCMDMX_INVALID_HANDLE);
   }
 
-  FDKmemcpy(&mpegMixDownInfo, &self->mpegMixDownInfo[self->frameDelay], sizeof(MPEG_MIXDOWN_INFO));
-  /* Maintain delay line */
-  for (slot = self->frameDelay; slot > 0; slot -= 1) {
-    FDKmemcpy(&self->mpegMixDownInfo[slot], &self->mpegMixDownInfo[slot-1], sizeof(MPEG_MIXDOWN_INFO));
-  }
-  FDKmemcpy(&dvbMixDownLevels, &self->dvbMixDownLevels[self->frameDelay], sizeof(DVB_MIXDOWN_LEVELS));
-  /* Maintain delay line */
-  for (slot = self->frameDelay; slot > 0; slot -= 1) {
-    FDKmemcpy(&self->dvbMixDownLevels[slot], &self->dvbMixDownLevels[slot-1], sizeof(DVB_MIXDOWN_LEVELS));
+  /* Init the output scaling */
+  dmxScale = 0;
+  if (pDmxOutScale != NULL) {
+    /* Avoid final scaling internally and hand it to the outside world. */
+    *pDmxOutScale = 0;
+    dmxMaxScale = PCMDMX_MAX_HEADROOM;
+  } else {
+    /* Apply the scaling internally. */
+    pDmxOutScale = &devNull;  /* redirect to temporal stack memory */
+    dmxMaxScale = 0;
   }
 
-  if (self->applyProcessing == 0) { return (errorStatus); }
+  pParam = &self->userParams;
+  numInChannels = *nChannels;
 
+  /* Perform some input sanity checks */
   if (pPcmBuf == NULL)     { return (PCMDMX_INVALID_ARGUMENT); }
   if (frameSize == 0)      { return (PCMDMX_INVALID_ARGUMENT); }
-  if (numInChannels == 0)  { return (PCMDMX_INVALID_ARGUMENT); }
+  if ( (numInChannels == 0)
+    || (numInChannels > PCM_DMX_MAX_IN_CHANNELS) )
+                           { return (PCMDMX_INVALID_ARGUMENT); }
+
+  /* Check on misconfiguration */
+  FDK_ASSERT( (pParam->numOutChannelsMax <= 0) \
+           || (pParam->numOutChannelsMax >= pParam->numOutChannelsMin));
+
+  /* Determine if the module has to do processing */
+  if (   (self->applyProcessing == 0)
+    &&  ((pParam->numOutChannelsMax <= 0)
+      || (pParam->numOutChannelsMax >= numInChannels))
+    &&   (pParam->numOutChannelsMin <= numInChannels) ) {
+    /* Nothing to do */
+    return (errorStatus);
+  }
 
-  if (self->numOutputChannels <= 0) {
+  /* Determine the number of output channels */
+  if ( (pParam->numOutChannelsMax > 0)
+    && (numInChannels > pParam->numOutChannelsMax) ) {
+    numOutChannels = pParam->numOutChannelsMax;
+  }
+  else if (numInChannels < pParam->numOutChannelsMin) {
+    numOutChannels = pParam->numOutChannelsMin;
+  }
+  else {
     numOutChannels = numInChannels;
-  } else {
-    numOutChannels = self->numOutputChannels;
   }
-  dualChannelMode = self->dualChannelMode;
+
+  dualChannelMode = pParam->dualChannelMode;
 
   /* Analyse input channel configuration and get channel offset
    * table that can be accessed with the fixed channel labels. */
-  inChMode = getChannelMode(
+  errorStatus = getChannelMode(
                    numInChannels,
                    channelType,
                    channelIndices,
-                   inOffsetTable
+                   inOffsetTable,
+                  &inChMode
                  );
-  if (inChMode == CH_MODE_UNDEFINED) {
+  if ( PCMDMX_IS_FATAL_ERROR(errorStatus)
+    || (inChMode == CH_MODE_UNDEFINED) ) {
     /* We don't need to restore because the channel
        configuration has not been changed. Just exit. */
     return (PCMDMX_INVALID_CH_CONFIG);
   }
 
+  /* Set input stride and offset */
+  if (fInterleaved) {
+    inStride  = numInChannels;
+    offset = 1;                /* Channel specific offset factor */
+  } else {
+    inStride  = 1;
+    offset = frameSize;        /* Channel specific offset factor */
+  }
+
+  /* Reset downmix meta data if necessary */
+  if ( (pParam->expiryFrame > 0)
+    && (++self->bsMetaData[0].expiryCount > pParam->expiryFrame) )
+  { /* The metadata read from bitstream is too old. */
+    PCMDMX_ERROR err = pcmDmx_Reset(self, PCMDMX_RESET_BS_DATA);
+    FDK_ASSERT(err == PCMDMX_OK);
+  }
+  FDKmemcpy(&bsMetaData, &self->bsMetaData[pParam->frameDelay], sizeof(DMX_BS_META_DATA));
+  /* Maintain delay line */
+  for (slot = pParam->frameDelay; slot > 0; slot -= 1) {
+    FDKmemcpy(&self->bsMetaData[slot], &self->bsMetaData[slot-1], sizeof(DMX_BS_META_DATA));
+  }
+
   /* - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - */
+#ifdef PCM_DOWNMIX_ENABLE
   if ( numInChannels > numOutChannels )
   { /* Apply downmix */
-    INT_PCM  *pInCF, *pInLF, *pInRF, *pInLO, *pInRO, *pInLR, *pInRR, *pOutL, *pOutR;
-    FIXP_SGL  flev, clev, slev;
-
-    UINT   sample;
-    int    inStride, outStride, offset;
-    int    useGuidedDownMix = 0;
-    UCHAR  outOffsetTable[PCM_DMX_MAX_CHANNELS];
-
-    /* Set I/O strides and offsets */
-    if (fInterleaved) {
-      inStride  = numInChannels;
-      outStride = TWO_CHANNEL;   /* The output of STAGE ONE is always STEREO !!!
-                                    STAGE TWO creates a downmix to mono if required. */
-      offset = 1;                /* Channel specific offset factor */
-    } else {
-      inStride  = 1;
-      outStride = 1;
-      offset = frameSize;        /* Channel specific offset factor */
+    INT_PCM  *pInPcm[PCM_DMX_MAX_IN_CHANNELS] = { NULL };
+    INT_PCM  *pOutPcm[PCM_DMX_MAX_OUT_CHANNELS] = { NULL };
+    FIXP_DMX  mixFactors[PCM_DMX_MAX_CHANNELS][PCM_DMX_MAX_CHANNELS];
+    UCHAR     outOffsetTable[PCM_DMX_MAX_CHANNELS];
+    UINT      sample;
+    int       chCfg = 0;
+    int       bypScale = 0;
+
+#if (PCM_DMX_MAX_IN_CHANNELS >= 7)
+    if (numInChannels > SIX_CHANNEL) {
+      AUDIO_CHANNEL_TYPE multiPurposeChType[2];
+
+      /* Get the type of the multipurpose channels */
+      multiPurposeChType[0] = channelType[inOffsetTable[LEFT_MULTIPRPS_CHANNEL]];
+      multiPurposeChType[1] = channelType[inOffsetTable[RIGHT_MULTIPRPS_CHANNEL]];
+
+      /* Check if the input configuration is one defined in the standard. */
+      switch (inChMode) {
+      case CH_MODE_5_0_2_1:  /* chCfg 7 || 14 */
+        /* Further analyse the input config to distinguish the two CH_MODE_5_0_2_1 configs. */
+        if ( (multiPurposeChType[0] == ACT_FRONT_TOP)
+          && (multiPurposeChType[1] == ACT_FRONT_TOP) ) {
+          chCfg = 14;
+        } else {
+          chCfg = 7;
+        }
+        break;
+      case CH_MODE_3_0_3_1:  /* chCfg 11 */
+        chCfg = 11;
+        break;
+      case CH_MODE_3_0_4_1:  /* chCfg 12 */
+        chCfg = 12;
+        break;
+      default:
+        chCfg = 0;  /* Not a known config */
+        break;
+      }
     }
+#endif
 
-    /* Get channel description and channel mapping for this
-     * stages number of output channels (always STEREO). */
+    /* Set this stages output stride and channel mode: */
+    outStride = (fInterleaved) ? numOutChannels : 1;
+    outChMode = outChModeTable[numOutChannels];
+
+    /* Get channel description and channel mapping for the desired output configuration. */
     getChannelDescription(
-            CH_MODE_2_0_0_0,
+            outChMode,
             channelMapping,
             channelType,
             channelIndices,
@@ -854,287 +2046,300 @@ PCMDMX_ERROR pcmDmx_ApplyFrame (
            );
     /* Now there is no way back because we modified the channel configuration! */
 
