[SCM] fdk-aac/master: Imported Upstream version 0.1.3

alessio at users.alioth.debian.org alessio at users.alioth.debian.org
Fri Jul 25 11:44:09 UTC 2014 

The following commit has been merged in the master branch:
commit 4e123ca68297162b1acfd6e9f14eb6efaeb74edb
Author: Alessio Treglia <alessio at debian.org>
Date:   Fri Jul 25 11:15:36 2014 +0100

    Imported Upstream version 0.1.3

diff --git a/.gitignore b/.gitignore
index 387e07b..a882555 100644
--- a/.gitignore
+++ b/.gitignore
@@ -3,6 +3,7 @@
 *.la
 .deps
 .libs
+.dirstamp
 Makefile
 Makefile.in
 aclocal.m4
diff --git a/ChangeLog b/ChangeLog
index b895989..0702fe0 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,3 +1,7 @@
+0.1.3
+ - Updated upstream sources, with a number of crash fixes and new features
+   (including support for encoding 7.1)
+
 0.1.2
  - Fix a few more crashes
  - Include dependency libs (such as -lm) in the pkg-config file
diff --git a/Makefile.am b/Makefile.am
index e995b0a..e179cf9 100644
--- a/Makefile.am
+++ b/Makefile.am
@@ -184,6 +184,7 @@ libfdk_aac_la_SOURCES = \
     $(PCMUTILS_SRC) $(FDK_SRC) $(SYS_SRC)
 
 EXTRA_DIST = \
+    $(top_srcdir)/autogen.sh \
     $(top_srcdir)/NOTICE \
     $(top_srcdir)/Android.mk \
     $(top_srcdir)/fdk-aac.sym \
diff --git a/autogen.sh b/autogen.sh
new file mode 100755
index 0000000..210ccb8
--- /dev/null
+++ b/autogen.sh
@@ -0,0 +1,2 @@
+#!/bin/sh
+autoreconf -fiv
diff --git a/configure.ac b/configure.ac
index 4fd969e..452f74f 100644
--- a/configure.ac
+++ b/configure.ac
@@ -1,7 +1,7 @@
 dnl                                               -*- Autoconf -*-
 dnl Process this file with autoconf to produce a configure script.
 
-AC_INIT([fdk-aac], [0.1.2], [http://sourceforge.net/projects/opencore-amr/])
+AC_INIT([fdk-aac], [0.1.3], [http://sourceforge.net/projects/opencore-amr/])
 AC_CONFIG_AUX_DIR(.)
 AC_CONFIG_MACRO_DIR([m4])
 AM_INIT_AUTOMAKE([tar-ustar foreign])
@@ -26,7 +26,7 @@ AC_CHECK_LIB([m], [sin])
 dnl soname version to use
 dnl goes by ‘current[:revision[:age]]’ with the soname ending up as
 dnl current.age.revision
-FDK_AAC_VERSION=0:3:0
+FDK_AAC_VERSION=0:4:0
 
 AS_IF([test x$enable_shared = xyes], [LIBS_PRIVATE=$LIBS], [LIBS_PUBLIC=$LIBS])
 AC_SUBST(FDK_AAC_VERSION)
diff --git a/documentation/aacEncoder.pdf b/documentation/aacEncoder.pdf
index 84100eb..e8420b1 100644
--- a/documentation/aacEncoder.pdf
+++ b/documentation/aacEncoder.pdf
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-/Next 239 0 R
 >> endobj
 231 0 obj <<
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 >> endobj
 227 0 obj <<
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 >> endobj
 223 0 obj <<
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 >> endobj
 219 0 obj <<
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 >> endobj
 215 0 obj <<
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 >> endobj
 211 0 obj <<
@@ -8342,9 +7810,7 @@ endobj
 /A 209 0 R
 /Parent 203 0 R
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-/First 215 0 R
-/Last 243 0 R
-/Count -8
+/Next 215 0 R
 >> endobj
 207 0 obj <<
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@@ -8355,33 +7821,33 @@ endobj
 203 0 obj <<
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-/Prev 183 0 R
-/Next 247 0 R
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-/Count -2
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 >> endobj
 199 0 obj <<
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 >> endobj
 195 0 obj <<
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 >> endobj
 191 0 obj <<
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-/First 195 0 R
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-/Count -2
 >> endobj
 187 0 obj <<
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@@ -8392,9 +7858,8 @@ endobj
 183 0 obj <<
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@@ -8402,34 +7867,37 @@ endobj
 179 0 obj <<
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-/Prev 175 0 R
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 >> endobj
 175 0 obj <<
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-/Prev 171 0 R
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 >> endobj
 171 0 obj <<
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 >> endobj
 167 0 obj <<
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 >> endobj
 163 0 obj <<
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 >> endobj
 159 0 obj <<
@@ -8437,9 +7905,7 @@ endobj
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-/First 163 0 R
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-/Count -5
+/Next 163 0 R
 >> endobj
 155 0 obj <<
 /Title 156 0 R
@@ -8450,36 +7916,36 @@ endobj
 151 0 obj <<
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-/Next 183 0 R
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-/Count -2
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 >> endobj
 147 0 obj <<
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 >> endobj
 143 0 obj <<
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 >> endobj
 139 0 obj <<
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-/Count -1
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 >> endobj
 135 0 obj <<
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@@ -8489,8 +7955,8 @@ endobj
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@@ -8499,110 +7965,109 @@ endobj
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 >> endobj
 123 0 obj <<
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 >> endobj
 119 0 obj <<
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 >> endobj
 115 0 obj <<
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 >> endobj
 111 0 obj <<
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 >> endobj
 107 0 obj <<
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 >> endobj
 103 0 obj <<
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 >> endobj
 99 0 obj <<
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 >> endobj
 95 0 obj <<
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 >> endobj
 91 0 obj <<
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 >> endobj
 87 0 obj <<
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 >> endobj
 83 0 obj <<
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 >> endobj
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 >> endobj
 75 0 obj <<
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 71 0 obj <<
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 >> endobj
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@@ -8622,10 +8087,10 @@ 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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-/Limits [(ENCODERBEHAVIOUR_BEHAVIOUR_BANDWIDTH) (Item.2)]
->> endobj
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->> endobj
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->> endobj
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-/Limits [(LIBUSE_channelMode) (LIBUSE_encReconf)]
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 >> 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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 >> endobj
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 >> endobj
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 >> endobj
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 trailer
-<< /Size 1417
-/Root 1415 0 R
-/Info 1416 0 R
-/ID [<6568D5E6D3D414DEC0C8E7A6930872CF> <6568D5E6D3D414DEC0C8E7A6930872CF>] >>
+<< /Size 1336
+/Root 1334 0 R
+/Info 1335 0 R
+/ID [<DF0BD4413253B81F550BF5F2F8C5F1D3> <DF0BD4413253B81F550BF5F2F8C5F1D3>] >>
 startxref
-508901
+491779
 %%EOF
diff --git a/libAACdec/include/aacdecoder_lib.h b/libAACdec/include/aacdecoder_lib.h
index 79b4ba1..60efe8d 100644
--- a/libAACdec/include/aacdecoder_lib.h
+++ b/libAACdec/include/aacdecoder_lib.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -504,7 +504,8 @@ typedef struct
   INT               aacSamplesPerFrame;  /*!< Samples per frame for the AAC core (from ASC). \n
                                               1024 or 960 for AAC-LC \n
                                               512 or 480 for AAC-LD and AAC-ELD         */
-
+  INT               aacNumChannels;      /*!< The number of audio channels after AAC core processing (before PS or MPS processing).
+                                              CAUTION: This are not the final number of output channels! */
   AUDIO_OBJECT_TYPE extAot;              /*!< Extension Audio Object Type (from ASC)   */
   INT               extSamplingRate;     /*!< Extension sampling rate in Hz (from ASC) */
 
diff --git a/libAACdec/src/aac_ram.cpp b/libAACdec/src/aac_ram.cpp
index 75c9a94..a38f57c 100644
--- a/libAACdec/src/aac_ram.cpp
+++ b/libAACdec/src/aac_ram.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACdec/src/aac_ram.h b/libAACdec/src/aac_ram.h
index 2130ca1..4527e27 100644
--- a/libAACdec/src/aac_ram.h
+++ b/libAACdec/src/aac_ram.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACdec/src/aac_rom.cpp b/libAACdec/src/aac_rom.cpp
index dd51919..fa04c42 100644
--- a/libAACdec/src/aac_rom.cpp
+++ b/libAACdec/src/aac_rom.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACdec/src/aac_rom.h b/libAACdec/src/aac_rom.h
index beec2ee..536d184 100644
--- a/libAACdec/src/aac_rom.h
+++ b/libAACdec/src/aac_rom.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACdec/src/aacdec_drc.cpp b/libAACdec/src/aacdec_drc.cpp
index ebc6975..2666454 100644
--- a/libAACdec/src/aacdec_drc.cpp
+++ b/libAACdec/src/aacdec_drc.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -135,10 +135,13 @@ void aacDecoder_drcInit (
   /* init params */
   pParams = &self->params;
   pParams->bsDelayEnable = 0;
-  pParams->cut   = FL2FXCONST_DBL(0.0f);
-  pParams->boost = FL2FXCONST_DBL(0.0f);
+  pParams->cut      = FL2FXCONST_DBL(0.0f);
+  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->expiryFrame = AACDEC_DRC_DFLT_EXPIRY_FRAMES;
+  pParams->applyHeavyCompression = 0;
 
   /* initial program ref level = target ref level */
   self->progRefLevel = pParams->targetRefLevel;
@@ -193,7 +196,9 @@ AAC_DECODER_ERROR aacDecoder_drcSetParam (
     if (self == NULL) {
       return AAC_DEC_INVALID_HANDLE;
     }
-    self->params.cut = (FIXP_DBL)((INT)(DRC_PARAM_QUANT_STEP>>DRC_PARAM_SCALE) * (INT)value);
+    self->params.usrCut = (FIXP_DBL)((INT)(DRC_PARAM_QUANT_STEP>>DRC_PARAM_SCALE) * (INT)value);
+    if (self->params.applyHeavyCompression == 0)
+      self->params.cut = self->params.usrCut;
     break;
   case DRC_BOOST_SCALE:
     /* set boost factor */
@@ -204,7 +209,9 @@ AAC_DECODER_ERROR aacDecoder_drcSetParam (
     if (self == NULL) {
       return AAC_DEC_INVALID_HANDLE;
     }
-    self->params.boost = (FIXP_DBL)((INT)(DRC_PARAM_QUANT_STEP>>DRC_PARAM_SCALE) * (INT)value);
+    self->params.usrBoost = (FIXP_DBL)((INT)(DRC_PARAM_QUANT_STEP>>DRC_PARAM_SCALE) * (INT)value);
+    if (self->params.applyHeavyCompression == 0)
+      self->params.boost = self->params.usrBoost;
     break;
   case TARGET_REF_LEVEL:
     if ( value >  MAX_REFERENCE_LEVEL
@@ -220,9 +227,11 @@ AAC_DECODER_ERROR aacDecoder_drcSetParam (
     else {
       /* ref_level must be between 0 and MAX_REFERENCE_LEVEL, inclusive */
       self->digitalNorm    = 1;
-      self->params.targetRefLevel = value;
-      self->progRefLevel = (SCHAR)value;  /* Set the program reference level equal to the target
-                                               level according to 4.5.2.7.3 of ISO/IEC 14496-3. */
+      if (self->params.targetRefLevel != (SCHAR)value) {
+        self->params.targetRefLevel = (SCHAR)value;
+        self->progRefLevel = (SCHAR)value;  /* Always set the program reference level equal to the
+                                               target level according to 4.5.2.7.3 of ISO/IEC 14496-3. */
+      }
     }
     break;
   case APPLY_HEAVY_COMPRESSION:
@@ -232,7 +241,19 @@ AAC_DECODER_ERROR aacDecoder_drcSetParam (
     if (self == NULL) {
       return AAC_DEC_INVALID_HANDLE;
     }
-    self->params.applyHeavyCompression = (UCHAR)value;
+    if (self->params.applyHeavyCompression != (UCHAR)value) {
+      if (value == 1) {
+        /* Disable scaling of DRC values by setting the max values */
+        self->params.boost = FL2FXCONST_DBL(1.0f/(float)(1<<DRC_PARAM_SCALE));
+        self->params.cut   = FL2FXCONST_DBL(1.0f/(float)(1<<DRC_PARAM_SCALE));
+      } else {
+        /* Restore the user params */
+        self->params.boost = self->params.usrBoost;
+        self->params.cut   = self->params.usrCut;
+      }
+      /* Store new parameter value */
+      self->params.applyHeavyCompression = (UCHAR)value;
+    }
     break;
   case DRC_BS_DELAY:
     if (value < 0 || value > 1) {
@@ -473,7 +494,7 @@ static int aacDecoder_drcParse (
     }
   }
   else {
-    pDrcBs->channelData.bandTop[0] = 255;
+    pDrcBs->channelData.bandTop[0] = (1024 >> 2) - 1;  /* ... comprising the whole spectrum. */;
   }
 
   pDrcBs->channelData.numBands = numBands;
@@ -627,10 +648,17 @@ static int aacDecoder_drcExtractAndMap (
 {
   CDrcPayload  threadBs[MAX_DRC_THREADS];
   CDrcPayload *validThreadBs[MAX_DRC_THREADS];
+  CDrcParams  *pParams;
   UINT backupBsPosition;
   int  i, thread, validThreads = 0;
   int  numExcludedChns[MAX_DRC_THREADS];
 
+  FDK_ASSERT(self != NULL);
+  FDK_ASSERT(hBs != NULL);
+  FDK_ASSERT(pAacDecoderStaticChannelInfo != NULL);
+
+  pParams = &self->params;
+
   self->numThreads = 0;
   backupBsPosition = FDKgetValidBits(hBs);
 
@@ -752,6 +780,7 @@ static int aacDecoder_drcExtractAndMap (
      */
     if (pThreadBs->progRefLevel >= 0) {
       self->progRefLevel = pThreadBs->progRefLevel;
+      self->prlExpiryCount = 0;  /* Got a new value -> Reset counter */
     }
 
     /* SCE, CPE and LFE */
@@ -769,6 +798,14 @@ static int aacDecoder_drcExtractAndMap (
     /* CCEs not supported by now */
   }
 
+  /* Increment and check expiry counter for the program reference level: */
+  if ( (pParams->expiryFrame > 0)
+    && (self->prlExpiryCount++ > pParams->expiryFrame) )
+  { /* The program reference level is too old, so set it back to the target level. */
+    self->progRefLevel = pParams->targetRefLevel;
+    self->prlExpiryCount = 0;
+  }
+
   return 0;
 }
 
diff --git a/libAACdec/src/aacdec_drc.h b/libAACdec/src/aacdec_drc.h
index 9c90e32..2ebae2c 100644
--- a/libAACdec/src/aacdec_drc.h
+++ b/libAACdec/src/aacdec_drc.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -143,6 +143,16 @@ int aacDecoder_drcProlog (
         UCHAR  channelMapping[],
         int    numChannels );
 
+/**
+ * \brief Apply DRC. If SBR is present, DRC data is handed over to the SBR decoder.
+ * \param self AAC decoder instance
+ * \param pSbrDec pointer to SBR decoder instance
+ * \param pAacDecoderChannelInfo AAC decoder channel instance to be processed
+ * \param pDrcDat DRC channel data
+ * \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
+ */
 void aacDecoder_drcApply (
         HANDLE_AAC_DRC          self,
         void                   *pSbrDec,
diff --git a/libAACdec/src/aacdec_drc_types.h b/libAACdec/src/aacdec_drc_types.h
index 3871b4b..1b5cd76 100644
--- a/libAACdec/src/aacdec_drc_types.h
+++ b/libAACdec/src/aacdec_drc_types.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -132,8 +132,10 @@ typedef struct
 
 typedef struct
 {
-  FIXP_DBL  cut;
-  FIXP_DBL  boost;
+  FIXP_DBL  cut;       /* The attenuation scale factor currently used.     */
+  FIXP_DBL  usrCut;    /* The latest attenuation scale factor set by user. */
+  FIXP_DBL  boost;     /* The boost scale factor currently used.           */
+  FIXP_DBL  usrBoost;  /* The latest boost scale factor set by user.       */
 
   UINT   expiryFrame;
   SCHAR  targetRefLevel;
@@ -154,6 +156,8 @@ typedef struct
   USHORT numThreads;                         /* The number of DRC data threads extracted from the found payload elements */
   SCHAR  progRefLevel;                       /* Program reference level for all channels */
 
+  UINT   prlExpiryCount;                     /* Counter that can be used to monitor the life time of the program reference level. */
+
   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/aacdec_hcr.cpp b/libAACdec/src/aacdec_hcr.cpp
index 5b86833..e314e27 100644
--- a/libAACdec/src/aacdec_hcr.cpp
+++ b/libAACdec/src/aacdec_hcr.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACdec/src/aacdec_hcr.h b/libAACdec/src/aacdec_hcr.h
index f8f3abf..6fc527b 100644
--- a/libAACdec/src/aacdec_hcr.h
+++ b/libAACdec/src/aacdec_hcr.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACdec/src/aacdec_hcr_bit.cpp b/libAACdec/src/aacdec_hcr_bit.cpp
index ab82ca3..df2685b 100644
--- a/libAACdec/src/aacdec_hcr_bit.cpp
+++ b/libAACdec/src/aacdec_hcr_bit.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACdec/src/aacdec_hcr_bit.h b/libAACdec/src/aacdec_hcr_bit.h
index c191b1d..8994ff1 100644
--- a/libAACdec/src/aacdec_hcr_bit.h
+++ b/libAACdec/src/aacdec_hcr_bit.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACdec/src/aacdec_hcr_types.h b/libAACdec/src/aacdec_hcr_types.h
index 54b3dbd..323ec4e 100644
--- a/libAACdec/src/aacdec_hcr_types.h
+++ b/libAACdec/src/aacdec_hcr_types.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACdec/src/aacdec_hcrs.cpp b/libAACdec/src/aacdec_hcrs.cpp
index 74ecb65..c0b2173 100644
--- a/libAACdec/src/aacdec_hcrs.cpp
+++ b/libAACdec/src/aacdec_hcrs.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACdec/src/aacdec_hcrs.h b/libAACdec/src/aacdec_hcrs.h
index 7a004c4..678ba26 100644
--- a/libAACdec/src/aacdec_hcrs.h
+++ b/libAACdec/src/aacdec_hcrs.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACdec/src/aacdec_pns.cpp b/libAACdec/src/aacdec_pns.cpp
index a3f7a78..541ef07 100644
--- a/libAACdec/src/aacdec_pns.cpp
+++ b/libAACdec/src/aacdec_pns.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACdec/src/aacdec_pns.h b/libAACdec/src/aacdec_pns.h
index c07656c..fc9bdcb 100644
--- a/libAACdec/src/aacdec_pns.h
+++ b/libAACdec/src/aacdec_pns.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACdec/src/aacdec_tns.cpp b/libAACdec/src/aacdec_tns.cpp
index 702196f..352f04a 100644
--- a/libAACdec/src/aacdec_tns.cpp
+++ b/libAACdec/src/aacdec_tns.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACdec/src/aacdec_tns.h b/libAACdec/src/aacdec_tns.h
index 499f763..f029d96 100644
--- a/libAACdec/src/aacdec_tns.h
+++ b/libAACdec/src/aacdec_tns.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACdec/src/aacdecoder.cpp b/libAACdec/src/aacdecoder.cpp
index 8d3c18d..3a2a561 100644
--- a/libAACdec/src/aacdecoder.cpp
+++ b/libAACdec/src/aacdecoder.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -183,7 +183,7 @@ void CAacDecoder_SyncQmfMode(HANDLE_AACDECODER self)
   if ( self->qmfModeCurr == NOT_DEFINED )
   {
     if ( (IS_LOWDELAY(self->streamInfo.aot) && (self->flags & AC_MPS_PRESENT))
-      || ( (self->ascChannels == 1)
+      || ( (self->streamInfo.aacNumChannels == 1)
         && ( (CAN_DO_PS(self->streamInfo.aot) && !(self->flags & AC_MPS_PRESENT))
           || (  IS_USAC(self->streamInfo.aot) &&  (self->flags & AC_MPS_PRESENT)) ) ) )
     {
@@ -196,7 +196,7 @@ void CAacDecoder_SyncQmfMode(HANDLE_AACDECODER self)
 
   /* Set SBR to current QMF mode. Error does not matter. */
   sbrDecoder_SetParam(self->hSbrDecoder, SBR_QMF_MODE, (self->qmfModeCurr == MODE_LP));
-  self->psPossible = ((CAN_DO_PS(self->streamInfo.aot) && self->aacChannels == 1 && ! (self->flags & AC_MPS_PRESENT))) && self->qmfModeCurr == MODE_HQ ;
+  self->psPossible = ((CAN_DO_PS(self->streamInfo.aot) && self->streamInfo.aacNumChannels == 1 && ! (self->flags & AC_MPS_PRESENT))) && self->qmfModeCurr == MODE_HQ ;
   FDK_ASSERT( ! ( (self->flags & AC_MPS_PRESENT) && self->psPossible ) );
 }
 
@@ -373,7 +373,7 @@ static AAC_DECODER_ERROR CDataStreamElement_Read (
 
   {
     INT readBits, dataBits = count<<3;
-    
+
     /* Move to the beginning of the data junk */
     FDKpushBack(bs, dataStart-FDKgetValidBits(bs));
 
@@ -394,23 +394,26 @@ static AAC_DECODER_ERROR CDataStreamElement_Read (
   \brief Read Program Config Element
 
   \bs Bitstream Handle
-  \count Pointer to program config element.
+  \pTp Transport decoder handle for CRC handling
+  \pce Pointer to PCE buffer
+  \channelConfig Current channel configuration
+  \alignAnchor Anchor for byte alignment
 
-  \return  Error code
+  \return  PCE status (-1: fail, 0: no new PCE, 1: PCE updated, 2: PCE updated need re-config).
 */
-static AAC_DECODER_ERROR CProgramConfigElement_Read (
+static int CProgramConfigElement_Read (
     HANDLE_FDK_BITSTREAM bs,
     HANDLE_TRANSPORTDEC  pTp,
     CProgramConfig      *pce,
-    UINT                 channelConfig,
-    UINT                 alignAnchor )
+    const UINT           channelConfig,
+    const UINT           alignAnchor )
 {
-  AAC_DECODER_ERROR error = AAC_DEC_OK;
+  int pceStatus = 0;
   int crcReg;
 
   /* read PCE to temporal buffer first */
   C_ALLOC_SCRATCH_START(tmpPce, CProgramConfig, 1);
-  
+
   CProgramConfig_Init(tmpPce);
   CProgramConfig_Reset(tmpPce);
 
@@ -421,22 +424,43 @@ static AAC_DECODER_ERROR CProgramConfigElement_Read (
   transportDec_CrcEndReg(pTp, crcReg);
 
   if (  CProgramConfig_IsValid(tmpPce)
-    && ( (channelConfig == 6 && (tmpPce->NumChannels == 6))
-      || (channelConfig == 5 && (tmpPce->NumChannels == 5))
-      || (channelConfig == 0 && (tmpPce->NumChannels == pce->NumChannels)) )
-    && (tmpPce->NumFrontChannelElements == 2)
-    && (tmpPce->NumSideChannelElements  == 0)
-    && (tmpPce->NumBackChannelElements  == 1)
     && (tmpPce->Profile == 1) )
-  { /* Copy the complete PCE including metadata. */
-    FDKmemcpy(pce, tmpPce, sizeof(CProgramConfig));
+  {
+    if ( !pce->isValid && (channelConfig > 0) ) {
+      /* Create a standard channel config PCE to compare with */
+      CProgramConfig_GetDefault( pce, channelConfig );
+    }
+
+    if (pce->isValid) {
+      /* Compare the new and the old PCE (tags ignored) */
+      switch ( CProgramConfig_Compare( pce, tmpPce ) )
+      {
+      case 1:  /* Channel configuration not changed. Just new metadata. */
+        FDKmemcpy(pce, tmpPce, sizeof(CProgramConfig));    /* Store the complete PCE */
+        pceStatus = 1;                                     /* New PCE but no change of config */
+        break;
+      case 2:  /* The number of channels are identical but not the config */
+        if (channelConfig == 0) {
+          FDKmemcpy(pce, tmpPce, sizeof(CProgramConfig));  /* Store the complete PCE */
+          pceStatus = 2;                                   /* Decoder needs re-configuration */
+        }
+        break;
+      case -1:  /* The channel configuration is completely different */
+        pceStatus = -1;  /* Not supported! */
+        break;
+      case 0:  /* Nothing to do because PCE matches the old one exactly. */
+      default:
+        /* pceStatus = 0; */
+        break;
+      }
+    }
   }
 
   C_ALLOC_SCRATCH_END(tmpPce, CProgramConfig, 1);
 
-  return error;
+  return pceStatus;
 }
-#endif
+#endif /* TP_PCE_ENABLE */
 
 /*!
   \brief Parse Extension Payload
@@ -591,7 +615,7 @@ AAC_DECODER_ERROR CAacDecoder_ExtPayloadParse (HANDLE_AACDECODER self,
     {                             /* ... created to circumvent the missing length in ER-Syntax. */
       int bitCnt, len = FDKreadBits(hBs, 4);
       *count -= 4;
-      
+
       if (len == 15) {
         int add_len = FDKreadBits(hBs, 8);
         *count -= 8;
@@ -609,9 +633,7 @@ AAC_DECODER_ERROR CAacDecoder_ExtPayloadParse (HANDLE_AACDECODER self,
         /* Check NOTE 2: The extension_payload() included here must
                          not have extension_type == EXT_DATA_LENGTH. */
         error = AAC_DEC_PARSE_ERROR;
-        goto bail;
-      }
-      else {
+      } else {
         /* rewind and call myself again. */
         FDKpushBack(hBs, 4);
 
@@ -622,7 +644,7 @@ AAC_DECODER_ERROR CAacDecoder_ExtPayloadParse (HANDLE_AACDECODER self,
                  &bitCnt,
                   previous_element,
                   elIndex,
-                  1 );      /* Treat same as fill element */
+                  0 );
 
         *count -= len - bitCnt;
       }
@@ -754,8 +776,12 @@ LINKSPEC_CPP void CAacDecoder_Close(HANDLE_AACDECODER self)
 
   for (ch=0; ch<(6); ch++) {
     if (self->pAacDecoderStaticChannelInfo[ch] != NULL) {
-      FreeOverlapBuffer (&self->pAacDecoderStaticChannelInfo[ch]->pOverlapBuffer);
-      FreeAacDecoderStaticChannelInfo (&self->pAacDecoderStaticChannelInfo[ch]);
+      if (self->pAacDecoderStaticChannelInfo[ch]->pOverlapBuffer != NULL) {
+        FreeOverlapBuffer (&self->pAacDecoderStaticChannelInfo[ch]->pOverlapBuffer);
+      }
+      if (self->pAacDecoderStaticChannelInfo[ch] != NULL) {
+        FreeAacDecoderStaticChannelInfo (&self->pAacDecoderStaticChannelInfo[ch]);
+      }
     }
     if (self->pAacDecoderChannelInfo[ch] != NULL) {
       FreeAacDecoderChannelInfo (&self->pAacDecoderChannelInfo[ch]);
@@ -768,8 +794,12 @@ LINKSPEC_CPP void CAacDecoder_Close(HANDLE_AACDECODER self)
     FreeDrcInfo(&self->hDrcInfo);
   }
 
-  FreeWorkBufferCore1 (&self->aacCommonData.workBufferCore1);
-  FreeWorkBufferCore2 (&self->aacCommonData.workBufferCore2);
+  if (self->aacCommonData.workBufferCore1 != NULL) {
+    FreeWorkBufferCore1 (&self->aacCommonData.workBufferCore1);
+  }
+  if (self->aacCommonData.workBufferCore2 != NULL) {
+    FreeWorkBufferCore2 (&self->aacCommonData.workBufferCore2);
+  }
 
   FreeAacDecoder ( &self);
 }
@@ -994,12 +1024,14 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_Init(HANDLE_AACDECODER self, const CS
          CPns_InitPns(&self->pAacDecoderChannelInfo[ch]->data.aac.PnsData, &self->aacCommonData.pnsInterChannelData, &self->aacCommonData.pnsCurrentSeed, self->aacCommonData.pnsRandomSeed);
        }
 
+       if (ascChannels > self->aacChannels)
+       {
+         /* Make allocated channel count persistent in decoder context. */
+         self->aacChannels = ascChannels;
+       }
 
        HcrInitRom(&self->aacCommonData.overlay.aac.erHcrInfo);
        setHcrType(&self->aacCommonData.overlay.aac.erHcrInfo, ID_SCE);
-
-       /* Make allocated channel count persistent in decoder context. */
-       self->aacChannels = ascChannels;
     }
 
     /* Make amount of signalled channels persistent in decoder context. */
@@ -1009,8 +1041,10 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_Init(HANDLE_AACDECODER self, const CS
   /* Update structures */
   if (ascChanged) {
 
-     /* Things to be done for each channel, which do not involved allocating memory. */
-     for (ch = 0; ch < ascChannels; ch++) {
+     /* Things to be done for each channel, which do not involve allocating memory.
+        Doing these things only on the channels needed for the current configuration
+        (ascChannels) could lead to memory access violation later (error concealment). */
+     for (ch = 0; ch < self->aacChannels; ch++) {
        switch (self->streamInfo.aot) {
          case AOT_ER_AAC_ELD:
          case AOT_ER_AAC_LD:
@@ -1241,10 +1275,10 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
           else {
             self->frameOK = 0;
           }
-          /* Create SBR element for SBR for upsampling. */
-          if ( (type == ID_LFE)
-            && ( (self->flags & AC_SBR_PRESENT)
-              || (self->sbrEnabled == 1) ) )
+          /* Create SBR element for SBR for upsampling for LFE elements,
+             and if SBR was explicitly signaled, because the first frame(s)
+             may not contain SBR payload (broken encoder, bit errors). */
+          if ( (self->flags & AC_SBR_PRESENT) || (self->sbrEnabled == 1) )
           {
             SBR_ERROR sbrError;
 
@@ -1254,7 +1288,7 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
                     self->streamInfo.extSamplingRate,
                     self->streamInfo.aacSamplesPerFrame,
                     self->streamInfo.aot,
-                    ID_LFE,
+                    type,
                     previous_element_index
                     );
             if (sbrError != SBRDEC_OK) {
@@ -1394,26 +1428,34 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
 
 #ifdef TP_PCE_ENABLE
       case ID_PCE:
-
-        if ( CProgramConfigElement_Read( bs,
+        {
+          int result = CProgramConfigElement_Read(
+                                    bs,
                                     self->hInput,
                                     pce,
                                     self->streamInfo.channelConfig,
-                                    auStartAnchor ) )
-        { /* Built element table */
-          int elIdx = CProgramConfig_GetElementTable(pce, self->elements, 7);
-          /* Reset the remaining tabs */
-          for ( ; elIdx<7; elIdx++) {
-            self->elements[elIdx] = ID_NONE;
-          }
-          /* Make new number of channel persistant */
-          self->ascChannels = pce->NumChannels;
-          /* If PCE is not first element conceal this frame to avoid inconsistencies */
-          if ( element_count != 0 ) {
+                                    auStartAnchor );
+          if ( result < 0 ) {
+            /* Something went wrong */
+            ErrorStatus = AAC_DEC_PARSE_ERROR;
             self->frameOK = 0;
           }
+          else if ( result > 1 ) {
+            /* Built element table */
+            int elIdx = CProgramConfig_GetElementTable(pce, self->elements, 7);
+            /* Reset the remaining tabs */
+            for ( ; elIdx<7; elIdx++) {
+              self->elements[elIdx] = ID_NONE;
+            }
+            /* Make new number of channel persistant */
+            self->ascChannels = pce->NumChannels;
+            /* If PCE is not first element conceal this frame to avoid inconsistencies */
+            if ( element_count != 0 ) {
+              self->frameOK = 0;
+            }
+          }
+          pceRead = (result>=0) ? 1 : 0;
         }
-        pceRead = 1;
         break;
 #endif /* TP_PCE_ENABLE */
 
@@ -1573,7 +1615,7 @@ LINKSPEC_CPP AAC_DECODER_ERROR CAacDecoder_DecodeFrame(
   }
 
   /* Update number of output channels */
-  self->streamInfo.numChannels = aacChannels;
+  self->streamInfo.aacNumChannels = aacChannels;
 
  #ifdef TP_PCE_ENABLE
   if (pceRead == 1 && CProgramConfig_IsValid(pce)) {
diff --git a/libAACdec/src/aacdecoder.h b/libAACdec/src/aacdecoder.h
index f153c84..16351e6 100644
--- a/libAACdec/src/aacdecoder.h
+++ b/libAACdec/src/aacdecoder.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACdec/src/aacdecoder_lib.cpp b/libAACdec/src/aacdecoder_lib.cpp
index a3db39e..ec8f41e 100644
--- a/libAACdec/src/aacdecoder_lib.cpp
+++ b/libAACdec/src/aacdecoder_lib.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -109,8 +109,8 @@ amm-info at iis.fraunhofer.de
 
 /* Decoder library info */
 #define AACDECODER_LIB_VL0 2
-#define AACDECODER_LIB_VL1 4
-#define AACDECODER_LIB_VL2 7
+#define AACDECODER_LIB_VL1 5
+#define AACDECODER_LIB_VL2 5
 #define AACDECODER_LIB_TITLE "AAC Decoder Lib"
 #define AACDECODER_LIB_BUILD_DATE __DATE__
 #define AACDECODER_LIB_BUILD_TIME __TIME__
@@ -261,7 +261,7 @@ setConcealMethod ( const HANDLE_AACDECODER  self,   /*!< Handle of the decoder i
   HANDLE_SBRDECODER hSbrDec = NULL;
   HANDLE_AAC_DRC hDrcInfo = NULL;
   HANDLE_PCM_DOWNMIX hPcmDmx = NULL;
-  CConcealmentMethod backupMethod;
+  CConcealmentMethod backupMethod = ConcealMethodNone;
   int backupDelay = 0;
   int bsDelay = 0;
 
@@ -396,11 +396,15 @@ aacDecoder_SetParam ( const HANDLE_AACDECODER  self,   /*!< Handle of the decode
   AAC_DECODER_ERROR errorStatus = AAC_DEC_OK;
   CConcealParams  *pConcealData = NULL;
   HANDLE_AAC_DRC hDrcInfo = NULL;
+  HANDLE_PCM_DOWNMIX hPcmDmx = NULL;
 
   /* check decoder handle */
   if (self != NULL) {
     pConcealData = &self->concealCommonData;
     hDrcInfo = self->hDrcInfo;
+    hPcmDmx = self->hPcmUtils;
+  } else {
+    errorStatus = AAC_DEC_INVALID_HANDLE;
   }
 
   /* configure the subsystems */
@@ -417,11 +421,14 @@ aacDecoder_SetParam ( const HANDLE_AACDECODER  self,   /*!< Handle of the decode
     break;
 
   case AAC_PCM_OUTPUT_CHANNELS:
+    if (value < -1 || value > (6)) {
+      return AAC_DEC_SET_PARAM_FAIL;
+    }
     {
       PCMDMX_ERROR err;
 
       err = pcmDmx_SetParam (
-              self->hPcmUtils,
+              hPcmDmx,
               NUMBER_OF_OUTPUT_CHANNELS,
               value );
 
@@ -441,7 +448,7 @@ aacDecoder_SetParam ( const HANDLE_AACDECODER  self,   /*!< Handle of the decode
       PCMDMX_ERROR err;
 
       err = pcmDmx_SetParam (
-              self->hPcmUtils,
+              hPcmDmx,
               DUAL_CHANNEL_DOWNMIX_MODE,
               value );
 
@@ -459,10 +466,14 @@ aacDecoder_SetParam ( const HANDLE_AACDECODER  self,   /*!< Handle of the decode
   case AAC_PCM_OUTPUT_CHANNEL_MAPPING:
     switch (value) {
       case 0:
-        self->channelOutputMapping = channelMappingTablePassthrough;
+        if (self != NULL) {
+          self->channelOutputMapping = channelMappingTablePassthrough;
+        }
         break;
       case 1:
-        self->channelOutputMapping = channelMappingTableWAV;
+        if (self != NULL) {
+          self->channelOutputMapping = channelMappingTableWAV;
+        }
         break;
       default:
         errorStatus = AAC_DEC_SET_PARAM_FAIL;
@@ -472,6 +483,9 @@ aacDecoder_SetParam ( const HANDLE_AACDECODER  self,   /*!< Handle of the decode
 
 
   case AAC_QMF_LOWPOWER:
+    if (value < -1 || value > 1) {
+      return AAC_DEC_SET_PARAM_FAIL;
+    }
     if (self == NULL) {
       return AAC_DEC_INVALID_HANDLE;
     }
@@ -794,8 +808,8 @@ LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_DecodeFrame(
       /* Export data into streaminfo structure */
       self->streamInfo.sampleRate = self->streamInfo.aacSampleRate;
       self->streamInfo.frameSize  = self->streamInfo.aacSamplesPerFrame;
-      self->streamInfo.numChannels = self->aacChannels;
     }
+    self->streamInfo.numChannels = self->streamInfo.aacNumChannels;
 
 
 
@@ -832,7 +846,7 @@ LINKSPEC_CPP AAC_DECODER_ERROR aacDecoder_DecodeFrame(
                                     pTimeData,
                                    &self->streamInfo.numChannels,
                                    &self->streamInfo.sampleRate,
-                                    self->channelOutputMapping[self->aacChannels-1],
+                                    self->channelOutputMapping[self->streamInfo.numChannels-1],
                                     interleaved,
                                     self->frameOK,
                                    &self->psPossible);
diff --git a/libAACdec/src/arm/block_arm.cpp b/libAACdec/src/arm/block_arm.cpp
index 4c0fcdd..fbc1bf3 100644
--- a/libAACdec/src/arm/block_arm.cpp
+++ b/libAACdec/src/arm/block_arm.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACdec/src/block.cpp b/libAACdec/src/block.cpp
index 1f82780..9d703cc 100644
--- a/libAACdec/src/block.cpp
+++ b/libAACdec/src/block.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -589,7 +589,6 @@ AAC_DECODER_ERROR  CBlock_ReadSpectralData(HANDLE_FDK_BITSTREAM bs,
   {
     H_HCR_INFO hHcr = &pAacDecoderChannelInfo->pComData->overlay.aac.erHcrInfo;
     int hcrStatus = 0;
-    int hcrConcealWholeFrame = 0;
 
     /* advanced Huffman decoding starts here (HCR decoding :) */
     if ( pAacDecoderChannelInfo->pDynData->specificTo.aac.lenOfReorderedSpectralData != 0 ) {
@@ -598,24 +597,19 @@ AAC_DECODER_ERROR  CBlock_ReadSpectralData(HANDLE_FDK_BITSTREAM bs,
       hcrStatus = HcrInit(hHcr, pAacDecoderChannelInfo, pSamplingRateInfo, bs);
 
       if (hcrStatus != 0) {
-#if HCR_ERROR_CONCEALMENT
-        hcrConcealWholeFrame = 1;
-        return AAC_DEC_DECODE_FRAME_ERROR;  /* concealment is muting in the first step, therefore return now */
-        // hcr decoding is not skipped because of returning above
-#else
         return AAC_DEC_DECODE_FRAME_ERROR;
-#endif
       }
 
       /* HCR decoding short */
       hcrStatus = HcrDecoder(hHcr, pAacDecoderChannelInfo, pSamplingRateInfo, bs);
 
-
+      if (hcrStatus != 0) {
 #if HCR_ERROR_CONCEALMENT
-      HcrMuteErroneousLines(hHcr);
+        HcrMuteErroneousLines(hHcr);
 #else
-      return AAC_DEC_DECODE_FRAME_ERROR;
+        return AAC_DEC_DECODE_FRAME_ERROR;
 #endif /* HCR_ERROR_CONCEALMENT */
+      }
 
       FDKpushFor (bs, pAacDecoderChannelInfo->pDynData->specificTo.aac.lenOfReorderedSpectralData);
     }
diff --git a/libAACdec/src/block.h b/libAACdec/src/block.h
index f97b85c..f9394f6 100644
--- a/libAACdec/src/block.h
+++ b/libAACdec/src/block.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACdec/src/channel.cpp b/libAACdec/src/channel.cpp
index 21517f5..5475079 100644
--- a/libAACdec/src/channel.cpp
+++ b/libAACdec/src/channel.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACdec/src/channel.h b/libAACdec/src/channel.h
index 4c428c8..1146998 100644
--- a/libAACdec/src/channel.h
+++ b/libAACdec/src/channel.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACdec/src/channelinfo.cpp b/libAACdec/src/channelinfo.cpp
index 6e520c0..76d5895 100644
--- a/libAACdec/src/channelinfo.cpp
+++ b/libAACdec/src/channelinfo.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACdec/src/channelinfo.h b/libAACdec/src/channelinfo.h
index 9be07da..e092ab3 100644
--- a/libAACdec/src/channelinfo.h
+++ b/libAACdec/src/channelinfo.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACdec/src/conceal.cpp b/libAACdec/src/conceal.cpp
index dc5d99f..c26051c 100644
--- a/libAACdec/src/conceal.cpp
+++ b/libAACdec/src/conceal.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -441,7 +441,7 @@ AAC_DECODER_ERROR
 
   /* set confort noise level which will be inserted while in state 'muting' */
   if (comfNoiseLevel != AACDEC_CONCEAL_PARAM_NOT_SPECIFIED) {
-    if ( (comfNoiseLevel < 0) 
+    if ( (comfNoiseLevel < -1)
       || (comfNoiseLevel > 127) ) {
       return AAC_DEC_SET_PARAM_FAIL;
     }
@@ -1527,8 +1527,13 @@ static void
       {
       case ConcealState_Ok:
         if (!frameOk) {
-          /* change to state SINGLE-FRAME-LOSS */
-          pConcealmentInfo->concealState   = ConcealState_Single;
+          if (pConcealCommonData->numFadeOutFrames > 0) {
+            /* change to state SINGLE-FRAME-LOSS */
+            pConcealmentInfo->concealState   = ConcealState_Single;
+          } else {
+            /* change to state MUTE */
+            pConcealmentInfo->concealState = ConcealState_Mute;
+          }
           pConcealmentInfo->cntFadeFrames  = 0;
           pConcealmentInfo->cntValidFrames = 0;
         }
@@ -1561,11 +1566,16 @@ static void
       case ConcealState_FadeOut:
         pConcealmentInfo->cntFadeFrames += 1;  /* used to address the fade-out factors */
         if (pConcealmentInfo->cntValidFrames > pConcealCommonData->numMuteReleaseFrames) {
-          /* change to state FADE-IN */
-          pConcealmentInfo->concealState  = ConcealState_FadeIn;
-          pConcealmentInfo->cntFadeFrames = findEquiFadeFrame( pConcealCommonData,
-                                                               pConcealmentInfo->cntFadeFrames-1,
-                                                               0 /* FadeOut -> FadeIn */);
+          if (pConcealCommonData->numFadeInFrames > 0) {
+            /* change to state FADE-IN */
+            pConcealmentInfo->concealState  = ConcealState_FadeIn;
+            pConcealmentInfo->cntFadeFrames = findEquiFadeFrame( pConcealCommonData,
+                                                                 pConcealmentInfo->cntFadeFrames-1,
+                                                                 0 /* FadeOut -> FadeIn */);
+          } else {
+            /* change to state OK */
+            pConcealmentInfo->concealState = ConcealState_Ok;
+          }
         } else {
           if (pConcealmentInfo->cntFadeFrames >= pConcealCommonData->numFadeOutFrames) {
             /* change to state MUTE */
@@ -1576,9 +1586,14 @@ static void
 
       case ConcealState_Mute:
         if (pConcealmentInfo->cntValidFrames > pConcealCommonData->numMuteReleaseFrames) {
-          /* change to state FADE-IN */
-          pConcealmentInfo->concealState = ConcealState_FadeIn;
-          pConcealmentInfo->cntFadeFrames = pConcealCommonData->numFadeInFrames - 1;
+          if (pConcealCommonData->numFadeInFrames > 0) {
+            /* change to state FADE-IN */
+            pConcealmentInfo->concealState = ConcealState_FadeIn;
+            pConcealmentInfo->cntFadeFrames = pConcealCommonData->numFadeInFrames - 1;
+          } else {
+            /* change to state OK */
+            pConcealmentInfo->concealState = ConcealState_Ok;
+          }
         }
         break;
 
@@ -1590,11 +1605,16 @@ static void
             pConcealmentInfo->concealState = ConcealState_Ok;
           }
         } else {
-          /* change to state FADE-OUT */
-          pConcealmentInfo->concealState  = ConcealState_FadeOut;
-          pConcealmentInfo->cntFadeFrames = findEquiFadeFrame( pConcealCommonData,
-                                                               pConcealmentInfo->cntFadeFrames+1,
-                                                               1 /* FadeIn -> FadeOut */);
+          if (pConcealCommonData->numFadeOutFrames > 0) {
+            /* change to state FADE-OUT */
+            pConcealmentInfo->concealState  = ConcealState_FadeOut;
+            pConcealmentInfo->cntFadeFrames = findEquiFadeFrame( pConcealCommonData,
+                                                                 pConcealmentInfo->cntFadeFrames+1,
+                                                                 1 /* FadeIn -> FadeOut */);
+          } else {
+            /* change to state MUTE */
+            pConcealmentInfo->concealState = ConcealState_Mute;
+          }
         }
         break;
 
@@ -1625,8 +1645,13 @@ static void
       case ConcealState_Ok:
         if (!(pConcealmentInfo->prevFrameOk[1] ||
              (pConcealmentInfo->prevFrameOk[0] && !pConcealmentInfo->prevFrameOk[1] && frameOk))) {
-          /* Fade out only if the energy interpolation algorithm can not be applied! */
-          pConcealmentInfo->concealState   = ConcealState_FadeOut;
+          if (pConcealCommonData->numFadeOutFrames > 0) {
+            /* Fade out only if the energy interpolation algorithm can not be applied! */
+            pConcealmentInfo->concealState   = ConcealState_FadeOut;
+          } else {
+            /* change to state MUTE */
+            pConcealmentInfo->concealState = ConcealState_Mute;
+          }
           pConcealmentInfo->cntFadeFrames  = 0;
           pConcealmentInfo->cntValidFrames = 0;
         }
@@ -1640,11 +1665,16 @@ static void
         pConcealmentInfo->cntFadeFrames += 1;
 
         if (pConcealmentInfo->cntValidFrames > pConcealCommonData->numMuteReleaseFrames) {
-          /* change to state FADE-IN */
-          pConcealmentInfo->concealState  = ConcealState_FadeIn;
-          pConcealmentInfo->cntFadeFrames = findEquiFadeFrame( pConcealCommonData,
-                                                               pConcealmentInfo->cntFadeFrames-1,
-                                                               0 /* FadeOut -> FadeIn */);
+          if (pConcealCommonData->numFadeInFrames > 0) {
+            /* change to state FADE-IN */
+            pConcealmentInfo->concealState  = ConcealState_FadeIn;
+            pConcealmentInfo->cntFadeFrames = findEquiFadeFrame( pConcealCommonData,
+                                                                 pConcealmentInfo->cntFadeFrames-1,
+                                                                 0 /* FadeOut -> FadeIn */);
+          } else {
+            /* change to state OK */
+            pConcealmentInfo->concealState = ConcealState_Ok;
+          }
         } else {
           if (pConcealmentInfo->cntFadeFrames >= pConcealCommonData->numFadeOutFrames) {
             /* change to state MUTE */
@@ -1655,9 +1685,14 @@ static void
 
       case ConcealState_Mute:
         if (pConcealmentInfo->cntValidFrames > pConcealCommonData->numMuteReleaseFrames) {
-          /* change to state FADE-IN */
-          pConcealmentInfo->concealState = ConcealState_FadeIn;
-          pConcealmentInfo->cntFadeFrames = pConcealCommonData->numFadeInFrames - 1;
+          if (pConcealCommonData->numFadeInFrames > 0) {
+            /* change to state FADE-IN */
+            pConcealmentInfo->concealState = ConcealState_FadeIn;
+            pConcealmentInfo->cntFadeFrames = pConcealCommonData->numFadeInFrames - 1;
+          } else {
+            /* change to state OK */
+            pConcealmentInfo->concealState = ConcealState_Ok;
+          }
         }
         break;
 
@@ -1670,11 +1705,16 @@ static void
             pConcealmentInfo->concealState = ConcealState_Ok;
           }
         } else {
-          /* change to state FADE-OUT */
-          pConcealmentInfo->concealState  = ConcealState_FadeOut;
-          pConcealmentInfo->cntFadeFrames = findEquiFadeFrame( pConcealCommonData,
-                                                               pConcealmentInfo->cntFadeFrames+1,
-                                                               1 /* FadeIn -> FadeOut */);
+          if (pConcealCommonData->numFadeOutFrames > 0) {
+            /* change to state FADE-OUT */
+            pConcealmentInfo->concealState  = ConcealState_FadeOut;
+            pConcealmentInfo->cntFadeFrames = findEquiFadeFrame( pConcealCommonData,
+                                                                 pConcealmentInfo->cntFadeFrames+1,
+                                                                 1 /* FadeIn -> FadeOut */);
+          } else {
+            /* change to state MUTE */
+            pConcealmentInfo->concealState = ConcealState_Mute;
+          }
         }
         break;
       } /* End switch(pConcealmentInfo->concealState) */
diff --git a/libAACdec/src/conceal.h b/libAACdec/src/conceal.h
index 2abc26a..20e674f 100644
--- a/libAACdec/src/conceal.h
+++ b/libAACdec/src/conceal.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACdec/src/conceal_types.h b/libAACdec/src/conceal_types.h
index c1cc6f2..31bc645 100644
--- a/libAACdec/src/conceal_types.h
+++ b/libAACdec/src/conceal_types.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACdec/src/debug.h b/libAACdec/src/debug.h
index 9486c66..e903291 100644
--- a/libAACdec/src/debug.h
+++ b/libAACdec/src/debug.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACdec/src/ldfiltbank.cpp b/libAACdec/src/ldfiltbank.cpp
index 8f719c8..c08cc41 100644
--- a/libAACdec/src/ldfiltbank.cpp
+++ b/libAACdec/src/ldfiltbank.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACdec/src/ldfiltbank.h b/libAACdec/src/ldfiltbank.h
index 97c71b3..d1f9402 100644
--- a/libAACdec/src/ldfiltbank.h
+++ b/libAACdec/src/ldfiltbank.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACdec/src/overlapadd.h b/libAACdec/src/overlapadd.h
index 08bef69..1469be7 100644
--- a/libAACdec/src/overlapadd.h
+++ b/libAACdec/src/overlapadd.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACdec/src/pulsedata.cpp b/libAACdec/src/pulsedata.cpp
index 4edfd39..0f2f0ec 100644
--- a/libAACdec/src/pulsedata.cpp
+++ b/libAACdec/src/pulsedata.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACdec/src/pulsedata.h b/libAACdec/src/pulsedata.h
index 7b10666..fa97d99 100644
--- a/libAACdec/src/pulsedata.h
+++ b/libAACdec/src/pulsedata.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACdec/src/rvlc.cpp b/libAACdec/src/rvlc.cpp
index 7b533a5..16f0bf5 100644
--- a/libAACdec/src/rvlc.cpp
+++ b/libAACdec/src/rvlc.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACdec/src/rvlc.h b/libAACdec/src/rvlc.h
index 7903659..18d5fa3 100644
--- a/libAACdec/src/rvlc.h
+++ b/libAACdec/src/rvlc.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACdec/src/rvlc_info.h b/libAACdec/src/rvlc_info.h
index e245090..63934af 100644
--- a/libAACdec/src/rvlc_info.h
+++ b/libAACdec/src/rvlc_info.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACdec/src/rvlcbit.cpp b/libAACdec/src/rvlcbit.cpp
index 9ddc5a6..6efbb93 100644
--- a/libAACdec/src/rvlcbit.cpp
+++ b/libAACdec/src/rvlcbit.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACdec/src/rvlcbit.h b/libAACdec/src/rvlcbit.h
index 176e816..02fba88 100644
--- a/libAACdec/src/rvlcbit.h
+++ b/libAACdec/src/rvlcbit.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACdec/src/rvlcconceal.cpp b/libAACdec/src/rvlcconceal.cpp
index 682e6eb..ae6b4da 100644
--- a/libAACdec/src/rvlcconceal.cpp
+++ b/libAACdec/src/rvlcconceal.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACdec/src/rvlcconceal.h b/libAACdec/src/rvlcconceal.h
index fdc5795..750cbcd 100644
--- a/libAACdec/src/rvlcconceal.h
+++ b/libAACdec/src/rvlcconceal.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACdec/src/stereo.cpp b/libAACdec/src/stereo.cpp
index 79ebc7d..4f28244 100644
--- a/libAACdec/src/stereo.cpp
+++ b/libAACdec/src/stereo.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACdec/src/stereo.h b/libAACdec/src/stereo.h
index 1ed0d9d..899b485 100644
--- a/libAACdec/src/stereo.h
+++ b/libAACdec/src/stereo.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/include/aacenc_lib.h b/libAACenc/include/aacenc_lib.h
index 862dcb5..307dfd4 100644
--- a/libAACenc/include/aacenc_lib.h
+++ b/libAACenc/include/aacenc_lib.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -316,7 +316,8 @@ if the parameter was not set from extern. The bitrate depends on the number of e
 channels and sampling rate and is determined as follows.
 \code
 AAC-LC (AOT_AAC_LC): 1.5 bits per sample
-HE-AAC (AOT_SBR): 0.625 bits per sample
+HE-AAC (AOT_SBR): 0.625 bits per sample (dualrate sbr)
+HE-AAC (AOT_SBR): 1.125 bits per sample (downsampled sbr)
 HE-AAC v2 (AOT_PS): 0.5 bits per sample
 \endcode
 
@@ -341,6 +342,33 @@ increase which might be significant. If workload is not an issue in the applicat
 we recommended to activate this feature.
 \code aacEncoder_SetParam(hAacEncoder, AACENC_AFTERBURNER, 1); \endcode
 
+\subsection encELD ELD Auto Configuration Mode
+For ELD configuration a so called auto configurator is available which configures SBR and the SBR ratio by itself.
+The configurator is used when the encoder parameter ::AACENC_SBR_MODE and ::AACENC_SBR_RATIO are not set explicitely.
+
+Based on sampling rate and chosen bitrate per channel a reasonable SBR configuration will be used.
+\verbatim
+------------------------------------------------------------
+  Sampling Rate  | Channel Bitrate |  SBR |       SBR Ratio
+-----------------+-----------------+------+-----------------
+ ]min, 16] kHz   |     min - 27999 |   on | downsampled SBR
+                 |   28000 -   max |  off |             ---
+-----------------+-----------------+------+-----------------
+ ]16 - 24] kHz   |     min - 39999 |   on | downsampled SBR
+                 |   40000 -   max |  off |             ---
+-----------------+-----------------+------+-----------------
+ ]24 - 32] kHz   |     min - 27999 |   on |    dualrate SBR
+                 |   28000 - 55999 |   on | downsampled SBR
+                 |   56000 -   max |  off |             ---
+-----------------+-----------------+------+-----------------
+ ]32 - 44.1] kHz |     min - 63999 |   on |    dualrate SBR
+                 |   64000 -   max |  off |             ---
+-----------------+-----------------+------+-----------------
+ ]44.1 - 48] kHz |     min - 63999 |   on |    dualrate SBR
+                 |   64000 - max   |  off |             ---
+------------------------------------------------------------
+\endverbatim
+
 
 \section audiochCfg Audio Channel Configuration
 The MPEG standard refers often to the so-called Channel Configuration. This Channel Configuration is used for a fixed Channel
@@ -349,16 +377,20 @@ For user defined Configurations the Channel Configuration is set to 0 and the Ch
 Program Config Element. The present Encoder implementation does not allow the user to configure this Channel Configuration from
 extern. The Encoder implementation supports fixed Channel Modes which are mapped to Channel Configuration as follow.
 \verbatim
---------------------------------------------------------------------
- ChannelMode     | ChCfg  | front_El | side_El  | back_El  | lfe_El
------------------+--------+----------+----------+----------+--------
-MODE_1           |      1 | SCE      |          |          |
-MODE_2           |      2 | CPE      |          |          |
-MODE_1_2         |      3 | SCE, CPE |          |          |
-MODE_1_2_1       |      4 | SCE, CPE |          | SCE      |
-MODE_1_2_2       |      5 | SCE, CPE |          | CPE      |
-MODE_1_2_2_1     |      6 | SCE, CPE |          | CPE      | LFE
---------------------------------------------------------------------
+-------------------------------------------------------------------------------
+ ChannelMode           | ChCfg  | front_El      | side_El  | back_El  | lfe_El
+-----------------------+--------+---------------+----------+----------+--------
+MODE_1                 |      1 | SCE           |          |          |
+MODE_2                 |      2 | CPE           |          |          |
+MODE_1_2               |      3 | SCE, CPE      |          |          |
+MODE_1_2_1             |      4 | SCE, CPE      |          | SCE      |
+MODE_1_2_2             |      5 | SCE, CPE      |          | CPE      |
+MODE_1_2_2_1           |      6 | SCE, CPE      |          | CPE      | LFE
+MODE_1_2_2_2_1         |      7 | SCE, CPE, CPE |          | CPE      | LFE
+-----------------------+--------+---------------+----------+----------+--------
+MODE_7_1_REAR_SURROUND |      0 | SCE, CPE      |          | CPE, CPE | LFE
+MODE_7_1_FRONT_CENTER  |      0 | SCE, CPE, CPE |          | CPE      | LFE
+-------------------------------------------------------------------------------
  - SCE: Single Channel Element.
  - CPE: Channel Pair.
  - SCE: Low Frequency Element.
@@ -374,16 +406,20 @@ Beside the Channel Element assignment the Channel Modes are resposible for audio
 of the audio data depends on the selected ::AACENC_CHANNELORDER which can be MPEG or WAV like order.\n
 Following Table describes the complete channel mapping for both Channel Order configurations.
 \verbatim
----------------------------------------------------------------------------------
-ChannelMode      |  MPEG-Channelorder            |  WAV-Channelorder
------------------+---+---+---+---+---+---+---+---+---+---+---+---+---+---+---+---
-MODE_1           | 0 |   |   |   |   |   |   |   | 0 |   |   |   |   |   |   |
-MODE_2           | 0 | 1 |   |   |   |   |   |   | 0 | 1 |   |   |   |   |   |
-MODE_1_2         | 0 | 1 | 2 |   |   |   |   |   | 2 | 0 | 1 |   |   |   |   |
-MODE_1_2_1       | 0 | 1 | 2 | 3 |   |   |   |   | 2 | 0 | 1 | 3 |   |   |   |
-MODE_1_2_2       | 0 | 1 | 2 | 3 | 4 |   |   |   | 2 | 0 | 1 | 3 | 4 |   |   |
-MODE_1_2_2_1     | 0 | 1 | 2 | 3 | 4 | 5 |   |   | 2 | 0 | 1 | 4 | 5 | 3 |   |
----------------------------------------------------------------------------------
+---------------------------------------------------------------------------------------
+ChannelMode            |  MPEG-Channelorder            |  WAV-Channelorder
+-----------------------+---+---+---+---+---+---+---+---+---+---+---+---+---+---+---+---
+MODE_1                 | 0 |   |   |   |   |   |   |   | 0 |   |   |   |   |   |   |
+MODE_2                 | 0 | 1 |   |   |   |   |   |   | 0 | 1 |   |   |   |   |   |
+MODE_1_2               | 0 | 1 | 2 |   |   |   |   |   | 2 | 0 | 1 |   |   |   |   |
+MODE_1_2_1             | 0 | 1 | 2 | 3 |   |   |   |   | 2 | 0 | 1 | 3 |   |   |   |
+MODE_1_2_2             | 0 | 1 | 2 | 3 | 4 |   |   |   | 2 | 0 | 1 | 3 | 4 |   |   |
+MODE_1_2_2_1           | 0 | 1 | 2 | 3 | 4 | 5 |   |   | 2 | 0 | 1 | 4 | 5 | 3 |   |
+MODE_1_2_2_2_1         | 0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 | 2 | 6 | 7 | 0 | 1 | 4 | 5 | 3
+-----------------------+---+---+---+---+---+---+---+---+---+---+---+---+---+---+---+---
+MODE_7_1_REAR_SURROUND | 0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 | 2 | 0 | 1 | 6 | 7 | 4 | 5 | 3
+MODE_7_1_FRONT_CENTER  | 0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 | 2 | 6 | 7 | 0 | 1 | 4 | 5 | 3
+---------------------------------------------------------------------------------------
 \endverbatim
 
 The denoted mapping is important for correct audio channel assignment when using MPEG or WAV ordering. The incoming audio
@@ -416,8 +452,9 @@ determines the maximum allowed bitrate for AAC-LC. For HE-AAC and HE-AAC v2 a li
 look-up table is used.
 
 A good working point in terms of audio quality, sampling rate and bitrate, is at 1 to 1.5
-bits/audio sample for AAC-LC, 0.625 bits/audio sample for HE-AAC and 0.5 bits/audio sample
-for HE-AAC v2. For example for one channel with a sampling frequency of 48 kHz, the range from
+bits/audio sample for AAC-LC, 0.625 bits/audio sample for dualrate HE-AAC, 1.125 bits/audio sample
+for downsampled HE-AAC and 0.5 bits/audio sample for HE-AAC v2.
+For example for one channel with a sampling frequency of 48 kHz, the range from
 48 kbit/s to 72 kbit/s achieves reasonable audio quality for AAC-LC.
 
 For HE-AAC and HE-AAC v2 the lowest possible audio input sampling frequency is 16 kHz because then the
@@ -434,7 +471,7 @@ quality at that bitrate than HE-AAC or HE-AAC v2.
 The following table provides an overview of recommended encoder configuration parameters
 which we determined by virtue of numerous listening tests.
 
-\subsection reommendedConfigLC AAC-LC, HE-AAC, HE-AACv2.
+\subsection reommendedConfigLC AAC-LC, HE-AAC, HE-AACv2 in Dualrate SBR mode.
 \verbatim
 -----------------------------------------------------------------------------------
 Audio Object Type  |  Bit Rate Range  |            Supported  | Preferred  | No. of
@@ -457,8 +494,8 @@ AAC LC + SBR       |  64000 - 128000  |  32.00, 44.10, 48.00  |     48.00  |
 -------------------+------------------+-----------------------+------------+-------
 AAC LC + SBR       |  64000 -  69999  |  32.00, 44.10, 48.00  |     32.00  | 5, 5.1
 AAC LC + SBR       |  70000 - 159999  |  32.00, 44.10, 48.00  |     44.10  | 5, 5.1
-AAC LC + SBR       | 160000 - 319999  |  32.00, 44.10, 48.00  |     48.00  | 5, 5.1
-AAC LC + SBR       | 320000 - 640000  |  64.00, 88.20, 96.00  |     96.00  | 5, 5.1
+AAC LC + SBR       | 160000 - 245999  |  32.00, 44.10, 48.00  |     48.00  |      5
+AAC LC + SBR       | 160000 - 265999  |  32.00, 44.10, 48.00  |     48.00  |    5.1
 -------------------+------------------+-----------------------+------------+-------
 AAC LC             |   8000 -  15999  | 11.025, 12.00, 16.00  |     12.00  |      1
 AAC LC             |  16000 -  23999  |                16.00  |     16.00  |      1
@@ -481,7 +518,7 @@ AAC LC             | 280000 - 800000  |  32.00, 44.10, 48.00  |     44.10  | 5,
 -----------------------------------------------------------------------------------
 \endverbatim \n
 
-\subsection reommendedConfigLD AAC-LD, AAC-ELD, AAC-ELD with SBR.
+\subsection reommendedConfigLD AAC-LD, AAC-ELD, AAC-ELD with SBR in Dualrate SBR mode.
 \verbatim
 -----------------------------------------------------------------------------------
 Audio Object Type  |  Bit Rate Range  |            Supported  | Preferred  | No. of
@@ -489,18 +526,20 @@ Audio Object Type  |  Bit Rate Range  |            Supported  | Preferred  | No.
                    |                  |                [kHz]  |      Rate  |
                    |                  |                       |     [kHz]  |
 -------------------+------------------+-----------------------+------------+-------
-ELD + SBR          |  16000 -  24999  |        32.00 - 44.10  |     32.00  |      1
+ELD + SBR          |  18000 -  24999  |        32.00 - 44.10  |     32.00  |      1
 ELD + SBR          |  25000 -  31999  |        32.00 - 48.00  |     32.00  |      1
 ELD + SBR          |  32000 -  64000  |        32.00 - 48.00  |     48.00  |      1
 -------------------+------------------+-----------------------+------------+-------
 ELD + SBR          |  32000 -  51999  |        32.00 - 48.00  |     44.10  |      2
 ELD + SBR          |  52000 - 128000  |        32.00 - 48.00  |     48.00  |      2
 -------------------+------------------+-----------------------+------------+-------
-ELD + SBR          |  72000 - 192000  |        44.10 - 48.00  |     48.00  |      3
+ELD + SBR          |  72000 - 160000  |        44.10 - 48.00  |     48.00  |      3
+-------------------+------------------+-----------------------+------------+-------
+ELD + SBR          |  96000 - 212000  |        44.10 - 48.00  |     48.00  |      4
 -------------------+------------------+-----------------------+------------+-------
-ELD + SBR          |  96000 - 256000  |        44.10 - 48.00  |     48.00  |      4
+ELD + SBR          | 120000 - 246000  |        44.10 - 48.00  |     48.00  |      5
 -------------------+------------------+-----------------------+------------+-------
-ELD + SBR          | 120000 - 320000  |        44.10 - 48.00  |     48.00  |      5
+ELD + SBR          | 120000 - 266000  |        44.10 - 48.00  |     48.00  |    5.1
 -------------------+------------------+-----------------------+------------+-------
 LD, ELD            |  16000 -  19999  |        16.00 - 24.00  |     16.00  |      1
 LD, ELD            |  20000 -  39999  |        16.00 - 32.00  |     24.00  |      1
@@ -531,13 +570,33 @@ LD, ELD            | 340000 - 960000  |        44.10 - 48.00  |     48.00  |
 -----------------------------------------------------------------------------------
 \endverbatim \n
 
+\subsection reommendedConfigELD AAC-ELD with SBR in Downsampled SBR mode.
+\verbatim
+-----------------------------------------------------------------------------------
+Audio Object Type  |  Bit Rate Range  |            Supported  | Preferred  | No. of
+                   |         [bit/s]  |       Sampling Rates  |    Sampl.  |  Chan.
+                   |                  |                [kHz]  |      Rate  |
+                   |                  |                       |     [kHz]  |
+-------------------+------------------+-----------------------+------------+-------
+ELD + SBR          |  18000 -  24999  |        16.00 - 22.05  |     22.05  |      1
+(downsampled SBR)  |  25000 -  35999  |        22.05 - 32.00  |     24.00  |      1
+                   |  36000 -  64000  |        32.00 - 48.00  |     32.00  |      1
+-----------------------------------------------------------------------------------
+\endverbatim \n
+
+
 \page ENCODERBEHAVIOUR Encoder Behaviour
 
 \section BEHAVIOUR_BANDWIDTH Bandwidth
 
 The FDK AAC encoder usually does not use the full frequency range of the input signal, but restricts the bandwidth
 according to certain library-internal settings. They can be changed in the table "bandWidthTable" in the
-file bandwidth.cpp (if available), or via command-line argument "-w" (see chapter \ref CommandLineUsage).
+file bandwidth.cpp (if available).
+
+The encoder API provides the ::AACENC_BANDWIDTH parameter to adjust the bandwidth explicitly.
+\code
+aacEncoder_SetParam(hAacEncoder, AACENC_BANDWIDTH, value);
+\endcode
 
 However it is not recommended to change these settings, because they are based on numerious listening
 tests and careful tweaks to ensure the best overall encoding quality.
@@ -638,7 +697,6 @@ an MPEG-2 AOT is choosen since PNS is an MPEG-4 AAC feature.
 If SBR is activated, the encoder automatically deactivates PNS internally. If TNS is disabled but PNS is allowed,
 the encoder deactivates PNS calculation internally.
 
-
 */
 
 #ifndef _AAC_ENC_LIB_H_
@@ -647,6 +705,9 @@ the encoder deactivates PNS calculation internally.
 #include "machine_type.h"
 #include "FDK_audio.h"
 
+#define AACENCODER_LIB_VL0 3
+#define AACENCODER_LIB_VL1 4
+#define AACENCODER_LIB_VL2 12
 
 /**
  *  AAC encoder error codes.
@@ -858,8 +919,9 @@ typedef enum
   AACENC_SAMPLERATE               = 0x0103,  /*!< Audio input data sampling rate. Encoder supports following sampling rates:
                                                   8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, 64000, 88200, 96000 */
 
-  AACENC_SBR_MODE                 = 0x0104,  /*!< Configure SBR independently of the chosen Audio Object Type ::AUDIO_OBJECT_TYPE:.
-                                                  This parameter is only available for ELD.
+  AACENC_SBR_MODE                 = 0x0104,  /*!< Configure SBR independently of the chosen Audio Object Type ::AUDIO_OBJECT_TYPE.
+                                                  This parameter is for ELD audio object type only.
+                                                  - -1: Use ELD SBR auto configurator (default).
                                                   - 0: Disable Spectral Band Replication.
                                                   - 1: Enable Spectral Band Replication. */
 
@@ -869,12 +931,21 @@ typedef enum
                                                   - 480: Optional length in LD/ELD configuration. */
 
   AACENC_CHANNELMODE              = 0x0106,  /*!< Set explicit channel mode. Channel mode must match with number of input channels.
-                                                  - 1-6: MPEG channel modes supported, see ::CHANNEL_MODE in FDK_audio.h. */
+                                                  - 1-7 and 33,34: MPEG channel modes supported, see ::CHANNEL_MODE in FDK_audio.h. */
 
   AACENC_CHANNELORDER             = 0x0107,  /*!< Input audio data channel ordering scheme:
                                                   - 0: MPEG channel ordering (e. g. 5.1: C, L, R, SL, SR, LFE). (default)
                                                   - 1: WAVE file format channel ordering (e. g. 5.1: L, R, C, LFE, SL, SR). */
 
+  AACENC_SBR_RATIO                = 0x0108,  /*!<  Controls activation of downsampled SBR. With downsampled SBR, the delay will be
+                                                   shorter. On the other hand, for achieving the same quality level, downsampled SBR
+                                                   needs more bits than dual-rate SBR.
+                                                   With downsampled SBR, the AAC encoder will work at the same sampling rate as the
+                                                   SBR encoder (single rate).
+                                                   Downsampled SBR is supported for AAC-ELD and HE-AACv1.
+                                                   - 1: Downsampled SBR (default for ELD).
+                                                   - 2: Dual-rate SBR   (default for HE-AAC). */
+
   AACENC_AFTERBURNER              = 0x0200,  /*!< This parameter controls the use of the afterburner feature.
                                                   The afterburner is a type of analysis by synthesis algorithm which increases the
                                                   audio quality but also the required processing power. It is recommended to always
@@ -906,14 +977,50 @@ typedef enum
                                                   - n: Frame count period. */
 
   AACENC_SIGNALING_MODE           = 0x0302,  /*!< Signaling mode of the extension AOT:
-                                                  - 0: Implicit backward compatible signaling. (default)
-                                                  - 1: Explicit SBR and implicit PS signaling.
-                                                  - 2: Explicit hierarchical signaling.
-
-                                                  The use of backward-compatible implicit signaling is recommended if the user specically
-                                                  aims at preserving compatibility with decoders only capable of decoding AAC-LC. Otherwise
-                                                  use non-backward-compatible explicit signaling.
-                                                  Bitstream formats ADTS and ADIF can only do implicit signaling. */
+                                                  - 0: Implicit backward compatible signaling (default for non-MPEG-4 based
+                                                       AOT's and for the transport formats ADIF and ADTS)
+                                                       - A stream that uses implicit signaling can be decoded by every AAC decoder, even AAC-LC-only decoders
+                                                       - An AAC-LC-only decoder will only decode the low-frequency part of the stream, resulting in a band-limited output
+                                                       - This method works with all transport formats
+                                                       - This method does not work with downsampled SBR
+                                                  - 1: Explicit backward compatible signaling
+                                                       - A stream that uses explicit backward compatible signaling can be decoded by every AAC decoder, even AAC-LC-only decoders
+                                                       - An AAC-LC-only decoder will only decode the low-frequency part of the stream, resulting in a band-limited output
+                                                       - A decoder not capable of decoding PS will only decode the AAC-LC+SBR part.
+                                                         If the stream contained PS, the result will be a a decoded mono downmix
+                                                       - This method does not work with ADIF or ADTS. For LOAS/LATM, it only works with AudioMuxVersion==1
+                                                       - This method does work with downsampled SBR
+                                                  - 2: Explicit hierarchical signaling (default for MPEG-4 based AOT's and for all transport formats excluding ADIF and ADTS)
+                                                       - A stream that uses explicit hierarchical signaling can be decoded only by HE-AAC decoders
+                                                       - An AAC-LC-only decoder will not decode a stream that uses explicit hierarchical signaling
+                                                       - A decoder not capable of decoding PS will not decode the stream at all if it contained PS
+                                                       - This method does not work with ADIF or ADTS. It works with LOAS/LATM and the MPEG-4 File format
+                                                       - This method does work with downsampled SBR
+
+                                                   For making sure that the listener always experiences the best audio quality,
+                                                   explicit hierarchical signaling should be used.
+                                                   This makes sure that only a full HE-AAC-capable decoder will decode those streams.
+                                                   The audio is played at full bandwidth.
+                                                   For best backwards compatibility, it is recommended to encode with implicit SBR signaling.
+                                                   A decoder capable of AAC-LC only will then only decode the AAC part, which means the decoded
+                                                   audio will sound band-limited.
+
+                                                   For MPEG-2 transport types (ADTS,ADIF), only implicit signaling is possible.
+
+                                                   For LOAS and LATM, explicit backwards compatible signaling only works together with AudioMuxVersion==1.
+                                                   The reason is that, for explicit backwards compatible signaling, additional information will be appended to the ASC.
+                                                   A decoder that is only capable of decoding AAC-LC will skip this part.
+                                                   Nevertheless, for jumping to the end of the ASC, it needs to know the ASC length.
+                                                   Transmitting the length of the ASC is a feature of AudioMuxVersion==1, it is not possible to transmit the
+                                                   length of the ASC with AudioMuxVersion==0, therefore an AAC-LC-only decoder will not be able to parse a
+                                                   LOAS/LATM stream that was being encoded with AudioMuxVersion==0.
+
+                                                   For downsampled SBR, explicit signaling is mandatory. The reason for this is that the
+                                                   extension sampling frequency (which is in case of SBR the sampling frequqncy of the SBR part)
+                                                   can only be signaled in explicit mode.
+
+                                                   For AAC-ELD, the SBR information is transmitted in the ELDSpecific Config, which is part of the
+                                                   AudioSpecificConfig. Therefore, the settings here will have no effect on AAC-ELD.*/
 