-    /* Set channel pointer for input */
-    pInCF = &pPcmBuf[inOffsetTable[CENTER_FRONT_CHANNEL]*offset];
-    pInLF = &pPcmBuf[inOffsetTable[LEFT_FRONT_CHANNEL]*offset];
-    pInRF = &pPcmBuf[inOffsetTable[RIGHT_FRONT_CHANNEL]*offset];
-    pInLO = &pPcmBuf[inOffsetTable[LEFT_OUTSIDE_CHANNEL]*offset];
-    pInRO = &pPcmBuf[inOffsetTable[RIGHT_OUTSIDE_CHANNEL]*offset];
-    pInLR = &pPcmBuf[inOffsetTable[LEFT_REAR_CHANNEL]*offset];
-    pInRR = &pPcmBuf[inOffsetTable[RIGHT_REAR_CHANNEL]*offset];
-
-    /* Set channel pointer for output
-       Caution: Different channel mapping compared to input */
-    pOutL = &pPcmBuf[outOffsetTable[LEFT_FRONT_CHANNEL]*offset];    /* LEFT_FRONT_CHANNEL  */
-    pOutR = &pPcmBuf[outOffsetTable[RIGHT_FRONT_CHANNEL]*offset];   /* RIGHT_FRONT_CHANNEL */
-
-    /* Set downmix levels: */
-    flev = ATTENUATION_FACTOR_1;    /* 0.707 */
-    clev = ATTENUATION_FACTOR_1;    /* 0.707 */
-    slev = ATTENUATION_FACTOR_1;    /* 0.707 */
-
-    if ( dvbMixDownLevels.mixLevelsAvail ) {
-      clev = dvbMixDownLevels.centerMixLevelValue;
-      slev = dvbMixDownLevels.surroundMixLevelValue;
-      useGuidedDownMix = 1;
-    }
-
-    /* FIRST STAGE:
-         Always downmix to 2 channel output: */
-    switch ( inChMode )
-    {
-    case CH_MODE_2_0_0_0:
-    case CH_MODE_2_0_0_1: 
-      /* 2/0 input: */
-      switch (dualChannelMode)
-      {
-      case CH1_MODE:  /* L' = 0.707 * Ch1;  R' = 0.707 * Ch1 */
-        for (sample = 0; sample < frameSize; sample++) {
-          *pOutL = *pOutR =
-            (INT_PCM)SATURATE_RIGHT_SHIFT(fMult((FIXP_PCM)*pInLF, flev), DFRACT_BITS-SAMPLE_BITS, SAMPLE_BITS);
-
-          pInLF += inStride;
-          pOutL += outStride; pOutR += outStride;
-        }
-        break;
-
-      case CH2_MODE:  /* L' = 0.707 * Ch2;  R' = 0.707 * Ch2 */
-        for (sample = 0; sample < frameSize; sample++) {
-          *pOutL = *pOutR =
-            (INT_PCM)SATURATE_RIGHT_SHIFT(fMult((FIXP_PCM)*pInRF, flev), DFRACT_BITS-SAMPLE_BITS, SAMPLE_BITS);
-
-          pInRF += inStride;
-          pOutL += outStride; pOutR += outStride;
+    /* Create the DMX matrix */
+    errorStatus = getMixFactors (
+                       (chCfg>0) ? 1 : 0,
+                       (chCfg>0) ? (PCM_DMX_CHANNEL_MODE)chCfg : inChMode,
+                       outChMode,
+                       pParam,
+                      &bsMetaData,
+                       mixFactors,
+                      &dmxScale
+                     );
+    /* No fatal errors can occur here. The function is designed to always return a valid matrix.
+       The error code is used to signal configurations and matrices that are not conform to any standard. */
+
+    /* Determine the final scaling */
+    bypScale = FDKmin(dmxMaxScale, dmxScale);
+    *pDmxOutScale += bypScale;
+    dmxScale -= bypScale;
+
+    { /* Set channel pointer for input. Remove empty cols. */
+      int inCh, outCh, map[PCM_DMX_MAX_CHANNELS];
+      ch = 0;
+      for (inCh=0; inCh < PCM_DMX_MAX_CHANNELS; inCh+=1) {
+        if (inOffsetTable[inCh] != 255) {
+          pInPcm[ch] = &pPcmBuf[inOffsetTable[inCh]*offset];
+          map[ch++]  = inCh;
         }
-        break;
-      case MIXED_MODE:  /* L' = 0.5*Ch1 + 0.5*Ch2;  R' = 0.5*Ch1 + 0.5*Ch2 */
-        for (sample = 0; sample < frameSize; sample++) {
-          *pOutL = *pOutR = (*pInLF >> 1) + (*pInRF >> 1);
-
-          pInLF += inStride;  pInRF += inStride;
-          pOutL += outStride; pOutR += outStride;
-        }
-        break;
-      default:
-      case STEREO_MODE:
-        /* nothing to do */
-        break;
       }
-      break;
-
-    case CH_MODE_3_0_0_0:
-      /* 3/0 input:  L' = L + 0.707*C;  R' = R + 0.707*C; */
-      for (sample = 0; sample < frameSize; sample++)
-      {
-        FIXP_DBL tCF = fMultDiv2((FIXP_PCM)*pInCF, clev);
-#if (SAMPLE_BITS == 32)
-        /* left channel */
-        *pOutL = (INT_PCM)SATURATE_LEFT_SHIFT((FX_PCM2FX_DBL((FIXP_PCM)*pInLF)>>1)+tCF, 1, SAMPLE_BITS);
-        /* right channel */
-        *pOutR = (INT_PCM)SATURATE_LEFT_SHIFT((FX_PCM2FX_DBL((FIXP_PCM)*pInRF)>>1)+tCF, 1, SAMPLE_BITS);
-#else
-        /* left channel */
-        *pOutL = (INT_PCM)SATURATE_RIGHT_SHIFT((FX_PCM2FX_DBL((FIXP_PCM)*pInLF)>>1)+tCF, DFRACT_BITS-SAMPLE_BITS-1, SAMPLE_BITS);
-        /* right channel */
-        *pOutR = (INT_PCM)SATURATE_RIGHT_SHIFT((FX_PCM2FX_DBL((FIXP_PCM)*pInRF)>>1)+tCF, DFRACT_BITS-SAMPLE_BITS-1, SAMPLE_BITS);
-#endif
-        pInLF += inStride;  pInRF += inStride;  pInCF  += inStride;
-        pOutL += outStride; pOutR += outStride;
+      FDK_ASSERT(ch == numInChannels);
+
+      /* Remove unused cols from factor matrix */
+      for (inCh=0; inCh < numInChannels; inCh+=1) {
+        if (inCh != map[inCh]) {
+          int outCh;
+          for (outCh=0; outCh < PCM_DMX_MAX_CHANNELS; outCh+=1) {
+            mixFactors[outCh][inCh] = mixFactors[outCh][map[inCh]];
+          }
+        }
       }
-      break;
 