   AACENC_TPSUBFRAMES              = 0x0303,  /*!< Number of sub frames in a transport frame for LOAS/LATM or ADTS (default 1).
                                                   - ADTS: Maximum number of sub frames restricted to 4.
diff --git a/libAACenc/src/aacEnc_ram.cpp b/libAACenc/src/aacEnc_ram.cpp
index 45bca93..be3eea2 100644
--- a/libAACenc/src/aacEnc_ram.cpp
+++ b/libAACenc/src/aacEnc_ram.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -89,7 +89,7 @@ amm-info at iis.fraunhofer.de
 ******************************************************************************/
 /*!
   \file
-  \brief  Memory layout   
+  \brief  Memory layout
   \author Markus Lohwasser
 */
 
@@ -119,12 +119,12 @@ C_ALLOC_MEM (Ram_aacEnc_AacEncoder,           AAC_ENC,          1)
      PsyInputBuffer.
 */
 
-C_ALLOC_MEM2 (Ram_aacEnc_PsyElement, PSY_ELEMENT, 1, (6))
+C_ALLOC_MEM2 (Ram_aacEnc_PsyElement, PSY_ELEMENT, 1, (8))
 
 C_ALLOC_MEM  (Ram_aacEnc_PsyInternal, PSY_INTERNAL, 1)
-C_ALLOC_MEM2 (Ram_aacEnc_PsyStatic,   PSY_STATIC,   1, (6))
+C_ALLOC_MEM2 (Ram_aacEnc_PsyStatic,   PSY_STATIC,   1, (8))
 
-C_ALLOC_MEM2 (Ram_aacEnc_PsyInputBuffer, INT_PCM, MAX_INPUT_BUFFER_SIZE, (6))
+C_ALLOC_MEM2 (Ram_aacEnc_PsyInputBuffer, INT_PCM, MAX_INPUT_BUFFER_SIZE, (8))
 
   PSY_DYNAMIC *GetRam_aacEnc_PsyDynamic (int n, UCHAR* dynamic_RAM) {
       FDK_ASSERT(dynamic_RAM!=0);
@@ -139,8 +139,8 @@ C_ALLOC_MEM2 (Ram_aacEnc_PsyInputBuffer, INT_PCM, MAX_INPUT_BUFFER_SIZE, (6))
 */
 C_ALLOC_MEM2 (Ram_aacEnc_PsyOut,         PSY_OUT, 1, (1))
 
-C_ALLOC_MEM2 (Ram_aacEnc_PsyOutElements, PSY_OUT_ELEMENT, 1, (1)*(6))
-C_ALLOC_MEM2 (Ram_aacEnc_PsyOutChannel,  PSY_OUT_CHANNEL, 1, (1)*(6))
+C_ALLOC_MEM2 (Ram_aacEnc_PsyOutElements, PSY_OUT_ELEMENT, 1, (1)*(8))
+C_ALLOC_MEM2 (Ram_aacEnc_PsyOutChannel,  PSY_OUT_CHANNEL, 1, (1)*(8))
 
 
 /*
@@ -155,8 +155,8 @@ C_ALLOC_MEM2 (Ram_aacEnc_PsyOutChannel,  PSY_OUT_CHANNEL, 1, (1)*(6))
 C_ALLOC_MEM (Ram_aacEnc_QCstate, QC_STATE, 1)
 C_ALLOC_MEM (Ram_aacEnc_AdjustThreshold, ADJ_THR_STATE, 1)
 
-C_ALLOC_MEM2 (Ram_aacEnc_AdjThrStateElement, ATS_ELEMENT, 1, (6))
-C_ALLOC_MEM2 (Ram_aacEnc_ElementBits, ELEMENT_BITS, 1, (6))
+C_ALLOC_MEM2 (Ram_aacEnc_AdjThrStateElement, ATS_ELEMENT, 1, (8))
+C_ALLOC_MEM2 (Ram_aacEnc_ElementBits, ELEMENT_BITS, 1, (8))
 C_ALLOC_MEM (Ram_aacEnc_BitCntrState, BITCNTR_STATE, 1)
 
   INT *GetRam_aacEnc_BitLookUp(int n, UCHAR* dynamic_RAM) {
@@ -175,7 +175,7 @@ C_ALLOC_MEM (Ram_aacEnc_BitCntrState, BITCNTR_STATE, 1)
 */
 
 C_ALLOC_MEM2 (Ram_aacEnc_QCout, QC_OUT, 1, (1))
-C_ALLOC_MEM2 (Ram_aacEnc_QCelement, QC_OUT_ELEMENT, 1, (1)*(6))
+C_ALLOC_MEM2 (Ram_aacEnc_QCelement, QC_OUT_ELEMENT, (1), (8))
   QC_OUT_CHANNEL *GetRam_aacEnc_QCchannel (int n, UCHAR* dynamic_RAM) {
       FDK_ASSERT(dynamic_RAM!=0);
       return ((QC_OUT_CHANNEL*) (dynamic_RAM + P_BUF_0 + n*sizeof(QC_OUT_CHANNEL)));
diff --git a/libAACenc/src/aacEnc_ram.h b/libAACenc/src/aacEnc_ram.h
index 918e279..cf7da7c 100644
--- a/libAACenc/src/aacEnc_ram.h
+++ b/libAACenc/src/aacEnc_ram.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -118,7 +118,7 @@ amm-info at iis.fraunhofer.de
 */
 
 /* define hBitstream size: max AAC framelength is 6144 bits/channel */
-/*#define BUFFER_BITSTR_SIZE ((6400*(6)/bbWordSize)    +((bbWordSize - 1) / bbWordSize))*/
+/*#define BUFFER_BITSTR_SIZE ((6400*(8)/bbWordSize)    +((bbWordSize - 1) / bbWordSize))*/
 
 struct AAC_ENC {
 
@@ -173,7 +173,7 @@ struct AAC_ENC {
  ++++++++++++++++++++++++++++++++++++++++++++
 */
 
-#define BUF_SIZE_0       ( ALIGN_SIZE(sizeof(QC_OUT_CHANNEL)*(6)) )
+#define BUF_SIZE_0       ( ALIGN_SIZE(sizeof(QC_OUT_CHANNEL)*(8)) )
 #define BUF_SIZE_1       ( ALIGN_SIZE(maxSize(sizeof(PSY_DYNAMIC), \
                                       (BIT_LOOK_UP_SIZE+MERGE_GAIN_LOOK_UP_SIZE))) )
 
diff --git a/libAACenc/src/aacEnc_rom.cpp b/libAACenc/src/aacEnc_rom.cpp
index 48ba668..0cdf5fe 100644
--- a/libAACenc/src/aacEnc_rom.cpp
+++ b/libAACenc/src/aacEnc_rom.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/aacEnc_rom.h b/libAACenc/src/aacEnc_rom.h
index f25d327..37e5012 100644
--- a/libAACenc/src/aacEnc_rom.h
+++ b/libAACenc/src/aacEnc_rom.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/aacenc.cpp b/libAACenc/src/aacenc.cpp
index d3f36aa..569662b 100644
--- a/libAACenc/src/aacenc.cpp
+++ b/libAACenc/src/aacenc.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -393,7 +393,7 @@ AAC_ENCODER_ERROR FDKaacEnc_Initialize(HANDLE_AAC_ENC      hAacEnc,
   /******************* sanity checks *******************/
 
   /* check config structure */
-  if (config->nChannels  < 1 || config->nChannels > (6)) {
+  if (config->nChannels  < 1 || config->nChannels > (8)) {
     return AAC_ENC_UNSUPPORTED_CHANNELCONFIG;
   }
 
@@ -558,6 +558,13 @@ AAC_ENCODER_ERROR FDKaacEnc_Initialize(HANDLE_AAC_ENC      hAacEnc,
   qcInit.channelMapping      = &hAacEnc->channelMapping;
   qcInit.sceCpe              = 0;
 
+  if ((config->bitrateMode>=1) && (config->bitrateMode<=5)) {
+      qcInit.averageBits     = (averageBitsPerFrame+7)&~7;
+      qcInit.bitRes          = MIN_BUFSIZE_PER_EFF_CHAN*cm->nChannelsEff;
+      qcInit.maxBits         = MIN_BUFSIZE_PER_EFF_CHAN*cm->nChannelsEff;
+      qcInit.minBits         = 0;
+  }
+  else
   {
       int maxBitres;
       qcInit.averageBits     = (averageBitsPerFrame+7)&~7;
@@ -571,6 +578,8 @@ AAC_ENCODER_ERROR FDKaacEnc_Initialize(HANDLE_AAC_ENC      hAacEnc,
       qcInit.minBits         = (config->minBitsPerFrame!=-1) ? fixMax(qcInit.minBits, config->minBitsPerFrame) : qcInit.minBits;
   }
 
+  qcInit.sampleRate          = config->sampleRate;
+  qcInit.advancedBitsToPe    = isLowDelay(config->audioObjectType) ? 1 : 0 ;
   qcInit.nSubFrames          = config->nSubFrames;
   qcInit.padding.paddingRest = config->sampleRate;
 
diff --git a/libAACenc/src/aacenc.h b/libAACenc/src/aacenc.h
index 517b0dc..ed167c2 100644
--- a/libAACenc/src/aacenc.h
+++ b/libAACenc/src/aacenc.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -113,7 +113,7 @@ typedef enum {
   /* initialization errors */
   aac_enc_init_error_start      = 0x2000,
   AAC_ENC_INVALID_HANDLE        = 0x2020,             /*!< The handle passed to the function call was invalid (probably NULL).        */
-  AAC_ENC_INVALID_FRAME_LENGTH  = 0x2080,             /*!< Invalid frame length (must be 1024 or 960).                                */
+  AAC_ENC_INVALID_FRAME_LENGTH  = 0x2080,             /*!< Invalid frame length.                                                      */
   AAC_ENC_INVALID_N_CHANNELS    = 0x20e0,             /*!< Invalid amount of audio input channels.                                    */
   AAC_ENC_INVALID_SFB_TABLE     = 0x2140,             /*!< Internal encoder error.                                                    */
 
@@ -154,7 +154,7 @@ typedef enum {
 
 #define ANC_DATA_BUFFERSIZE 1024      /* ancBuffer size */
 
-#define MAX_TOTAL_EXT_PAYLOADS  (((6) * (1)) + (2+2))
+#define MAX_TOTAL_EXT_PAYLOADS  (((8) * (1)) + (2+2))
 
 
 typedef enum {
@@ -205,6 +205,8 @@ struct AACENC_CONFIG {
   INT   maxBitsPerFrame;        /* maximum number of bits in AU */
   INT   bitreservoir;           /* size of bitreservoir */
 
+  UINT  sbrRatio;               /* sbr sampling rate ratio: dual- or single-rate */
+
   UCHAR useTns;                 /* flag: use temporal noise shaping */
   UCHAR usePns;                 /* flag: use perceptual noise substitution */
   UCHAR useIS;                  /* flag: use intensity coding */
diff --git a/libAACenc/src/aacenc_hcr.cpp b/libAACenc/src/aacenc_hcr.cpp
index 91c46a9..316623a 100644
--- a/libAACenc/src/aacenc_hcr.cpp
+++ b/libAACenc/src/aacenc_hcr.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/aacenc_hcr.h b/libAACenc/src/aacenc_hcr.h
index 257459c..934247a 100644
--- a/libAACenc/src/aacenc_hcr.h
+++ b/libAACenc/src/aacenc_hcr.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/aacenc_lib.cpp b/libAACenc/src/aacenc_lib.cpp
index a4291d5..95c08b3 100644
--- a/libAACenc/src/aacenc_lib.cpp
+++ b/libAACenc/src/aacenc_lib.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -97,8 +97,8 @@ amm-info at iis.fraunhofer.de
 
 /* Encoder library info */
 #define AACENCODER_LIB_VL0 3
-#define AACENCODER_LIB_VL1 3
-#define AACENCODER_LIB_VL2 3
+#define AACENCODER_LIB_VL1 4
+#define AACENCODER_LIB_VL2 12
 #define AACENCODER_LIB_TITLE "AAC Encoder"
 #define AACENCODER_LIB_BUILD_DATE __DATE__
 #define AACENCODER_LIB_BUILD_TIME __TIME__
@@ -118,10 +118,12 @@ amm-info at iis.fraunhofer.de
 #define SBL(fl)            (fl/8)                 /*!< Short block length (hardcoded to 8 short blocks per long block) */
 #define BSLA(fl)           (4*SBL(fl)+SBL(fl)/2)  /*!< AAC block switching look-ahead */
 #define DELAY_AAC(fl)      (fl+BSLA(fl))          /*!< MDCT + blockswitching */
-#define DELAY_AACELD(fl)   ( (fl) + ((fl)/2)  )   /*!< ELD FB delay */
+#define DELAY_AACELD(fl)   ((fl)/2)               /*!< ELD FB delay (no framing delay included) */
 
 #define INPUTBUFFER_SIZE (1537+100+2048)
 
+#define DEFAULT_HEADER_PERIOD_REPETITION_RATE  10 /*!< Default header repetition rate used in transport library and for SBR header. */
+
 ////////////////////////////////////////////////////////////////////////////////////
 /**
  * Flags to characterize encoder modules to be supported in present instance.
@@ -164,7 +166,8 @@ typedef struct {
 
     UCHAR             userMetaDataMode;      /*!< Meta data library configuration. */
 
-    UCHAR             userSbrEnabled;
+    UCHAR             userSbrEnabled;        /*!< Enable SBR for ELD. */
+    UINT              userSbrRatio;          /*!< SBR sampling rate ratio. Dual- or single-rate. */
 
 } USER_PARAM;
 
@@ -212,8 +215,8 @@ struct AACENCODER
 
     AACENC_EXT_PAYLOAD       extPayload [MAX_TOTAL_EXT_PAYLOADS];
     /* Extension payload */
-    UCHAR                    extPayloadData [(1)][(6)][MAX_PAYLOAD_SIZE];
-    UINT                     extPayloadSize [(1)][(6)]; /* payload sizes in bits */
+    UCHAR                    extPayloadData [(1)][(8)][MAX_PAYLOAD_SIZE];
+    UINT                     extPayloadSize [(1)][(8)]; /* payload sizes in bits */
 
     ULONG                    InitFlags;         /* internal status to treggier re-initialization */
 
@@ -227,12 +230,92 @@ struct AACENCODER
 
    UINT                      encoder_modis;
 
-   /* Capabity flags */
+   /* Capability flags */
    UINT                      CAPF_tpEnc;
 
 } ;
 
-////////////////////////////////////////////////////////////////////////////////////
+typedef struct
+{
+    ULONG               samplingRate;   /*!< Encoder output sampling rate. */
+    ULONG               bitrateRange;   /*!< Lower bitrate range for config entry. */
+
+    UCHAR               lowDelaySbr;    /*!< 0: ELD sbr off,
+                                             1: ELD sbr on */
+
+    UCHAR               downsampledSbr; /*!< 0: ELD with dualrate sbr,
+                                             1: ELD with downsampled sbr */
+
+} ELD_SBR_CONFIGURATOR;
+
+/**
+ * \brief  This table defines ELD/SBR default configurations.
+ */
+static const ELD_SBR_CONFIGURATOR eldSbrAutoConfigTab[] =
+{
+  { 48000,     0, 1, 0 },
+  { 48000, 64001, 0, 0 },
+
+  { 44100,     0, 1, 0 },
+  { 44100, 64001, 0, 0 },
+
+  { 32000,     0, 1, 0 },
+  { 32000, 28000, 1, 1 },
+  { 32000, 56000, 0, 0 },
+
+  { 24000,     0, 1, 1 },
+  { 24000, 40000, 0, 0 },
+
+  { 16000,     0, 1, 1 },
+  { 16000, 28000, 0, 0 }
+
+};
+
+/*
+ * \brief  Configure SBR for ELD configuration.
+ *
+ * This function finds default SBR configuration for ELD based on sampling rate and channel bitrate.
+ * Outputparameters are SBR on/off, and SBR ratio.
+ *
+ * \param samplingRate          Audio signal sampling rate.
+ * \param channelMode           Channel configuration to be used.
+ * \param totalBitrate          Overall bitrate.
+ * \param eldSbr                Pointer to eldSbr parameter, filled on return.
+ * \param eldSbrRatio           Pointer to eldSbrRatio parameter, filled on return.
+ *
+ * \return - AACENC_OK, all fine.
+ *         - AACENC_INVALID_CONFIG, on failure.
+ */
+static AACENC_ERROR eldSbrConfigurator(
+        const ULONG                      samplingRate,
+        const CHANNEL_MODE               channelMode,
+        const ULONG                      totalBitrate,
+        UINT * const                     eldSbr,
+        UINT * const                     eldSbrRatio
+        )
+{
+    AACENC_ERROR err = AACENC_OK;
+    int i, cfgIdx = -1;
+    const ULONG channelBitrate = totalBitrate / FDKaacEnc_GetChannelModeConfiguration(channelMode)->nChannelsEff;
+
+    for (i=0; i<(sizeof(eldSbrAutoConfigTab)/sizeof(ELD_SBR_CONFIGURATOR)); i++) {
+      if ( (samplingRate <= eldSbrAutoConfigTab[i].samplingRate)
+        && (channelBitrate >= eldSbrAutoConfigTab[i].bitrateRange) )
+      {
+        cfgIdx = i;
+      }
+    }
+
+    if (cfgIdx != -1) {
+      *eldSbr      = (eldSbrAutoConfigTab[cfgIdx].lowDelaySbr==0) ? 0 : 1;
+      *eldSbrRatio = (eldSbrAutoConfigTab[cfgIdx].downsampledSbr==0) ? 2 : 1;
+    }
+    else {
+      err = AACENC_INVALID_CONFIG; /* no default configuration for eld-sbr available. */
+    }
+
+    return err;
+}
 
 static inline INT isSbrActive(const HANDLE_AACENC_CONFIG hAacConfig)
 {
@@ -253,6 +336,70 @@ static inline INT isSbrActive(const HANDLE_AACENC_CONFIG hAacConfig)
     return ( sbrUsed );
 }
 
+static inline INT isPsActive(const AUDIO_OBJECT_TYPE audioObjectType)
+{
+    INT psUsed = 0;
+
+    if ( (audioObjectType==AOT_PS)
+      || (audioObjectType==AOT_MP2_PS)
+      || (audioObjectType==AOT_DABPLUS_PS)
+      || (audioObjectType==AOT_DRM_MPEG_PS) )
+    {
+        psUsed = 1;
+    }
+
+    return ( psUsed );
+}
+
+static SBR_PS_SIGNALING getSbrSignalingMode(
+        const AUDIO_OBJECT_TYPE          audioObjectType,
+        const TRANSPORT_TYPE             transportType,
+        const UCHAR                      transportSignaling,
+        const UINT                       sbrRatio
+        )
+
+{
+  SBR_PS_SIGNALING sbrSignaling;
+
+  if (transportType==TT_UNKNOWN || sbrRatio==0) {
+    sbrSignaling = SIG_UNKNOWN; /* Needed parameters have not been set */
+    return sbrSignaling;
+  } else {
+    sbrSignaling = SIG_IMPLICIT; /* default: implicit signaling */
+  }
+
+  if ((audioObjectType==AOT_AAC_LC)     || (audioObjectType==AOT_SBR)     || (audioObjectType==AOT_PS)    ||
+      (audioObjectType==AOT_MP2_AAC_LC) || (audioObjectType==AOT_MP2_SBR) || (audioObjectType==AOT_MP2_PS) ) {
+    switch (transportType) {
+      case TT_MP4_ADIF:
+      case TT_MP4_ADTS:
+        sbrSignaling = SIG_IMPLICIT; /* For MPEG-2 transport types, only implicit signaling is possible */
+        break;
+
+      case TT_MP4_RAW:
+      case TT_MP4_LATM_MCP1:
+      case TT_MP4_LATM_MCP0:
+      case TT_MP4_LOAS:
+      default:
+        if ( transportSignaling==0xFF ) {
+          /* Defaults */
+          if ( sbrRatio==1 ) {
+            sbrSignaling = SIG_EXPLICIT_HIERARCHICAL; /* For downsampled SBR, explicit signaling is mandatory */
+          } else {
+            sbrSignaling = SIG_IMPLICIT; /* For dual-rate SBR, implicit signaling is default */
+          }
+        } else {
+          /* User set parameters */
+          /* Attention: Backward compatible explicit signaling does only work with AMV1 for LATM/LOAS */
+          sbrSignaling = (SBR_PS_SIGNALING)transportSignaling;
+        }
+        break;
+    }
+  }
+
+  return sbrSignaling;
+}
+
 /****************************************************************************
                                Allocate Encoder
 ****************************************************************************/
@@ -266,8 +413,12 @@ C_ALLOC_MEM (_AacEncoder, AACENCODER, 1)
 /*
  * Map Encoder specific config structures to CODER_CONFIG.
  */
-static
-void FDKaacEnc_MapConfig(CODER_CONFIG *cc, USER_PARAM *extCfg, HANDLE_AACENC_CONFIG hAacConfig)
+static void FDKaacEnc_MapConfig(
+        CODER_CONFIG *const              cc,
+        const USER_PARAM *const          extCfg,
+        const SBR_PS_SIGNALING           sbrSignaling,
+        const HANDLE_AACENC_CONFIG       hAacConfig
+        )
 {
   AUDIO_OBJECT_TYPE transport_AOT = AOT_NULL_OBJECT;
   FDKmemclear(cc, sizeof(CODER_CONFIG));
@@ -304,17 +455,26 @@ void FDKaacEnc_MapConfig(CODER_CONFIG *cc, USER_PARAM *extCfg, HANDLE_AACENC_CON
   }
 
   /* Configure extension aot. */
-  if (extCfg->userTpSignaling==0) {
+  if (sbrSignaling==SIG_IMPLICIT) {
     cc->extAOT = AOT_NULL_OBJECT;  /* implicit */
   }
   else {
-    if ( (extCfg->userTpSignaling==1) && ( (transport_AOT==AOT_SBR) || (transport_AOT==AOT_PS) ) ) {
+    if ( (sbrSignaling==SIG_EXPLICIT_BW_COMPATIBLE) && ( (transport_AOT==AOT_SBR) || (transport_AOT==AOT_PS) ) ) {
       cc->extAOT = AOT_SBR;        /* explicit backward compatible */
     }
     else {
       cc->extAOT = transport_AOT;  /* explicit hierarchical */
     }
   }
+
+  if ( (transport_AOT==AOT_SBR) || (transport_AOT==AOT_PS) ) {
+    cc->sbrPresent=1;
+    if (transport_AOT==AOT_PS) {
+      cc->psPresent=1;
+    }
+  }
+  cc->sbrSignaling    = sbrSignaling;
+
   cc->extSamplingRate = extCfg->userSamplerate;
   cc->bitRate         = hAacConfig->bitRate;
   cc->noChannels      = hAacConfig->nChannels;
@@ -335,7 +495,7 @@ void FDKaacEnc_MapConfig(CODER_CONFIG *cc, USER_PARAM *extCfg, HANDLE_AACENC_CON
       case TT_MP4_ADTS:
       case TT_MP4_LOAS:
       case TT_MP4_LATM_MCP1:
-        cc->headerPeriod = 10;
+        cc->headerPeriod = DEFAULT_HEADER_PERIOD_REPETITION_RATE;
         break;
       default:
         cc->headerPeriod = 0;
@@ -351,7 +511,6 @@ void FDKaacEnc_MapConfig(CODER_CONFIG *cc, USER_PARAM *extCfg, HANDLE_AACENC_CON
     case AOT_MP2_SBR:
     case AOT_MP2_PS:
       cc->flags &= ~CC_MPEG_ID; /* Required for ADTS. */
-      //config->userTpSignaling=0;
       cc->extAOT = AOT_NULL_OBJECT;
       break;
     default:
@@ -411,7 +570,7 @@ AAC_ENCODER_ERROR aacEncDefaultConfig(HANDLE_AACENC_CONFIG hAacConfig,
     /* make reasonable default settings */
     FDKaacEnc_AacInitDefaultConfig (hAacConfig);
 
-    /* clear confure structure and copy default settings */
+    /* clear configuration structure and copy default settings */
     FDKmemclear(config, sizeof(USER_PARAM));
 
     /* copy encoder configuration settings */
@@ -438,7 +597,7 @@ AAC_ENCODER_ERROR aacEncDefaultConfig(HANDLE_AACENC_CONFIG hAacConfig,
     /* initialize transport parameters */
     config->userTpType         = TT_UNKNOWN;
     config->userTpAmxv         = 0;
-    config->userTpSignaling    = 0;    /* default, implicit signaling */
+    config->userTpSignaling    = 0xFF;    /* choose signaling automatically */
     config->userTpNsubFrames   = 1;
     config->userTpProtection   = 0;    /* not crc protected*/
     config->userTpHeaderPeriod = 0xFF; /* header period in auto mode */
@@ -447,6 +606,14 @@ AAC_ENCODER_ERROR aacEncDefaultConfig(HANDLE_AACENC_CONFIG hAacConfig,
 
     config->userAncDataRate    = 0;
 
+    /* SBR rate is set to 0 here, which means it should be set automatically
+       in FDKaacEnc_AdjustEncSettings() if the user did not set a rate
+       expilicitely. */
+    config->userSbrRatio = 0;
+
+    /* SBR enable set to -1 means to inquire ELD audio configurator for reasonable configuration. */
+    config->userSbrEnabled     = -1;
+
     return AAC_ENC_OK;
 }
 
@@ -481,6 +648,7 @@ INT aacEncoder_LimitBitrate(
         INT bitRate,
         const INT nSubFrames,
         const INT sbrActive,
+        const INT sbrDownSampleRate,
         const AUDIO_OBJECT_TYPE aot
         )
 {
@@ -490,8 +658,7 @@ INT aacEncoder_LimitBitrate(
   FDKaacEnc_InitChannelMapping(channelMode, CH_ORDER_MPEG, &cm);
 
   if (sbrActive) {
-    /* Assume SBR rate ratio of 2:1 */
-    coreSamplingRate = samplingRate / 2;
+    coreSamplingRate = samplingRate >>  (sbrEncoder_IsSingleRatePossible(aot) ? (sbrDownSampleRate-1):1);
   } else {
     coreSamplingRate = samplingRate;
   }
@@ -506,7 +673,7 @@ INT aacEncoder_LimitBitrate(
     bitRate = FDKmin(576000*nChannels, bitRate);
     /*bitRate = FDKmax(0*nChannels, bitRate);*/
   }
-  
+
 
   /* Limit bit rate in respect to the core coder */
   bitRate = FDKaacEnc_LimitBitrate(
@@ -525,56 +692,58 @@ INT aacEncoder_LimitBitrate(
   /* Limit bit rate in respect to available SBR modes if active */
   if (sbrActive)
   {
-    SBR_ELEMENT_INFO sbrElInfo[6];
-    INT sbrBitRate = 0;
-    int e, tooBig=-1;
+    int numIterations = 0;
+    INT initialBitrate, adjustedBitrate;
+    initialBitrate = adjustedBitrate = bitRate;
 
-    FDK_ASSERT(cm.nElements <= (6));
+    /* Find total bitrate which provides valid configuration for each SBR element. */
+    do {
+      int e;
+      SBR_ELEMENT_INFO sbrElInfo[(8)];
+      FDK_ASSERT(cm.nElements <= (8));
 
-    /* Get bit rate for each SBR element */
-    aacEncDistributeSbrBits(&cm, sbrElInfo, bitRate);
+      initialBitrate = adjustedBitrate;
 
-    for (e=0; e<cm.nElements; e++)
-    { 
-      INT sbrElementBitRateIn, sbrBitRateOut;
+      /* Get bit rate for each SBR element */
+      aacEncDistributeSbrBits(&cm, sbrElInfo, initialBitrate);
 
-      if (cm.elInfo[e].elType != ID_SCE && cm.elInfo[e].elType != ID_CPE) {
-        continue;
-      }
-      sbrElementBitRateIn = sbrElInfo[e].bitRate;
-      sbrBitRateOut = sbrEncoder_LimitBitRate(sbrElementBitRateIn , cm.elInfo[e].nChannelsInEl, coreSamplingRate, aot);
-      if (sbrBitRateOut == 0) {
-        return 0;
-      }
-      if (sbrElementBitRateIn < sbrBitRateOut) {
-        FDK_ASSERT(tooBig != 1);
-        tooBig = 0;
-        if (e == 0) {
-          sbrBitRate = 0;
-        }
-      }
-      if (sbrElementBitRateIn > sbrBitRateOut) {
-        FDK_ASSERT(tooBig != 0);
-        tooBig = 1;
-        if (e == 0) {
-          sbrBitRate = 5000000;
-        }
-      }
-      if (tooBig != -1)
+      for (e=0; e<cm.nElements; e++)
       {
-        INT sbrBitRateLimit = (INT)fDivNorm((FIXP_DBL)sbrBitRateOut, cm.elInfo[e].relativeBits);
-        if (tooBig) {
-          sbrBitRate = fMin(sbrBitRate, sbrBitRateLimit-16);
-          FDK_ASSERT( (INT)fMultNorm(cm.elInfo[e].relativeBits, (FIXP_DBL)sbrBitRate) < sbrBitRateOut);
-        } else {
-          sbrBitRate = fMax(sbrBitRate, sbrBitRateLimit+16);
-          FDK_ASSERT( (INT)fMultNorm(cm.elInfo[e].relativeBits, (FIXP_DBL)sbrBitRate) >= sbrBitRateOut);
+        INT sbrElementBitRateIn, sbrBitRateOut;
+
+        if (cm.elInfo[e].elType != ID_SCE && cm.elInfo[e].elType != ID_CPE) {
+          continue;
         }
-      }
-    }
-    if (tooBig != -1) {
-      bitRate = sbrBitRate;
-    }
+        sbrElementBitRateIn = sbrElInfo[e].bitRate;
+        sbrBitRateOut = sbrEncoder_LimitBitRate(sbrElementBitRateIn , cm.elInfo[e].nChannelsInEl, coreSamplingRate, aot);
+        if (sbrBitRateOut == 0) {
+          return 0;
+        }
+
+        /* If bitrates don't match, distribution and limiting needs to be determined again.
+           Abort element loop and restart with adapted bitrate. */
+        if (sbrElementBitRateIn != sbrBitRateOut) {
+
+          if (sbrElementBitRateIn < sbrBitRateOut) {
+            adjustedBitrate = fMax(initialBitrate, (INT)fDivNorm((FIXP_DBL)(sbrBitRateOut+8), cm.elInfo[e].relativeBits));
+            break;
+          }
+
+          if (sbrElementBitRateIn > sbrBitRateOut) {
+            adjustedBitrate = fMin(initialBitrate, (INT)fDivNorm((FIXP_DBL)(sbrBitRateOut-8), cm.elInfo[e].relativeBits));
+            break;
+          }
+
+        } /* sbrElementBitRateIn != sbrBitRateOut */
+
+      } /* elements */
+
+      numIterations++; /* restrict iteration to worst case of num elements */
+
+    } while ( (initialBitrate!=adjustedBitrate) && (numIterations<=cm.nElements) );
+
+    /* Unequal bitrates mean that no reasonable bitrate configuration found. */
+    bitRate = (initialBitrate==adjustedBitrate) ? adjustedBitrate : 0;
   }
 
   FDK_ASSERT(bitRate > 0);
@@ -624,26 +793,12 @@ AACENC_ERROR FDKaacEnc_AdjustEncSettings(HANDLE_AACENCODER hAacEncoder,
       case AOT_MP2_SBR:
       case AOT_MP2_PS:
           hAacConfig->usePns = 0;
-          if (config->userTpSignaling!=0) {
-            return AACENC_INVALID_CONFIG; /* only implicit signaling allowed */
-          }
       case AOT_AAC_LC:
       case AOT_SBR:
       case AOT_PS:
           config->userTpType = (config->userTpType!=TT_UNKNOWN) ? config->userTpType : TT_MP4_ADTS;
           hAacConfig->framelength = (config->userFramelength!=(UINT)-1) ? config->userFramelength : 1024;
-          if (hAacConfig->framelength != 1024 && hAacConfig->framelength != 960) {
-            return AACENC_INVALID_CONFIG;
-          }
-          break;
-      case AOT_ER_AAC_LC:
-          hAacConfig->epConfig = 0;
-          hAacConfig->syntaxFlags |= AC_ER;
-          hAacConfig->syntaxFlags |= ((config->userErTools & 0x1) ? AC_ER_VCB11 : 0);
-          hAacConfig->syntaxFlags |= ((config->userErTools & 0x2) ? AC_ER_HCR : 0);
-          config->userTpType = (config->userTpType!=TT_UNKNOWN) ? config->userTpType : TT_MP4_LOAS;
-          hAacConfig->framelength = (config->userFramelength!=(UINT)-1) ? config->userFramelength : 1024;
-          if (hAacConfig->framelength != 1024 && hAacConfig->framelength != 960) {
+          if (hAacConfig->framelength != 1024) {
             return AACENC_INVALID_CONFIG;
           }
           break;
@@ -665,7 +820,7 @@ AACENC_ERROR FDKaacEnc_AdjustEncSettings(HANDLE_AACENCODER hAacEncoder,
           hAacConfig->syntaxFlags |= ((config->userErTools & 0x1) ? AC_ER_VCB11 : 0);
           hAacConfig->syntaxFlags |= ((config->userErTools & 0x2) ? AC_ER_HCR : 0);
           hAacConfig->syntaxFlags |= ((config->userErTools & 0x4) ? AC_ER_RVLC : 0);
-          hAacConfig->syntaxFlags |= ((config->userSbrEnabled)    ? AC_SBR_PRESENT : 0);
+          hAacConfig->syntaxFlags |= ((config->userSbrEnabled==1)  ? AC_SBR_PRESENT : 0);
           config->userTpType = (config->userTpType!=TT_UNKNOWN) ? config->userTpType : TT_MP4_LOAS;
           hAacConfig->framelength = (config->userFramelength!=(UINT)-1) ? config->userFramelength : 512;
           if (hAacConfig->framelength != 512 && hAacConfig->framelength != 480) {
@@ -676,19 +831,6 @@ AACENC_ERROR FDKaacEnc_AdjustEncSettings(HANDLE_AACENCODER hAacEncoder,
           break;
     }
 
-    /* We need the frame length to call aacEncoder_LimitBitrate() */
-    hAacConfig->bitRate = aacEncoder_LimitBitrate(
-              NULL,
-              hAacConfig->sampleRate,
-              hAacConfig->framelength,
-              hAacConfig->nChannels,
-              hAacConfig->channelMode,
-              config->userBitrate,
-              hAacConfig->nSubFrames,
-              isSbrActive(hAacConfig),
-              hAacConfig->audioObjectType
-              );
-
     switch ( hAacConfig->audioObjectType ) {
       case AOT_ER_AAC_LD:
       case AOT_ER_AAC_ELD:
@@ -696,7 +838,7 @@ AACENC_ERROR FDKaacEnc_AdjustEncSettings(HANDLE_AACENCODER hAacEncoder,
           hAacConfig->bitrateMode = 0;
         }
         if (config->userBitrateMode==0) {
-          hAacConfig->bitreservoir = 50*config->nChannels; /* default, reduced bitreservoir */
+          hAacConfig->bitreservoir = 100*config->nChannels; /* default, reduced bitreservoir */
         }
         if (hAacConfig->bitrateMode!=0) {
           return AACENC_INVALID_CONFIG;
@@ -706,24 +848,7 @@ AACENC_ERROR FDKaacEnc_AdjustEncSettings(HANDLE_AACENCODER hAacEncoder,
         break;
     }
 
-    if (hAacConfig->epConfig >= 0) {
-        hAacConfig->syntaxFlags |= AC_ER;
-         if (((INT)hAacConfig->channelMode < 1) || ((INT)hAacConfig->channelMode > 7)) {
-           return AACENC_INVALID_CONFIG;        /* Cannel config 0 not supported. */
-         }
-    }
-
-    if ( FDKaacEnc_DetermineEncoderMode(&hAacConfig->channelMode, hAacConfig->nChannels) != AAC_ENC_OK) {
-        return AACENC_INVALID_CONFIG;        /* nChannels doesn't match chMode, this is just a check-up */
-    }
-
-    if ( (hAacConfig->nChannels > hAacEncoder->nMaxAacChannels)
-      || ( (FDKaacEnc_GetChannelModeConfiguration(hAacConfig->channelMode)->nChannelsEff > hAacEncoder->nMaxSbrChannels) &&
-            isSbrActive(hAacConfig) )
-         )
-    {
-        return AACENC_INVALID_CONFIG;      /* not enough channels allocated */
-    }
+    hAacConfig->bitRate = config->userBitrate;
 
     /* get bitrate in VBR configuration */
     if ( (hAacConfig->bitrateMode>=1) && (hAacConfig->bitrateMode<=5) ) {
@@ -734,25 +859,92 @@ AACENC_ERROR FDKaacEnc_AdjustEncSettings(HANDLE_AACENCODER hAacEncoder,
 
 
     /* Set default bitrate if no external bitrate declared. */
-    if (hAacConfig->bitRate==-1) {
+    if ( (hAacConfig->bitrateMode==0) && (config->userBitrate==(UINT)-1) ) {
         INT bitrate = FDKaacEnc_GetChannelModeConfiguration(hAacConfig->channelMode)->nChannelsEff * hAacConfig->sampleRate;
-        switch (hAacConfig->audioObjectType)
+
+        if ( isPsActive(hAacConfig->audioObjectType) ) {
+          hAacConfig->bitRate = (bitrate>>1);                  /* 0.5 bit per sample */
+        }
+        else if ( isSbrActive(hAacConfig) )
+        {
+          if ( (config->userSbrRatio==2) || ((config->userSbrRatio==0)&&(hAacConfig->audioObjectType!=AOT_ER_AAC_ELD)) ) {
+            hAacConfig->bitRate = (bitrate + (bitrate>>2))>>1; /* 0.625 bits per sample */
+          }
+          if ( (config->userSbrRatio==1) || ((config->userSbrRatio==0)&&(hAacConfig->audioObjectType==AOT_ER_AAC_ELD)) ) {
+            hAacConfig->bitRate = (bitrate + (bitrate>>3));    /* 1.125 bits per sample */
+          }
+        } else
         {
-            case AOT_AAC_LC:
                 hAacConfig->bitRate = bitrate + (bitrate>>1);        /* 1.5 bits per sample */
-                break;
-            case AOT_SBR:
-                hAacConfig->bitRate = (bitrate + (bitrate>>2))>>1;   /* 0.625 bits per sample */
-                break;
-            case AOT_PS:
-                hAacConfig->bitRate = (bitrate>>1);                  /* 0.5 bit per sample */
-                break;
-            default:
-                hAacConfig->bitRate = bitrate;
-                break;
         }
     }
 
+    /* Initialize SBR parameters */
+    if ( (hAacConfig->audioObjectType==AOT_ER_AAC_ELD)
+      && (config->userSbrEnabled == (UCHAR)-1) && (config->userSbrRatio==0) )
+    {
+      UINT eldSbr = 0;
+      UINT eldSbrRatio = 0;
+
+      if ( AACENC_OK!=(err=eldSbrConfigurator(
+            hAacConfig->sampleRate,
+            hAacConfig->channelMode,
+            hAacConfig->bitRate,
+           &eldSbr,
+           &eldSbrRatio)) )
+      {
+        return err;
+      }
+
+      hAacConfig->syntaxFlags |= ((eldSbr) ? AC_SBR_PRESENT : 0);
+      hAacConfig->sbrRatio = eldSbrRatio;
+    }
+    else
+    if ( (config->userSbrRatio==0) && (isSbrActive(hAacConfig)) ) {
+      /* Automatic SBR ratio configuration
+       * - downsampled SBR for ELD
+       * - otherwise always dualrate SBR
+       */
+        hAacConfig->sbrRatio = (hAacConfig->audioObjectType==AOT_ER_AAC_ELD) ? 1 : 2;
+    }
+    else {
+      /* SBR ratio has been set by the user, so use it. */
+      hAacConfig->sbrRatio = config->userSbrRatio;
+    }
+
+    {
+      UCHAR tpSignaling=getSbrSignalingMode(hAacConfig->audioObjectType, config->userTpType, config->userTpSignaling, hAacConfig->sbrRatio);
+
+      if ( (hAacConfig->audioObjectType==AOT_AAC_LC || hAacConfig->audioObjectType==AOT_SBR || hAacConfig->audioObjectType==AOT_PS) &&
+           (config->userTpType==TT_MP4_LATM_MCP1 || config->userTpType==TT_MP4_LATM_MCP0 || config->userTpType==TT_MP4_LOAS) &&
+           (tpSignaling==1) && (config->userTpAmxv==0) ) {
+             /* For backward compatible explicit signaling, AMV1 has to be active */
+             return AACENC_INVALID_CONFIG;
+      }
+
+      if ( (hAacConfig->audioObjectType==AOT_AAC_LC || hAacConfig->audioObjectType==AOT_SBR || hAacConfig->audioObjectType==AOT_PS) &&
+           (tpSignaling==0) && (hAacConfig->sbrRatio==1)) {
+             /* Downsampled SBR has to be signaled explicitely (for transmission of SBR sampling fequency) */
+             return AACENC_INVALID_CONFIG;
+      }
+    }
+
+
+
+    /* We need the frame length to call aacEncoder_LimitBitrate() */
+    hAacConfig->bitRate = aacEncoder_LimitBitrate(
+              NULL,
+              hAacConfig->sampleRate,
+              hAacConfig->framelength,
+              hAacConfig->nChannels,
+              hAacConfig->channelMode,
+              hAacConfig->bitRate,
+              hAacConfig->nSubFrames,
+              isSbrActive(hAacConfig),
+              hAacConfig->sbrRatio,
+              hAacConfig->audioObjectType
+              );
+
     /* Configure PNS */
     if ( ((hAacConfig->bitrateMode>=1) && (hAacConfig->bitrateMode<=5)) /* VBR without PNS. */
         || (hAacConfig->useTns == 0) )                                  /* TNS required. */
@@ -760,12 +952,32 @@ AACENC_ERROR FDKaacEnc_AdjustEncSettings(HANDLE_AACENCODER hAacEncoder,
         hAacConfig->usePns = 0;
     }
 
+    if (hAacConfig->epConfig >= 0) {
+        hAacConfig->syntaxFlags |= AC_ER;
+         if (((INT)hAacConfig->channelMode < 1) || ((INT)hAacConfig->channelMode > 7)) {
+           return AACENC_INVALID_CONFIG;        /* Cannel config 0 not supported. */
+         }
+    }
+
+    if ( FDKaacEnc_DetermineEncoderMode(&hAacConfig->channelMode, hAacConfig->nChannels) != AAC_ENC_OK) {
+        return AACENC_INVALID_CONFIG;        /* nChannels doesn't match chMode, this is just a check-up */
+    }
+
+    if ( (hAacConfig->nChannels > hAacEncoder->nMaxAacChannels)
+      || ( (FDKaacEnc_GetChannelModeConfiguration(hAacConfig->channelMode)->nChannelsEff > hAacEncoder->nMaxSbrChannels) &&
+            isSbrActive(hAacConfig) )
+         )
+    {
+        return AACENC_INVALID_CONFIG;      /* not enough channels allocated */
+    }
+
     /* Meta data restriction. */
     switch (hAacConfig->audioObjectType)
     {
       /* Allow metadata support */
       case AOT_AAC_LC:
       case AOT_SBR:
+      case AOT_PS:
         hAacEncoder->metaDataAllowed = 1;
         if (((INT)hAacConfig->channelMode < 1) || ((INT)hAacConfig->channelMode > 7)) {
           config->userMetaDataMode = 0;
@@ -849,10 +1061,8 @@ static AACENC_ERROR aacEncInit(HANDLE_AACENCODER  hAacEncoder,
         ((InitFlags & AACENC_INIT_CONFIG) || (InitFlags & AACENC_INIT_STATES)) )
     {
         INT sbrError;
-        SBR_ELEMENT_INFO sbrElInfo[(6)];
+        SBR_ELEMENT_INFO sbrElInfo[(8)];
         CHANNEL_MAPPING channelMapping;
-        
-        AUDIO_OBJECT_TYPE aot = hAacConfig->audioObjectType;
 
         if ( FDKaacEnc_InitChannelMapping(hAacConfig->channelMode,
                                           hAacConfig->channelOrder,
@@ -862,7 +1072,7 @@ static AACENC_ERROR aacEncInit(HANDLE_AACENCODER  hAacEncoder,
         }
 
         /* Check return value and if the SBR encoder can handle enough elements */
-        if (channelMapping.nElements > (6)) {
+        if (channelMapping.nElements > (8)) {
             return AACENC_INIT_ERROR;
         }
 
@@ -881,15 +1091,17 @@ static AACENC_ERROR aacEncInit(HANDLE_AACENCODER  hAacEncoder,
                                 &aacBufferOffset,
                                 &hAacConfig->nChannels,
                                 &hAacConfig->sampleRate,
+                                &hAacConfig->sbrRatio,
                                 &frameLength,
-                                &hAacConfig->audioObjectType,
+                                 hAacConfig->audioObjectType,
                                 &hAacEncoder->nDelay,
                                  (hAacConfig->audioObjectType == AOT_ER_AAC_ELD) ? 1 : TRANS_FAC,
+                                 (config->userTpHeaderPeriod!=0xFF) ? config->userTpHeaderPeriod : DEFAULT_HEADER_PERIOD_REPETITION_RATE,
                                  initFlag
                                 );
 
         /* Suppress AOT reconfiguration and check error status. */
-        if ( sbrError || (hAacConfig->audioObjectType!=aot) ) {
+        if (sbrError) {
             return AACENC_INIT_SBR_ERROR;
         }
 
@@ -915,7 +1127,11 @@ static AACENC_ERROR aacEncInit(HANDLE_AACENCODER  hAacEncoder,
     {
         UINT flags = 0;
 
-        FDKaacEnc_MapConfig(&hAacEncoder->coderConfig, config, hAacConfig);
+        FDKaacEnc_MapConfig(
+                &hAacEncoder->coderConfig,
+                config,
+                getSbrSignalingMode(hAacConfig->audioObjectType, config->userTpType, config->userTpSignaling, hAacConfig->sbrRatio),
+                hAacConfig);
 
         /* create flags for transport encoder */
         if (config->userTpAmxv == 1) {
@@ -958,7 +1174,7 @@ static AACENC_ERROR aacEncInit(HANDLE_AACENCODER  hAacEncoder,
         INT inputDataDelay = DELAY_AAC(hAacConfig->framelength);
 
         if ( isSbrActive(hAacConfig) && hSbrEncoder!=NULL) {
-            inputDataDelay = 2*inputDataDelay + sbrEncoder_GetInputDataDelay(*hSbrEncoder);
+          inputDataDelay = hAacConfig->sbrRatio*inputDataDelay + sbrEncoder_GetInputDataDelay(*hSbrEncoder);
         }
 
         if ( FDK_MetadataEnc_Init(hAacEncoder->hMetadataEnc,
@@ -1033,8 +1249,8 @@ AACENC_ERROR aacEncOpen(
 
     /* Determine max channel configuration. */
     if (maxChannels==0) {
-        hAacEncoder->nMaxAacChannels = (6);
-        hAacEncoder->nMaxSbrChannels = (6);
+        hAacEncoder->nMaxAacChannels = (8);
+        hAacEncoder->nMaxSbrChannels = (8);
     }
     else {
         hAacEncoder->nMaxAacChannels = (maxChannels&0x00FF);
@@ -1042,15 +1258,15 @@ AACENC_ERROR aacEncOpen(
             hAacEncoder->nMaxSbrChannels = (maxChannels&0xFF00) ? (maxChannels>>8) : hAacEncoder->nMaxAacChannels;
         }
 
-        if ( (hAacEncoder->nMaxAacChannels>(6)) || (hAacEncoder->nMaxSbrChannels>(6)) ) {
+        if ( (hAacEncoder->nMaxAacChannels>(8)) || (hAacEncoder->nMaxSbrChannels>(8)) ) {
             err = AACENC_INVALID_CONFIG;
             goto bail;
         }
     } /* maxChannels==0 */
 
     /* Max number of elements could be tuned any more. */
-    hAacEncoder->nMaxAacElements = fixMin((6), hAacEncoder->nMaxAacChannels);
-    hAacEncoder->nMaxSbrElements = fixMin((6), hAacEncoder->nMaxSbrChannels);
+    hAacEncoder->nMaxAacElements = fixMin((8), hAacEncoder->nMaxAacChannels);
+    hAacEncoder->nMaxSbrElements = fixMin((8), hAacEncoder->nMaxSbrChannels);
     hAacEncoder->nMaxSubFrames = (1);
 
 
@@ -1108,7 +1324,7 @@ AACENC_ERROR aacEncOpen(
         goto bail;
     }
     else {
-        C_ALLOC_SCRATCH_START(pLibInfo, LIB_INFO, FDK_MODULE_LAST); 
+        C_ALLOC_SCRATCH_START(pLibInfo, LIB_INFO, FDK_MODULE_LAST);
 
         FDKinitLibInfo( pLibInfo);
         transportEnc_GetLibInfo( pLibInfo );
@@ -1343,8 +1559,12 @@ AACENC_ERROR aacEncEncode(
         for (i=0; i<(INT)nMetaDataExtensions; i++) {  /* Get meta data extension payload. */
             hAacEncoder->extPayload[nExtensions++] = pMetaDataExtPayload[i];
         }
-        if (matrix_mixdown_idx!=-1) {            /* Set matrix mixdown coefficient. */
-          UINT pceValue = (UINT)( (1<<3) | ((matrix_mixdown_idx&0x2)<<1) | 1 );
+
+        if ( (matrix_mixdown_idx!=-1)
+          && ((hAacEncoder->extParam.userChannelMode==MODE_1_2_2)||(hAacEncoder->extParam.userChannelMode==MODE_1_2_2_1)) )
+        {
+          /* Set matrix mixdown coefficient. */
+          UINT pceValue = (UINT)( (1<<3) | ((matrix_mixdown_idx&0x3)<<1) | 1 );
           if (hAacEncoder->extParam.userPceAdditions != pceValue) {
             hAacEncoder->extParam.userPceAdditions = pceValue;
             hAacEncoder->InitFlags |= AACENC_INIT_TRANSPORT;
@@ -1375,7 +1595,7 @@ AACENC_ERROR aacEncEncode(
         }
         else {
             /* Add SBR extension payload */
-            for (i = 0; i < (6); i++) {
+            for (i = 0; i < (8); i++) {
                 if (hAacEncoder->extPayloadSize[nPayload][i] > 0) {
                     hAacEncoder->extPayload[nExtensions].pData    = hAacEncoder->extPayloadData[nPayload][i];
                     {
@@ -1573,7 +1793,6 @@ AACENC_ERROR aacEncoder_SetParam(
                 }
               case AOT_AAC_LC:
               case AOT_MP2_AAC_LC:
-              case AOT_ER_AAC_LC:
               case AOT_ER_AAC_LD:
               case AOT_ER_AAC_ELD:
                 if (!(hAacEncoder->encoder_modis & (ENC_MODE_FLAG_AAC))) {
@@ -1636,7 +1855,7 @@ AACENC_ERROR aacEncoder_SetParam(
             }
             if ( (pConfig->nElements > hAacEncoder->nMaxAacElements)
               || (pConfig->nChannelsEff > hAacEncoder->nMaxAacChannels)
-              || !((value>=1) && (value<=6))
+              || !(((value>=1) && (value<=7))||((value>=33) && (value<=34)))
                 )
             {
                 err = AACENC_INVALID_CONFIG;
@@ -1691,6 +1910,16 @@ AACENC_ERROR aacEncoder_SetParam(
           }
         }
         break;
+    case AACENC_SBR_RATIO:
+        if (settings->userSbrRatio != value) {
+            if (! ((value==0) || (value==1) || (value==2)) ) {
+              err = AACENC_INVALID_CONFIG;
+              break;
+            }
+            settings->userSbrRatio = value;
+            hAacEncoder->InitFlags |= AACENC_INIT_CONFIG | AACENC_INIT_STATES | AACENC_INIT_TRANSPORT;
+        }
+        break;
     case AACENC_SBR_MODE:
         if (settings->userSbrEnabled != value) {
             settings->userSbrEnabled = value;
@@ -1812,7 +2041,7 @@ UINT aacEncoder_GetParam(
         value = (UINT)hAacEncoder->aacConfig.bitrateMode;
         break;
     case AACENC_SAMPLERATE:
-        value = (UINT)settings->userSamplerate;
+        value = (UINT)hAacEncoder->coderConfig.extSamplingRate;
         break;
     case AACENC_CHANNELMODE:
         value = (UINT)hAacEncoder->aacConfig.channelMode;
@@ -1829,6 +2058,9 @@ UINT aacEncoder_GetParam(
     case AACENC_GRANULE_LENGTH:
         value = (UINT)hAacEncoder->aacConfig.framelength;
        break;
+    case AACENC_SBR_RATIO:
+        value = isSbrActive(&hAacEncoder->aacConfig) ? hAacEncoder->aacConfig.sbrRatio : 0;
+        break;
     case AACENC_SBR_MODE:
         value = (UINT) (hAacEncoder->aacConfig.syntaxFlags & AC_SBR_PRESENT) ? 1 : 0;
         break;
@@ -1836,7 +2068,7 @@ UINT aacEncoder_GetParam(
         value = (UINT)settings->userTpType;
         break;
     case AACENC_SIGNALING_MODE:
-        value = (UINT)settings->userTpSignaling;
+        value = (UINT)getSbrSignalingMode(hAacEncoder->aacConfig.audioObjectType, settings->userTpType, settings->userTpSignaling, hAacEncoder->aacConfig.sbrRatio);
         break;
     case AACENC_PROTECTION:
         value = (UINT)settings->userTpProtection;
diff --git a/libAACenc/src/aacenc_pns.cpp b/libAACenc/src/aacenc_pns.cpp
index 4d81268..b9640d9 100644
--- a/libAACenc/src/aacenc_pns.cpp
+++ b/libAACenc/src/aacenc_pns.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/aacenc_pns.h b/libAACenc/src/aacenc_pns.h
index ce82071..3bda9de 100644
--- a/libAACenc/src/aacenc_pns.h
+++ b/libAACenc/src/aacenc_pns.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/aacenc_tns.cpp b/libAACenc/src/aacenc_tns.cpp
index 00b2bca..85aea65 100644
--- a/libAACenc/src/aacenc_tns.cpp
+++ b/libAACenc/src/aacenc_tns.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -83,7 +83,7 @@ amm-info at iis.fraunhofer.de
 
 /******************************** MPEG Audio Encoder **************************
 
-   Initial author:       Alex Groeschel
+   Initial author:       Alex Groeschel, Tobias Chalupka
    contents/description: Temporal noise shaping
 
 ******************************************************************************/
@@ -281,7 +281,6 @@ static INT getTnsMaxBands(
   int maxBandsTabSize = 0;
 
   switch (granuleLength) {
-    case 960:
     case 1024:
       pMaxBandsTab = tnsMaxBandsTab1024;
       maxBandsTabSize = sizeof(tnsMaxBandsTab1024)/sizeof(TNS_MAX_TAB_ENTRY);
@@ -404,7 +403,6 @@ AAC_ENCODER_ERROR FDKaacEnc_InitTnsConfiguration(INT bitRate,
   tC->lpcStopLine    = pC->sfbOffset[tC->lpcStopBand];
 
   switch (granuleLength) {
-    case 960:
     case 1024:
       /* TNS start line: skip lower MDCT lines to prevent artifacts due to filter mismatch */
       tC->lpcStartBand[LOFILT]   = (blockType == SHORT_WINDOW) ? 0 : ((sampleRate < 18783) ? 4 : 8);
diff --git a/libAACenc/src/aacenc_tns.h b/libAACenc/src/aacenc_tns.h
index 519fd69..f2b731f 100644
--- a/libAACenc/src/aacenc_tns.h
+++ b/libAACenc/src/aacenc_tns.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/adj_thr.cpp b/libAACenc/src/adj_thr.cpp
index a779357..6433633 100644
--- a/libAACenc/src/adj_thr.cpp
+++ b/libAACenc/src/adj_thr.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -123,6 +123,117 @@ static const FIXP_DBL  SnrLdMin4 = (FIXP_DBL)0x02000000; /*FL2FXCONST_DBL(FDKlog
 static const FIXP_DBL  SnrLdMin5 = (FIXP_DBL)0xfc000000; /*FL2FXCONST_DBL(FDKlog(0.25) /FDKlog(2.0)/LD_DATA_SCALING);*/
 
 
+/*
+The bits2Pe factors are choosen for the case that some times
+the crash recovery strategy will be activated once.
+*/
+
+typedef struct {
+  INT                 bitrate;
+  LONG                bits2PeFactor_mono;
+  LONG                bits2PeFactor_mono_slope;
+  LONG                bits2PeFactor_stereo;
+  LONG                bits2PeFactor_stereo_slope;
+  LONG                bits2PeFactor_mono_scfOpt;
+  LONG                bits2PeFactor_mono_scfOpt_slope;
+  LONG                bits2PeFactor_stereo_scfOpt;
+  LONG                bits2PeFactor_stereo_scfOpt_slope;
+
+} BIT_PE_SFAC;
+
+typedef struct {
+  const INT           sampleRate;
+  const BIT_PE_SFAC * pPeTab;
+  const INT           nEntries;
+
+} BITS2PE_CFG_TAB;
+
+static const BIT_PE_SFAC S_Bits2PeTab16000[] = {
+  { 10000, 0x228F5C29, 0x02FEF55D, 0x1D70A3D7, 0x09BC9D6D, 0x228F5C29, 0x02FEF55D, 0x1C28F5C3, 0x0CBB92CA},
+  { 24000, 0x23D70A3D, 0x029F16B1, 0x2199999A, 0x07DD4413, 0x23D70A3D, 0x029F16B1, 0x2199999A, 0x07DD4413},
+  { 32000, 0x247AE148, 0x11B1D92B, 0x23851EB8, 0x01F75105, 0x247AE148, 0x110A137F, 0x23851EB8, 0x01F75105},
+  { 48000, 0x2D1EB852, 0x6833C600, 0x247AE148, 0x014F8B59, 0x2CCCCCCD, 0x68DB8BAC, 0x247AE148, 0x01F75105},
+  { 64000, 0x60000000, 0x00000000, 0x251EB852, 0x154C985F, 0x60000000, 0x00000000, 0x2570A3D7, 0x154C985F},
+  { 96000, 0x60000000, 0x00000000, 0x39EB851F, 0x088509C0, 0x60000000, 0x00000000, 0x3A3D70A4, 0x088509C0},
+  {128000, 0x60000000, 0x00000000, 0x423D70A4, 0x18A43BB4, 0x60000000, 0x00000000, 0x428F5C29, 0x181E03F7},
+  {148000, 0x60000000, 0x00000000, 0x5147AE14, 0x00000000, 0x60000000, 0x00000000, 0x5147AE14, 0x00000000}
+};
+
+static const BIT_PE_SFAC S_Bits2PeTab22050[] = {
+  { 16000, 0x1a8f5c29, 0x1797cc3a, 0x128f5c29, 0x18e75793, 0x175c28f6, 0x221426fe, 0x00000000, 0x5a708ede},
+  { 24000, 0x2051eb85, 0x092ccf6c, 0x18a3d70a, 0x13a92a30, 0x1fae147b, 0xbcbe61d,  0x16147ae1, 0x18e75793},
+  { 32000, 0x228f5c29, 0x029f16b1, 0x1d70a3d7, 0x088509c0, 0x228f5c29, 0x29f16b1,  0x1c28f5c3, 0x0b242071},
+  { 48000, 0x23d70a3d, 0x014f8b59, 0x2199999a, 0x03eea20a, 0x23d70a3d, 0x14f8b59,  0x2199999a, 0x03eea20a},
+  { 64000, 0x247ae148, 0x08d8ec96, 0x23851eb8, 0x00fba882, 0x247ae148, 0x88509c0,  0x23851eb8, 0x00fba882},
+  { 96000, 0x2d1eb852, 0x3419e300, 0x247ae148, 0x00a7c5ac, 0x2ccccccd, 0x346dc5d6, 0x247ae148, 0x00fba882},
+  {128000, 0x60000000, 0x00000000, 0x251eb852, 0x029f16b1, 0x60000000, 0x00000000, 0x2570a3d7, 0x009f16b1},
+  {148000, 0x60000000, 0x00000000, 0x26b851ec, 0x00000000, 0x60000000, 0x00000000, 0x270a3d71, 0x00000000}
+};
+
+static const BIT_PE_SFAC S_Bits2PeTab24000[] = {
+  { 16000, 0x19eb851f, 0x13a92a30, 0x1147ae14, 0x164840e1, 0x1999999a, 0x12599ed8, 0x00000000, 0x46c764ae},
+  { 24000, 0x1eb851ec, 0x0d1b7176, 0x16b851ec, 0x18e75793, 0x1e147ae1, 0x0fba8827, 0x1147ae14, 0x2c9081c3},
+  { 32000, 0x21eb851f, 0x049667b6, 0x1ccccccd, 0x07357e67, 0x21eb851f, 0x03eea20a, 0x1c28f5c3, 0x07357e67},
+  { 48000, 0x2428f5c3, 0x014f8b59, 0x2051eb85, 0x053e2d62, 0x23d70a3d, 0x01f75105, 0x1fae147b, 0x07357e67},
+  { 64000, 0x24cccccd, 0x05e5f30e, 0x22e147ae, 0x01a36e2f, 0x24cccccd, 0x05e5f30e, 0x23333333, 0x014f8b59},
+  { 96000, 0x2a8f5c29, 0x24b33db0, 0x247ae148, 0x00fba882, 0x2a8f5c29, 0x26fe718b, 0x247ae148, 0x00fba882},
+  {128000, 0x4e666666, 0x1cd5f99c, 0x2570a3d7, 0x010c6f7a, 0x50a3d70a, 0x192a7371, 0x2570a3d7, 0x010c6f7a},
+  {148000, 0x60000000, 0x00000000, 0x26147ae1, 0x00000000, 0x60000000, 0x00000000, 0x26147ae1, 0x00000000}
+};
+
+static const BIT_PE_SFAC S_Bits2PeTab32000[] = {
+  { 16000, 0x1199999a, 0x20c49ba6, 0x00000000, 0x4577d955, 0x00000000, 0x60fe4799, 0x00000000, 0x00000000},
+  { 24000, 0x1999999a, 0x0fba8827, 0x10f5c28f, 0x1b866e44, 0x17ae147b, 0x0fba8827, 0x00000000, 0x4d551d69},
+  { 32000, 0x1d70a3d7, 0x07357e67, 0x17ae147b, 0x09d49518, 0x1b851eb8, 0x0a7c5ac4, 0x12e147ae, 0x110a137f},
+  { 48000, 0x20f5c28f, 0x049667b6, 0x1c7ae148, 0x053e2d62, 0x20a3d70a, 0x053e2d62, 0x1b333333, 0x05e5f30e},
+  { 64000, 0x23333333, 0x029f16b1, 0x1f0a3d71, 0x02f2f987, 0x23333333, 0x029f16b1, 0x1e147ae1, 0x03eea20a},
+  { 96000, 0x25c28f5c, 0x2c3c9eed, 0x21eb851f, 0x01f75105, 0x25c28f5c, 0x0a7c5ac4, 0x21eb851f, 0x01a36e2f},
+  {128000, 0x50f5c28f, 0x18a43bb4, 0x23d70a3d, 0x010c6f7a, 0x30000000, 0x168b5cc0, 0x23851eb8, 0x0192a737},
+  {148000, 0x60000000, 0x00000000, 0x247ae148, 0x00dfb23b, 0x3dc28f5c, 0x300f4aaf, 0x247ae148, 0x01bf6476},
+  {160000, 0x60000000, 0xb15b5740, 0x24cccccd, 0x053e2d62, 0x4f5c28f6, 0xbefd0072, 0x251eb852, 0x04fb1184},
+  {200000, 0x00000000, 0x00000000, 0x2b333333, 0x0836be91, 0x00000000, 0x00000000, 0x2b333333, 0x0890390f},
+  {320000, 0x00000000, 0x00000000, 0x4947ae14, 0x00000000, 0x00000000, 0x00000000, 0x4a8f5c29, 0x00000000}
+};
+
+static const BIT_PE_SFAC S_Bits2PeTab44100[] = {
+  { 16000, 0x10a3d70a, 0x1797cc3a, 0x00000000, 0x00000000, 0x00000000, 0x59210386, 0x00000000, 0x00000000},
+  { 24000, 0x16666666, 0x1797cc3a, 0x00000000, 0x639d5e4a, 0x15c28f5c, 0x12599ed8, 0x00000000, 0x5bc01a37},
+  { 32000, 0x1c28f5c3, 0x049667b6, 0x1851eb85, 0x049667b6, 0x1a3d70a4, 0x088509c0, 0x16666666, 0x053e2d62},
+  { 48000, 0x1e666666, 0x05e5f30e, 0x1a8f5c29, 0x049667b6, 0x1e666666, 0x05e5f30e, 0x18f5c28f, 0x05e5f30e},
+  { 64000, 0x2147ae14, 0x0346dc5d, 0x1ccccccd, 0x02f2f987, 0x2147ae14, 0x02f2f987, 0x1bd70a3d, 0x039abf34},
+  { 96000, 0x247ae148, 0x068db8bb, 0x1fae147b, 0x029f16b1, 0x2428f5c3, 0x0639d5e5, 0x1f5c28f6, 0x029f16b1},
+  {128000, 0x2ae147ae, 0x1b435265, 0x223d70a4, 0x0192a737, 0x2a3d70a4, 0x1040bfe4, 0x21eb851f, 0x0192a737},
+  {148000, 0x3b851eb8, 0x2832069c, 0x23333333, 0x00dfb23b, 0x3428f5c3, 0x2054c288, 0x22e147ae, 0x00dfb23b},
+  {160000, 0x4a3d70a4, 0xc32ebe5a, 0x23851eb8, 0x01d5c316, 0x40000000, 0xcb923a2b, 0x23333333, 0x01d5c316},
+  {200000, 0x00000000, 0x00000000, 0x25c28f5c, 0x0713f078, 0x00000000, 0x00000000, 0x2570a3d7, 0x072a4f17},
+  {320000, 0x00000000, 0x00000000, 0x3fae147b, 0x00000000, 0x00000000, 0x00000000, 0x3fae147b, 0x00000000}
+};
+
+static const BIT_PE_SFAC S_Bits2PeTab48000[] = {
+  { 16000, 0x0f5c28f6, 0x31ceaf25, 0x00000000, 0x00000000, 0x00000000, 0x74a771c9, 0x00000000, 0x00000000},
+  { 24000, 0x1b851eb8, 0x029f16b1, 0x00000000, 0x663c74fb, 0x1c7ae148, 0xe47991bd, 0x00000000, 0x49667b5f},
+  { 32000, 0x1c28f5c3, 0x029f16b1, 0x18f5c28f, 0x07357e67, 0x15c28f5c, 0x0f12c27a, 0x11eb851f, 0x13016484},
+  { 48000, 0x1d70a3d7, 0x053e2d62, 0x1c7ae148, 0xfe08aefc, 0x1d1eb852, 0x068db8bb, 0x1b333333, 0xfeb074a8},
+  { 64000, 0x20000000, 0x03eea20a, 0x1b851eb8, 0x0346dc5d, 0x2051eb85, 0x0346dc5d, 0x1a8f5c29, 0x039abf34},
+  { 96000, 0x23d70a3d, 0x053e2d62, 0x1eb851ec, 0x029f16b1, 0x23851eb8, 0x04ea4a8c, 0x1e147ae1, 0x02f2f987},
+  {128000, 0x28f5c28f, 0x14727dcc, 0x2147ae14, 0x0218def4, 0x2851eb85, 0x0e27e0f0, 0x20f5c28f, 0x0218def4},
+  {148000, 0x3570a3d7, 0x1cd5f99c, 0x228f5c29, 0x01bf6476, 0x30f5c28f, 0x18777e75, 0x223d70a4, 0x01bf6476},
+  {160000, 0x40000000, 0xcb923a2b, 0x23333333, 0x0192a737, 0x39eb851f, 0xd08d4bae, 0x22e147ae, 0x0192a737},
+  {200000, 0x00000000, 0x00000000, 0x251eb852, 0x06775a1b, 0x00000000, 0x00000000, 0x24cccccd, 0x06a4175a},
+  {320000, 0x00000000, 0x00000000, 0x3ccccccd, 0x00000000, 0x00000000, 0x00000000, 0x3d1eb852, 0x00000000}
+};
+
+static const BITS2PE_CFG_TAB bits2PeConfigTab[] = {
+  { 16000, S_Bits2PeTab16000, sizeof(S_Bits2PeTab16000)/sizeof(BIT_PE_SFAC) },
+  { 22050, S_Bits2PeTab22050, sizeof(S_Bits2PeTab22050)/sizeof(BIT_PE_SFAC) },
+  { 24000, S_Bits2PeTab24000, sizeof(S_Bits2PeTab24000)/sizeof(BIT_PE_SFAC) },
+  { 32000, S_Bits2PeTab32000, sizeof(S_Bits2PeTab32000)/sizeof(BIT_PE_SFAC) },
+  { 44100, S_Bits2PeTab44100, sizeof(S_Bits2PeTab44100)/sizeof(BIT_PE_SFAC) },
+  { 48000, S_Bits2PeTab48000, sizeof(S_Bits2PeTab48000)/sizeof(BIT_PE_SFAC) }
+};
+
+
+
 /* values for avoid hole flag */
 enum _avoid_hole_state {
     NO_AH              =0,
@@ -135,6 +246,99 @@ enum _avoid_hole_state {
 #define Q_BITFAC    (24)   /* Q scaling used in FDKaacEnc_bitresCalcBitFac() calculation */
 #define Q_AVGBITS   (17)   /* scale bit values */
 
+
+/*****************************************************************************
+    functionname: FDKaacEnc_InitBits2PeFactor
+    description:  retrieve bits2PeFactor from table
+*****************************************************************************/
+static void FDKaacEnc_InitBits2PeFactor(
+        FIXP_DBL *bits2PeFactor_m,
+        INT *bits2PeFactor_e,
+        const INT bitRate,
+        const INT nChannels,
+        const INT sampleRate,
+        const INT advancedBitsToPe,
+        const INT invQuant
+        )
+{
+  /* default bits2pe factor */
+  FIXP_DBL bit2PE_m = FL2FXCONST_DBL(1.18f/(1<<(1)));
+  INT      bit2PE_e = 1;
+
+  /* make use of advanced bits to pe factor table */
+  if (advancedBitsToPe) {
+
+    int i;
+    const BIT_PE_SFAC *peTab = NULL;
+    INT size = 0;
+
+
+    /* Get correct table entry */
+    for (i=0; i<(INT)(sizeof(bits2PeConfigTab)/sizeof(BITS2PE_CFG_TAB)); i++) {
+      if (sampleRate >= bits2PeConfigTab[i].sampleRate) {
+        peTab = bits2PeConfigTab[i].pPeTab;
+        size  = bits2PeConfigTab[i].nEntries;
+      }
+    }
+
+    if ( (peTab!=NULL) && (size!=0) ) {
+
+      INT startB      = -1;
+      LONG startPF    = 0;
+      LONG peSlope    = 0;
+
+      /* stereo or mono mode and invQuant used or not */
+      for (i=0; i<size-1; i++)
+      {
+        if ((peTab[i].bitrate<=bitRate) && ((peTab[i+1].bitrate>bitRate) || ((i==size-2)) ))
+        {
+          if (nChannels==1)
+          {
+            startPF = (!invQuant) ? peTab[i].bits2PeFactor_mono   : peTab[i].bits2PeFactor_mono_scfOpt;
+            peSlope = (!invQuant) ? peTab[i].bits2PeFactor_mono_slope : peTab[i].bits2PeFactor_mono_scfOpt_slope;
+            /*endPF   = (!invQuant) ? peTab[i+1].bits2PeFactor_mono : peTab[i+1].bits2PeFactor_mono_scfOpt;
+            endB=peTab[i+1].bitrate;*/
+            startB=peTab[i].bitrate;
+            break;
+          }
+          else
+          {
+            startPF = (!invQuant) ? peTab[i].bits2PeFactor_stereo   : peTab[i].bits2PeFactor_stereo_scfOpt;
+            peSlope = (!invQuant) ? peTab[i].bits2PeFactor_stereo_slope : peTab[i].bits2PeFactor_stereo_scfOpt_slope;
+            /*endPF   = (!invQuant) ? peTab[i+1].bits2PeFactor_stereo : peTab[i+1].bits2PeFactor_stereo_scfOpt;
+            endB=peTab[i+1].bitrate;*/
+            startB=peTab[i].bitrate;
+            break;
+          }
+        }
+      } /* for i */
+
+      /* if a configuration is available */
+      if (startB!=-1) {
+        /* linear interpolate to actual PEfactor */
+        FIXP_DBL peFac = fMult((FIXP_DBL)(bitRate-startB)<<14, (FIXP_DBL)peSlope) << 2;
+        FIXP_DBL bit2PE = peFac + (FIXP_DBL)startPF; /* startPF_float = startPF << 2 */
+
+        /* sanity check if bits2pe value is high enough */
+        if ( bit2PE >= (FL2FXCONST_DBL(0.35f) >> 2) ) {
+          bit2PE_m = bit2PE;
+          bit2PE_e = 2; /*  table is fixed scaled */
+        }
+      } /* br */
+    } /* sr */
+  } /* advancedBitsToPe */
+
+
+  /* return bits2pe factor */
+  *bits2PeFactor_m = bit2PE_m;
+  *bits2PeFactor_e = bit2PE_e;
+}
+
+
+/*****************************************************************************
+functionname: FDKaacEnc_bits2pe2
+description:  convert from bits to pe
+*****************************************************************************/
 static INT FDKaacEnc_bits2pe2(
         const INT                 bits,
         const FIXP_DBL            factor_m,
@@ -450,22 +654,23 @@ void FDKaacEnc_calcWeighting(PE_DATA *peData,
             FIXP_DBL nrgSum14, nrgSum12, nrgSum34, nrgTotal;
             FIXP_DBL nrgFacLd_14, nrgFacLd_12, nrgFacLd_34;
             INT usePatch, exePatch;
-            int sfb, nLinesSum = 0;
+            int sfb, sfbGrp, nLinesSum = 0;
 
             nrgSum14 = nrgSum12 = nrgSum34 = nrgTotal = FL2FXCONST_DBL(0.f);
 