-    /* 2/1 input: not supported!
-    case CH_MODE_2_0_1_0: */
-
-    case CH_MODE_3_0_1_0:
-      if (useGuidedDownMix) {
-        /* 3/1 input:  L' = L + clev*C + 0.707*slev*S;  R' = R + clev*C + 0.707*slev*S; */
-        slev = FX_DBL2FX_SGL(fMult(flev, slev));  /* 0.707*slef */
-
-        for (sample = 0; sample < frameSize; sample++)
-        {
-          FIXP_DBL tCF = fMultDiv2((FIXP_PCM)*pInCF, clev) >> 1;
-          FIXP_DBL tLR = fMultDiv2((FIXP_PCM)*pInLR, slev) >> 1;
-#if (SAMPLE_BITS == 32)
-          /* left channel */
-          *pOutL = (INT_PCM)SATURATE_LEFT_SHIFT((FX_PCM2FX_DBL((FIXP_PCM)*pInLF)>>2)+tCF+tLR, 2, SAMPLE_BITS);
-          /* right channel */
-          *pOutR = (INT_PCM)SATURATE_LEFT_SHIFT((FX_PCM2FX_DBL((FIXP_PCM)*pInRF)>>2)+tCF+tLR, 2, SAMPLE_BITS);
-#else
-          /* left channel */
-          *pOutL = (INT_PCM)SATURATE_RIGHT_SHIFT((FX_PCM2FX_DBL((FIXP_PCM)*pInLF)>>2)+tCF-tLR, DFRACT_BITS-SAMPLE_BITS-2, SAMPLE_BITS);
-          /* right channel */
-          *pOutR = (INT_PCM)SATURATE_RIGHT_SHIFT((FX_PCM2FX_DBL((FIXP_PCM)*pInRF)>>2)+tCF+tLR, DFRACT_BITS-SAMPLE_BITS-2, SAMPLE_BITS);
-#endif
-          pInLF += inStride;  pInRF += inStride;  pInCF  += inStride;  pInLR  += inStride;
-          pOutL += outStride; pOutR += outStride;
-        }
-      } else {
-        /* 3/1 input:  L' = L + 0.707*C - 0.707*S;  R' = R + 0.707*C + 0.707*S */
-        for (sample = 0; sample < frameSize; sample++)
-        {
-          FIXP_DBL tCF = fMultDiv2((FIXP_PCM)*pInCF, clev) >> 1;
-          FIXP_DBL tLR = fMultDiv2((FIXP_PCM)*pInLR, slev) >> 1;
-#if (SAMPLE_BITS == 32)
-          /* left channel */
-          *pOutL = (INT_PCM)SATURATE_LEFT_SHIFT((FX_PCM2FX_DBL((FIXP_PCM)*pInLF)>>2)+tCF-tLR, 2, SAMPLE_BITS);
-          /* right channel */
-          *pOutR = (INT_PCM)SATURATE_LEFT_SHIFT((FX_PCM2FX_DBL((FIXP_PCM)*pInRF)>>2)+tCF+tLR, 2, SAMPLE_BITS);
-#else
-          /* left channel */
-          *pOutL = (INT_PCM)SATURATE_RIGHT_SHIFT((FX_PCM2FX_DBL((FIXP_PCM)*pInLF)>>2)+tCF-tLR, DFRACT_BITS-SAMPLE_BITS-2, SAMPLE_BITS);
-          /* right channel */
-          *pOutR = (INT_PCM)SATURATE_RIGHT_SHIFT((FX_PCM2FX_DBL((FIXP_PCM)*pInRF)>>2)+tCF+tLR, DFRACT_BITS-SAMPLE_BITS-2, SAMPLE_BITS);
-#endif
-          pInLF += inStride;  pInRF += inStride;  pInCF  += inStride;  pInLR  += inStride;
-          pOutL += outStride; pOutR += outStride;
+      /* Set channel pointer for output. Remove empty cols. */
+      ch = 0;
+      for (outCh=0; outCh < PCM_DMX_MAX_CHANNELS; outCh+=1) {
+        if (outOffsetTable[outCh] != 255) {
+          pOutPcm[ch] = &pPcmBuf[outOffsetTable[outCh]*offset];
+          map[ch++]  = outCh;
         }
       }
-      break;
-
-    /* 2/2 input: not supported!
-    case CH_MODE_2_0_2_0: */
+      FDK_ASSERT(ch == numOutChannels);
 
-    case CH_MODE_3_0_2_0:   /* 5.0ch input */
-    case CH_MODE_3_0_2_1:   /* 5.1ch input */
-      if (useGuidedDownMix) {
-        /* 3/2 input:  L' = L + clev*C + slev*Ls;  R' = R + clev*C + slev*Rs; */
-        for (sample = 0; sample < frameSize; sample++)
-        {
-          FIXP_DBL tCF = fMultDiv2((FIXP_PCM)*pInCF, clev) >> 1;
-          FIXP_DBL tLR = fMultDiv2((FIXP_PCM)*pInLR, slev) >> 1;
-          FIXP_DBL tRR = fMultDiv2((FIXP_PCM)*pInRR, slev) >> 1;
-#if (SAMPLE_BITS == 32)
-          /* left channel */
-          *pOutL = (INT_PCM)SATURATE_LEFT_SHIFT((FX_PCM2FX_DBL((FIXP_PCM)*pInLF)>>2)+tCF+tLR, 2, SAMPLE_BITS);
-          /* right channel */
-          *pOutR = (INT_PCM)SATURATE_LEFT_SHIFT((FX_PCM2FX_DBL((FIXP_PCM)*pInRF)>>2)+tCF+tRR, 2, SAMPLE_BITS);
-#else
-          /* left channel */
-          *pOutL = (INT_PCM)SATURATE_RIGHT_SHIFT((FX_PCM2FX_DBL((FIXP_PCM)*pInLF)>>2)+tCF+tLR, DFRACT_BITS-SAMPLE_BITS-2, SAMPLE_BITS);
-          /* right channel */
-          *pOutR = (INT_PCM)SATURATE_RIGHT_SHIFT((FX_PCM2FX_DBL((FIXP_PCM)*pInRF)>>2)+tCF+tRR, DFRACT_BITS-SAMPLE_BITS-2, SAMPLE_BITS);
-#endif
-          pInLF += inStride;  pInRF += inStride;  pInCF  += inStride;  pInLR  += inStride;  pInRR  += inStride;
-          pOutL += outStride; pOutR += outStride;
+      /* Remove unused rows from factor matrix */
+      for (outCh=0; outCh < numOutChannels; outCh+=1) {
+        if (outCh != map[outCh]) {
+          FDKmemcpy(&mixFactors[outCh], &mixFactors[map[outCh]], PCM_DMX_MAX_CHANNELS*sizeof(FIXP_DMX));
         }
       }
-      else if (mpegMixDownInfo.mixdownAvailable) {
-        /* 3/2 input: L' = (1.707+A)^-1 * [L+0.707*C+A*Ls]; R'= (1.707+A)^-1 * [R+0.707*C+A*Rs]; */
-        FIXP_SGL mtrxMixDwnCoef    = mpegMixDownIdx2Coef[mpegMixDownInfo.matrixMixdownIdx];
-        FIXP_SGL mtrxMixDwnPreFact = mpegMixDownIdx2PreFact[mpegMixDownInfo.matrixMixdownIdx];
-        clev = FX_DBL2FX_SGL(fMult(mtrxMixDwnPreFact, flev /* 0.707 */));
-        flev = mtrxMixDwnPreFact;
-        slev = FX_DBL2FX_SGL(fMult(mtrxMixDwnPreFact, mtrxMixDwnCoef));
-
-        for (sample = 0; sample < frameSize; sample++)
-        {
-          FIXP_DBL tCF = fMultDiv2((FIXP_PCM)*pInCF, clev);
-          FIXP_DBL tLF = fMultDiv2((FIXP_PCM)*pInLF, flev);
-          FIXP_DBL tRF = fMultDiv2((FIXP_PCM)*pInRF, flev);
-          FIXP_DBL tLR = fMultDiv2((FIXP_PCM)*pInLR, slev);
-          FIXP_DBL tRR = fMultDiv2((FIXP_PCM)*pInRR, slev);
-
-#if (SAMPLE_BITS == 32)
-          /* left channel */
-          *pOutL = (INT_PCM)SATURATE_LEFT_SHIFT(tLF+tCF+tLR, 1, SAMPLE_BITS);
-          /* right channel */
-          *pOutR = (INT_PCM)SATURATE_LEFT_SHIFT(tRF+tCF+tRR, 1, SAMPLE_BITS);
-#else
-          /* left channel */
-          *pOutL = (INT_PCM)SATURATE_RIGHT_SHIFT(tLF+tCF+tLR, DFRACT_BITS-SAMPLE_BITS-1, SAMPLE_BITS);
-          /* right channel */
-          *pOutR = (INT_PCM)SATURATE_RIGHT_SHIFT(tRF+tCF+tRR, DFRACT_BITS-SAMPLE_BITS-1, SAMPLE_BITS);
-#endif
+    }
 
-          pInLF += inStride;  pInRF += inStride;  pInCF  += inStride;  pInLR  += inStride;  pInRR  += inStride;
-          pOutL += outStride; pOutR += outStride;
-        }
+    /* Sample processing loop */
+    for (sample = 0; sample < frameSize; sample++)
+    {
+      FIXP_PCM tIn[PCM_DMX_MAX_IN_CHANNELS];
+      FIXP_DBL tOut[PCM_DMX_MAX_OUT_CHANNELS] = { (FIXP_DBL)0 };
+      int inCh, outCh;
+
+      /* Preload all input samples */
+      for (inCh=0; inCh < numInChannels; inCh+=1) {
+        tIn[inCh] = (FIXP_PCM)*pInPcm[inCh];
+        pInPcm[inCh] += inStride;
       }
-      else {
-        /* 3/2 input:  L' = L + 0.707*C - 0.707*Ls - 0.707*Rs;  R' = R + 0.707*C + 0.707*Ls + 0.707*Rs */
-        for (sample = 0; sample < frameSize; sample++)
-        {
-          FIXP_DBL tCF = fMultDiv2((FIXP_PCM)*pInCF, clev) >> 2;
-          FIXP_DBL tLR = fMultDiv2((FIXP_PCM)*pInLR, slev) >> 2;
-          FIXP_DBL tRR = fMultDiv2((FIXP_PCM)*pInRR, slev) >> 2;
-#if (SAMPLE_BITS == 32)
-          /* left channel */
-          *pOutL = (INT_PCM)SATURATE_LEFT_SHIFT((FX_PCM2FX_DBL((FIXP_PCM)*pInLF)>>3)+tCF-tLR-tRR, 3, SAMPLE_BITS);
-          /* right channel */
-          *pOutR = (INT_PCM)SATURATE_LEFT_SHIFT((FX_PCM2FX_DBL((FIXP_PCM)*pInRF)>>3)+tCF+tLR+tRR, 3, SAMPLE_BITS);
+      /* Apply downmix coefficients to input samples and accumulate for output */
+      for (outCh=0; outCh < numOutChannels; outCh+=1) {
+        for (inCh=0; inCh < numInChannels; inCh+=1) {
+          tOut[outCh] += fMult(tIn[inCh], mixFactors[outCh][inCh]);
+        }
+        /* Write sample */
+#if (SAMPLE_BITS == DFRACT_BITS)
+        *pOutPcm[outCh] = (INT_PCM)SATURATE_LEFT_SHIFT(tOut[outCh], dmxScale, SAMPLE_BITS);
 #else
-          /* left channel */
-          *pOutL = (INT_PCM)SATURATE_RIGHT_SHIFT((FX_PCM2FX_DBL((FIXP_PCM)*pInLF)>>3)+tCF-tLR-tRR, DFRACT_BITS-SAMPLE_BITS-3, SAMPLE_BITS);
-          /* right channel */
-          *pOutR = (INT_PCM)SATURATE_RIGHT_SHIFT((FX_PCM2FX_DBL((FIXP_PCM)*pInRF)>>3)+tCF+tLR+tRR, DFRACT_BITS-SAMPLE_BITS-3, SAMPLE_BITS);
+        *pOutPcm[outCh] = (INT_PCM)SATURATE_RIGHT_SHIFT(tOut[outCh], DFRACT_BITS-SAMPLE_BITS-dmxScale, SAMPLE_BITS);
 #endif
-          pInLF += inStride;  pInRF += inStride;  pInCF  += inStride;  pInLR  += inStride;  pInRR  += inStride;
-          pOutL += outStride; pOutR += outStride;
-        }
+        pOutPcm[outCh] += outStride;
       }
-      break;
-
-    default:
-      errorStatus = PCMDMX_INVALID_MODE;
-      break;
     }
 
-    /* SECOND STAGE:
-         If desired create a mono donwmix:
-         Note: Input are always two channels! */
-    if (numOutChannels == 1)
-    {
-      INT_PCM *pOutC;
-      FIXP_SGL mlev;
-
-      if (useGuidedDownMix) mlev = FL2FXCONST_SGL(1.0f); else mlev = flev;
+    /* Update the number of output channels */
+    *nChannels = numOutChannels;
 
-      /* Output of STAGE ONE = Input of STAGE TWO */
-      FDKmemcpy(inOffsetTable, outOffsetTable, PCM_DMX_MAX_CHANNELS*sizeof(UCHAR));
+  } /* - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - */
+  else
+#endif /* PCM_DOWNMIX_ENABLE */
+#ifdef PCM_CHANNEL_EXTENSION_ENABLE
+  if ( numInChannels < numOutChannels )
+  { /* Apply rudimentary upmix */
+    /* Set up channel pointer */
+    UINT     sample;
+    UCHAR    outOffsetTable[PCM_DMX_MAX_CHANNELS];
+
+    /* FIRST STAGE
+         Create a stereo/dual channel signal */
+    if (numInChannels == ONE_CHANNEL)
+    {
+      INT_PCM  *pInPcm[PCM_DMX_MAX_CHANNELS];
+      INT_PCM  *pOutLF, *pOutRF;
 
-      /* Set I/O strides and offsets */
-      inStride  = outStride;          /* output from STAGE ONE */
-      outStride = numOutChannels;     /* final output */
+      /* Set this stages output stride and channel mode: */
+      outStride = (fInterleaved) ? TWO_CHANNEL : 1;
+      outChMode = outChModeTable[TWO_CHANNEL];
 
       /* Get channel description and channel mapping for this
-       * stages number of output channels (always MONO). */
+       * stages number of output channels (always STEREO). */
       getChannelDescription(
-              CH_MODE_1_0_0_0,
+              outChMode,
               channelMapping,
               channelType,
               channelIndices,
               outOffsetTable
              );
+      /* Now there is no way back because we modified the channel configuration! */
 
-      /* Set input channel pointer. */
-      pInLF = &pPcmBuf[inOffsetTable[LEFT_FRONT_CHANNEL]*offset];
-      pInRF = &pPcmBuf[inOffsetTable[RIGHT_FRONT_CHANNEL]*offset];
+      /* Set input channel pointer. The first channel is always at index 0. */
+      pInPcm[CENTER_FRONT_CHANNEL] = &pPcmBuf[(frameSize-1)*inStride];  /* Considering input mapping could lead to a invalid pointer
+                                                                           here if the channel is not declared to be a front channel. */
 
-      /* Set output channel pointer */
-      pOutC = &pPcmBuf[outOffsetTable[CENTER_FRONT_CHANNEL]*offset];
+      /* Set output channel pointer (for this stage). */
+      pOutLF = &pPcmBuf[outOffsetTable[LEFT_FRONT_CHANNEL]*offset+(frameSize-1)*outStride];
+      pOutRF = &pPcmBuf[outOffsetTable[RIGHT_FRONT_CHANNEL]*offset+(frameSize-1)*outStride];
 
-      /* C' = 0.707*L + 0.707*R */
+      /* 1/0 input: */
       for (sample = 0; sample < frameSize; sample++) {
-#if (SAMPLE_BITS == 32)
-        *pOutC =
-          (INT_PCM)SATURATE_LEFT_SHIFT(fMultDiv2((FIXP_PCM)*pInLF,mlev)+fMultDiv2((FIXP_PCM)*pInRF,mlev), 1, SAMPLE_BITS);
-#else
-        *pOutC =
-          (INT_PCM)SATURATE_RIGHT_SHIFT(fMultDiv2((FIXP_PCM)*pInLF,mlev)+fMultDiv2((FIXP_PCM)*pInRF,mlev), DFRACT_BITS-SAMPLE_BITS-1, SAMPLE_BITS);
-#endif
+        /* L' = C;  R' = C; */
+        *pOutLF = *pOutRF = *pInPcm[CENTER_FRONT_CHANNEL];
 
-        pInLF += inStride; pInRF += inStride;
-        pOutC += 1;
+        pInPcm[CENTER_FRONT_CHANNEL] -= inStride;
+        pOutLF -= outStride; pOutRF -= outStride;
       }
-      /* Finished STAGE TWO */
+
+      /* Prepare for next stage: */
+      inStride = outStride;
+      inChMode = outChMode;
+      FDKmemcpy(inOffsetTable, outOffsetTable, PCM_DMX_MAX_CHANNELS*sizeof(UCHAR));
     }
 
-    /* Update the number of output channels */
-    *nChannels = self->numOutputChannels;
+#if (PCM_DMX_MAX_OUT_CHANNELS > 2)
+    /* SECOND STAGE
+         Extend with zero channels to achieved the desired number of output channels. */
+    if (numOutChannels > TWO_CHANNEL)
+    {
+      INT_PCM *pIn[PCM_DMX_MAX_CHANNELS]  = { NULL };
+      INT_PCM *pOut[PCM_DMX_MAX_CHANNELS] = { NULL };
+      AUDIO_CHANNEL_TYPE  inChTypes[PCM_DMX_MAX_CHANNELS];
+      UCHAR    inChIndices[PCM_DMX_MAX_CHANNELS];
+      UCHAR    numChPerGrp[2][PCM_DMX_MAX_CHANNEL_GROUPS];
+      int      nContentCh = 0;  /* Number of channels with content */
+      int      nEmptyCh = 0;    /* Number of channels with content */
+      int      ch, chGrp, isCompatible = 1;
+
+      /* Do not change the signalling which is the channel types and indices.
+         Just reorder and add channels. So first save the input signalling. */
+      FDKmemcpy(inChTypes, channelType, PCM_DMX_MAX_CHANNELS*sizeof(AUDIO_CHANNEL_TYPE));
+      FDKmemcpy(inChIndices, channelIndices, PCM_DMX_MAX_CHANNELS*sizeof(UCHAR));
+
+      /* Set this stages output stride and channel mode: */
+      outStride = (fInterleaved) ? numOutChannels : 1;
+      outChMode = outChModeTable[numOutChannels];
+
+      /* Check if input channel config can be easily mapped to the desired output config. */
+      for (chGrp = 0; chGrp < PCM_DMX_MAX_CHANNEL_GROUPS; chGrp += 1) {
+        numChPerGrp[IN][chGrp] = (inChMode >> (chGrp*4)) & 0xF;
+        numChPerGrp[OUT][chGrp] = (outChMode >> (chGrp*4)) & 0xF;
+
+        if (numChPerGrp[IN][chGrp] > numChPerGrp[OUT][chGrp]) {
+          isCompatible = 0;
+          break;
+        }
+      }
 