             /* calculate flatness of audible spectrum, i.e. spectrum above masking threshold. */
-            for (sfb = 0; sfb < psyOutChan->sfbCnt; sfb++) {
-
-                FIXP_DBL nrgFac12 = CalcInvLdData(psyOutChan->sfbEnergyLdData[sfb]>>1);   /* nrg^(1/2) */
-                FIXP_DBL nrgFac14 = CalcInvLdData(psyOutChan->sfbEnergyLdData[sfb]>>2);   /* nrg^(1/4) */
+            for (sfbGrp = 0;sfbGrp < psyOutChannel[ch]->sfbCnt; sfbGrp+=psyOutChannel[ch]->sfbPerGroup) {
+              for (sfb=0; sfb<psyOutChannel[ch]->maxSfbPerGroup; sfb++) {
+                FIXP_DBL nrgFac12 = CalcInvLdData(psyOutChan->sfbEnergyLdData[sfbGrp+sfb]>>1); /* nrg^(1/2) */
+                FIXP_DBL nrgFac14 = CalcInvLdData(psyOutChan->sfbEnergyLdData[sfbGrp+sfb]>>2); /* nrg^(1/4) */
 
                 /* maximal number of bands is 64, results scaling factor 6 */
-                nLinesSum += peData->peChannelData[ch].sfbNLines[sfb];                    /* relevant lines */
-                nrgTotal  += ( psyOutChan->sfbEnergy[sfb] >> 6 );                         /* sum up nrg */
+                nLinesSum += peData->peChannelData[ch].sfbNLines[sfbGrp+sfb];             /* relevant lines */
+                nrgTotal  += ( psyOutChan->sfbEnergy[sfbGrp+sfb] >> 6 );                  /* sum up nrg */
                 nrgSum12  += ( nrgFac12 >> 6 );                                           /* sum up nrg^(2/4) */
                 nrgSum14  += ( nrgFac14 >> 6 );                                           /* sum up nrg^(1/4) */
                 nrgSum34  += ( fMult(nrgFac14, nrgFac12) >> 6 );                          /* sum up nrg^(3/4) */
+              }
             }
 
             nrgTotal = CalcLdData(nrgTotal);                                              /* get ld64 of total nrg */
@@ -479,32 +684,35 @@ void FDKaacEnc_calcWeighting(PE_DATA *peData,
             usePatch = (adjThrStateElement->chaosMeasureEnFac[ch] > FL2FXCONST_DBL(0.78125f));
             exePatch = ((usePatch) && (adjThrStateElement->lastEnFacPatch[ch]));
 
-            for (sfb = 0; sfb < psyOutChan->sfbCnt; sfb++) {
+            for (sfbGrp = 0;sfbGrp < psyOutChannel[ch]->sfbCnt; sfbGrp+=psyOutChannel[ch]->sfbPerGroup) {
+              for (sfb=0; sfb<psyOutChannel[ch]->maxSfbPerGroup; sfb++) {
+
                 INT sfbExePatch;
 
                 /* for MS coupled SFBs, also execute patch in side channel if done in mid channel */
-                if ((ch == 1) && (toolsInfo->msMask[sfb])) {
+                if ((ch == 1) && (toolsInfo->msMask[sfbGrp+sfb])) {
                     sfbExePatch = exePatchM;
                 }
                 else {
                     sfbExePatch = exePatch;
                 }
 
-                if ( (sfbExePatch) && (psyOutChan->sfbEnergy[sfb]>FL2FXCONST_DBL(0.f)) )
+                if ( (sfbExePatch) && (psyOutChan->sfbEnergy[sfbGrp+sfb]>FL2FXCONST_DBL(0.f)) )
                 {
                     /* execute patch based on spectral flatness calculated above */
                     if (adjThrStateElement->chaosMeasureEnFac[ch] > FL2FXCONST_DBL(0.8125f)) {
-                        qcOutChannel[ch]->sfbEnFacLd[sfb] = ( (nrgFacLd_14 + (psyOutChan->sfbEnergyLdData[sfb]+(psyOutChan->sfbEnergyLdData[sfb]>>1)))>>1 ); /* sfbEnergy^(3/4) */
+                        qcOutChannel[ch]->sfbEnFacLd[sfbGrp+sfb] = ( (nrgFacLd_14 + (psyOutChan->sfbEnergyLdData[sfbGrp+sfb]+(psyOutChan->sfbEnergyLdData[sfbGrp+sfb]>>1)))>>1 ); /* sfbEnergy^(3/4) */
                     }
                     else if (adjThrStateElement->chaosMeasureEnFac[ch] > FL2FXCONST_DBL(0.796875f)) {
-                        qcOutChannel[ch]->sfbEnFacLd[sfb] = ( (nrgFacLd_12 + psyOutChan->sfbEnergyLdData[sfb])>>1 );          /* sfbEnergy^(2/4) */
+                        qcOutChannel[ch]->sfbEnFacLd[sfbGrp+sfb] = ( (nrgFacLd_12 + psyOutChan->sfbEnergyLdData[sfbGrp+sfb])>>1 );          /* sfbEnergy^(2/4) */
                     }
                     else {
-                        qcOutChannel[ch]->sfbEnFacLd[sfb] = ( (nrgFacLd_34 + (psyOutChan->sfbEnergyLdData[sfb]>>1))>>1 );     /* sfbEnergy^(1/4) */
+                        qcOutChannel[ch]->sfbEnFacLd[sfbGrp+sfb] = ( (nrgFacLd_34 + (psyOutChan->sfbEnergyLdData[sfbGrp+sfb]>>1))>>1 );     /* sfbEnergy^(1/4) */
                     }
-                    qcOutChannel[ch]->sfbEnFacLd[sfb] = fixMin(qcOutChannel[ch]->sfbEnFacLd[sfb],(FIXP_DBL)0);
+                    qcOutChannel[ch]->sfbEnFacLd[sfbGrp+sfb] = fixMin(qcOutChannel[ch]->sfbEnFacLd[sfbGrp+sfb],(FIXP_DBL)0);
 
                 }
+              }
             } /* sfb loop */
 
             adjThrStateElement->lastEnFacPatch[ch] = usePatch;
@@ -735,7 +943,6 @@ static FIXP_DBL FDKaacEnc_calcChaosMeasure(PSY_OUT_CHANNEL *psyOutChannel,
   return chaosMeasure;
 }
 
-
 /* apply reduction formula for VBR-mode */
 static void FDKaacEnc_reduceThresholdsVBR(QC_OUT_CHANNEL* qcOutChannel[(2)],
                                 PSY_OUT_CHANNEL* psyOutChannel[(2)],
@@ -923,7 +1130,6 @@ static void FDKaacEnc_reduceThresholdsVBR(QC_OUT_CHANNEL* qcOutChannel[(2)],
   }
 }
 
-
 /*****************************************************************************
 functionname: FDKaacEnc_correctThresh
 description:  if pe difference deltaPe between desired pe and real pe is small enough,
@@ -931,12 +1137,12 @@ the difference can be distributed among the scale factor bands.
 New thresholds can be derived from this pe-difference
 *****************************************************************************/
 static void FDKaacEnc_correctThresh(CHANNEL_MAPPING* cm,
-                          QC_OUT_ELEMENT*  qcElement[(6)],
-                          PSY_OUT_ELEMENT* psyOutElement[(6)],
-                          UCHAR            ahFlag[(6)][(2)][MAX_GROUPED_SFB],
-                          FIXP_DBL         thrExp[(6)][(2)][MAX_GROUPED_SFB],
-                          const            FIXP_DBL redVal[(6)],
-                          const            SCHAR redValScaling[(6)],
+                          QC_OUT_ELEMENT*  qcElement[(8)],
+                          PSY_OUT_ELEMENT* psyOutElement[(8)],
+                          UCHAR            ahFlag[(8)][(2)][MAX_GROUPED_SFB],
+                          FIXP_DBL         thrExp[(8)][(2)][MAX_GROUPED_SFB],
+                          const            FIXP_DBL redVal[(8)],
+                          const            SCHAR redValScaling[(8)],
                           const            INT deltaPe,
                           const            INT processElements,
                           const            INT elementOffset)
@@ -947,8 +1153,8 @@ static void FDKaacEnc_correctThresh(CHANNEL_MAPPING* cm,
    PE_CHANNEL_DATA *peChanData;
    FIXP_DBL thrFactorLdData;
    FIXP_DBL sfbEnLdData, sfbThrLdData, sfbThrReducedLdData;
-   FIXP_DBL *sfbPeFactorsLdData[(6)][(2)];
-   FIXP_DBL sfbNActiveLinesLdData[(2)][MAX_GROUPED_SFB];
+   FIXP_DBL *sfbPeFactorsLdData[(8)][(2)];
+   FIXP_DBL sfbNActiveLinesLdData[(8)][(2)][MAX_GROUPED_SFB];
    INT      normFactorInt;
    FIXP_DBL normFactorLdData;
 
@@ -979,13 +1185,13 @@ static void FDKaacEnc_correctThresh(CHANNEL_MAPPING* cm,
             for (sfb=0; sfb<psyOutChan->maxSfbPerGroup; sfb++) {
 
              if ( peChanData->sfbNActiveLines[sfbGrp+sfb] == 0 ) {
-                sfbNActiveLinesLdData[ch][sfbGrp+sfb] = FL2FXCONST_DBL(-1.0f);
+                sfbNActiveLinesLdData[elementId][ch][sfbGrp+sfb] = FL2FXCONST_DBL(-1.0f);
              }
              else {
                 /* Both CalcLdInt and CalcLdData can be used!
                  * No offset has to be subtracted, because sfbNActiveLinesLdData
                  * is shorted while thrFactor calculation */
-                sfbNActiveLinesLdData[ch][sfbGrp+sfb] = CalcLdInt(peChanData->sfbNActiveLines[sfbGrp+sfb]);
+                sfbNActiveLinesLdData[elementId][ch][sfbGrp+sfb] = CalcLdInt(peChanData->sfbNActiveLines[sfbGrp+sfb]);
              }
              if ( ((ahFlag[elementId][ch][sfbGrp+sfb] < AH_ACTIVE) || (deltaPe > 0)) &&
                    peChanData->sfbNActiveLines[sfbGrp+sfb] != 0 )
@@ -1002,14 +1208,14 @@ static void FDKaacEnc_correctThresh(CHANNEL_MAPPING* cm,
                                                - (FIXP_DBL)(minScale<<(DFRACT_BITS-1-LD_DATA_SHIFT));
 
                    if (sumLd < FL2FXCONST_DBL(0.f)) {
-                      sfbPeFactorsLdData[elementId][ch][sfbGrp+sfb] = sfbNActiveLinesLdData[ch][sfbGrp+sfb] - sumLd;
+                      sfbPeFactorsLdData[elementId][ch][sfbGrp+sfb] = sfbNActiveLinesLdData[elementId][ch][sfbGrp+sfb] - sumLd;
                    }
                    else {
-                     if ( sfbNActiveLinesLdData[ch][sfbGrp+sfb] > (FL2FXCONST_DBL(-1.f) + sumLd) ) {
-                       sfbPeFactorsLdData[elementId][ch][sfbGrp+sfb] = sfbNActiveLinesLdData[ch][sfbGrp+sfb] - sumLd;
+                     if ( sfbNActiveLinesLdData[elementId][ch][sfbGrp+sfb] > (FL2FXCONST_DBL(-1.f) + sumLd) ) {
+                       sfbPeFactorsLdData[elementId][ch][sfbGrp+sfb] = sfbNActiveLinesLdData[elementId][ch][sfbGrp+sfb] - sumLd;
                      }
                      else {
-                      sfbPeFactorsLdData[elementId][ch][sfbGrp+sfb] = sfbNActiveLinesLdData[ch][sfbGrp+sfb];
+                      sfbPeFactorsLdData[elementId][ch][sfbGrp+sfb] = sfbNActiveLinesLdData[elementId][ch][sfbGrp+sfb];
                      }
                    }
 
@@ -1050,7 +1256,7 @@ static void FDKaacEnc_correctThresh(CHANNEL_MAPPING* cm,
                  }
                  else {
                    /* new threshold */
-                   FIXP_DBL tmp = CalcInvLdData(sfbPeFactorsLdData[elementId][ch][sfbGrp+sfb] + normFactorLdData - sfbNActiveLinesLdData[ch][sfbGrp+sfb] - FL2FXCONST_DBL((float)LD_DATA_SHIFT/LD_DATA_SCALING));
+                   FIXP_DBL tmp = CalcInvLdData(sfbPeFactorsLdData[elementId][ch][sfbGrp+sfb] + normFactorLdData - sfbNActiveLinesLdData[elementId][ch][sfbGrp+sfb] - FL2FXCONST_DBL((float)LD_DATA_SHIFT/LD_DATA_SCALING));
 
                    /* limit thrFactor to 60dB */
                    tmp = (deltaPe<0) ? tmp : (-tmp);
@@ -1102,9 +1308,9 @@ static void FDKaacEnc_correctThresh(CHANNEL_MAPPING* cm,
                   reducing minSnr
 *****************************************************************************/
 void FDKaacEnc_reduceMinSnr(CHANNEL_MAPPING* cm,
-                            QC_OUT_ELEMENT*  qcElement[(6)],
-                            PSY_OUT_ELEMENT* psyOutElement[(6)],
-                            UCHAR            ahFlag[(6)][(2)][MAX_GROUPED_SFB],
+                            QC_OUT_ELEMENT*  qcElement[(8)],
+                            PSY_OUT_ELEMENT* psyOutElement[(8)],
+                            UCHAR            ahFlag[(8)][(2)][MAX_GROUPED_SFB],
                             const            INT desiredPe,
                             INT*             redPeGlobal,
                             const            INT processElements,
@@ -1204,10 +1410,10 @@ bail:
                   bands have to be quantized to zero
 *****************************************************************************/
 static void FDKaacEnc_allowMoreHoles(CHANNEL_MAPPING* cm,
-                           QC_OUT_ELEMENT*  qcElement[(6)],
-                           PSY_OUT_ELEMENT* psyOutElement[(6)],
-                           ATS_ELEMENT*     AdjThrStateElement[(6)],
-                           UCHAR            ahFlag[(6)][(2)][MAX_GROUPED_SFB],
+                           QC_OUT_ELEMENT*  qcElement[(8)],
+                           PSY_OUT_ELEMENT* psyOutElement[(8)],
+                           ATS_ELEMENT*     AdjThrStateElement[(8)],
+                           UCHAR            ahFlag[(8)][(2)][MAX_GROUPED_SFB],
                            const INT        desiredPe,
                            const INT        currentPe,
                            const int        processElements,
@@ -1439,17 +1645,17 @@ functionname: FDKaacEnc_adaptThresholdsToPe
 description:  two guesses for the reduction value and one final correction of the thresholds
 *****************************************************************************/
 static void FDKaacEnc_adaptThresholdsToPe(CHANNEL_MAPPING*  cm,
-                                ATS_ELEMENT*      AdjThrStateElement[(6)],
-                                QC_OUT_ELEMENT*   qcElement[(6)],
-                                PSY_OUT_ELEMENT*  psyOutElement[(6)],
+                                ATS_ELEMENT*      AdjThrStateElement[(8)],
+                                QC_OUT_ELEMENT*   qcElement[(8)],
+                                PSY_OUT_ELEMENT*  psyOutElement[(8)],
                                 const INT         desiredPe,
                                 const INT         processElements,
                                 const INT         elementOffset)
 {
-   FIXP_DBL redValue[(6)];
-   SCHAR    redValScaling[(6)];
-   UCHAR    pAhFlag[(6)][(2)][MAX_GROUPED_SFB];
-   FIXP_DBL pThrExp[(6)][(2)][MAX_GROUPED_SFB];
+   FIXP_DBL redValue[(8)];
+   SCHAR    redValScaling[(8)];
+   UCHAR    pAhFlag[(8)][(2)][MAX_GROUPED_SFB];
+   FIXP_DBL pThrExp[(8)][(2)][MAX_GROUPED_SFB];
    int iter;
 
    INT constPartGlobal, noRedPeGlobal, nActiveLinesGlobal, redPeGlobal;
@@ -1683,7 +1889,6 @@ static void FDKaacEnc_adaptThresholdsToPe(CHANNEL_MAPPING*  cm,
 
 }
 
-
 /* similar to FDKaacEnc_adaptThresholdsToPe(), for  VBR-mode */
 void FDKaacEnc_AdaptThresholdsVBR(QC_OUT_CHANNEL* qcOutChannel[(2)],
                                PSY_OUT_CHANNEL* psyOutChannel[(2)],
@@ -1692,8 +1897,14 @@ void FDKaacEnc_AdaptThresholdsVBR(QC_OUT_CHANNEL* qcOutChannel[(2)],
                                PE_DATA *peData,
                                const INT nChannels)
 {
-   UCHAR    pAhFlag[(2)][MAX_GROUPED_SFB];
-   FIXP_DBL pThrExp[(2)][MAX_GROUPED_SFB];
+   UCHAR    (*pAhFlag)[MAX_GROUPED_SFB];
+   FIXP_DBL (*pThrExp)[MAX_GROUPED_SFB];
+
+   /* allocate scratch memory */
+   C_ALLOC_SCRATCH_START(_pAhFlag, UCHAR, (2)*MAX_GROUPED_SFB)
+   C_ALLOC_SCRATCH_START(_pThrExp, FIXP_DBL, (2)*MAX_GROUPED_SFB)
+   pAhFlag = (UCHAR(*)[MAX_GROUPED_SFB])_pAhFlag;
+   pThrExp = (FIXP_DBL(*)[MAX_GROUPED_SFB])_pThrExp;
 
    /* thresholds to the power of redExp */
    FDKaacEnc_calcThreshExp(pThrExp, qcOutChannel, psyOutChannel, nChannels);
@@ -1711,6 +1922,9 @@ void FDKaacEnc_AdaptThresholdsVBR(QC_OUT_CHANNEL* qcOutChannel[(2)],
                        AdjThrStateElement->vbrQualFactor,
                        &AdjThrStateElement->chaosMeasureOld);
 
+   /* free scratch memory */
+   C_ALLOC_SCRATCH_END(_pThrExp, FIXP_DBL, (2)*MAX_GROUPED_SFB)
+   C_ALLOC_SCRATCH_END(_pAhFlag, UCHAR, (2)*MAX_GROUPED_SFB)
 }
 
 
@@ -2002,95 +2216,112 @@ bail:
 functionname: FDKaacEnc_AdjThrInit
 description:  initialize ADJ_THR_STATE
 *****************************************************************************/
-void FDKaacEnc_AdjThrInit(ADJ_THR_STATE   *hAdjThr,
-    const INT       meanPe,
-    ELEMENT_BITS    *elBits[(6)],
-    INT             nElements,
-    FIXP_DBL        vbrQualFactor)
+void FDKaacEnc_AdjThrInit(
+        ADJ_THR_STATE   *hAdjThr,
+        const INT       meanPe,
+        ELEMENT_BITS    *elBits[(8)],
+        INT             invQuant,
+        INT             nElements,
+        INT             nChannelsEff,
+        INT             sampleRate,
+        INT             advancedBitsToPe,
+        FIXP_DBL        vbrQualFactor
+        )
 {
-    INT i;
-
-    FIXP_DBL POINT8 = FL2FXCONST_DBL(0.8f);
-    FIXP_DBL POINT6 = FL2FXCONST_DBL(0.6f);
+  INT i;
+
+  FIXP_DBL POINT8 = FL2FXCONST_DBL(0.8f);
+  FIXP_DBL POINT6 = FL2FXCONST_DBL(0.6f);
+
+  /* common for all elements: */
+  /* parameters for bitres control */
+  hAdjThr->bresParamLong.clipSaveLow   = (FIXP_DBL)0x1999999a; /* FL2FXCONST_DBL(0.2f); */
+  hAdjThr->bresParamLong.clipSaveHigh  = (FIXP_DBL)0x7999999a; /* FL2FXCONST_DBL(0.95f); */
+  hAdjThr->bresParamLong.minBitSave    = (FIXP_DBL)0xf999999a; /* FL2FXCONST_DBL(-0.05f); */
+  hAdjThr->bresParamLong.maxBitSave    = (FIXP_DBL)0x26666666; /* FL2FXCONST_DBL(0.3f); */
+  hAdjThr->bresParamLong.clipSpendLow  = (FIXP_DBL)0x1999999a; /* FL2FXCONST_DBL(0.2f); */
+  hAdjThr->bresParamLong.clipSpendHigh = (FIXP_DBL)0x7999999a; /* FL2FXCONST_DBL(0.95f); */
+  hAdjThr->bresParamLong.minBitSpend   = (FIXP_DBL)0xf3333333; /* FL2FXCONST_DBL(-0.10f); */
+  hAdjThr->bresParamLong.maxBitSpend   = (FIXP_DBL)0x33333333; /* FL2FXCONST_DBL(0.4f); */
+
+  hAdjThr->bresParamShort.clipSaveLow   = (FIXP_DBL)0x199999a0; /* FL2FXCONST_DBL(0.2f); */
+  hAdjThr->bresParamShort.clipSaveHigh  = (FIXP_DBL)0x5fffffff; /* FL2FXCONST_DBL(0.75f); */
+  hAdjThr->bresParamShort.minBitSave    = (FIXP_DBL)0x00000000; /* FL2FXCONST_DBL(0.0f); */
+  hAdjThr->bresParamShort.maxBitSave    = (FIXP_DBL)0x199999a0; /* FL2FXCONST_DBL(0.2f); */
+  hAdjThr->bresParamShort.clipSpendLow  = (FIXP_DBL)0x199999a0; /* FL2FXCONST_DBL(0.2f); */
+  hAdjThr->bresParamShort.clipSpendHigh = (FIXP_DBL)0x5fffffff; /* FL2FXCONST_DBL(0.75f); */
+  hAdjThr->bresParamShort.minBitSpend   = (FIXP_DBL)0xf9999998; /* FL2FXCONST_DBL(-0.05f); */
+  hAdjThr->bresParamShort.maxBitSpend   = (FIXP_DBL)0x40000000; /* FL2FXCONST_DBL(0.5f); */
+
+  /* specific for each element: */
+  for (i=0; i<nElements; i++) {
+    ATS_ELEMENT* atsElem = hAdjThr->adjThrStateElem[i];
+    MINSNR_ADAPT_PARAM *msaParam = &atsElem->minSnrAdaptParam;
+    INT chBitrate = elBits[i]->chBitrateEl;
 
-    /* common for all elements: */
     /* parameters for bitres control */
-    hAdjThr->bresParamLong.clipSaveLow   = (FIXP_DBL)0x1999999a; /* FL2FXCONST_DBL(0.2f); */
-    hAdjThr->bresParamLong.clipSaveHigh  = (FIXP_DBL)0x7999999a; /* FL2FXCONST_DBL(0.95f); */
-    hAdjThr->bresParamLong.minBitSave    = (FIXP_DBL)0xf999999a; /* FL2FXCONST_DBL(-0.05f); */
-    hAdjThr->bresParamLong.maxBitSave    = (FIXP_DBL)0x26666666; /* FL2FXCONST_DBL(0.3f); */
-    hAdjThr->bresParamLong.clipSpendLow  = (FIXP_DBL)0x1999999a; /* FL2FXCONST_DBL(0.2f); */
-    hAdjThr->bresParamLong.clipSpendHigh = (FIXP_DBL)0x7999999a; /* FL2FXCONST_DBL(0.95f); */
-    hAdjThr->bresParamLong.minBitSpend   = (FIXP_DBL)0xf3333333; /* FL2FXCONST_DBL(-0.10f); */
-    hAdjThr->bresParamLong.maxBitSpend   = (FIXP_DBL)0x33333333; /* FL2FXCONST_DBL(0.4f); */
-
-    hAdjThr->bresParamShort.clipSaveLow   = (FIXP_DBL)0x199999a0; /* FL2FXCONST_DBL(0.2f); */
-    hAdjThr->bresParamShort.clipSaveHigh  = (FIXP_DBL)0x5fffffff; /* FL2FXCONST_DBL(0.75f); */
-    hAdjThr->bresParamShort.minBitSave    = (FIXP_DBL)0x00000000; /* FL2FXCONST_DBL(0.0f); */
-    hAdjThr->bresParamShort.maxBitSave    = (FIXP_DBL)0x199999a0; /* FL2FXCONST_DBL(0.2f); */
-    hAdjThr->bresParamShort.clipSpendLow  = (FIXP_DBL)0x199999a0; /* FL2FXCONST_DBL(0.2f); */
-    hAdjThr->bresParamShort.clipSpendHigh = (FIXP_DBL)0x5fffffff; /* FL2FXCONST_DBL(0.75f); */
-    hAdjThr->bresParamShort.minBitSpend   = (FIXP_DBL)0xf9999998; /* FL2FXCONST_DBL(-0.05f); */
-    hAdjThr->bresParamShort.maxBitSpend   = (FIXP_DBL)0x40000000; /* FL2FXCONST_DBL(0.5f); */
-
-    /* specific for each element: */
-    for (i=0; i<nElements; i++) {
-        ATS_ELEMENT* atsElem = hAdjThr->adjThrStateElem[i];
-        MINSNR_ADAPT_PARAM *msaParam = &atsElem->minSnrAdaptParam;
-        INT chBitrate = elBits[i]->chBitrateEl;
+    atsElem->peMin = fMultI(POINT8, meanPe) >> 1;
+    atsElem->peMax = fMultI(POINT6, meanPe);
 
-        /* parameters for bitres control */
-        atsElem->peMin = fMultI(POINT8, meanPe) >> 1;
-        atsElem->peMax = fMultI(POINT6, meanPe);
+    /* for use in FDKaacEnc_reduceThresholdsVBR */
+    atsElem->chaosMeasureOld = FL2FXCONST_DBL(0.3f);
 
-        /* for use in FDKaacEnc_reduceThresholdsVBR */
-        atsElem->chaosMeasureOld = FL2FXCONST_DBL(0.3f);
+    /* additional pe offset to correct pe2bits for low bitrates */
+    atsElem->peOffset = 0;
 
-        /* additional pe offset to correct pe2bits for low bitrates */
-        atsElem->peOffset = 0;
+    /* vbr initialisation */
+    atsElem->vbrQualFactor = vbrQualFactor;
+    if (chBitrate < 32000)
+    {
+      atsElem->peOffset = fixMax(50, 100-fMultI((FIXP_DBL)0x666667, chBitrate));
+    }
 
-        /* vbr initialisation */
-        atsElem->vbrQualFactor = vbrQualFactor;
-        if (chBitrate < 32000)
-        {
-            atsElem->peOffset = fixMax(50, 100-fMultI((FIXP_DBL)0x666667, chBitrate));
-        }
+    /* avoid hole parameters */
+    if (chBitrate > 20000) {
+      atsElem->ahParam.modifyMinSnr = TRUE;
+      atsElem->ahParam.startSfbL = 15;
+      atsElem->ahParam.startSfbS = 3;
+    }
+    else {
+      atsElem->ahParam.modifyMinSnr = FALSE;
+      atsElem->ahParam.startSfbL = 0;
+      atsElem->ahParam.startSfbS = 0;
+    }
 
-        /* avoid hole parameters */
-        if (chBitrate > 20000) {
-            atsElem->ahParam.modifyMinSnr = TRUE;
-            atsElem->ahParam.startSfbL = 15;
-            atsElem->ahParam.startSfbS = 3;
-        }
-        else {
-            atsElem->ahParam.modifyMinSnr = FALSE;
-            atsElem->ahParam.startSfbL = 0;
-            atsElem->ahParam.startSfbS = 0;
-        }
+    /* minSnr adaptation */
+    msaParam->maxRed = FL2FXCONST_DBL(0.00390625f); /* 0.25f/64.0f */
+    /* start adaptation of minSnr for avgEn/sfbEn > startRatio */
+    msaParam->startRatio = FL2FXCONST_DBL(0.05190512648f); /* ld64(10.0f) */
+    /* maximum minSnr reduction to minSnr^maxRed is reached for
+       avgEn/sfbEn >= maxRatio */
+    /* msaParam->maxRatio = 1000.0f; */
+    /*msaParam->redRatioFac = ((float)1.0f - msaParam->maxRed) / ((float)10.0f*log10(msaParam->startRatio/msaParam->maxRatio)/log10(2.0f)*(float)0.3010299956f);*/
+    msaParam->redRatioFac = FL2FXCONST_DBL(-0.375f); /* -0.0375f * 10.0f */
+    /*msaParam->redOffs = (float)1.0f - msaParam->redRatioFac * (float)10.0f * log10(msaParam->startRatio)/log10(2.0f) * (float)0.3010299956f;*/
+    msaParam->redOffs = FL2FXCONST_DBL(0.021484375); /* 1.375f/64.0f */
+
+    /* init pe correction */
+    atsElem->peCorrectionFactor_m = FL2FXCONST_DBL(0.5f); /* 1.0 */
+    atsElem->peCorrectionFactor_e = 1;
+
+    atsElem->dynBitsLast = -1;
+    atsElem->peLast = 0;
+
+    /* init bits to pe factor */
+
+    /* init bits2PeFactor */
+    FDKaacEnc_InitBits2PeFactor(
+              &atsElem->bits2PeFactor_m,
+              &atsElem->bits2PeFactor_e,
+              chBitrate,       /* bitrate/channel*/
+              nChannelsEff,    /* number of channels */
+              sampleRate,
+              advancedBitsToPe,
+              invQuant
+              );
+
+  } /* for nElements */
 
-    		  /* minSnr adaptation */
-        msaParam->maxRed = FL2FXCONST_DBL(0.00390625f); /* 0.25f/64.0f */
-        /* start adaptation of minSnr for avgEn/sfbEn > startRatio */
-        msaParam->startRatio = FL2FXCONST_DBL(0.05190512648f); /* ld64(10.0f) */
-        /* maximum minSnr reduction to minSnr^maxRed is reached for
-           avgEn/sfbEn >= maxRatio */
-        /* msaParam->maxRatio = 1000.0f; */
-        /*msaParam->redRatioFac = ((float)1.0f - msaParam->maxRed) / ((float)10.0f*log10(msaParam->startRatio/msaParam->maxRatio)/log10(2.0f)*(float)0.3010299956f);*/
-        msaParam->redRatioFac = FL2FXCONST_DBL(-0.375f); /* -0.0375f * 10.0f */
-        /*msaParam->redOffs = (float)1.0f - msaParam->redRatioFac * (float)10.0f * log10(msaParam->startRatio)/log10(2.0f) * (float)0.3010299956f;*/
-        msaParam->redOffs = FL2FXCONST_DBL(0.021484375); /* 1.375f/64.0f */
-
-        /* init pe correction */
-        atsElem->peCorrectionFactor_m = FL2FXCONST_DBL(0.5f); /* 1.0 */
-        atsElem->peCorrectionFactor_e = 1;
-
-        atsElem->dynBitsLast = -1;
-        atsElem->peLast = 0;
-
-        /* init bits to pe factor */
-        atsElem->bits2PeFactor_m = FL2FXCONST_DBL(1.18f/(1<<(1)));
-        atsElem->bits2PeFactor_e = 1;
-    }
 }
 
 
@@ -2154,6 +2385,67 @@ static void FDKaacEnc_FDKaacEnc_calcPeCorrection(
 }
 
 
+static void FDKaacEnc_calcPeCorrectionLowBitRes(
+        FIXP_DBL *const           correctionFac_m,
+        INT *const                correctionFac_e,
+        const INT                 peLast,
+        const INT                 bitsLast,
+        const INT                 bitresLevel,
+        const INT                 nChannels,
+        const FIXP_DBL            bits2PeFactor_m,
+        const INT                 bits2PeFactor_e
+        )
+{
+  /* tuning params */
+  const FIXP_DBL amp     = FL2FXCONST_DBL(0.005);
+  const FIXP_DBL maxDiff = FL2FXCONST_DBL(0.25f);
+
+  if (bitsLast > 0) {
+
+    /* Estimate deviation of granted and used dynamic bits in previous frame, in PE units */
+    const int bitsBalLast = peLast - FDKaacEnc_bits2pe2(
+          bitsLast,
+          bits2PeFactor_m,
+          bits2PeFactor_e);
+
+    /* reserve n bits per channel */
+    int headroom = (bitresLevel>=50*nChannels) ? 0 : (100*nChannels);
+
+    /* in PE units */
+    headroom = FDKaacEnc_bits2pe2(
+          headroom,
+          bits2PeFactor_m,
+          bits2PeFactor_e);
+
+    /*
+     * diff = amp * ((bitsBalLast - headroom) / (bitresLevel + headroom)
+     * diff = max ( min ( diff, maxDiff, -maxDiff)) / 2
+     */
+    FIXP_DBL denominator = (FIXP_DBL)FDKaacEnc_bits2pe2(bitresLevel, bits2PeFactor_m, bits2PeFactor_e) + (FIXP_DBL)headroom;
+
+    int scaling = 0;
+    FIXP_DBL diff = (bitsBalLast>=headroom)
+         ?  fMult(amp, fDivNorm( (FIXP_DBL)(bitsBalLast - headroom), denominator, &scaling))
+         : -fMult(amp, fDivNorm(-(FIXP_DBL)(bitsBalLast - headroom), denominator, &scaling)) ;
+
+    scaling -= 1; /* divide by 2 */
+
+    diff = (scaling<=0) ? FDKmax( FDKmin (diff>>(-scaling), maxDiff>>1), -maxDiff>>1)
+                        : FDKmax( FDKmin (diff, maxDiff>>(1+scaling)), -maxDiff>>(1+scaling)) << scaling;
+
+    /*
+     * corrFac += diff
+     * corrFac = max ( min ( corrFac/2.f, 1.f/2.f, 0.75f/2.f ) )
+     */
+    *correctionFac_m = FDKmax(FDKmin((*correctionFac_m)+diff, FL2FXCONST_DBL(1.0f/2.f)), FL2FXCONST_DBL(0.75f/2.f)) ;
+    *correctionFac_e = 1;
+  }
+  else {
+    *correctionFac_m = FL2FXCONST_DBL(0.75/2.f);
+    *correctionFac_e = 1;
+  }
+}
+
 void FDKaacEnc_DistributeBits(ADJ_THR_STATE *adjThrState,
     ATS_ELEMENT       *AdjThrStateElement,
     PSY_OUT_CHANNEL   *psyOutChannel[(2)],
@@ -2166,7 +2458,7 @@ void FDKaacEnc_DistributeBits(ADJ_THR_STATE *adjThrState,
     const INT         bitresBits,
     const INT         maxBitresBits,
     const FIXP_DBL    maxBitFac,
-    const INT         bitDistributenMode)
+    const INT         bitDistributionMode)
 {
   FIXP_DBL bitFactor;
   INT noRedPe = peData->pe;
@@ -2184,7 +2476,7 @@ void FDKaacEnc_DistributeBits(ADJ_THR_STATE *adjThrState,
   }
 
   if (grantedDynBits >= 1) {
-    if (bitDistributenMode!=0) {
+    if (bitDistributionMode!=0) {
       *grantedPe = FDKaacEnc_bits2pe2(grantedDynBits, AdjThrStateElement->bits2PeFactor_m, AdjThrStateElement->bits2PeFactor_e);
     }
     else
@@ -2208,16 +2500,32 @@ void FDKaacEnc_DistributeBits(ADJ_THR_STATE *adjThrState,
   }
 
   /* correction of pe value */
-  {
-    FDKaacEnc_FDKaacEnc_calcPeCorrection(
-       &AdjThrStateElement->peCorrectionFactor_m,
-       &AdjThrStateElement->peCorrectionFactor_e,
-        fixMin(*grantedPe, noRedPe),
-        AdjThrStateElement->peLast,
-        AdjThrStateElement->dynBitsLast,
-        AdjThrStateElement->bits2PeFactor_m,
-        AdjThrStateElement->bits2PeFactor_e
+  switch (bitDistributionMode) {
+  case 2:
+  case 1:
+    FDKaacEnc_calcPeCorrectionLowBitRes(
+           &AdjThrStateElement->peCorrectionFactor_m,
+           &AdjThrStateElement->peCorrectionFactor_e,
+            AdjThrStateElement->peLast,
+            AdjThrStateElement->dynBitsLast,
+            bitresBits,
+            nChannels,
+            AdjThrStateElement->bits2PeFactor_m,
+            AdjThrStateElement->bits2PeFactor_e
         );
+    break;
+  case 0:
+  default:
+      FDKaacEnc_FDKaacEnc_calcPeCorrection(
+           &AdjThrStateElement->peCorrectionFactor_m,
+           &AdjThrStateElement->peCorrectionFactor_e,
+            fixMin(*grantedPe, noRedPe),
+            AdjThrStateElement->peLast,
+            AdjThrStateElement->dynBitsLast,
+            AdjThrStateElement->bits2PeFactor_m,
+            AdjThrStateElement->bits2PeFactor_e
+            );
+    break;
   }
 
   *grantedPeCorr = (INT)(fMult((FIXP_DBL)(*grantedPe<<Q_AVGBITS), AdjThrStateElement->peCorrectionFactor_m) >> (Q_AVGBITS-AdjThrStateElement->peCorrectionFactor_e));
@@ -2232,10 +2540,10 @@ void FDKaacEnc_DistributeBits(ADJ_THR_STATE *adjThrState,
 functionname: FDKaacEnc_AdjustThresholds
 description:  adjust thresholds
 *****************************************************************************/
-void FDKaacEnc_AdjustThresholds(ATS_ELEMENT*        AdjThrStateElement[(6)],
-                                QC_OUT_ELEMENT*     qcElement[(6)],
+void FDKaacEnc_AdjustThresholds(ATS_ELEMENT*        AdjThrStateElement[(8)],
+                                QC_OUT_ELEMENT*     qcElement[(8)],
                                 QC_OUT*             qcOut,
-                                PSY_OUT_ELEMENT*    psyOutElement[(6)],
+                                PSY_OUT_ELEMENT*    psyOutElement[(8)],
                                 INT                 CBRbitrateMode,
                                 CHANNEL_MAPPING*    cm)
 {
@@ -2291,20 +2599,19 @@ void FDKaacEnc_AdjustThresholds(ATS_ELEMENT*        AdjThrStateElement[(6)],
         }  /* -end- element loop */
 