+      if ( isCompatible ) {
+        /* Get new channel description and channel
+         * mapping for the desired output channel mode. */
+        getChannelDescription(
+                outChMode,
+                channelMapping,
+                channelType,
+                channelIndices,
+                outOffsetTable
+               );
+        /* If the input config has a back center channel but the output
+           config has not, copy it to left and right (if available). */
+        if (  (numChPerGrp[IN][CH_GROUP_REAR]%2)
+          && !(numChPerGrp[OUT][CH_GROUP_REAR]%2) ) {
+          if (numChPerGrp[IN][CH_GROUP_REAR] == 1) {
+            inOffsetTable[RIGHT_REAR_CHANNEL] = inOffsetTable[LEFT_REAR_CHANNEL];
+          } else if (numChPerGrp[IN][CH_GROUP_REAR] == 3) {
+            inOffsetTable[RIGHT_MULTIPRPS_CHANNEL] = inOffsetTable[LEFT_MULTIPRPS_CHANNEL];
+          }
+        }
+      }
+      else {
+        /* Just copy and extend the original config */
+        FDKmemcpy(outOffsetTable, inOffsetTable, PCM_DMX_MAX_CHANNELS*sizeof(UCHAR));
+      }
+
+      /* Set I/O channel pointer.
+         Note: The following assignment algorithm clears the channel offset tables.
+               Thus they can not be used afterwards. */
+      for (ch = 0; ch < PCM_DMX_MAX_CHANNELS; ch+=1) {
+        if ( (outOffsetTable[ch] < 255)
+          && (inOffsetTable[ch] < 255) )
+        { /* Set I/O pointer: */
+          pIn[nContentCh] = &pPcmBuf[inOffsetTable[ch]*offset+(frameSize-1)*inStride];
+          pOut[nContentCh] = &pPcmBuf[outOffsetTable[ch]*offset+(frameSize-1)*outStride];
+          /* Update signalling */
+          channelType[outOffsetTable[ch]] = inChTypes[inOffsetTable[ch]];
+          channelIndices[outOffsetTable[ch]] = inChIndices[inOffsetTable[ch]];
+          inOffsetTable[ch] = 255;
+          outOffsetTable[ch] = 255;
+          nContentCh += 1;
+        }
+      }
+      if ( isCompatible ) {
+        /* Assign the remaining input channels.
+           This is just a safety appliance. We should never need it. */
+        for (ch = 0; ch < PCM_DMX_MAX_CHANNELS; ch+=1) {
+          if (inOffsetTable[ch] < 255) {
+            int  outCh;
+            for (outCh = 0 ; outCh < PCM_DMX_MAX_CHANNELS; outCh += 1) {
+              if (outOffsetTable[outCh] < 255) {
+                break;
+              }
+            }
+            /* Set I/O pointer: */
+            pIn[nContentCh] = &pPcmBuf[inOffsetTable[ch]*offset+(frameSize-1)*inStride];
+            pOut[nContentCh] = &pPcmBuf[outOffsetTable[outCh]*offset+(frameSize-1)*outStride];
+            /* Update signalling */
+            channelType[outOffsetTable[outCh]] = inChTypes[inOffsetTable[ch]];
+            channelIndices[outOffsetTable[outCh]] = inChIndices[inOffsetTable[ch]];
+            inOffsetTable[ch] = 255;
+            outOffsetTable[outCh] = 255;
+            nContentCh += 1;
+          }
+        }
+        /* Set the remaining output channel pointer */
+        for (ch = 0; ch < PCM_DMX_MAX_CHANNELS; ch+=1) {
+          if (outOffsetTable[ch] < 255) {
+            pOut[nContentCh+nEmptyCh] = &pPcmBuf[outOffsetTable[ch]*offset+(frameSize-1)*outStride];
+            /* Expand output signalling */
+            channelType[outOffsetTable[ch]] = ACT_NONE;
+            channelIndices[outOffsetTable[ch]] = nEmptyCh;
+            outOffsetTable[ch] = 255;
+            nEmptyCh += 1;
+          }
+        }
+      }
+      else {
+        /* Set the remaining output channel pointer */
+        for (ch = nContentCh; ch < numOutChannels; ch+=1) {
+          pOut[ch] = &pPcmBuf[ch*offset+(frameSize-1)*outStride];
+          /* Expand output signalling */
+          channelType[ch] = ACT_NONE;
+          channelIndices[ch] = nEmptyCh;
+          nEmptyCh += 1;
+        }
+      }
+
+      /* First copy the channels that have signal */
+      for (sample = 0; sample < frameSize; sample+=1) {
+        INT_PCM tIn[PCM_DMX_MAX_CHANNELS];
+        /* Read all channel samples */
+        for (ch = 0; ch < nContentCh; ch+=1) {
+          tIn[ch] = *pIn[ch];
+          pIn[ch] -= inStride;
+        }
+        /* Write all channel samples */
+        for (ch = 0; ch < nContentCh; ch+=1) {
+          *pOut[ch] = tIn[ch];
+          pOut[ch] -= outStride;
+        }
+      }
+
+      /* Clear all the other channels */
+      for (sample = 0; sample < frameSize; sample++) {
+        for (ch = nContentCh; ch < numOutChannels; ch+=1) {
+          *pOut[ch] = (INT_PCM)0;
+          pOut[ch] -= outStride;
+        }
+      }
+    }
+#endif  /* if (PCM_DMX_MAX_OUT_CHANNELS > 2) */
+
+    /* update the number of output channels */
+    *nChannels = numOutChannels;
   } /* - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - */
   else
+#endif /* PCM_CHANNEL_EXTENSION_ENABLE */
   if ( numInChannels == numOutChannels )
   {
     /* Don't need to change the channel description here */
@@ -1143,8 +2348,9 @@ PCMDMX_ERROR pcmDmx_ApplyFrame (
     {
     case 2:
       { /* Set up channel pointer */
-        INT_PCM *pInLF, *pInRF, *pOutL, *pOutR;
-        FIXP_SGL flev;
+        INT_PCM  *pInPcm[PCM_DMX_MAX_CHANNELS];
+        INT_PCM  *pOutL, *pOutR;
+        FIXP_DMX  flev;
 
         UINT sample;
         int inStride, outStride, offset;
@@ -1160,41 +2366,41 @@ PCMDMX_ERROR pcmDmx_ApplyFrame (
         }
 
         /* Set input channel pointer */
-        pInLF = &pPcmBuf[inOffsetTable[LEFT_FRONT_CHANNEL]*offset];
-        pInRF = &pPcmBuf[inOffsetTable[RIGHT_FRONT_CHANNEL]*offset];
+        pInPcm[LEFT_FRONT_CHANNEL]  = &pPcmBuf[inOffsetTable[LEFT_FRONT_CHANNEL]*offset];
+        pInPcm[RIGHT_FRONT_CHANNEL] = &pPcmBuf[inOffsetTable[RIGHT_FRONT_CHANNEL]*offset];
 
         /* Set output channel pointer (same as input) */
-        pOutL  =  pInLF;
-        pOutR  =  pInRF;
+        pOutL  =  pInPcm[LEFT_FRONT_CHANNEL];
+        pOutR  =  pInPcm[RIGHT_FRONT_CHANNEL];
 
         /* Set downmix levels: */
-        flev = ATTENUATION_FACTOR_1;    /* 0.707 */
+        flev = FL2FXCONST_DMX(0.70710678f);
         /* 2/0 input: */
         switch (dualChannelMode)
         {
         case CH1_MODE:  /* L' = 0.707 * Ch1;  R' = 0.707 * Ch1 */
           for (sample = 0; sample < frameSize; sample++) {
             *pOutL = *pOutR =
-              (INT_PCM)SATURATE_RIGHT_SHIFT(fMult((FIXP_PCM)*pInLF, flev), DFRACT_BITS-SAMPLE_BITS, SAMPLE_BITS);
+              (INT_PCM)SATURATE_RIGHT_SHIFT(fMult((FIXP_PCM)*pInPcm[LEFT_FRONT_CHANNEL], flev), DFRACT_BITS-SAMPLE_BITS, SAMPLE_BITS);
 