     }
-        for (i=0; i<cm->nElements; i++) {
-            int ch,sfb,sfbGrp;
-            /* no weighting of threholds and energies for mlout */
-            /* weight energies and thresholds */
-            for (ch=0; ch<cm->elInfo[i].nChannelsInEl; ch++) {
-                QC_OUT_CHANNEL* pQcOutCh = qcElement[i]->qcOutChannel[ch];
-                for (sfbGrp = 0;sfbGrp < psyOutElement[i]->psyOutChannel[ch]->sfbCnt; sfbGrp+=psyOutElement[i]->psyOutChannel[ch]->sfbPerGroup) {
-                    for (sfb=0; sfb<psyOutElement[i]->psyOutChannel[ch]->maxSfbPerGroup; sfb++) {
-                        pQcOutCh->sfbThresholdLdData[sfb+sfbGrp] += pQcOutCh->sfbEnFacLd[sfb+sfbGrp];
-                    }
+    for (i=0; i<cm->nElements; i++) {
+        int ch,sfb,sfbGrp;
+        /* no weighting of threholds and energies for mlout */
+        /* weight energies and thresholds */
+        for (ch=0; ch<cm->elInfo[i].nChannelsInEl; ch++) {
+            QC_OUT_CHANNEL* pQcOutCh = qcElement[i]->qcOutChannel[ch];
+            for (sfbGrp = 0;sfbGrp < psyOutElement[i]->psyOutChannel[ch]->sfbCnt; sfbGrp+=psyOutElement[i]->psyOutChannel[ch]->sfbPerGroup) {
+                for (sfb=0; sfb<psyOutElement[i]->psyOutChannel[ch]->maxSfbPerGroup; sfb++) {
+                    pQcOutCh->sfbThresholdLdData[sfb+sfbGrp] += pQcOutCh->sfbEnFacLd[sfb+sfbGrp];
                 }
             }
         }
-
+    }
 }
 
 void FDKaacEnc_AdjThrClose(ADJ_THR_STATE** phAdjThr)
@@ -2313,7 +2620,7 @@ void FDKaacEnc_AdjThrClose(ADJ_THR_STATE** phAdjThr)
     ADJ_THR_STATE* hAdjThr = *phAdjThr;
 
     if (hAdjThr!=NULL) {
-      for (i=0; i<(6); i++) {
+      for (i=0; i<(8); i++) {
         if (hAdjThr->adjThrStateElem[i]!=NULL) {
           FreeRam_aacEnc_AdjThrStateElement(&hAdjThr->adjThrStateElem[i]);
         }
diff --git a/libAACenc/src/adj_thr.h b/libAACenc/src/adj_thr.h
index 83d4c49..69b1dcc 100644
--- a/libAACenc/src/adj_thr.h
+++ b/libAACenc/src/adj_thr.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -98,23 +98,28 @@ amm-info at iis.fraunhofer.de
 #include "interface.h"
 
 
-
-void FDKaacEnc_peCalculation(PE_DATA *peData,
-                             PSY_OUT_CHANNEL* psyOutChannel[(2)],
-                             QC_OUT_CHANNEL* qcOutChannel[(2)],
-                             struct TOOLSINFO *toolsInfo,
-                             ATS_ELEMENT* adjThrStateElement,
-                             const INT nChannels);
+void FDKaacEnc_peCalculation(
+        PE_DATA *peData,
+        PSY_OUT_CHANNEL* psyOutChannel[(2)],
+        QC_OUT_CHANNEL* qcOutChannel[(2)],
+        struct TOOLSINFO *toolsInfo,
+        ATS_ELEMENT* adjThrStateElement,
+        const INT nChannels
+        );
 
 INT  FDKaacEnc_AdjThrNew(ADJ_THR_STATE** phAdjThr,
                          INT             nElements);
 
 void FDKaacEnc_AdjThrInit(ADJ_THR_STATE *hAdjThr,
                 const INT peMean,
-                ELEMENT_BITS* elBits[(6)],
+                ELEMENT_BITS* elBits[(8)],
+                INT invQuant,
                 INT nElements,
+                INT nChannelsEff,
+                INT sampleRate,
+                INT advancedBitsToPe,
                 FIXP_DBL vbrQualFactor);
-
+ 
 
 void FDKaacEnc_DistributeBits(ADJ_THR_STATE *adjThrState,
     ATS_ELEMENT       *AdjThrStateElement,
@@ -128,12 +133,12 @@ void FDKaacEnc_DistributeBits(ADJ_THR_STATE *adjThrState,
     const INT         bitresBits,
     const INT         maxBitresBits,
     const FIXP_DBL    maxBitFac,
-    const INT         bitDistributenMode);
+    const INT         bitDistributionMode);
 
-void FDKaacEnc_AdjustThresholds(ATS_ELEMENT* AdjThrStateElement[(6)],
-    QC_OUT_ELEMENT*   qcElement[(6)],
+void FDKaacEnc_AdjustThresholds(ATS_ELEMENT* AdjThrStateElement[(8)],
+    QC_OUT_ELEMENT*   qcElement[(8)],
     QC_OUT*           qcOut,
-    PSY_OUT_ELEMENT*  psyOutElement[(6)],
+    PSY_OUT_ELEMENT*  psyOutElement[(8)],
     INT               CBRbitrateMode,
     CHANNEL_MAPPING*  cm);
 
diff --git a/libAACenc/src/adj_thr_data.h b/libAACenc/src/adj_thr_data.h
index d209a51..3eb7678 100644
--- a/libAACenc/src/adj_thr_data.h
+++ b/libAACenc/src/adj_thr_data.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -144,7 +144,7 @@ typedef struct {
 
 typedef struct {
   BRES_PARAM bresParamLong, bresParamShort;
-  ATS_ELEMENT* adjThrStateElem[(6)];
+  ATS_ELEMENT* adjThrStateElem[(8)];
 } ADJ_THR_STATE;
 
 #endif
diff --git a/libAACenc/src/band_nrg.cpp b/libAACenc/src/band_nrg.cpp
index 0e46b45..861f7a8 100644
--- a/libAACenc/src/band_nrg.cpp
+++ b/libAACenc/src/band_nrg.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/band_nrg.h b/libAACenc/src/band_nrg.h
index cf4c4cb..540a8ef 100644
--- a/libAACenc/src/band_nrg.h
+++ b/libAACenc/src/band_nrg.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/bandwidth.cpp b/libAACenc/src/bandwidth.cpp
index f6ca8ef..6fc7d87 100644
--- a/libAACenc/src/bandwidth.cpp
+++ b/libAACenc/src/bandwidth.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -204,7 +204,6 @@ static INT GetBandwidthEntry(
   INT bwTabSize = 0;
 
   switch (frameLength) {
-    case 960:
     case 1024:
       pBwTab = bandWidthTable;
       bwTabSize = sizeof(bandWidthTable)/sizeof(BANDWIDTH_TAB);
@@ -253,7 +252,6 @@ static INT GetBandwidthEntry(
           chanBitRate < pBwTab[i+1].chanBitRate)
       {
         switch (frameLength) {
-          case 960:
           case 1024:
             bandwidth = (entryNo==0)
               ? pBwTab[i].bandWidthMono
@@ -318,6 +316,8 @@ AAC_ENCODER_ERROR FDKaacEnc_DetermineBandWidth(INT* bandWidth,
       case MODE_1_2_2:
       case MODE_1_2_2_1:
       case MODE_1_2_2_2_1:
+      case MODE_7_1_REAR_SURROUND:
+      case MODE_7_1_FRONT_CENTER:
         *bandWidth = bandWidthTableVBR[bitrateMode].bandWidth2AndMoreChan;
         break;
       default:
@@ -348,6 +348,8 @@ AAC_ENCODER_ERROR FDKaacEnc_DetermineBandWidth(INT* bandWidth,
       case MODE_1_2_2:    /* sce + cpe + cpe */
       case MODE_1_2_2_1:  /* (5.1) sce + cpe + cpe + lfe */
       case MODE_1_2_2_2_1: /* (7.1) sce + cpe + cpe + cpe + lfe */
+      case MODE_7_1_REAR_SURROUND:
+      case MODE_7_1_FRONT_CENTER:
         entryNo = 1;      /* use stereo bandwith settings */
         break;
 
diff --git a/libAACenc/src/bandwidth.h b/libAACenc/src/bandwidth.h
index 61c7f93..2e92453 100644
--- a/libAACenc/src/bandwidth.h
+++ b/libAACenc/src/bandwidth.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/bit_cnt.cpp b/libAACenc/src/bit_cnt.cpp
index e89710e..926ee49 100644
--- a/libAACenc/src/bit_cnt.cpp
+++ b/libAACenc/src/bit_cnt.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/bit_cnt.h b/libAACenc/src/bit_cnt.h
index 8650566..7c4b59e 100644
--- a/libAACenc/src/bit_cnt.h
+++ b/libAACenc/src/bit_cnt.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/bitenc.cpp b/libAACenc/src/bitenc.cpp
index d2cb5af..8e477aa 100644
--- a/libAACenc/src/bitenc.cpp
+++ b/libAACenc/src/bitenc.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -1360,7 +1360,7 @@ AAC_ENCODER_ERROR FDKaacEnc_WriteBitstream(HANDLE_TRANSPORTENC hTpEnc,
 
   if ( (syntaxFlags & AC_ER) && !(syntaxFlags & AC_DRM) )
   {
-    UCHAR channelElementExtensionWritten[(6)][(1)]; /* 0: extension not touched, 1: extension already written */
+    UCHAR channelElementExtensionWritten[(8)][(1)]; /* 0: extension not touched, 1: extension already written */
 
     FDKmemclear(channelElementExtensionWritten, sizeof(channelElementExtensionWritten));
 
diff --git a/libAACenc/src/bitenc.h b/libAACenc/src/bitenc.h
index 337ce38..498be7c 100644
--- a/libAACenc/src/bitenc.h
+++ b/libAACenc/src/bitenc.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/block_switch.cpp b/libAACenc/src/block_switch.cpp
index 96fcb08..7b3e275 100644
--- a/libAACenc/src/block_switch.cpp
+++ b/libAACenc/src/block_switch.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -83,7 +83,7 @@ amm-info at iis.fraunhofer.de
 
 /*****************************  MPEG-4 AAC Encoder  **************************
 
-   Author(s):   M. Werner
+   Author(s):   M. Werner, Tobias Chalupka
    Description: Block switching
 
 ******************************************************************************/
@@ -100,9 +100,11 @@ amm-info at iis.fraunhofer.de
 
 static FIXP_DBL FDKaacEnc_GetWindowEnergy(const FIXP_DBL in[], const INT blSwWndIdx);
 
-static void FDKaacEnc_CalcWindowEnergy( BLOCK_SWITCHING_CONTROL *RESTRICT blockSwitchingControl,
-                              INT                      windowLen);
-
+static void FDKaacEnc_CalcWindowEnergy(
+        BLOCK_SWITCHING_CONTROL *RESTRICT blockSwitchingControl,
+        INT                      windowLen,
+        const INT_PCM           *pTimeSignal
+        );
 
 /****************** Constants *****************************/
 /*                                                LONG         START        SHORT         STOP         LOWOV                  */
@@ -145,20 +147,9 @@ static const FIXP_DBL minAttackNrg = (FL2FXCONST_DBL(1e+6f*NORM_PCM_ENERGY)>>BLO
 
 /**************** internal function prototypes ***********/
 
-static INT FDKaacEnc_GetWindowIndex(INT blockSwWindowIndex);
-
-static FIXP_DBL FDKaacEnc_GetWindowEnergy(const FIXP_DBL in[], const INT shortWndIdx);
-
-static void FDKaacEnc_CalcWindowEnergy( BLOCK_SWITCHING_CONTROL *RESTRICT blockSwitchingControl,
-                                        INT windowLen);
-
-
-
 /****************** Routines ****************************/
 void FDKaacEnc_InitBlockSwitching(BLOCK_SWITCHING_CONTROL *blockSwitchingControl, INT isLowDelay)
 {
-  /* note: the pointer to timeSignal can be zeroed here, because it is initialized for every call
-           to FDKaacEnc_BlockSwitching anew */
   FDKmemclear (blockSwitchingControl, sizeof(BLOCK_SWITCHING_CONTROL));
 
   if (isLowDelay)
@@ -214,7 +205,7 @@ static const INT chgWndSqLkAhd[2][2][N_BLOCKTYPES] =
   /*attack   */   {START_WINDOW,  SHORT_WINDOW,  SHORT_WINDOW,  START_WINDOW, WRONG_WINDOW, WRONG_WINDOW} }  /* attack      */
 };
 
-int FDKaacEnc_BlockSwitching(BLOCK_SWITCHING_CONTROL *blockSwitchingControl, const INT granuleLength, const int isLFE)
+int FDKaacEnc_BlockSwitching(BLOCK_SWITCHING_CONTROL *blockSwitchingControl, const INT granuleLength, const int isLFE, const INT_PCM *pTimeSignal)
 {
     UINT i;
     FIXP_DBL enM1, enMax;
@@ -263,7 +254,7 @@ int FDKaacEnc_BlockSwitching(BLOCK_SWITCHING_CONTROL *blockSwitchingControl, con
 
 
     /* Calculate unfiltered and filtered energies in subwindows and combine to segments */
-    FDKaacEnc_CalcWindowEnergy(blockSwitchingControl, granuleLength>>(nBlockSwitchWindows==4? 2:3 ));
+    FDKaacEnc_CalcWindowEnergy(blockSwitchingControl, granuleLength>>(nBlockSwitchWindows==4? 2:3 ), pTimeSignal);
 
     /* now calculate if there is an attack */
 
@@ -335,8 +326,7 @@ static FIXP_DBL FDKaacEnc_GetWindowEnergy(const FIXP_DBL in[], const INT blSwWnd
 
 }
 
-
-static void FDKaacEnc_CalcWindowEnergy(BLOCK_SWITCHING_CONTROL *RESTRICT blockSwitchingControl, INT windowLen)
+static void FDKaacEnc_CalcWindowEnergy(BLOCK_SWITCHING_CONTROL *RESTRICT blockSwitchingControl, INT windowLen, const INT_PCM *pTimeSignal)
 {
     INT  i;
     UINT w;
@@ -344,8 +334,6 @@ static void FDKaacEnc_CalcWindowEnergy(BLOCK_SWITCHING_CONTROL *RESTRICT blockSw
     FIXP_SGL hiPassCoeff0 = hiPassCoeff[0];
     FIXP_SGL hiPassCoeff1 = hiPassCoeff[1];
 
-    INT_PCM *timeSignal = blockSwitchingControl->timeSignal;
-
     /* sum up scalarproduct of timesignal as windowed Energies */
     for (w=0; w < blockSwitchingControl->nBlockSwitchWindows; w++) {
 
@@ -361,9 +349,9 @@ static void FDKaacEnc_CalcWindowEnergy(BLOCK_SWITCHING_CONTROL *RESTRICT blockSw
             FIXP_DBL tempUnfiltered, tempFiltred, t1, t2;
             /* tempUnfiltered is scaled with 1 to prevent overflows during calculation of tempFiltred */
 #if SAMPLE_BITS == DFRACT_BITS
-            tempUnfiltered = (FIXP_DBL) *timeSignal++ >> 1;
+            tempUnfiltered = (FIXP_DBL) *pTimeSignal++ >> 1;
 #else
-            tempUnfiltered = (FIXP_DBL) *timeSignal++ << (DFRACT_BITS-SAMPLE_BITS-1);
+            tempUnfiltered = (FIXP_DBL) *pTimeSignal++ << (DFRACT_BITS-SAMPLE_BITS-1);
 #endif
             t1 = fMultDiv2(hiPassCoeff1, tempUnfiltered-temp_iirState0);
             t2 = fMultDiv2(hiPassCoeff0, temp_iirState1);
diff --git a/libAACenc/src/block_switch.h b/libAACenc/src/block_switch.h
index 179e16b..e94b6f5 100644
--- a/libAACenc/src/block_switch.h
+++ b/libAACenc/src/block_switch.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -107,7 +107,6 @@ amm-info at iis.fraunhofer.de
 
 /****************** Structures ***************************/
 typedef struct{
-  INT_PCM *timeSignal;
   INT   lastWindowSequence;
   INT   windowShape;
   INT   lastWindowShape;
@@ -136,7 +135,7 @@ typedef struct{
 
 void FDKaacEnc_InitBlockSwitching(BLOCK_SWITCHING_CONTROL *blockSwitchingControl, INT isLowDelay);
 
-int FDKaacEnc_BlockSwitching(BLOCK_SWITCHING_CONTROL *blockSwitchingControl, const INT granuleLength, const int isLFE);
+int FDKaacEnc_BlockSwitching(BLOCK_SWITCHING_CONTROL *blockSwitchingControl, const INT granuleLength, const int isLFE, const INT_PCM *pTimeSignal);
 
 int FDKaacEnc_SyncBlockSwitching(
       BLOCK_SWITCHING_CONTROL *blockSwitchingControlLeft,
diff --git a/libAACenc/src/channel_map.cpp b/libAACenc/src/channel_map.cpp
index 6eddd47..99ed2b5 100644
--- a/libAACenc/src/channel_map.cpp
+++ b/libAACenc/src/channel_map.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -124,7 +124,7 @@ amm-info at iis.fraunhofer.de
 typedef struct {
 
   CHANNEL_MODE  encoderMode;
-  INT           channel_assignment[/*(6)*/12];
+  INT           channel_assignment[/*(8)*/12];
 
 } CHANNEL_ASSIGNMENT_INFO_TAB;
 
@@ -139,6 +139,8 @@ static const CHANNEL_ASSIGNMENT_INFO_TAB assignmentInfoTabMpeg[] =
      { MODE_1_2_2,                   { 0, 1, 2, 3, 4,-1,-1,-1,-1,-1,-1,-1} },  /*  5ch       */
      { MODE_1_2_2_1,                 { 0, 1, 2, 3, 4, 5,-1,-1,-1,-1,-1,-1} },  /*  5.1ch     */
      { MODE_1_2_2_2_1,               { 0, 1, 2, 3, 4, 5, 6, 7,-1,-1,-1,-1} },  /*  7.1ch     */
+     { MODE_7_1_REAR_SURROUND,       { 0, 1, 2, 3, 4, 5, 6, 7,-1,-1,-1,-1} },  /*  7.1ch     */
+     { MODE_7_1_FRONT_CENTER,        { 0, 1, 2, 3, 4, 5, 6, 7,-1,-1,-1,-1} }   /*  7.1ch     */
 };
 
 static const CHANNEL_ASSIGNMENT_INFO_TAB assignmentInfoTabWav[] =
@@ -150,7 +152,9 @@ static const CHANNEL_ASSIGNMENT_INFO_TAB assignmentInfoTabWav[] =
      { MODE_1_2_1,                   { 2, 0, 1, 3,-1,-1,-1,-1,-1,-1,-1,-1} },  /*  4ch       */
      { MODE_1_2_2,                   { 2, 0, 1, 3, 4,-1,-1,-1,-1,-1,-1,-1} },  /*  5ch       */
      { MODE_1_2_2_1,                 { 2, 0, 1, 4, 5, 3,-1,-1,-1,-1,-1,-1} },  /*  5.1ch     */
-     { MODE_1_2_2_2_1,               { 2, 0, 1, 6, 7, 4, 5, 3,-1,-1,-1,-1} },  /*  7.1ch     */
+     { MODE_1_2_2_2_1,               { 2, 6, 7, 0, 1, 4, 5, 3,-1,-1,-1,-1} },  /*  7.1ch     */
+     { MODE_7_1_REAR_SURROUND,       { 2, 0, 1, 6, 7, 4, 5, 3,-1,-1,-1,-1} },  /*  7.1ch     */
+     { MODE_7_1_FRONT_CENTER,        { 2, 6, 7, 0, 1, 4, 5, 3,-1,-1,-1,-1} },  /*  7.1ch     */
 };
 
 static const CHANNEL_ASSIGNMENT_INFO_TAB assignmentInfoTabWg4[] =
@@ -177,6 +181,8 @@ static const CHANNEL_MODE_CONFIG_TAB channelModeConfig[] =
    { MODE_1_2_2,                    5,  5,  3 },   /* SCE,CPE,CPE         */
    { MODE_1_2_2_1,                  6,  5,  4 },   /* SCE,CPE,CPE,LFE     */
    { MODE_1_2_2_2_1,                8,  7,  5 },   /* SCE,CPE,CPE,CPE,LFE */
+   { MODE_7_1_REAR_SURROUND,        8,  7,  5 },
+   { MODE_7_1_FRONT_CENTER,         8,  7,  5 },
 };
 
 #define MAX_MODES (sizeof(assignmentInfoTabWav)/sizeof(CHANNEL_ASSIGNMENT_INFO_TAB))
@@ -329,6 +335,8 @@ AAC_ENCODER_ERROR FDKaacEnc_InitChannelMapping(CHANNEL_MODE mode, CHANNEL_ORDER
       break;
 
     case MODE_1_2_2_2_1:
+    case MODE_7_1_REAR_SURROUND:
+    case MODE_7_1_FRONT_CENTER:
       /* (7.1) sce + cpe + cpe + cpe + lfe */
       FDKaacEnc_initElement(&cm->elInfo[0], ID_SCE, &count, mode, co, it_cnt, FL2FXCONST_DBL(0.18f));
       FDKaacEnc_initElement(&cm->elInfo[1], ID_CPE, &count, mode, co, it_cnt, FL2FXCONST_DBL(0.26f));
@@ -336,14 +344,13 @@ AAC_ENCODER_ERROR FDKaacEnc_InitChannelMapping(CHANNEL_MODE mode, CHANNEL_ORDER
       FDKaacEnc_initElement(&cm->elInfo[3], ID_CPE, &count, mode, co, it_cnt, FL2FXCONST_DBL(0.26f));
       FDKaacEnc_initElement(&cm->elInfo[4], ID_LFE, &count, mode, co, it_cnt, FL2FXCONST_DBL(0.04f));
       break;
-
     default:
       //*chMap=0;
       return AAC_ENC_UNSUPPORTED_CHANNELCONFIG;
   };
 
 
-  FDK_ASSERT(cm->nElements<=(6));
+  FDK_ASSERT(cm->nElements<=(8));
 
 
   return AAC_ENC_OK;
@@ -456,19 +463,18 @@ AAC_ENCODER_ERROR FDKaacEnc_InitElementBits(QC_STATE *hQC,
 
     break;
   }
-
-  case MODE_1_2_2_2_1:{
-      /* (7.1) sce + cpe + cpe + cpe + lfe */
-    hQC->elementBits[0]->relativeBitsEl = cm->elInfo[0].relativeBits;
-    hQC->elementBits[1]->relativeBitsEl = cm->elInfo[1].relativeBits;
-    hQC->elementBits[2]->relativeBitsEl = cm->elInfo[2].relativeBits;
-    hQC->elementBits[3]->relativeBitsEl = cm->elInfo[3].relativeBits;
-    hQC->elementBits[4]->relativeBitsEl = cm->elInfo[4].relativeBits;
-    FIXP_DBL sceRate  = cm->elInfo[0].relativeBits;
-    FIXP_DBL cpe1Rate = cm->elInfo[1].relativeBits;
-    FIXP_DBL cpe2Rate = cm->elInfo[2].relativeBits;
-    FIXP_DBL cpe3Rate = cm->elInfo[3].relativeBits;
-    FIXP_DBL lfeRate  = cm->elInfo[4].relativeBits;
+  case MODE_7_1_REAR_SURROUND:
+  case MODE_7_1_FRONT_CENTER:
+  case MODE_1_2_2_2_1: {
+    int cpe3Idx = 3;
+    int lfeIdx  = 4;
+
+    /* (7.1) sce + cpe + cpe + cpe + lfe */
+    FIXP_DBL sceRate  = hQC->elementBits[0]->relativeBitsEl = cm->elInfo[0].relativeBits;
+    FIXP_DBL cpe1Rate = hQC->elementBits[1]->relativeBitsEl = cm->elInfo[1].relativeBits;
+    FIXP_DBL cpe2Rate = hQC->elementBits[2]->relativeBitsEl = cm->elInfo[2].relativeBits;
+    FIXP_DBL cpe3Rate = hQC->elementBits[cpe3Idx]->relativeBitsEl = cm->elInfo[cpe3Idx].relativeBits;
+    FIXP_DBL lfeRate  = hQC->elementBits[lfeIdx]->relativeBitsEl = cm->elInfo[lfeIdx].relativeBits;
 
     int maxBitsTot = maxChannelBits * 7; /* LFE does not add to bit reservoir */
     int sc = CountLeadingBits(fixMax(maxChannelBits,averageBitsTot));
@@ -480,17 +486,16 @@ AAC_ENCODER_ERROR FDKaacEnc_InitElementBits(QC_STATE *hQC,
     hQC->elementBits[0]->chBitrateEl = fMult(sceRate,  (FIXP_DBL)(bitrateTot<<sc_brTot))>>sc_brTot;
     hQC->elementBits[1]->chBitrateEl = fMult(cpe1Rate, (FIXP_DBL)(bitrateTot<<sc_brTot))>>(sc_brTot+1);
     hQC->elementBits[2]->chBitrateEl = fMult(cpe2Rate, (FIXP_DBL)(bitrateTot<<sc_brTot))>>(sc_brTot+1);
-    hQC->elementBits[3]->chBitrateEl = fMult(cpe3Rate, (FIXP_DBL)(bitrateTot<<sc_brTot))>>(sc_brTot+1);
-    hQC->elementBits[4]->chBitrateEl = fMult(lfeRate,  (FIXP_DBL)(bitrateTot<<sc_brTot))>>sc_brTot;
+    hQC->elementBits[cpe3Idx]->chBitrateEl = fMult(cpe3Rate, (FIXP_DBL)(bitrateTot<<sc_brTot))>>(sc_brTot+1);
+    hQC->elementBits[lfeIdx]->chBitrateEl = fMult(lfeRate,  (FIXP_DBL)(bitrateTot<<sc_brTot))>>sc_brTot;
 
     hQC->elementBits[0]->maxBitsEl = maxChannelBits;
     hQC->elementBits[1]->maxBitsEl = 2*maxChannelBits;
     hQC->elementBits[2]->maxBitsEl = 2*maxChannelBits;
-    hQC->elementBits[3]->maxBitsEl = 2*maxChannelBits;
-    hQC->elementBits[4]->maxBitsEl = maxLfeBits;
+    hQC->elementBits[cpe3Idx]->maxBitsEl = 2*maxChannelBits;
+    hQC->elementBits[lfeIdx]->maxBitsEl = maxLfeBits;
     break;
   }
-
   default:
     return AAC_ENC_UNSUPPORTED_CHANNELCONFIG;
   }
@@ -532,6 +537,8 @@ ELEMENT_MODE FDKaacEnc_GetMonoStereoMode(const CHANNEL_MODE mode){
   case MODE_1_2_2:
   case MODE_1_2_2_1:
   case MODE_1_2_2_2_1:
+  case MODE_7_1_REAR_SURROUND:
+  case MODE_7_1_FRONT_CENTER:
     monoStereoSetting = EL_MODE_STEREO;
     break;
   default:            /* error */
diff --git a/libAACenc/src/channel_map.h b/libAACenc/src/channel_map.h
index 6d135d2..2cfb486 100644
--- a/libAACenc/src/channel_map.h
+++ b/libAACenc/src/channel_map.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/chaosmeasure.cpp b/libAACenc/src/chaosmeasure.cpp
index 9d6d77e..4e56e9e 100644
--- a/libAACenc/src/chaosmeasure.cpp
+++ b/libAACenc/src/chaosmeasure.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/chaosmeasure.h b/libAACenc/src/chaosmeasure.h
index 732cb09..44301c5 100644
--- a/libAACenc/src/chaosmeasure.h
+++ b/libAACenc/src/chaosmeasure.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/dyn_bits.cpp b/libAACenc/src/dyn_bits.cpp
index 8cac2ef..0c07109 100644
--- a/libAACenc/src/dyn_bits.cpp
+++ b/libAACenc/src/dyn_bits.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/dyn_bits.h b/libAACenc/src/dyn_bits.h
index e7f219b..ae78a4c 100644
--- a/libAACenc/src/dyn_bits.h
+++ b/libAACenc/src/dyn_bits.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/grp_data.cpp b/libAACenc/src/grp_data.cpp
index 4355295..465865f 100644
--- a/libAACenc/src/grp_data.cpp
+++ b/libAACenc/src/grp_data.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/grp_data.h b/libAACenc/src/grp_data.h
index eddd694..f061855 100644
--- a/libAACenc/src/grp_data.h
+++ b/libAACenc/src/grp_data.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/intensity.cpp b/libAACenc/src/intensity.cpp
index 632620a..6d807f7 100644
--- a/libAACenc/src/intensity.cpp
+++ b/libAACenc/src/intensity.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -450,23 +450,24 @@ FDKaacEnc_finalizeIntensityDecision(const FIXP_DBL *hrrErr,
                                     const INT       maxSfbPerGroup)
 {
   INT sfb,sfboffs, j;
-  INT startIsSfb = 0;
-  INT inIsBlock;
-  INT currentIsSfbCount;
-  FIXP_DBL overallHrrError;
   FIXP_DBL isScaleLast = FL2FXCONST_DBL(0.0f);
-  FIXP_DBL isRegionLoudness;
+  INT isStartValueFound = 0;
 
   for (sfboffs = 0; sfboffs < sfbCnt; sfboffs += sfbPerGroup) {
-    inIsBlock = 0;
-    currentIsSfbCount = 0;
-    overallHrrError = FL2FXCONST_DBL(0.0f);
-    isRegionLoudness = FL2FXCONST_DBL(0.0f);
+    INT startIsSfb = 0;
+    INT inIsBlock = 0;
+    INT currentIsSfbCount = 0;
+    FIXP_DBL overallHrrError = FL2FXCONST_DBL(0.0f);
+    FIXP_DBL isRegionLoudness = FL2FXCONST_DBL(0.0f);
+
     for (sfb = 0; sfb < maxSfbPerGroup; sfb++) {
       if (isMask[sfboffs + sfb] == 1) {
         if (currentIsSfbCount == 0) {
           startIsSfb = sfboffs + sfb;
+        }
+        if (isStartValueFound==0) {
           isScaleLast = realIsScale[sfboffs + sfb];
+          isStartValueFound = 1;
         }
         inIsBlock = 1;
         currentIsSfbCount++;
@@ -510,6 +511,14 @@ FDKaacEnc_finalizeIntensityDecision(const FIXP_DBL *hrrErr,
           for(j = startIsSfb; j <= sfboffs + sfb; j++) {
             isMask[j] = 0;
           }
+          isScaleLast = FL2FXCONST_DBL(0.0f);
+          isStartValueFound = 0;
+          for (j=0; j < startIsSfb; j++) {
+            if (isMask[j]!=0) {
+              isScaleLast = realIsScale[j];
+              isStartValueFound = 1;
+            }
+          }
         }
         currentIsSfbCount = 0;
         overallHrrError = FL2FXCONST_DBL(0.0f);
@@ -584,7 +593,6 @@ void FDKaacEnc_IntensityStereoProcessing(
   FIXP_DBL realIsScale[MAX_GROUPED_SFB];
   INTENSITY_PARAMETERS isParams;
   INT isMask[MAX_GROUPED_SFB];
-  INT lastValIs = 0, delta;
 
   FDKmemclear((void*)isBook,sfbCnt*sizeof(INT));
   FDKmemclear((void*)isMask,sfbCnt*sizeof(INT));
@@ -740,14 +748,6 @@ void FDKaacEnc_IntensityStereoProcessing(
         isScale[sfb+sfboffs] = (INT)(((realIsScale[sfb+sfboffs]>>1)+FL2FXCONST_DBL(0.5f/(1<<(REAL_SCALE_SF+LD_DATA_SHIFT+1))))>>(DFRACT_BITS-1-REAL_SCALE_SF-LD_DATA_SHIFT-1));
       }
 
-      delta = isScale[sfb+sfboffs] - lastValIs;
-      if (delta < -CODE_BOOK_SCF_LAV) {
-          isScale[sfb+sfboffs] = lastValIs - CODE_BOOK_SCF_LAV;
-      } else if (delta + CODE_BOOK_SCF_LAV >= (INT)(sizeof(FDKaacEnc_huff_ltabscf)/sizeof(FDKaacEnc_huff_ltabscf[0]))) {
-          isScale[sfb+sfboffs] = lastValIs + sizeof(FDKaacEnc_huff_ltabscf)/sizeof(FDKaacEnc_huff_ltabscf[0]) - CODE_BOOK_SCF_LAV - 1;
-      }
-      lastValIs = isScale[sfb+sfboffs];
-
       sfbEnergyRight[sfb+sfboffs] = FL2FXCONST_DBL(0.0f);
       sfbEnergyLdDataRight[sfb+sfboffs] = FL2FXCONST_DBL(-1.0f);
       sfbThresholdRight[sfb+sfboffs] = FL2FXCONST_DBL(0.0f);
diff --git a/libAACenc/src/intensity.h b/libAACenc/src/intensity.h
index 12be8bc..2acc292 100644
--- a/libAACenc/src/intensity.h
+++ b/libAACenc/src/intensity.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/interface.h b/libAACenc/src/interface.h
index a1c3a96..51fb72a 100644
--- a/libAACenc/src/interface.h
+++ b/libAACenc/src/interface.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -92,6 +92,7 @@ amm-info at iis.fraunhofer.de
 #define _INTERFACE_H
 
 #include "common_fix.h"
+#include "FDK_audio.h"
 
 #include "psy_data.h"
 #include "aacenc_tns.h"
@@ -155,9 +156,14 @@ typedef struct {
 
 typedef struct {
 
-  PSY_OUT_ELEMENT* psyOutElement[(6)];
-  PSY_OUT_CHANNEL* pPsyOutChannels[(6)];
+  PSY_OUT_ELEMENT* psyOutElement[(8)];
+  PSY_OUT_CHANNEL* pPsyOutChannels[(8)];
 
 }PSY_OUT;
 
+inline int isLowDelay( AUDIO_OBJECT_TYPE aot )
+{
+  return (aot==AOT_ER_AAC_LD || aot==AOT_ER_AAC_ELD);
+}
+
 #endif /* _INTERFACE_H */
diff --git a/libAACenc/src/line_pe.cpp b/libAACenc/src/line_pe.cpp
index ba7a25d..f3c0dab 100644
--- a/libAACenc/src/line_pe.cpp
+++ b/libAACenc/src/line_pe.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/line_pe.h b/libAACenc/src/line_pe.h
index 2fcc958..3d5cfd5 100644
--- a/libAACenc/src/line_pe.h
+++ b/libAACenc/src/line_pe.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/metadata_compressor.cpp b/libAACenc/src/metadata_compressor.cpp
index 852c8bc..876de57 100644
--- a/libAACenc/src/metadata_compressor.cpp
+++ b/libAACenc/src/metadata_compressor.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -421,7 +421,7 @@ static FIXP_DBL tc2Coeff(
    result = f2Pow(-exponent, DFRACT_BITS-1-METADATA_FRACT_BITS, &e_res);
 
    /* result = 1.0 - exp(-1.0/((t) * (f))) */
-   result = FL2FXCONST_DBL(1.0f) - scaleValue(result, e_res);
+   result = (FIXP_DBL)MAXVAL_DBL - scaleValue(result, e_res);
 
    return result;
 }
@@ -539,14 +539,25 @@ INT FDK_DRC_Generator_Initialize(
         drcComp->channelIdx[RS]  = channelMapping.elInfo[2].ChannelIndex[1];
         break;
     case MODE_1_2_2_2_1: /* 7.1 ch */
-        drcComp->channelIdx[L]   = channelMapping.elInfo[1].ChannelIndex[0];
-        drcComp->channelIdx[R]   = channelMapping.elInfo[1].ChannelIndex[1];
-        drcComp->channelIdx[C]   = channelMapping.elInfo[0].ChannelIndex[0];
-        drcComp->channelIdx[LFE] = channelMapping.elInfo[4].ChannelIndex[0];
-        drcComp->channelIdx[LS]  = channelMapping.elInfo[2].ChannelIndex[0];
-        drcComp->channelIdx[RS]  = channelMapping.elInfo[2].ChannelIndex[1];
-        drcComp->channelIdx[LS2] = channelMapping.elInfo[3].ChannelIndex[0];
-        drcComp->channelIdx[RS2] = channelMapping.elInfo[3].ChannelIndex[1];
+    case MODE_7_1_FRONT_CENTER:
+        drcComp->channelIdx[L]   = channelMapping.elInfo[2].ChannelIndex[0]; /* l */
+        drcComp->channelIdx[R]   = channelMapping.elInfo[2].ChannelIndex[1]; /* r */
+        drcComp->channelIdx[C]   = channelMapping.elInfo[0].ChannelIndex[0]; /* c */
+        drcComp->channelIdx[LFE] = channelMapping.elInfo[4].ChannelIndex[0]; /* lfe */
+        drcComp->channelIdx[LS]  = channelMapping.elInfo[3].ChannelIndex[0]; /* ls */
+        drcComp->channelIdx[RS]  = channelMapping.elInfo[3].ChannelIndex[1]; /* rs */
+        drcComp->channelIdx[LS2] = channelMapping.elInfo[1].ChannelIndex[0]; /* lc */
+        drcComp->channelIdx[RS2] = channelMapping.elInfo[1].ChannelIndex[1]; /* rc */
+        break;
+    case MODE_7_1_REAR_SURROUND:
+        drcComp->channelIdx[L]   = channelMapping.elInfo[1].ChannelIndex[0]; /* l */
+        drcComp->channelIdx[R]   = channelMapping.elInfo[1].ChannelIndex[1]; /* r */
+        drcComp->channelIdx[C]   = channelMapping.elInfo[0].ChannelIndex[0]; /* c */
+        drcComp->channelIdx[LFE] = channelMapping.elInfo[4].ChannelIndex[0]; /* lfe */
+        drcComp->channelIdx[LS]  = channelMapping.elInfo[3].ChannelIndex[0]; /* lrear */
+        drcComp->channelIdx[RS]  = channelMapping.elInfo[3].ChannelIndex[1]; /* rrear */
+        drcComp->channelIdx[LS2] = channelMapping.elInfo[2].ChannelIndex[0]; /* ls */
+        drcComp->channelIdx[RS2] = channelMapping.elInfo[2].ChannelIndex[1]; /* rs */
         break;
     case MODE_1_1:
     case MODE_1_1_1_1:
@@ -832,12 +843,12 @@ INT FDK_DRC_Generator_Calc(
                     FIXP_DBL accu;
 