-            pInLF += inStride;
+            pInPcm[LEFT_FRONT_CHANNEL] += inStride;
             pOutL += outStride; pOutR += outStride;
           }
           break;
         case CH2_MODE:  /* L' = 0.707 * Ch2;  R' = 0.707 * Ch2 */
           for (sample = 0; sample < frameSize; sample++) {
             *pOutL = *pOutR =
-              (INT_PCM)SATURATE_RIGHT_SHIFT(fMult((FIXP_PCM)*pInRF, flev), DFRACT_BITS-SAMPLE_BITS, SAMPLE_BITS);
+              (INT_PCM)SATURATE_RIGHT_SHIFT(fMult((FIXP_PCM)*pInPcm[RIGHT_FRONT_CHANNEL], flev), DFRACT_BITS-SAMPLE_BITS, SAMPLE_BITS);
 
-            pInRF += inStride;
+            pInPcm[RIGHT_FRONT_CHANNEL] += inStride;
             pOutL += outStride; pOutR += outStride;
           }
           break;
         case MIXED_MODE:  /* L' = 0.5*Ch1 + 0.5*Ch2;  R' = 0.5*Ch1 + 0.5*Ch2 */
           for (sample = 0; sample < frameSize; sample++) {
-            *pOutL = *pOutR = (*pInLF >> 1) + (*pInRF >> 1);
+            *pOutL = *pOutR = (*pInPcm[LEFT_FRONT_CHANNEL] >> 1) + (*pInPcm[RIGHT_FRONT_CHANNEL] >> 1);
 
-            pInLF += inStride;  pInRF += inStride;
+            pInPcm[LEFT_FRONT_CHANNEL] += inStride;  pInPcm[RIGHT_FRONT_CHANNEL] += inStride;
             pOutL += outStride; pOutR += outStride;
           }
           break;
@@ -1254,23 +2460,37 @@ PCMDMX_ERROR pcmDmx_GetLibInfo( LIB_INFO *info )
   if (i == FDK_MODULE_LAST) {
     return PCMDMX_UNKNOWN;
   }
-  info += i;
 
   /* Add the library info */
-  info->module_id  = FDK_PCMDMX;
-  info->version    = LIB_VERSION(PCMDMX_LIB_VL0, PCMDMX_LIB_VL1, PCMDMX_LIB_VL2);
-  LIB_VERSION_STRING(info);
-  info->build_date = PCMDMX_LIB_BUILD_DATE;
-  info->build_time = PCMDMX_LIB_BUILD_TIME;
-  info->title      = PCMDMX_LIB_TITLE;
+  info[i].module_id  = FDK_PCMDMX;
+  info[i].version    = LIB_VERSION(PCMDMX_LIB_VL0, PCMDMX_LIB_VL1, PCMDMX_LIB_VL2);
+  LIB_VERSION_STRING(info+i);
+  info[i].build_date = PCMDMX_LIB_BUILD_DATE;
+  info[i].build_time = PCMDMX_LIB_BUILD_TIME;
+  info[i].title      = PCMDMX_LIB_TITLE;
 
   /* Set flags */
-  info->flags = 0
+  info[i].flags = 0
+#ifdef PCM_DOWNMIX_ENABLE
       | CAPF_DMX_BLIND   /* At least blind downmixing is possible */
+ #ifdef PCE_METADATA_ENABLE
       | CAPF_DMX_PCE     /* Guided downmix with data from MPEG-2/4 Program Config Elements (PCE). */
+  #ifdef ARIB_MIXDOWN_ENABLE
+      | CAPF_DMX_ARIB    /* PCE guided downmix with slightly different equations and levels. */
+  #endif
+ #endif /* PCE_METADATA_ENABLE */
+ #ifdef DSE_METADATA_ENABLE
       | CAPF_DMX_DVB     /* Guided downmix with data from DVB ancillary data fields. */
+ #endif
+#endif /* PCM_DOWNMIX_ENABLE */
+#ifdef PCM_CHANNEL_EXTENSION_ENABLE
+      | CAPF_DMX_CH_EXP  /* Simple upmixing by dublicating channels or adding zero channels. */
+#endif
       ;
 
+  /* Add lib info for FDK tools (if not yet done). */
+  FDK_toolsGetLibInfo(info);
+
   return PCMDMX_OK;
 }
 
diff --git a/libSBRdec/include/sbrdecoder.h b/libSBRdec/include/sbrdecoder.h
index 13707e0..174fb5c 100644
--- a/libSBRdec/include/sbrdecoder.h
+++ b/libSBRdec/include/sbrdecoder.h
@@ -145,6 +145,8 @@ typedef enum
   SBR_SYSTEM_BITSTREAM_DELAY,          /*!< System: Switch to enable an additional SBR bitstream delay of one frame. */
   SBR_QMF_MODE,                        /*!< Set QMF mode, either complex or low power. */
   SBR_LD_QMF_TIME_ALIGN,               /*!< Set QMF type, either LD-MPS or CLDFB. Relevant for ELD streams only. */
+  SBR_FLUSH_DATA,                      /*!< Set internal state to flush the decoder with the next process call. */
+  SBR_CLEAR_HISTORY,                   /*!< Clear all internal states (delay lines, QMF states, ...). */
   SBR_BS_INTERRUPTION                  /*!< Signal bit stream interruption. Value is ignored. */
 } SBRDEC_PARAM;
 
@@ -308,7 +310,7 @@ SBR_ERROR sbrDecoder_Apply ( HANDLE_SBRDECODER    self,
                              INT_PCM             *timeData,
                              int                 *numChannels,
                              int                 *sampleRate,
-                             const UCHAR          channelMapping[(6)],
+                             const UCHAR          channelMapping[(8)],
                              const int            interleaved,
                              const int            coreDecodedOk,
                              UCHAR               *psDecoded );
@@ -329,6 +331,13 @@ SBR_ERROR sbrDecoder_Close ( HANDLE_SBRDECODER *self );
  */
 INT sbrDecoder_GetLibInfo( LIB_INFO *info );    
 
+/**
+ * \brief       Determine the modules output signal delay in samples.
+ * \param self  SBR decoder handle.
+ * \return      The number of samples signal delay added by the module.
+ */
+UINT sbrDecoder_GetDelay( const HANDLE_SBRDECODER self );
+
 
 #ifdef __cplusplus
 }
diff --git a/libSBRdec/src/env_extr.h b/libSBRdec/src/env_extr.h
index 5db6d3d..ab6b704 100644
--- a/libSBRdec/src/env_extr.h
+++ b/libSBRdec/src/env_extr.h
@@ -179,6 +179,9 @@ typedef FREQ_BAND_DATA *HANDLE_FREQ_BAND_DATA;
 #define SBRDEC_LOW_POWER      16  /* Flag indicating that Low Power QMF mode shall be used. */
 #define SBRDEC_PS_DECODED     32  /* Flag indicating that PS was decoded and rendered. */
 #define SBRDEC_LD_MPS_QMF    512  /* Flag indicating that the LD-MPS QMF shall be used. */
+#define SBRDEC_DOWNSAMPLE   8192  /* Flag indicating that the downsampling mode is used. */
+#define SBRDEC_FLUSH       16384  /* Flag is used to flush all elements in use. */
+#define SBRDEC_FORCE_RESET 32768  /* Flag is used to force a reset of all elements in use. */
 
 #define SBRDEC_HDR_STAT_RESET  1
 #define SBRDEC_HDR_STAT_UPDATE 2
diff --git a/libSBRdec/src/sbr_ram.cpp b/libSBRdec/src/sbr_ram.cpp
index 6ae941f..c1c2499 100644
--- a/libSBRdec/src/sbr_ram.cpp
+++ b/libSBRdec/src/sbr_ram.cpp
@@ -107,19 +107,19 @@ amm-info@iis.fraunhofer.de
 /*! SBR Decoder main structure */
 C_ALLOC_MEM(Ram_SbrDecoder, struct SBR_DECODER_INSTANCE, 1)
 /*! SBR Decoder element data  <br>
-  Dimension: (4) */
-C_ALLOC_MEM2(Ram_SbrDecElement, SBR_DECODER_ELEMENT, 1, (4))
+  Dimension: (8) */
+C_ALLOC_MEM2(Ram_SbrDecElement, SBR_DECODER_ELEMENT, 1, (8))
 /*! SBR Decoder individual channel data  <br>
-  Dimension: (6) */
-C_ALLOC_MEM2(Ram_SbrDecChannel, SBR_CHANNEL, 1, (6)+1)
+  Dimension: (8) */
+C_ALLOC_MEM2(Ram_SbrDecChannel, SBR_CHANNEL, 1, (8)+1)
 
 /*! Filter states for QMF-synthesis.  <br>
-  Dimension: #(6) * (#QMF_FILTER_STATE_SYN_SIZE-#(64)) */
-C_AALLOC_MEM2_L(Ram_sbr_QmfStatesSynthesis, FIXP_QSS, (640)-(64), (6)+1, SECT_DATA_L1)
+  Dimension: #(8) * (#QMF_FILTER_STATE_SYN_SIZE-#(64)) */
+C_AALLOC_MEM2_L(Ram_sbr_QmfStatesSynthesis, FIXP_QSS, (640)-(64), (8)+1, SECT_DATA_L1)
 
 /*! Delayed spectral data needed for the dynamic framing of SBR.
   For mp3PRO, 1/3 of a frame is buffered (#(6) 6) */
-C_AALLOC_MEM2(Ram_sbr_OverlapBuffer, FIXP_DBL, 2 * (6) * (64), (6)+1)
+C_AALLOC_MEM2(Ram_sbr_OverlapBuffer, FIXP_DBL, 2 * (6) * (64), (8)+1)
 
 /*! Static Data of PS */
 
diff --git a/libSBRdec/src/sbr_ram.h b/libSBRdec/src/sbr_ram.h
index 8fc2dae..f12631d 100644
--- a/libSBRdec/src/sbr_ram.h
+++ b/libSBRdec/src/sbr_ram.h
@@ -118,8 +118,8 @@ typedef struct
 
 struct SBR_DECODER_INSTANCE
 {
-  SBR_DECODER_ELEMENT  *pSbrElement[(4)];
-  SBR_HEADER_DATA       sbrHeader[(4)][(1)+1];      /* Sbr header for each individual channel of an element */
+  SBR_DECODER_ELEMENT  *pSbrElement[(8)];
+  SBR_HEADER_DATA       sbrHeader[(8)][(1)+1];      /* Sbr header for each individual channel of an element */
 
   FIXP_DBL *workBuffer1;
   FIXP_DBL *workBuffer2;
diff --git a/libSBRdec/src/sbrdec_drc.h b/libSBRdec/src/sbrdec_drc.h
index 872c6a0..7eed53a 100644
--- a/libSBRdec/src/sbrdec_drc.h
+++ b/libSBRdec/src/sbrdec_drc.h
@@ -95,7 +95,7 @@ amm-info@iis.fraunhofer.de
 
 
 
-#define SBRDEC_MAX_DRC_CHANNELS  (6)
+#define SBRDEC_MAX_DRC_CHANNELS  (8)
 #define SBRDEC_MAX_DRC_BANDS     ( 16 )
 
 typedef struct
diff --git a/libSBRdec/src/sbrdecoder.cpp b/libSBRdec/src/sbrdecoder.cpp
index 26b2ea2..16b0bbc 100644
--- a/libSBRdec/src/sbrdecoder.cpp
+++ b/libSBRdec/src/sbrdecoder.cpp
@@ -137,7 +137,7 @@ amm-info@iis.fraunhofer.de
 /* Decoder library info */
 #define SBRDECODER_LIB_VL0 2
 #define SBRDECODER_LIB_VL1 2
-#define SBRDECODER_LIB_VL2 3
+#define SBRDECODER_LIB_VL2 6
 #define SBRDECODER_LIB_TITLE "SBR Decoder"
 #define SBRDECODER_LIB_BUILD_DATE __DATE__
 #define SBRDECODER_LIB_BUILD_TIME __TIME__
@@ -251,8 +251,10 @@ SBR_ERROR sbrDecoder_ResetElement (
 
   if ( sampleRateIn == sampleRateOut ) {
     synDownsampleFac = 2;
+    self->flags |=  SBRDEC_DOWNSAMPLE;
   } else {
     synDownsampleFac = 1;
+    self->flags &= ~SBRDEC_DOWNSAMPLE;
   }
 
   self->synDownsampleFac = synDownsampleFac;
@@ -428,7 +430,7 @@ SBR_ERROR sbrDecoder_InitElement (
   int nSbrElementsStart = self->numSbrElements;
 
   /* Check core codec AOT */
-  if (! sbrDecoder_isCoreCodecValid(coreCodec) || elementIndex >= (4)) {
+  if (! sbrDecoder_isCoreCodecValid(coreCodec) || elementIndex >= (8)) {
     sbrError = SBRDEC_UNSUPPORTED_CONFIG;
     goto bail;
   }
@@ -444,6 +446,7 @@ SBR_ERROR sbrDecoder_InitElement (
      && self->coreCodec == coreCodec
      && self->pSbrElement[elementIndex] != NULL
      && self->pSbrElement[elementIndex]->elementID == elementID
+     && !(self->flags & SBRDEC_FORCE_RESET)
      )
   {
      /* Nothing to do */
@@ -550,8 +553,9 @@ bail:
     if (nSbrElementsStart < self->numSbrElements) {
       /* Free the memory allocated for this element */
       sbrDecoder_DestroyElement( self, elementIndex );
-    } else if (self->pSbrElement[elementIndex] != NULL) {
-      /* Set error flag to trigger concealment */
+    } else if ( (self->pSbrElement[elementIndex] != NULL)
+             && (elementIndex < (8)))
+    { /* Set error flag to trigger concealment */
       self->pSbrElement[elementIndex]->frameErrorFlag[self->pSbrElement[elementIndex]->useFrameSlot] = 1;
     }
   }
@@ -615,7 +619,7 @@ INT sbrDecoder_Header (
   SBR_ERROR sbrError = SBRDEC_OK;
   int headerIndex;
 
-  if ( self == NULL || elementIndex > (4) )
+  if ( self == NULL || elementIndex > (8) )
   {
     return SBRDEC_UNSUPPORTED_CONFIG;
   }
@@ -728,6 +732,24 @@ SBR_ERROR sbrDecoder_SetParam (HANDLE_SBRDECODER   self,
       }
     }
     break;
+  case SBR_FLUSH_DATA:
+    if (value != 0) {
+      if (self == NULL) {
+        errorStatus = SBRDEC_NOT_INITIALIZED;
+      } else {
+        self->flags |= SBRDEC_FLUSH;
+      }
+    }
+    break;
+  case SBR_CLEAR_HISTORY:
+    if (value != 0) {
+      if (self == NULL) {
+        errorStatus = SBRDEC_NOT_INITIALIZED;
+      } else {
+        self->flags |= SBRDEC_FORCE_RESET;
+      }
+    }
+    break;
   case SBR_BS_INTERRUPTION:
     {
       int elementIndex;
@@ -738,7 +760,8 @@ SBR_ERROR sbrDecoder_SetParam (HANDLE_SBRDECODER   self,
       }
 
       /* Loop over SBR elements */
-      for (elementIndex = 0; elementIndex < self->numSbrElements; elementIndex++)
+      for (elementIndex = 0; elementIndex < self->numSbrElements; elementIndex++) {
+      if (self->pSbrElement[elementIndex] != NULL)
       {
         HANDLE_SBR_HEADER_DATA hSbrHeader;
         int headerIndex = getHeaderSlot(self->pSbrElement[elementIndex]->useFrameSlot,
@@ -750,7 +773,7 @@ SBR_ERROR sbrDecoder_SetParam (HANDLE_SBRDECODER   self,
            This switches off bitstream parsing until a new header arrives. */
         hSbrHeader->syncState = UPSAMPLING;
         hSbrHeader->status   |= SBRDEC_HDR_STAT_UPDATE;
-      }
+      } }
     }
     break;
   default:
@@ -767,7 +790,7 @@ SBRDEC_DRC_CHANNEL * sbrDecoder_drcGetChannel( const HANDLE_SBRDECODER self, con
   SBRDEC_DRC_CHANNEL *pSbrDrcChannelData = NULL;
   int elementIndex, elChanIdx=0, numCh=0;
 
-  for (elementIndex = 0; (elementIndex < (4)) && (numCh <= channel); elementIndex++)
+  for (elementIndex = 0; (elementIndex < (8)) && (numCh <= channel); elementIndex++)
   {
     SBR_DECODER_ELEMENT *pSbrElement = self->pSbrElement[elementIndex];
     int c, elChannels;
@@ -829,7 +852,7 @@ SBR_ERROR sbrDecoder_drcFeedChannel ( HANDLE_SBRDECODER  self,
   if (self == NULL) {
     return SBRDEC_NOT_INITIALIZED;
   }
-  if (ch > (6) || pNextFact_mag == NULL) {
+  if (ch > (8) || pNextFact_mag == NULL) {
     return SBRDEC_SET_PARAM_FAIL;
   }
 