                     /* drcComp->smoothLevel[i] = (1-alpha) * drcComp->smoothLevel[i] + alpha * level; */
-                    accu =  fMult((FL2FXCONST_DBL(1.f)-alpha), drcComp->smoothLevel[i]);
+                    accu =  fMult(((FIXP_DBL)MAXVAL_DBL-alpha), drcComp->smoothLevel[i]);
                     accu += fMult(alpha,level);
                     drcComp->smoothLevel[i] = accu;
 
                     /* drcComp->smoothGain[i]  = (1-alpha) * drcComp->smoothGain[i] + alpha * gain; */
-                    accu =  fMult((FL2FXCONST_DBL(1.f)-alpha), drcComp->smoothGain[i]);
+                    accu =  fMult(((FIXP_DBL)MAXVAL_DBL-alpha), drcComp->smoothGain[i]);
                     accu += fMult(alpha,gain);
                     drcComp->smoothGain[i] = accu;
                 }
@@ -941,7 +952,7 @@ INT FDK_DRC_Generator_Calc(
             if ((drcComp->channelIdx[LS] >= 0) && (drcComp->channelIdx[LS2] >= 0)) tmp = fMult(FL2FXCONST_DBL(0.707f), tmp);                                     /* 7.1ch */
             /*if ((drcComp->channelIdx[RS] >= 0) && (drcComp->channelIdx[RS2] >= 0)) tmp *=0.707f;*/                                                             /* 7.1ch */
             if (drcComp->channelIdx[S] >= 0) tmp += fMultDiv2(slev, fMult(FL2FXCONST_DBL(0.7f), (FIXP_PCM)pSamples[drcComp->channelIdx[S]]))>>(DOWNMIX_SHIFT-1); /* S */
-            if (drcComp->channelIdx[C] >= 0) tmp += fMultDiv2(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[C]])>>(DOWNMIX_SHIFT-1);                              /* C */
+            if (drcComp->channelIdx[C] >= 0) tmp += fMult(clev, (FIXP_PCM)pSamples[drcComp->channelIdx[C]])>>(DOWNMIX_SHIFT-1);                                  /* C (2*clev) */
             tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[L]])>>DOWNMIX_SHIFT);                                                                   /* L */
             tmp += (FX_PCM2FX_DBL((FIXP_PCM)pSamples[drcComp->channelIdx[R]])>>DOWNMIX_SHIFT);                                                                   /* R */
 
@@ -973,7 +984,7 @@ INT FDK_DRC_Generator_Calc(
        *         + 0.2f*2^(-METADATA_FRACT_BITS) + drcComp->smoothGain[i]
        */
       peak[i] = fMult((FIXP_DBL)(10<<(METADATA_FRACT_BITS+LD_DATA_SHIFT)), fMult( FL2FX_DBL(2*0.30102999566398119521373889472449f), ld_peak));
-      peak[i] += (FL2FX_DBL(0.2f)>>METADATA_INT_BITS);           /* add a little bit headroom */
+      peak[i] += (FL2FX_DBL(0.5f)>>METADATA_INT_BITS);           /* add a little bit headroom */
       peak[i] +=  drcComp->smoothGain[i];
     }
 
diff --git a/libAACenc/src/metadata_compressor.h b/libAACenc/src/metadata_compressor.h
index c77e79e..ff639b5 100644
--- a/libAACenc/src/metadata_compressor.h
+++ b/libAACenc/src/metadata_compressor.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/metadata_main.cpp b/libAACenc/src/metadata_main.cpp
index 45763a1..e920793 100644
--- a/libAACenc/src/metadata_main.cpp
+++ b/libAACenc/src/metadata_main.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/metadata_main.h b/libAACenc/src/metadata_main.h
index f747f9f..bfc8ae1 100644
--- a/libAACenc/src/metadata_main.h
+++ b/libAACenc/src/metadata_main.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/ms_stereo.cpp b/libAACenc/src/ms_stereo.cpp
index ab0cb1c..306d490 100644
--- a/libAACenc/src/ms_stereo.cpp
+++ b/libAACenc/src/ms_stereo.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/ms_stereo.h b/libAACenc/src/ms_stereo.h
index 5657ae2..2f3addb 100644
--- a/libAACenc/src/ms_stereo.h
+++ b/libAACenc/src/ms_stereo.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/noisedet.cpp b/libAACenc/src/noisedet.cpp
index 178a2ad..f3c51de 100644
--- a/libAACenc/src/noisedet.cpp
+++ b/libAACenc/src/noisedet.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/noisedet.h b/libAACenc/src/noisedet.h
index bccf4ee..8d5e365 100644
--- a/libAACenc/src/noisedet.h
+++ b/libAACenc/src/noisedet.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/pns_func.h b/libAACenc/src/pns_func.h
index d1d8fb1..efa44ef 100644
--- a/libAACenc/src/pns_func.h
+++ b/libAACenc/src/pns_func.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/pnsparam.cpp b/libAACenc/src/pnsparam.cpp
index 3426ac3..afc5bdd 100644
--- a/libAACenc/src/pnsparam.cpp
+++ b/libAACenc/src/pnsparam.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/pnsparam.h b/libAACenc/src/pnsparam.h
index 53a2704..08bb83e 100644
--- a/libAACenc/src/pnsparam.h
+++ b/libAACenc/src/pnsparam.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/pre_echo_control.cpp b/libAACenc/src/pre_echo_control.cpp
index b1f9041..3dfd8ed 100644
--- a/libAACenc/src/pre_echo_control.cpp
+++ b/libAACenc/src/pre_echo_control.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/pre_echo_control.h b/libAACenc/src/pre_echo_control.h
index c2743d7..9224db0 100644
--- a/libAACenc/src/pre_echo_control.h
+++ b/libAACenc/src/pre_echo_control.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/psy_configuration.cpp b/libAACenc/src/psy_configuration.cpp
index 96f6a71..4393fa1 100644
--- a/libAACenc/src/psy_configuration.cpp
+++ b/libAACenc/src/psy_configuration.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/psy_configuration.h b/libAACenc/src/psy_configuration.h
index 3d8ad0b..3629246 100644
--- a/libAACenc/src/psy_configuration.h
+++ b/libAACenc/src/psy_configuration.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/psy_const.h b/libAACenc/src/psy_const.h
index 0195931..d9c9f43 100644
--- a/libAACenc/src/psy_const.h
+++ b/libAACenc/src/psy_const.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -98,7 +98,6 @@ amm-info at iis.fraunhofer.de
 
 #define FRAME_MAXLEN_SHORT    ((1024)/TRANS_FAC)
 #define FRAME_LEN_SHORT_128   ((1024)/TRANS_FAC)
-#define FRAME_LEN_SHORT_120   (FRAME_LEN_LONG_960/TRANS_FAC)
 
 /* Filterbank type*/
 enum FB_TYPE {
diff --git a/libAACenc/src/psy_data.h b/libAACenc/src/psy_data.h
index 2219f24..7183955 100644
--- a/libAACenc/src/psy_data.h
+++ b/libAACenc/src/psy_data.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/psy_main.cpp b/libAACenc/src/psy_main.cpp
index 672619e..59193c7 100644
--- a/libAACenc/src/psy_main.cpp
+++ b/libAACenc/src/psy_main.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -116,11 +116,6 @@ static const FIXP_DBL fadeOutFactor[FADE_OUT_LEN] = {1840644096, 1533870080, 122
 /* forward definitions */
 
 
-static inline int isLowDelay( AUDIO_OBJECT_TYPE aot )
-{
-  return (aot==AOT_ER_AAC_LD || aot==AOT_ER_AAC_ELD);
-}
-
 /*****************************************************************************
 
     functionname: FDKaacEnc_PsyNew
@@ -513,28 +508,28 @@ AAC_ENCODER_ERROR FDKaacEnc_psyMain(INT                 channels,
 
       for(ch = 0; ch < channels; ch++)
       {
-          C_ALLOC_SCRATCH_START(timeSignal, INT_PCM, (1024));
-          psyStatic[ch]->blockSwitchingControl.timeSignal = timeSignal;
+          C_ALLOC_SCRATCH_START(pTimeSignal, INT_PCM, (1024))
 
           /* deinterleave input data and use for block switching */
-          FDKaacEnc_deinterleaveInputBuffer( psyStatic[ch]->blockSwitchingControl.timeSignal,
+          FDKaacEnc_deinterleaveInputBuffer( pTimeSignal,
                                             &pInput[chIdx[ch]],
                                              psyConf->granuleLength,
                                              totalChannels);
 
 
           FDKaacEnc_BlockSwitching (&psyStatic[ch]->blockSwitchingControl,
-                                     psyConf->granuleLength
-                                    ,psyStatic[ch]->isLFE
+                                     psyConf->granuleLength,
+                                     psyStatic[ch]->isLFE,
+                                     pTimeSignal
                                    );
 
 
             /* fill up internal input buffer, to 2xframelength samples */
             FDKmemcpy(psyStatic[ch]->psyInputBuffer+blockSwitchingOffset,
-                      psyStatic[ch]->blockSwitchingControl.timeSignal,
+                      pTimeSignal,
                       (2*psyConf->granuleLength-blockSwitchingOffset)*sizeof(INT_PCM));
 
-            C_ALLOC_SCRATCH_END(timeSignal, INT_PCM, (1024));
+            C_ALLOC_SCRATCH_END(pTimeSignal, INT_PCM, (1024))
       }
 
       /* synch left and right block type */
@@ -1345,7 +1340,7 @@ void FDKaacEnc_PsyClose(PSY_INTERNAL   **phPsyInternal,
 
       if (hPsyInternal)
       {
-        for (i=0; i<(6); i++) {
+        for (i=0; i<(8); i++) {
           if (hPsyInternal->pStaticChannels[i]) {
             if (hPsyInternal->pStaticChannels[i]->psyInputBuffer)
               FreeRam_aacEnc_PsyInputBuffer(&hPsyInternal->pStaticChannels[i]->psyInputBuffer);  /* AUDIO INPUT BUFFER */
@@ -1354,7 +1349,7 @@ void FDKaacEnc_PsyClose(PSY_INTERNAL   **phPsyInternal,
           }
         }
 
-        for (i=0; i<(6); i++) {
+        for (i=0; i<(8); i++) {
           if (hPsyInternal->psyElement[i])
             FreeRam_aacEnc_PsyElement(&hPsyInternal->psyElement[i]);                             /* PSY_ELEMENT */
         }
@@ -1368,12 +1363,12 @@ void FDKaacEnc_PsyClose(PSY_INTERNAL   **phPsyInternal,
       for (n=0; n<(1); n++) {
         if (phPsyOut[n])
         {
-          for (i=0; i<(6); i++) {
+          for (i=0; i<(8); i++) {
             if (phPsyOut[n]->pPsyOutChannels[i])
               FreeRam_aacEnc_PsyOutChannel(&phPsyOut[n]->pPsyOutChannels[i]);                  /* PSY_OUT_CHANNEL */
           }
 
-          for (i=0; i<(6); i++) {
+          for (i=0; i<(8); i++) {
             if (phPsyOut[n]->psyOutElement[i])
               FreeRam_aacEnc_PsyOutElements(&phPsyOut[n]->psyOutElement[i]);                   /* PSY_OUT_ELEMENTS */
           }
diff --git a/libAACenc/src/psy_main.h b/libAACenc/src/psy_main.h
index 9670f4a..7bdcc38 100644
--- a/libAACenc/src/psy_main.h
+++ b/libAACenc/src/psy_main.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -117,8 +117,8 @@ typedef struct  {
 typedef struct  {
 
   PSY_CONFIGURATION  psyConf[2]; /* LONG / SHORT */
-  PSY_ELEMENT*       psyElement[(6)];
-  PSY_STATIC*        pStaticChannels[(6)];
+  PSY_ELEMENT*       psyElement[(8)];
+  PSY_STATIC*        pStaticChannels[(8)];
   PSY_DYNAMIC*       psyDynamic;
   INT                granuleLength;
 
diff --git a/libAACenc/src/qc_data.h b/libAACenc/src/qc_data.h
index d37ea92..a9309c8 100644
--- a/libAACenc/src/qc_data.h
+++ b/libAACenc/src/qc_data.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -127,7 +127,7 @@ typedef struct {
   INT nChannels;
   INT nChannelsEff;
   INT nElements;
-  ELEMENT_INFO elInfo[(6)];
+  ELEMENT_INFO elInfo[(8)];
 } CHANNEL_MAPPING;
 
 typedef struct {
@@ -143,6 +143,8 @@ struct QC_INIT{
   INT maxBits;     /* maximum number of bits in reservoir  */
   INT averageBits; /* average number of bits we should use */
   INT bitRes;
+  INT sampleRate;       /* output sample rate                   */
+  INT advancedBitsToPe; /* if set, calc bits2PE factor depending on samplerate */
   INT staticBits;  /* Bits per frame consumed by transport layers. */
   QCDATA_BR_MODE bitrateMode;
   INT meanPe;
@@ -215,8 +217,8 @@ typedef struct
 
 typedef struct
 {
-  QC_OUT_ELEMENT    *qcElement[(6)];
-  QC_OUT_CHANNEL    *pQcOutChannels[(6)];
+  QC_OUT_ELEMENT    *qcElement[(8)];
+  QC_OUT_CHANNEL    *pQcOutChannels[(8)];
   QC_OUT_EXTENSION   extension[(2+2)];  /* global extension payload */
   INT          nExtensions;       /* number of extension payloads for this AU */
   INT          maxDynBits;        /* maximal allowed dynamic bits in frame */
@@ -252,7 +254,7 @@ typedef struct
   INT minBitsPerFrame;   /* minimal allowd bits per fram, superframing - DRM */
   INT nElements;
   QCDATA_BR_MODE bitrateMode;
-  INT bitDistributenMode; /* 0: full bitreservoir, 1: reduced bitreservoir, 2: disabled bitreservoir */
+  INT bitDistributionMode; /* 0: full bitreservoir, 1: reduced bitreservoir, 2: disabled bitreservoir */
   INT bitResTot;
   INT bitResTotMax;
   INT maxIterations;      /* Maximum number of allowed iterations before FDKaacEnc_crashRecovery() is applied. */
@@ -263,7 +265,7 @@ typedef struct
 
   PADDING padding;
 
-  ELEMENT_BITS  *elementBits[(6)];
+  ELEMENT_BITS  *elementBits[(8)];
   BITCNTR_STATE *hBitCounter;
   ADJ_THR_STATE *hAdjThr;
 
diff --git a/libAACenc/src/qc_main.cpp b/libAACenc/src/qc_main.cpp
index d7e76c7..b74510a 100644
--- a/libAACenc/src/qc_main.cpp
+++ b/libAACenc/src/qc_main.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -382,10 +382,10 @@ AAC_ENCODER_ERROR FDKaacEnc_QCInit(QC_STATE *hQC,
   if ( isConstantBitrateMode(hQC->bitrateMode) ) {
     INT bitresPerChannel = (hQC->bitResTotMax / init->channelMapping->nChannelsEff);
     /* 0: full bitreservoir, 1: reduced bitreservoir, 2: disabled bitreservoir */
-    hQC->bitDistributenMode = (bitresPerChannel>50) ? 0 : (bitresPerChannel>0) ? 1 : 2;
+    hQC->bitDistributionMode = (bitresPerChannel>100) ? 0 : (bitresPerChannel>0) ? 1 : 2;
   }
   else {
-    hQC->bitDistributenMode = 0; /* full bitreservoir */
+    hQC->bitDistributionMode = 0; /* full bitreservoir */
   }
 
 
@@ -420,11 +420,17 @@ AAC_ENCODER_ERROR FDKaacEnc_QCInit(QC_STATE *hQC,
       break;
   }
 
-  FDKaacEnc_AdjThrInit(hQC->hAdjThr,
-             init->meanPe,
-             hQC->elementBits,                 /* or channelBitrates, was: channelBitrate */
-             init->channelMapping->nElements,
-             hQC->vbrQualFactor);
+  FDKaacEnc_AdjThrInit(
+        hQC->hAdjThr,
+        init->meanPe,
+        hQC->elementBits,                 /* or channelBitrates, was: channelBitrate */
+        hQC->invQuant,
+        init->channelMapping->nElements,
+        init->channelMapping->nChannelsEff,
+        init->sampleRate,                 /* output sample rate */
+        init->advancedBitsToPe,           /* if set, calc bits2PE factor depending on samplerate */
+        hQC->vbrQualFactor
+        );
 
   return AAC_ENC_OK;
 }
@@ -505,7 +511,7 @@ AAC_ENCODER_ERROR FDKaacEnc_AdjustBitrate(QC_STATE        *RESTRICT hQC,
 }
 
 static AAC_ENCODER_ERROR FDKaacEnc_distributeElementDynBits(QC_STATE*         hQC,
-                                                  QC_OUT_ELEMENT*   qcElement[(6)],
+                                                  QC_OUT_ELEMENT*   qcElement[(8)],
                                                   CHANNEL_MAPPING*  cm,
                                                   INT               codeBits)
 {
@@ -604,7 +610,7 @@ static AAC_ENCODER_ERROR FDKaacEnc_prepareBitDistribution(QC_STATE*            h
                                                 PSY_OUT**            psyOut,
                                                 QC_OUT**             qcOut,
                                                 CHANNEL_MAPPING*     cm,
-                                                QC_OUT_ELEMENT*      qcElement[(1)][(6)],
+                                                QC_OUT_ELEMENT*      qcElement[(1)][(8)],
                                                 INT                  avgTotalBits,
                                                 INT                 *totalAvailableBits,
                                                 INT                 *avgTotalDynBits)
@@ -655,7 +661,7 @@ static AAC_ENCODER_ERROR FDKaacEnc_prepareBitDistribution(QC_STATE*            h
                                          hQC->elementBits[i]->bitResLevelEl,
                                          hQC->elementBits[i]->maxBitResBitsEl,
                                          hQC->maxBitFac,
-                                         hQC->bitDistributenMode);
+                                         hQC->bitDistributionMode);
 
                 *totalAvailableBits += hQC->elementBits[i]->bitResLevelEl;
         /* get total corrected granted PE */
@@ -671,7 +677,7 @@ static AAC_ENCODER_ERROR FDKaacEnc_prepareBitDistribution(QC_STATE*            h
 
 ////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
 static AAC_ENCODER_ERROR FDKaacEnc_updateUsedDynBits(INT*               sumDynBitsConsumed,
-                                            QC_OUT_ELEMENT*    qcElement[(6)],
+                                            QC_OUT_ELEMENT*    qcElement[(8)],
                                             CHANNEL_MAPPING*   cm)
 {
   INT i;
@@ -714,7 +720,7 @@ static INT FDKaacEnc_getTotalConsumedDynBits(QC_OUT** qcOut,
 }
 
 static INT FDKaacEnc_getTotalConsumedBits(QC_OUT**          qcOut,
-                                QC_OUT_ELEMENT*   qcElement[(1)][(6)],
+                                QC_OUT_ELEMENT*   qcElement[(1)][(8)],
                                 CHANNEL_MAPPING*  cm,
                                 INT               globHdrBits,
                                 INT               nSubFrames)
@@ -814,7 +820,7 @@ AAC_ENCODER_ERROR FDKaacEnc_QCMain(QC_STATE* RESTRICT         hQC,
 
       /*-------------------------------------------- */
       /* helper pointer */
-      QC_OUT_ELEMENT*  qcElement[(1)][(6)];
+      QC_OUT_ELEMENT*  qcElement[(1)][(8)];
 
       /* work on a copy of qcChannel and qcElement */
       for (i=0; i<cm->nElements; i++)
@@ -891,10 +897,10 @@ AAC_ENCODER_ERROR FDKaacEnc_QCMain(QC_STATE* RESTRICT         hQC,
       } /* -end- sub frame counter */
 
       /*-------------------------------------------- */
-      INT iterations[(1)][(6)];
-      INT chConstraintsFulfilled[(1)][(6)][(2)];
-      INT calculateQuant[(1)][(6)][(2)];
-      INT constraintsFulfilled[(1)][(6)];
+      INT iterations[(1)][(8)];
+      INT chConstraintsFulfilled[(1)][(8)][(2)];
+      INT calculateQuant[(1)][(8)][(2)];
+      INT constraintsFulfilled[(1)][(8)];
       /*-------------------------------------------- */
 
 
@@ -1241,7 +1247,7 @@ static AAC_ENCODER_ERROR FDKaacEnc_reduceBitConsumption(int*             iterati
 
 AAC_ENCODER_ERROR FDKaacEnc_updateFillBits(CHANNEL_MAPPING*          cm,
                                            QC_STATE*                 qcKernel,
-                                           ELEMENT_BITS* RESTRICT    elBits[(6)],
+                                           ELEMENT_BITS* RESTRICT    elBits[(8)],
                                            QC_OUT**                  qcOut)
 {
   switch (qcKernel->bitrateMode) {
@@ -1597,10 +1603,10 @@ void  FDKaacEnc_QCClose (QC_STATE  **phQCstate, QC_OUT **phQC)
     for (n=0;n<(1);n++) {
       if (phQC[n] != NULL) {
         QC_OUT    *hQC      = phQC[n];
-        for (i=0; i<(6); i++) {
+        for (i=0; i<(8); i++) {
         }
 
-        for (i=0; i<(6); i++) {
+        for (i=0; i<(8); i++) {
           if (hQC->qcElement[i])
             FreeRam_aacEnc_QCelement(&hQC->qcElement[i]);
         }
@@ -1620,7 +1626,7 @@ void  FDKaacEnc_QCClose (QC_STATE  **phQCstate, QC_OUT **phQC)
       if (hQCstate->hBitCounter != NULL)
         FDKaacEnc_BCClose(&hQCstate->hBitCounter);
 
-      for (i=0; i<(6); i++) {
+      for (i=0; i<(8); i++) {
         if (hQCstate->elementBits[i]!=NULL) {
           FreeRam_aacEnc_ElementBits(&hQCstate->elementBits[i]);
         }
diff --git a/libAACenc/src/qc_main.h b/libAACenc/src/qc_main.h
index dadac8e..4e8c042 100644
--- a/libAACenc/src/qc_main.h
+++ b/libAACenc/src/qc_main.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -140,7 +140,7 @@ AAC_ENCODER_ERROR FDKaacEnc_QCMain(QC_STATE* RESTRICT         hQC,
 
 AAC_ENCODER_ERROR FDKaacEnc_updateFillBits(CHANNEL_MAPPING*          cm,
                                            QC_STATE*                 qcKernel,
-                                           ELEMENT_BITS* RESTRICT    elBits[(6)],
+                                           ELEMENT_BITS* RESTRICT    elBits[(8)],
                                            QC_OUT**                  qcOut);
 
 
diff --git a/libAACenc/src/quantize.cpp b/libAACenc/src/quantize.cpp
index 1f7402b..5380e35 100644
--- a/libAACenc/src/quantize.cpp
+++ b/libAACenc/src/quantize.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -127,10 +127,7 @@ static void FDKaacEnc_quantizeLines(INT      gain,
       accu = fMultDiv2(FDKaacEnc_mTab_3_4[tabIndex],FDKaacEnc_quantTableE[totalShift&3]);
       totalShift = (16-4)-(3*(totalShift>>2));
       FDK_ASSERT(totalShift >=0); /* MAX_QUANT_VIOLATION */
-      if (totalShift < 32)
-          accu>>=totalShift;
-      else
-          accu = 0;
+      accu >>= fixMin(totalShift,DFRACT_BITS-1);
       quaSpectrum[line] = (SHORT)(-((LONG)(k + accu) >> (DFRACT_BITS-1-16)));
     }
     else if(accu > FL2FXCONST_DBL(0.0f))
@@ -143,10 +140,7 @@ static void FDKaacEnc_quantizeLines(INT      gain,
       accu = fMultDiv2(FDKaacEnc_mTab_3_4[tabIndex],FDKaacEnc_quantTableE[totalShift&3]);
       totalShift = (16-4)-(3*(totalShift>>2));
       FDK_ASSERT(totalShift >=0); /* MAX_QUANT_VIOLATION */
-      if (totalShift < 32)
-          accu>>=totalShift;
-      else
-          accu = 0;
+      accu >>= fixMin(totalShift,DFRACT_BITS-1);
       quaSpectrum[line] = (SHORT)((LONG)(k + accu) >> (DFRACT_BITS-1-16));
     }
     else
diff --git a/libAACenc/src/quantize.h b/libAACenc/src/quantize.h
index 72dd851..975b98e 100644
--- a/libAACenc/src/quantize.h
+++ b/libAACenc/src/quantize.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/sf_estim.cpp b/libAACenc/src/sf_estim.cpp
index c5512cb..72b75a6 100644
--- a/libAACenc/src/sf_estim.cpp
+++ b/libAACenc/src/sf_estim.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/sf_estim.h b/libAACenc/src/sf_estim.h
index 3338a26..b5ac000 100644
--- a/libAACenc/src/sf_estim.h
+++ b/libAACenc/src/sf_estim.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/spreading.cpp b/libAACenc/src/spreading.cpp
index 5141b6e..852da1e 100644
--- a/libAACenc/src/spreading.cpp
+++ b/libAACenc/src/spreading.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/spreading.h b/libAACenc/src/spreading.h
index 078cc7f..e1b506c 100644
--- a/libAACenc/src/spreading.h
+++ b/libAACenc/src/spreading.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/tns_func.h b/libAACenc/src/tns_func.h
index 8f9bd26..6ee0edb 100644
--- a/libAACenc/src/tns_func.h
+++ b/libAACenc/src/tns_func.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/tonality.cpp b/libAACenc/src/tonality.cpp
index befff74..7246a34 100644
--- a/libAACenc/src/tonality.cpp
+++ b/libAACenc/src/tonality.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/tonality.h b/libAACenc/src/tonality.h
index dcd82a8..fbe78ee 100644
--- a/libAACenc/src/tonality.h
+++ b/libAACenc/src/tonality.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/transform.cpp b/libAACenc/src/transform.cpp
index fb57b14..690b82e 100644
--- a/libAACenc/src/transform.cpp
+++ b/libAACenc/src/transform.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libAACenc/src/transform.h b/libAACenc/src/transform.h
index 212f5f0..5053174 100644
--- a/libAACenc/src/transform.h
+++ b/libAACenc/src/transform.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -105,7 +105,7 @@ amm-info at iis.fraunhofer.de
  *        LONG_WINDOW, START_WINDOW, SHORT_WINDOW or STOP_WINDOW.
  * \param windowShape index indicating the window slope type to be used.
  *        Values allowed are either SINE_WINDOW or KBD_WINDOW.
- * \param frameLength length of the block. Either 1024 or 960.
+ * \param frameLength length of the block.
  * \param mdctData_e pointer to an INT where the exponent of the frequency
  *        domain output data is stored into.
  * \return 0 in case of success, non-zero in case of error (inconsistent parameters).
diff --git a/libFDK/include/FDK_archdef.h b/libFDK/include/FDK_archdef.h
index a963f55..da63aee 100644
--- a/libFDK/include/FDK_archdef.h
+++ b/libFDK/include/FDK_archdef.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/include/FDK_bitbuffer.h b/libFDK/include/FDK_bitbuffer.h
index 963ada8..65fa04b 100644
--- a/libFDK/include/FDK_bitbuffer.h
+++ b/libFDK/include/FDK_bitbuffer.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/include/FDK_bitstream.h b/libFDK/include/FDK_bitstream.h
index 3ef4799..fc8d7de 100644
--- a/libFDK/include/FDK_bitstream.h
+++ b/libFDK/include/FDK_bitstream.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/include/FDK_core.h b/libFDK/include/FDK_core.h
index ef0c032..10a6a8e 100644
--- a/libFDK/include/FDK_core.h
+++ b/libFDK/include/FDK_core.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/include/FDK_crc.h b/libFDK/include/FDK_crc.h
index b5b8d6b..b1fed88 100644
--- a/libFDK/include/FDK_crc.h
+++ b/libFDK/include/FDK_crc.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/include/FDK_hybrid.h b/libFDK/include/FDK_hybrid.h
index 3d355d6..63f2d70 100644
--- a/libFDK/include/FDK_hybrid.h
+++ b/libFDK/include/FDK_hybrid.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/include/FDK_tools_rom.h b/libFDK/include/FDK_tools_rom.h
index e9e126e..9361c81 100644
--- a/libFDK/include/FDK_tools_rom.h
+++ b/libFDK/include/FDK_tools_rom.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -171,6 +171,9 @@ extern const FIXP_WTP sin_twiddle_L64[];
 
 extern const FIXP_QTW qmf_phaseshift_cos32[32];
 extern const FIXP_QTW qmf_phaseshift_sin32[32];
+/* Adapted analysis post-twiddles for down-sampled HQ SBR */
+extern const FIXP_QTW qmf_phaseshift_cos_downsamp32[32];
+extern const FIXP_QTW qmf_phaseshift_sin_downsamp32[32];
 extern const FIXP_QTW qmf_phaseshift_cos64[64];
 extern const FIXP_QTW qmf_phaseshift_sin64[64];
 
diff --git a/libFDK/include/FDK_trigFcts.h b/libFDK/include/FDK_trigFcts.h
index d2ca4ee..c32c0f2 100644
--- a/libFDK/include/FDK_trigFcts.h
+++ b/libFDK/include/FDK_trigFcts.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/include/abs.h b/libFDK/include/abs.h
index d3576cb..0837e3c 100644
--- a/libFDK/include/abs.h
+++ b/libFDK/include/abs.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/include/arm/clz_arm.h b/libFDK/include/arm/clz_arm.h
index b33b800..f43a7b1 100644
--- a/libFDK/include/arm/clz_arm.h
+++ b/libFDK/include/arm/clz_arm.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/include/arm/cplx_mul.h b/libFDK/include/arm/cplx_mul.h
index 2416f98..8b48031 100644
--- a/libFDK/include/arm/cplx_mul.h
+++ b/libFDK/include/arm/cplx_mul.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/include/arm/fixmadd_arm.h b/libFDK/include/arm/fixmadd_arm.h
index e12d211..111147b 100644
--- a/libFDK/include/arm/fixmadd_arm.h
+++ b/libFDK/include/arm/fixmadd_arm.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/include/arm/fixmul_arm.h b/libFDK/include/arm/fixmul_arm.h
index d42d677..2f04705 100644
--- a/libFDK/include/arm/fixmul_arm.h
+++ b/libFDK/include/arm/fixmul_arm.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/include/arm/scale.h b/libFDK/include/arm/scale.h
index 41f94a4..cbb81dc 100644
--- a/libFDK/include/arm/scale.h
+++ b/libFDK/include/arm/scale.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/include/arm/scramble.h b/libFDK/include/arm/scramble.h
index ddf34bd..82d0338 100644
--- a/libFDK/include/arm/scramble.h
+++ b/libFDK/include/arm/scramble.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/include/autocorr2nd.h b/libFDK/include/autocorr2nd.h
index 227d52b..c2e9595 100644
--- a/libFDK/include/autocorr2nd.h
+++ b/libFDK/include/autocorr2nd.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/include/clz.h b/libFDK/include/clz.h
index 38c5073..fc612e1 100644
--- a/libFDK/include/clz.h
+++ b/libFDK/include/clz.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/include/common_fix.h b/libFDK/include/common_fix.h
index a39548e..79c87ac 100644
--- a/libFDK/include/common_fix.h
+++ b/libFDK/include/common_fix.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/include/cplx_mul.h b/libFDK/include/cplx_mul.h
index 68a49e1..093ffa6 100644
--- a/libFDK/include/cplx_mul.h
+++ b/libFDK/include/cplx_mul.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/include/dct.h b/libFDK/include/dct.h
index 59b7319..af1edd3 100644
--- a/libFDK/include/dct.h
+++ b/libFDK/include/dct.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/include/fft.h b/libFDK/include/fft.h
index 5bd49d1..616f350 100644
--- a/libFDK/include/fft.h
+++ b/libFDK/include/fft.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/include/fft_rad2.h b/libFDK/include/fft_rad2.h
index 97ddf5f..90d14da 100644
--- a/libFDK/include/fft_rad2.h
+++ b/libFDK/include/fft_rad2.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/include/fixmadd.h b/libFDK/include/fixmadd.h
index 71413ee..851b7a2 100644
--- a/libFDK/include/fixmadd.h
+++ b/libFDK/include/fixmadd.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/include/fixminmax.h b/libFDK/include/fixminmax.h
index a6aa623..5a52824 100644
--- a/libFDK/include/fixminmax.h
+++ b/libFDK/include/fixminmax.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/include/fixmul.h b/libFDK/include/fixmul.h
index 31c92b4..ea4bc6c 100644
--- a/libFDK/include/fixmul.h
+++ b/libFDK/include/fixmul.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/include/fixpoint_math.h b/libFDK/include/fixpoint_math.h
index ae554cb..df141d3 100644
--- a/libFDK/include/fixpoint_math.h
+++ b/libFDK/include/fixpoint_math.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/include/mdct.h b/libFDK/include/mdct.h
index 0d04e4b..e1c2d74 100644
--- a/libFDK/include/mdct.h
+++ b/libFDK/include/mdct.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/include/mips/abs_mips.h b/libFDK/include/mips/abs_mips.h
index e2ad607..3b9f876 100644
--- a/libFDK/include/mips/abs_mips.h
+++ b/libFDK/include/mips/abs_mips.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/include/mips/clz_mips.h b/libFDK/include/mips/clz_mips.h
index 48d465b..ff40a09 100644
--- a/libFDK/include/mips/clz_mips.h
+++ b/libFDK/include/mips/clz_mips.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/include/mips/cplx_mul.h b/libFDK/include/mips/cplx_mul.h
index fb777ce..e05d2b6 100644
--- a/libFDK/include/mips/cplx_mul.h
+++ b/libFDK/include/mips/cplx_mul.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/include/mips/fixmadd_mips.h b/libFDK/include/mips/fixmadd_mips.h
index 564268b..a9575fb 100644
--- a/libFDK/include/mips/fixmadd_mips.h
+++ b/libFDK/include/mips/fixmadd_mips.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/include/mips/fixmul_mips.h b/libFDK/include/mips/fixmul_mips.h
index 62d059e..73693e2 100644
--- a/libFDK/include/mips/fixmul_mips.h
+++ b/libFDK/include/mips/fixmul_mips.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/include/mips/scale.h b/libFDK/include/mips/scale.h
index 9279391..edf29da 100644
--- a/libFDK/include/mips/scale.h
+++ b/libFDK/include/mips/scale.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/include/mips/scramble.h b/libFDK/include/mips/scramble.h
index 95c4d74..3f2fe6e 100644
--- a/libFDK/include/mips/scramble.h
+++ b/libFDK/include/mips/scramble.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/include/qmf.h b/libFDK/include/qmf.h
index 16d2b70..be69477 100644
--- a/libFDK/include/qmf.h
+++ b/libFDK/include/qmf.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -137,6 +137,8 @@ amm-info at iis.fraunhofer.de
 #define QMF_FLAG_MPSLDFB     16
 /* Complex Low Delay Filter Bank used in MPEG Surround Encoder allows a optimized calculation of the modulation in qmfForwardModulationHQ() */
 #define QMF_FLAG_MPSLDFB_OPTIMIZE_MODULATION  32
+/* Flag to indicate HE-AAC down-sampled SBR mode (decoder) -> adapt analysis post twiddling */
+#define QMF_FLAG_DOWNSAMPLED  64
 
 
 typedef struct
diff --git a/libFDK/include/scale.h b/libFDK/include/scale.h
index d60ed2c..1587f9f 100644
--- a/libFDK/include/scale.h
+++ b/libFDK/include/scale.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/include/scramble.h b/libFDK/include/scramble.h
index 552081a..ceaf527 100644
--- a/libFDK/include/scramble.h
+++ b/libFDK/include/scramble.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/include/x86/abs_x86.h b/libFDK/include/x86/abs_x86.h
index 64add9f..96374ec 100644
--- a/libFDK/include/x86/abs_x86.h
+++ b/libFDK/include/x86/abs_x86.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/include/x86/clz_x86.h b/libFDK/include/x86/clz_x86.h
index db7970d..1bc43e5 100644
--- a/libFDK/include/x86/clz_x86.h
+++ b/libFDK/include/x86/clz_x86.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/include/x86/fixmul_x86.h b/libFDK/include/x86/fixmul_x86.h
index d10bc9a..7b74cc7 100644
--- a/libFDK/include/x86/fixmul_x86.h
+++ b/libFDK/include/x86/fixmul_x86.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/src/FDK_bitbuffer.cpp b/libFDK/src/FDK_bitbuffer.cpp
index 113c819..680ceae 100644
--- a/libFDK/src/FDK_bitbuffer.cpp
+++ b/libFDK/src/FDK_bitbuffer.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/src/FDK_core.cpp b/libFDK/src/FDK_core.cpp
index 8a07aa9..6db648d 100644
--- a/libFDK/src/FDK_core.cpp
+++ b/libFDK/src/FDK_core.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -92,8 +92,8 @@ amm-info at iis.fraunhofer.de
 