@@ -874,7 +897,7 @@ void sbrDecoder_drcDisable ( HANDLE_SBRDECODER  self,
   SBRDEC_DRC_CHANNEL *pSbrDrcChannelData = NULL;
 
   if ( (self == NULL)
-    || (ch > (6))
+    || (ch > (8))
     || (self->numSbrElements == 0)
     || (self->numSbrChannels == 0) ) {
     return;
@@ -1119,6 +1142,10 @@ SBR_ERROR sbrDecoder_Parse(
         }
       }
     }
+  } else {
+    /* The returned bit count will not be the actual payload size since we did not
+       parse the frame data. Return an error so that the caller can react respectively. */
+    errorStatus = SBRDEC_PARSE_ERROR;
   }
 
   if (!fDoDecodeSbrData) {
@@ -1198,6 +1225,15 @@ sbrDecoder_DecodeElement (
   int  stereo = (hSbrElement->elementID == ID_CPE) ? 1 : 0;
   int  numElementChannels = hSbrElement->nChannels; /* Number of channels of the current SBR element */
 
+  if (self->flags & SBRDEC_FLUSH) {
+    /* Move frame pointer to the next slot which is up to be decoded/applied next */
+    hSbrElement->useFrameSlot = (hSbrElement->useFrameSlot+1) % (self->numDelayFrames+1);
+    /* Update header and frame data pointer because they have already been set */
+    hSbrHeader = &self->sbrHeader[elementIndex][hSbrElement->useHeaderSlot[hSbrElement->useFrameSlot]];
+    hFrameDataLeft  = &hSbrElement->pSbrChannel[0]->frameData[hSbrElement->useFrameSlot];
+    hFrameDataRight = &hSbrElement->pSbrChannel[1]->frameData[hSbrElement->useFrameSlot];
+  }
+
   /* Update the header error flag */
   hSbrHeader->frameErrorFlag = hSbrElement->frameErrorFlag[hSbrElement->useFrameSlot];
 
@@ -1375,7 +1411,7 @@ SBR_ERROR sbrDecoder_Apply ( HANDLE_SBRDECODER   self,
                              INT_PCM            *timeData,
                              int                *numChannels,
                              int                *sampleRate,
-                             const UCHAR         channelMapping[(6)],
+                             const UCHAR         channelMapping[(8)],
                              const int           interleaved,
                              const int           coreDecodedOk,
                              UCHAR              *psDecoded )
@@ -1472,6 +1508,10 @@ SBR_ERROR sbrDecoder_Apply ( HANDLE_SBRDECODER   self,
 
 
 
+  /* Clear reset and flush flag because everything seems to be done successfully. */
+  self->flags &= ~SBRDEC_FORCE_RESET;
+  self->flags &= ~SBRDEC_FLUSH;
+
 bail:
 
   return errorStatus;
@@ -1496,7 +1536,7 @@ SBR_ERROR sbrDecoder_Close ( HANDLE_SBRDECODER *pSelf )
       FreeRam_SbrDecWorkBuffer2(&self->workBuffer2);
     }
 
-    for (i = 0; i < (4); i++) {
+    for (i = 0; i < (8); i++) {
       sbrDecoder_DestroyElement( self, i );
     }
 
@@ -1544,3 +1584,34 @@ INT sbrDecoder_GetLibInfo( LIB_INFO *info )
   return 0;
 }
 
+
+UINT sbrDecoder_GetDelay( const HANDLE_SBRDECODER self )
+{
+  UINT outputDelay = 0;
+
+  if ( self != NULL) {
+    UINT flags = self->flags;
+
+    /* See chapter 1.6.7.2 of ISO/IEC 14496-3 for the GA-SBR figures below. */
+
+    /* Are we initialized? */
+    if ( (self->numSbrChannels > 0)
+      && (self->numSbrElements > 0) )
+    {
+      /* Add QMF synthesis delay */
+      if ( (flags & SBRDEC_ELD_GRID)
+        && IS_LOWDELAY(self->coreCodec) ) {
+        /* Low delay SBR: */
+        {
+          outputDelay += (flags & SBRDEC_DOWNSAMPLE) ? 32 : 64;   /* QMF synthesis */
+        }
+      }
+      else if (!IS_USAC(self->coreCodec)) {
+        /* By the method of elimination this is the GA (AAC-LC, HE-AAC, ...) branch: */
+        outputDelay += (flags & SBRDEC_DOWNSAMPLE) ? 481 : 962;
+      }
+    }
+  }
+
+  return (outputDelay);
+}
diff --git a/libSYS/include/FDK_audio.h b/libSYS/include/FDK_audio.h
index c8d9c19..0660f4c 100644
--- a/libSYS/include/FDK_audio.h
+++ b/libSYS/include/FDK_audio.h
@@ -232,6 +232,14 @@ typedef enum
 
 } AUDIO_OBJECT_TYPE;
 
+#define IS_USAC(aot) \
+  ((aot) == AOT_USAC \
+|| (aot) == AOT_RSVD50)
+
+#define IS_LOWDELAY(aot) \
+  ((aot) == AOT_ER_AAC_LD \
+|| (aot) == AOT_ER_AAC_ELD)
+
 /** Channel Mode ( 1-7 equals MPEG channel configurations, others are arbitrary). */
 typedef enum {
   MODE_INVALID                  = -1,
@@ -263,17 +271,29 @@ typedef enum {
 
 } CHANNEL_MODE;
 
-/** Speaker description tags */
+/**
+ * Speaker description tags.
+ * Do not change the enumeration values unless it keeps the following segmentation:
+ * - Bit 0-3: Horizontal postion (0: none, 1: front, 2: side, 3: back, 4: lfe)
+ * - Bit 4-7: Vertical position (0: normal, 1: top, 2: bottom)
+ */
 typedef enum {
-  ACT_NONE,
-  ACT_FRONT,
-  ACT_SIDE,
-  ACT_BACK,
-  ACT_LFE,
-  ACT_FRONT_TOP,
-  ACT_SIDE_TOP,
-  ACT_BACK_TOP,
-  ACT_TOP /* Ts */
+  ACT_NONE         = 0x00,
+  ACT_FRONT        = 0x01,  /*!< Front speaker position (at normal height) */
+  ACT_SIDE         = 0x02,  /*!< Side speaker position (at normal height) */
+  ACT_BACK         = 0x03,  /*!< Back speaker position (at normal height) */
+  ACT_LFE          = 0x04,  /*!< Low frequency effect speaker postion (front) */
+
+  ACT_TOP          = 0x10,  /*!< Top speaker area (for combination with speaker positions) */
+  ACT_FRONT_TOP    = 0x11,  /*!< Top front speaker = (ACT_FRONT|ACT_TOP) */
+  ACT_SIDE_TOP     = 0x12,  /*!< Top side speaker  = (ACT_SIDE |ACT_TOP) */
+  ACT_BACK_TOP     = 0x13,  /*!< Top back speaker  = (ACT_BACK |ACT_TOP) */
+
+  ACT_BOTTOM       = 0x20,  /*!< Bottom speaker area (for combination with speaker positions) */
+  ACT_FRONT_BOTTOM = 0x21,  /*!< Bottom front speaker = (ACT_FRONT|ACT_BOTTOM) */
+  ACT_SIDE_BOTTOM  = 0x22,  /*!< Bottom side speaker  = (ACT_SIDE |ACT_BOTTOM) */
+  ACT_BACK_BOTTOM  = 0x23   /*!< Bottom back speaker  = (ACT_BACK |ACT_BOTTOM) */
+
 } AUDIO_CHANNEL_TYPE;
 
 typedef enum
diff --git a/libSYS/src/genericStds.cpp b/libSYS/src/genericStds.cpp
index ef1d4c2..d0b4a90 100644
--- a/libSYS/src/genericStds.cpp
+++ b/libSYS/src/genericStds.cpp
@@ -99,7 +99,7 @@ amm-info@iis.fraunhofer.de
 /* library info */
 #define SYS_LIB_VL0 1
 #define SYS_LIB_VL1 3
-#define SYS_LIB_VL2 4
+#define SYS_LIB_VL2 6
 #define SYS_LIB_TITLE "System Integration Library"
 #define SYS_LIB_BUILD_DATE __DATE__
 #define SYS_LIB_BUILD_TIME __TIME__