 /* FDK tools library info */
 #define FDK_TOOLS_LIB_VL0 2
-#define FDK_TOOLS_LIB_VL1 2
-#define FDK_TOOLS_LIB_VL2 8
+#define FDK_TOOLS_LIB_VL1 3
+#define FDK_TOOLS_LIB_VL2 2
 #define FDK_TOOLS_LIB_TITLE "FDK Tools"
 #define FDK_TOOLS_LIB_BUILD_DATE __DATE__
 #define FDK_TOOLS_LIB_BUILD_TIME __TIME__
diff --git a/libFDK/src/FDK_crc.cpp b/libFDK/src/FDK_crc.cpp
index 1b03903..17d47ad 100644
--- a/libFDK/src/FDK_crc.cpp
+++ b/libFDK/src/FDK_crc.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/src/FDK_hybrid.cpp b/libFDK/src/FDK_hybrid.cpp
index 3d3ab7b..20816f5 100644
--- a/libFDK/src/FDK_hybrid.cpp
+++ b/libFDK/src/FDK_hybrid.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/src/FDK_tools_rom.cpp b/libFDK/src/FDK_tools_rom.cpp
index 4fc80ce..c6e517e 100644
--- a/libFDK/src/FDK_tools_rom.cpp
+++ b/libFDK/src/FDK_tools_rom.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -1462,6 +1462,26 @@ const FIXP_PFT qmf_64[QMF640_PFT_TABLE_SIZE+QMF_NO_POLY] =
   QFC(0xfe4d1be3), QFC(0xd1c58ace), QFC(0x2e3a7532), QFC(0x01b2e41d), QFC(0x00000000),
 };
 
+RAM_ALIGN
+LNK_SECTION_CONSTDATA
+const FIXP_QTW qmf_phaseshift_cos_downsamp32[] =
+{
+  QTC(0x7fd8878e), QTC(0x7e9d55fc), QTC(0x7c29fbee), QTC(0x78848414), QTC(0x73b5ebd1), QTC(0x6dca0d14), QTC(0x66cf8120), QTC(0x5ed77c8a),
+  QTC(0x55f5a4d2), QTC(0x4c3fdff4), QTC(0x41ce1e65), QTC(0x36ba2014), QTC(0x2b1f34eb), QTC(0x1f19f97b), QTC(0x12c8106f), QTC(0x0647d97c),
+  QTC(0xf9b82684), QTC(0xed37ef91), QTC(0xe0e60685), QTC(0xd4e0cb15), QTC(0xc945dfec), QTC(0xbe31e19b), QTC(0xb3c0200c), QTC(0xaa0a5b2e),
+  QTC(0xa1288376), QTC(0x99307ee0), QTC(0x9235f2ec), QTC(0x8c4a142f), QTC(0x877b7bec), QTC(0x83d60412), QTC(0x8162aa04), QTC(0x80277872),
+};
+
+RAM_ALIGN
+LNK_SECTION_CONSTDATA
+const FIXP_QTW qmf_phaseshift_sin_downsamp32[] =
+{
+  QTC(0x0647d97c), QTC(0x12c8106f), QTC(0x1f19f97b), QTC(0x2b1f34eb), QTC(0x36ba2014), QTC(0x41ce1e65), QTC(0x4c3fdff4), QTC(0x55f5a4d2),
+  QTC(0x5ed77c8a), QTC(0x66cf8120), QTC(0x6dca0d14), QTC(0x73b5ebd1), QTC(0x78848414), QTC(0x7c29fbee), QTC(0x7e9d55fc), QTC(0x7fd8878e),
+  QTC(0x7fd8878e), QTC(0x7e9d55fc), QTC(0x7c29fbee), QTC(0x78848414), QTC(0x73b5ebd1), QTC(0x6dca0d14), QTC(0x66cf8120), QTC(0x5ed77c8a),
+  QTC(0x55f5a4d2), QTC(0x4c3fdff4), QTC(0x41ce1e65), QTC(0x36ba2014), QTC(0x2b1f34eb), QTC(0x1f19f97b), QTC(0x12c8106f), QTC(0x0647d97c),
+};
+
 #else /* QMF_NO_POLY == 5 */
 #endif /* QMF_NO_POLY==5 */
 
diff --git a/libFDK/src/FDK_trigFcts.cpp b/libFDK/src/FDK_trigFcts.cpp
index 2f8655c..1f3a017 100644
--- a/libFDK/src/FDK_trigFcts.cpp
+++ b/libFDK/src/FDK_trigFcts.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/src/arm/autocorr2nd.cpp b/libFDK/src/arm/autocorr2nd.cpp
index 18c4e43..cc76e30 100644
--- a/libFDK/src/arm/autocorr2nd.cpp
+++ b/libFDK/src/arm/autocorr2nd.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/src/arm/dct_arm.cpp b/libFDK/src/arm/dct_arm.cpp
index 187c843..59b773e 100644
--- a/libFDK/src/arm/dct_arm.cpp
+++ b/libFDK/src/arm/dct_arm.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/src/arm/fft_rad2_arm.cpp b/libFDK/src/arm/fft_rad2_arm.cpp
index 34d71c3..a6e2e42 100644
--- a/libFDK/src/arm/fft_rad2_arm.cpp
+++ b/libFDK/src/arm/fft_rad2_arm.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/src/arm/qmf_arm.cpp b/libFDK/src/arm/qmf_arm.cpp
index 39795d7..0c0ce80 100644
--- a/libFDK/src/arm/qmf_arm.cpp
+++ b/libFDK/src/arm/qmf_arm.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/src/arm/scale_arm.cpp b/libFDK/src/arm/scale_arm.cpp
index 02f3b71..e2837ef 100644
--- a/libFDK/src/arm/scale_arm.cpp
+++ b/libFDK/src/arm/scale_arm.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/src/autocorr2nd.cpp b/libFDK/src/autocorr2nd.cpp
index e73ef14..b7eae7c 100644
--- a/libFDK/src/autocorr2nd.cpp
+++ b/libFDK/src/autocorr2nd.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/src/dct.cpp b/libFDK/src/dct.cpp
index 5dc9e78..3c755bd 100644
--- a/libFDK/src/dct.cpp
+++ b/libFDK/src/dct.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/src/fft.cpp b/libFDK/src/fft.cpp
index bbf3d99..653a71a 100644
--- a/libFDK/src/fft.cpp
+++ b/libFDK/src/fft.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/src/fft_rad2.cpp b/libFDK/src/fft_rad2.cpp
index 4eaa27e..9cdadb5 100644
--- a/libFDK/src/fft_rad2.cpp
+++ b/libFDK/src/fft_rad2.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/src/fixpoint_math.cpp b/libFDK/src/fixpoint_math.cpp
index 000820c..30283ff 100644
--- a/libFDK/src/fixpoint_math.cpp
+++ b/libFDK/src/fixpoint_math.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -242,30 +242,72 @@ FIXP_DBL mul_dbl_sgl_rnd (const FIXP_DBL op1, const FIXP_SGL op2)
 
 *****************************************************************************/
 LNK_SECTION_CODE_L1
-FIXP_DBL CalcInvLdData(FIXP_DBL op)
+/* This table is used for lookup 2^x with   */
+/* x in range [0...1.0[ in steps of 1/32 */
+LNK_SECTION_DATA_L1 static const UINT exp2_tab_long[32]={
+0x40000000,0x4166C34C,0x42D561B4,0x444C0740,
+0x45CAE0F2,0x47521CC6,0x48E1E9BA,0x4A7A77D4,
+0x4C1BF829,0x4DC69CDD,0x4F7A9930,0x51382182,
+0x52FF6B55,0x54D0AD5A,0x56AC1F75,0x5891FAC1,
+0x5A82799A,0x5C7DD7A4,0x5E8451D0,0x60962665,
+0x62B39509,0x64DCDEC3,0x6712460B,0x69540EC9,
+0x6BA27E65,0x6DFDDBCC,0x70666F76,0x72DC8374,
+0x75606374,0x77F25CCE,0x7A92BE8B,0x7D41D96E
+// 0x80000000
+};
+
+/* This table is used for lookup 2^x with   */
+/* x in range [0...1/32[ in steps of 1/1024 */
+LNK_SECTION_DATA_L1 static const UINT exp2w_tab_long[32]={
+0x40000000,0x400B1818,0x4016321B,0x40214E0C,
+0x402C6BE9,0x40378BB4,0x4042AD6D,0x404DD113,
+0x4058F6A8,0x40641E2B,0x406F479E,0x407A7300,
+0x4085A051,0x4090CF92,0x409C00C4,0x40A733E6,
+0x40B268FA,0x40BD9FFF,0x40C8D8F5,0x40D413DD,
+0x40DF50B8,0x40EA8F86,0x40F5D046,0x410112FA,
+0x410C57A2,0x41179E3D,0x4122E6CD,0x412E3152,
+0x41397DCC,0x4144CC3B,0x41501CA0,0x415B6EFB,
+// 0x4166C34C,
+};
+/* This table is used for lookup 2^x with   */
+/* x in range [0...1/1024[ in steps of 1/32768 */
+LNK_SECTION_DATA_L1 static const UINT exp2x_tab_long[32]={
+0x40000000,0x400058B9,0x4000B173,0x40010A2D,
+0x400162E8,0x4001BBA3,0x4002145F,0x40026D1B,
+0x4002C5D8,0x40031E95,0x40037752,0x4003D011,
+0x400428CF,0x4004818E,0x4004DA4E,0x4005330E,
+0x40058BCE,0x4005E48F,0x40063D51,0x40069613,
+0x4006EED5,0x40074798,0x4007A05B,0x4007F91F,
+0x400851E4,0x4008AAA8,0x4009036E,0x40095C33,
+0x4009B4FA,0x400A0DC0,0x400A6688,0x400ABF4F,
+//0x400B1818
+};
+
+LNK_SECTION_CODE_L1 FIXP_DBL CalcInvLdData(FIXP_DBL x)
 {
-  FIXP_DBL result_m;
+  int set_zero = (x <  FL2FXCONST_DBL(-31.0/64.0))? 0 : 1;
+  int set_max  = (x >= FL2FXCONST_DBL( 31.0/64.0)) | (x == FL2FXCONST_DBL(0.0));
 
-  if ( op == FL2FXCONST_DBL(0.0f) ) {
-    result_m = (FIXP_DBL)MAXVAL_DBL;
-  }
-  else if ( op < FL2FXCONST_DBL(0.0f) ) {
-    result_m = f2Pow(op, LD_DATA_SHIFT);
-  }
-  else {
-    int result_e;
+  FIXP_SGL frac = (FIXP_SGL)(LONG)(x & 0x3FF);
+  UINT index3 = (UINT)(LONG)(x >> 10) & 0x1F;
+  UINT index2 = (UINT)(LONG)(x >> 15) & 0x1F; 
+  UINT index1 = (UINT)(LONG)(x >> 20) & 0x1F;
+  int exp  = (x >  FL2FXCONST_DBL(0.0f)) ? (31 - (int)(x>>25)) : (int)(-(x>>25));
 
-    result_m = f2Pow(op, LD_DATA_SHIFT, &result_e);
-    result_e = fixMin(fixMax(result_e+1-(DFRACT_BITS-1), -(DFRACT_BITS-1)), (DFRACT_BITS-1)); /* rounding and saturation */
+  UINT lookup1 = exp2_tab_long[index1]*set_zero;
+  UINT lookup2 = exp2w_tab_long[index2];
+  UINT lookup3 = exp2x_tab_long[index3];
+  UINT lookup3f = lookup3 + (UINT)(LONG)fMultDiv2((FIXP_DBL)(0x0016302F),(FIXP_SGL)frac);
+  
+  UINT lookup12 = (UINT)(LONG)fMult((FIXP_DBL)lookup1,  (FIXP_DBL) lookup2);
+  UINT lookup   = (UINT)(LONG)fMult((FIXP_DBL)lookup12, (FIXP_DBL) lookup3f);
 
-    if ( (result_e>0) && ( result_m > (((FIXP_DBL)MAXVAL_DBL)>>result_e) ) ) {
-      result_m = (FIXP_DBL)MAXVAL_DBL; /* saturate to max representable value */
-    }
-    else {
-      result_m = (scaleValue(result_m, result_e)+(FIXP_DBL)1)>>1; /* descale result + rounding */
-    }
-  }
-  return result_m;
+  FIXP_DBL retVal = (lookup<<3) >> exp;
+
+  if (set_max)
+    retVal=FL2FXCONST_DBL(1.0f);
+
+  return retVal;
 }
 
 
diff --git a/libFDK/src/mdct.cpp b/libFDK/src/mdct.cpp
index 7b9864f..9a29aa1 100644
--- a/libFDK/src/mdct.cpp
+++ b/libFDK/src/mdct.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/src/mips/fft_rad2_mips.cpp b/libFDK/src/mips/fft_rad2_mips.cpp
index 8518c79..f93ee98 100644
--- a/libFDK/src/mips/fft_rad2_mips.cpp
+++ b/libFDK/src/mips/fft_rad2_mips.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/src/mips/mips_fft_twiddles.cpp b/libFDK/src/mips/mips_fft_twiddles.cpp
index 080c61e..e15a09b 100644
--- a/libFDK/src/mips/mips_fft_twiddles.cpp
+++ b/libFDK/src/mips/mips_fft_twiddles.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/src/mips/scale.cpp b/libFDK/src/mips/scale.cpp
index 9e5964a..88a3a00 100644
--- a/libFDK/src/mips/scale.cpp
+++ b/libFDK/src/mips/scale.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libFDK/src/qmf.cpp b/libFDK/src/qmf.cpp
index 314b9fe..54526dd 100644
--- a/libFDK/src/qmf.cpp
+++ b/libFDK/src/qmf.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -1017,8 +1017,14 @@ qmfInitFilterBank (HANDLE_QMF_FILTER_BANK h_Qmf,     /*!< Handle to return */
         break;
       case 32:
         h_Qmf->p_filter = qmf_64;
+        if (flags & QMF_FLAG_DOWNSAMPLED) {
+          h_Qmf->t_cos = qmf_phaseshift_cos_downsamp32;
+          h_Qmf->t_sin = qmf_phaseshift_sin_downsamp32;
+        }
+        else {
         h_Qmf->t_cos = qmf_phaseshift_cos32;
         h_Qmf->t_sin = qmf_phaseshift_sin32;
+        }
         h_Qmf->p_stride = 2;
         h_Qmf->FilterSize = 640;
         h_Qmf->filterScale = 0;
diff --git a/libFDK/src/scale.cpp b/libFDK/src/scale.cpp
index 96c0165..5829443 100644
--- a/libFDK/src/scale.cpp
+++ b/libFDK/src/scale.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libMpegTPDec/include/mpegFileRead.h b/libMpegTPDec/include/mpegFileRead.h
index d45bee2..1fbfb58 100644
--- a/libMpegTPDec/include/mpegFileRead.h
+++ b/libMpegTPDec/include/mpegFileRead.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libMpegTPDec/include/tp_data.h b/libMpegTPDec/include/tp_data.h
index 6e8218a..5269858 100644
--- a/libMpegTPDec/include/tp_data.h
+++ b/libMpegTPDec/include/tp_data.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libMpegTPDec/include/tpdec_lib.h b/libMpegTPDec/include/tpdec_lib.h
index 7317204..fb4c41d 100644
--- a/libMpegTPDec/include/tpdec_lib.h
+++ b/libMpegTPDec/include/tpdec_lib.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -186,6 +186,27 @@ int  CProgramConfig_IsValid ( const CProgramConfig *pPce );
 void  CProgramConfig_Read ( CProgramConfig       *pPce,
                             HANDLE_FDK_BITSTREAM  bs,
                             UINT                  alignAnchor );
+
+/*!
+  \brief               Compare two Program Config Elements.
+  \param pPce1         Pointer to first Program Config Element structure.
+  \param pPce2         Pointer to second Program Config Element structure.
+  \return              -1 if PCEs are completely different,
+                        0 if PCEs are completely equal,
+                        1 if PCEs are different but have the same channel config,
+                        2 if PCEs have different channel config but same number of channels.
+*/
+int CProgramConfig_Compare ( const CProgramConfig * const pPce1,
+                             const CProgramConfig * const pPce2 );
+
+/*!
+  \brief               Get a Program Config Element that matches the predefined MPEG-4 channel configurations 1-14.
+  \param pPce          Program Config Element structure.
+  \param channelConfig MPEG-4 channel configuration.
+  \return              void
+*/
+void CProgramConfig_GetDefault ( CProgramConfig *pPce,
+                                 const UINT channelConfig );
 #endif /* TP_PCE_ENABLE */
 
 /**
@@ -206,7 +227,7 @@ void  CProgramConfig_Read ( CProgramConfig       *pPce,
  */
 int CProgramConfig_LookupElement(
         CProgramConfig *pPce,
-        const UINT      channelConfig,
+        UINT            channelConfig,
         const UINT      tag,
         const UINT      channelIdx,
         UCHAR           chMapping[],
diff --git a/libMpegTPDec/src/mpegFileFormat.h b/libMpegTPDec/src/mpegFileFormat.h
index 7ddaa0d..b0cfe9e 100644
--- a/libMpegTPDec/src/mpegFileFormat.h
+++ b/libMpegTPDec/src/mpegFileFormat.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libMpegTPDec/src/tpdec_adif.cpp b/libMpegTPDec/src/tpdec_adif.cpp
index b6d6087..237e881 100644
--- a/libMpegTPDec/src/tpdec_adif.cpp
+++ b/libMpegTPDec/src/tpdec_adif.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libMpegTPDec/src/tpdec_adif.h b/libMpegTPDec/src/tpdec_adif.h
index 7b9b071..d6780a7 100644
--- a/libMpegTPDec/src/tpdec_adif.h
+++ b/libMpegTPDec/src/tpdec_adif.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libMpegTPDec/src/tpdec_adts.cpp b/libMpegTPDec/src/tpdec_adts.cpp
index 92e2e59..c455681 100644
--- a/libMpegTPDec/src/tpdec_adts.cpp
+++ b/libMpegTPDec/src/tpdec_adts.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -374,12 +374,12 @@ int adtsRead_GetRawDataBlockLength(
       length = -1; /* raw data block length is unknown */
     } else {
       if (blockNum < 0 || blockNum > 3) {
-        return TRANSPORTDEC_INVALID_PARAMETER;
+        length = -1;
       }
       length = (pAdts->rawDataBlockDist[blockNum] << 3) - 16;
     }
   }
-  if (blockNum == 0) {
+  if (blockNum == 0 && length > 0) {
     length -= pAdts->bs.num_pce_bits;
   }
   return length;
diff --git a/libMpegTPDec/src/tpdec_adts.h b/libMpegTPDec/src/tpdec_adts.h
index def6aa0..a3b83a5 100644
--- a/libMpegTPDec/src/tpdec_adts.h
+++ b/libMpegTPDec/src/tpdec_adts.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libMpegTPDec/src/tpdec_asc.cpp b/libMpegTPDec/src/tpdec_asc.cpp
index d6e1a0b..4d7cc02 100644
--- a/libMpegTPDec/src/tpdec_asc.cpp
+++ b/libMpegTPDec/src/tpdec_asc.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -205,6 +205,145 @@ void CProgramConfig_Read(
 
   pPce->isValid = 1;
 }
+
+/*
+ * Compare two program configurations.
+ * Returns the result of the comparison:
+ *  -1 - completely different
+ *   0 - completely equal
+ *   1 - different but same channel configuration
+ *   2 - different channel configuration but same number of channels
+ */
+int CProgramConfig_Compare ( const CProgramConfig * const pPce1,
+                             const CProgramConfig * const pPce2 )
+{
+  int result = 0;  /* Innocent until proven false. */
+
+  if (FDKmemcmp(pPce1, pPce2, sizeof(CProgramConfig)) != 0)
+  { /* Configurations are not completely different.
+       So look into details and analyse the channel configurations: */
+    result = -1;
+
+    if (pPce1->NumChannels == pPce2->NumChannels)
+    { /* Now the logic changes. We first assume to have the same channel configuration
+         and then prove if this assumption is true. */
+      result = 1;
+
+      /* Front channels */
+      if (pPce1->NumFrontChannelElements != pPce2->NumFrontChannelElements) {
+        result = 2;  /* different number of front channel elements */
+      } else {
+        int el, numCh1 = 0, numCh2 = 0;
+        for (el = 0; el < pPce1->NumFrontChannelElements; el += 1) {
+          numCh1 += pPce1->FrontElementIsCpe[el] ? 2 : 1;
+          numCh2 += pPce2->FrontElementIsCpe[el] ? 2 : 1;
+        }
+        if (numCh1 != numCh2) {
+          result = 2;  /* different number of front channels */
+        }
+      }
+      /* Side channels */
+      if (pPce1->NumSideChannelElements != pPce2->NumSideChannelElements) {
+        result = 2;  /* different number of side channel elements */
+      } else {
+        int el, numCh1 = 0, numCh2 = 0;
+        for (el = 0; el < pPce1->NumSideChannelElements; el += 1) {
+          numCh1 += pPce1->SideElementIsCpe[el] ? 2 : 1;
+          numCh2 += pPce2->SideElementIsCpe[el] ? 2 : 1;
+        }
+        if (numCh1 != numCh2) {
+          result = 2;  /* different number of side channels */
+        }
+      }
+      /* Back channels */
+      if (pPce1->NumBackChannelElements != pPce2->NumBackChannelElements) {
+        result = 2;  /* different number of back channel elements */
+      } else {
+        int el, numCh1 = 0, numCh2 = 0;
+        for (el = 0; el < pPce1->NumBackChannelElements; el += 1) {
+          numCh1 += pPce1->BackElementIsCpe[el] ? 2 : 1;
+          numCh2 += pPce2->BackElementIsCpe[el] ? 2 : 1;
+        }
+        if (numCh1 != numCh2) {
+          result = 2;  /* different number of back channels */
+        }
+      }
+      /* LFE channels */
+      if (pPce1->NumLfeChannelElements != pPce2->NumLfeChannelElements) {
+        result = 2;  /* different number of lfe channels */
+      }
+      /* LFEs are always SCEs so we don't need to count the channels. */
+    }
+  }
+
+  return result;
+}
+
+void CProgramConfig_GetDefault( CProgramConfig *pPce,
+                                const UINT channelConfig )
+{
+  FDK_ASSERT(pPce != NULL);
+
+  /* Init PCE */
+  CProgramConfig_Init(pPce);
+  pPce->Profile = 1;  /* Set AAC LC because it is the only supported object type. */
+
+  switch (channelConfig) {
+  /* - - - - - - - - - - - - - - - - - - - - - - - - - - - - */
+  case 6:   /* 3/0/2.1ch */
+    pPce->NumLfeChannelElements   += 1;
+    pPce->NumChannels             += 1;
+  case 5:   /* 3/0/2.0ch */
+  case 4:   /* 3/0/1.0ch */
+    pPce->NumBackChannelElements  += 1;
+    pPce->BackElementIsCpe[0]      = (channelConfig>4) ? 1 : 0;
+    pPce->NumChannels             += (channelConfig>4) ? 2 : 1;
+    pPce->NumEffectiveChannels    += (channelConfig>4) ? 2 : 1;
+  case 3:   /* 3/0/0.0ch */
+    pPce->NumFrontChannelElements += 1;
+    pPce->FrontElementIsCpe[1]     = 1;
+    pPce->NumChannels             += 2;
+    pPce->NumEffectiveChannels    += 2;
+  case 1:   /* 1/0/0.0ch */
+    pPce->NumFrontChannelElements += 1;
+    pPce->FrontElementIsCpe[0]     = 0;
+    pPce->NumChannels             += 1;
+    pPce->NumEffectiveChannels    += 1;
+    pPce->isValid                  = 1;
+    break;
+  /* - - - - - - - - - - - - - - - - - - - - - - - - - - - - */
+  case 2:   /* 2/0/0.ch */
+    pPce->NumFrontChannelElements  = 1;
+    pPce->FrontElementIsCpe[0]     = 1;
+    pPce->NumChannels             += 2;
+    pPce->NumEffectiveChannels    += 2;
+    pPce->isValid                  = 1;
+    break;
+  /* - - - - - - - - - - - - - - - - - - - - - - - - - - - - */
+  default:
+    pPce->isValid                  = 0;   /* To be explicit! */
+    break;
+  }
+
+  if (pPce->isValid) {
+    /* Create valid element instance tags */
+    int el, elTagSce = 0, elTagCpe = 0;
+
+    for (el = 0; el < pPce->NumFrontChannelElements; el += 1) {
+      pPce->FrontElementTagSelect[el] = (pPce->FrontElementIsCpe) ? elTagCpe++ : elTagSce++;
+    }
+    for (el = 0; el < pPce->NumSideChannelElements; el += 1) {
+      pPce->SideElementTagSelect[el] = (pPce->SideElementIsCpe) ? elTagCpe++ : elTagSce++;
+    }
+    for (el = 0; el < pPce->NumBackChannelElements; el += 1) {
+      pPce->BackElementTagSelect[el] = (pPce->BackElementIsCpe) ? elTagCpe++ : elTagSce++;
+    }
+    elTagSce = 0;
+    for (el = 0; el < pPce->NumLfeChannelElements; el += 1) {
+      pPce->LfeElementTagSelect[el] = elTagSce++;
+    }
+  }
+}
 #endif /* TP_PCE_ENABLE */
 
 /**
@@ -267,7 +406,7 @@ void getImplicitAudioChannelTypeAndIndex(
 
 int CProgramConfig_LookupElement(
         CProgramConfig *pPce,
-        const UINT      channelConfig,
+        UINT            channelConfig,
         const UINT      tag,
         const UINT      channelIdx,
         UCHAR           chMapping[],
@@ -289,7 +428,13 @@ int CProgramConfig_LookupElement(
       *elMapping = pPce->elCounter;
       if (elList[pPce->elCounter] != elType) {
         /* Not in the list */
-        return 0;
+        if ( (channelConfig == 2) && (elType == ID_SCE) )
+        { /* This scenario occurs with HE-AAC v2 streams of buggy encoders.
+             Due to other decoder implementations decoding of these kind of streams is desired. */
+          channelConfig = 1;
+        } else {
+          return 0;
+        }
       }
       /* Assume all front channels */
       getImplicitAudioChannelTypeAndIndex(&chType[channelIdx], &chIndex[channelIdx], channelConfig, channelIdx);
@@ -583,18 +728,18 @@ static INT ld_sbr_header( const CSAudioSpecificConfig *asc,
   }
 
   switch ( channelConfiguration ) {
+    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:
     case 5:
       error |= cb->cbSbr(cb->cbSbrData, hBs, asc->m_samplingFrequency, asc->m_extensionSamplingFrequency, asc->m_samplesPerFrame, AOT_ER_AAC_ELD, ID_CPE, i++);
     case 3:
       error |= cb->cbSbr(cb->cbSbrData, hBs, asc->m_samplingFrequency, asc->m_extensionSamplingFrequency, asc->m_samplesPerFrame, AOT_ER_AAC_ELD, ID_CPE, i++);
       break;
 
-    case 7:
-      error |= cb->cbSbr(cb->cbSbrData, hBs, asc->m_samplingFrequency, asc->m_extensionSamplingFrequency, asc->m_samplesPerFrame, AOT_ER_AAC_ELD, ID_SCE, i++);
-    case 6:
-      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++);
       break;
   }
 
@@ -689,6 +834,62 @@ bail:
 #endif /* TP_ELD_ENABLE */
 
 
+static
+TRANSPORTDEC_ERROR AudioSpecificConfig_ExtensionParse(CSAudioSpecificConfig *self, HANDLE_FDK_BITSTREAM bs, CSTpCallBacks *cb)
+{
+  TP_ASC_EXTENSION_ID  lastAscExt, ascExtId = ASCEXT_UNKOWN;
+  INT  bitsAvailable = (INT)FDKgetValidBits(bs);
+
+  while (bitsAvailable >= 11)
+  {
+    lastAscExt = ascExtId;
+    ascExtId   = (TP_ASC_EXTENSION_ID)FDKreadBits(bs, 11);
+    bitsAvailable -= 11;
+
+    switch (ascExtId) {
+    case ASCEXT_SBR:    /* 0x2b7 */
+      if ( (self->m_extensionAudioObjectType != AOT_SBR) && (bitsAvailable >= 5) ) {
+        self->m_extensionAudioObjectType = getAOT(bs);
+
+        if ( (self->m_extensionAudioObjectType == AOT_SBR)
+          || (self->m_extensionAudioObjectType == AOT_ER_BSAC) )
+        { /* Get SBR extension configuration */
+          self->m_sbrPresentFlag = FDKreadBits(bs, 1);
+          bitsAvailable -= 1;
+
+          if ( self->m_sbrPresentFlag == 1 ) {
+            self->m_extensionSamplingFrequency = getSampleRate(bs, &self->m_extensionSamplingFrequencyIndex, 4);
+
+            if ((INT)self->m_extensionSamplingFrequency <= 0) {
+              return TRANSPORTDEC_PARSE_ERROR;
+            }
+          }
+          if ( self->m_extensionAudioObjectType == AOT_ER_BSAC ) {
+            self->m_extensionChannelConfiguration = FDKreadBits(bs, 4);
+            bitsAvailable -= 4;
+          }
+        }
+        /* Update counter because of variable length fields (AOT and sampling rate) */
+        bitsAvailable = (INT)FDKgetValidBits(bs);
+      }
+      break;
+    case ASCEXT_PS:     /* 0x548 */
+      if ( (lastAscExt == ASCEXT_SBR)
+        && (self->m_extensionAudioObjectType == AOT_SBR)
+        && (bitsAvailable > 0) )
+      { /* Get PS extension configuration */
+        self->m_psPresentFlag = FDKreadBits(bs, 1);
+        bitsAvailable -= 1;
+      }
+      break;
+    default:
+      /* Just ignore anything. */
+      return TRANSPORTDEC_OK;
+    }
+  }
+
+  return TRANSPORTDEC_OK;
+}
 
 /*
  * API Functions
@@ -839,6 +1040,9 @@ TRANSPORTDEC_ERROR AudioSpecificConfig_Parse(
       break;
   }
 
+  if (fExplicitBackwardCompatible) {
+    ErrorStatus = AudioSpecificConfig_ExtensionParse(self, bs, cb);
+  }
 
   return (ErrorStatus);
 }
diff --git a/libMpegTPDec/src/tpdec_latm.cpp b/libMpegTPDec/src/tpdec_latm.cpp
index 99b905c..5710d77 100644
--- a/libMpegTPDec/src/tpdec_latm.cpp
+++ b/libMpegTPDec/src/tpdec_latm.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -120,7 +120,8 @@ TRANSPORTDEC_ERROR CLatmDemux_ReadAudioMuxElement(
         CLatmDemux *pLatmDemux,
         int m_muxConfigPresent,
         CSTpCallBacks *pTpDecCallbacks,
-        CSAudioSpecificConfig *pAsc
+        CSAudioSpecificConfig *pAsc,
+        int *pfConfigFound
         )
 {
   TRANSPORTDEC_ERROR ErrorStatus = TRANSPORTDEC_OK;
@@ -129,12 +130,17 @@ TRANSPORTDEC_ERROR CLatmDemux_ReadAudioMuxElement(
     pLatmDemux->m_useSameStreamMux = FDKreadBits(bs,1);
 
     if (!pLatmDemux->m_useSameStreamMux) {
-      if ((ErrorStatus = CLatmDemux_ReadStreamMuxConfig(bs, pLatmDemux, pTpDecCallbacks, pAsc))) {
+      if ((ErrorStatus = CLatmDemux_ReadStreamMuxConfig(bs, pLatmDemux, pTpDecCallbacks, pAsc, pfConfigFound))) {
         return (ErrorStatus);
       }
     }
   }
 
+  /* If there was no configuration read, its not possible to parse PayloadLengthInfo below. */
+  if (! *pfConfigFound) {
+    return TRANSPORTDEC_SYNC_ERROR;
+  }
+
   if (pLatmDemux->m_AudioMuxVersionA == 0) {
     /* Do only once per call, because parsing and decoding is done in-line. */
     if ((ErrorStatus = CLatmDemux_ReadPayloadLengthInfo(bs,pLatmDemux))) {
@@ -154,6 +160,7 @@ TRANSPORTDEC_ERROR CLatmDemux_Read(
         TRANSPORT_TYPE tt,
         CSTpCallBacks *pTpDecCallbacks,
         CSAudioSpecificConfig *pAsc,
+        int *pfConfigFound,
         const INT ignoreBufferFullness
         )
 {
@@ -168,7 +175,7 @@ TRANSPORTDEC_ERROR CLatmDemux_Read(
     return TRANSPORTDEC_NOT_ENOUGH_BITS;
   }
 
-  if ((ErrorStatus = CLatmDemux_ReadAudioMuxElement(bs, pLatmDemux, (tt != TT_MP4_LATM_MCP0), pTpDecCallbacks, pAsc)))
+  if ((ErrorStatus = CLatmDemux_ReadAudioMuxElement(bs, pLatmDemux, (tt != TT_MP4_LATM_MCP0), pTpDecCallbacks, pAsc, pfConfigFound)))
     return (ErrorStatus);
 
   if (!ignoreBufferFullness)
@@ -205,7 +212,8 @@ TRANSPORTDEC_ERROR CLatmDemux_ReadStreamMuxConfig(
         HANDLE_FDK_BITSTREAM bs,
         CLatmDemux *pLatmDemux,
         CSTpCallBacks *pTpDecCallbacks,
-        CSAudioSpecificConfig *pAsc
+        CSAudioSpecificConfig *pAsc,
+        int * pfConfigFound
         )
 {
   LATM_LAYER_INFO *p_linfo = NULL;
@@ -272,6 +280,7 @@ TRANSPORTDEC_ERROR CLatmDemux_ReadStreamMuxConfig(
             if ((ErrorStatus = AudioSpecificConfig_Parse(&pAsc[TPDEC_TRACKINDEX(prog,lay)], &tmpBs, 1, pTpDecCallbacks))) {
               return (ErrorStatus);
             }
+            *pfConfigFound = 1;
 
             /* The field p_linfo->m_ascLen could be wrong, so check if */
             if ( 0 > (INT)FDKgetValidBits(&tmpBs)) {
@@ -292,6 +301,7 @@ TRANSPORTDEC_ERROR CLatmDemux_ReadStreamMuxConfig(
             if (cbError != 0) {
               return TRANSPORTDEC_UNKOWN_ERROR;
             }
+            *pfConfigFound = 1;
           }
         }
 
@@ -377,7 +387,7 @@ TRANSPORTDEC_ERROR CLatmDemux_ReadPayloadLengthInfo(HANDLE_FDK_BITSTREAM bs, CLa
   else {
     ErrorStatus = TRANSPORTDEC_PARSE_ERROR; //AAC_DEC_LATM_TIMEFRAMING;
   }
-  if (pLatmDemux->m_audioMuxLengthBytes > 0 && totalPayloadBits > pLatmDemux->m_audioMuxLengthBytes*8) {
+  if (pLatmDemux->m_audioMuxLengthBytes > (UINT)0 && totalPayloadBits > (int)pLatmDemux->m_audioMuxLengthBytes*8) {
     return TRANSPORTDEC_PARSE_ERROR;
   }
   return (ErrorStatus);
diff --git a/libMpegTPDec/src/tpdec_latm.h b/libMpegTPDec/src/tpdec_latm.h
index f1fcc19..6dc13e8 100644
--- a/libMpegTPDec/src/tpdec_latm.h
+++ b/libMpegTPDec/src/tpdec_latm.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -144,14 +144,25 @@ TRANSPORTDEC_ERROR CLatmDemux_Read(
         TRANSPORT_TYPE tt,
         CSTpCallBacks *pTpDecCallbacks,
         CSAudioSpecificConfig *pAsc,
+        int *pfConfigFound,
         const INT ignoreBufferFullness
         );
 
+/**
+ * \brief Read StreamMuxConfig
+ * \param bs bit stream handle as data source
+ * \param pLatmDemux pointer to CLatmDemux struct of current LATM context
+ * \param pTpDecCallbacks Call back structure for configuration callbacks
+ * \param pAsc pointer to a ASC for configuration storage
+ * \param pfConfigFound pointer to a flag which is set to 1 if a configuration was found and processed successfully
+ * \return error code
+ */
 TRANSPORTDEC_ERROR CLatmDemux_ReadStreamMuxConfig(
         HANDLE_FDK_BITSTREAM bs,
         CLatmDemux *pLatmDemux,
         CSTpCallBacks *pTpDecCallbacks,
-        CSAudioSpecificConfig *pAsc
+        CSAudioSpecificConfig *pAsc,
+        int * pfConfigFound
         );
 
 TRANSPORTDEC_ERROR CLatmDemux_ReadPayloadLengthInfo(HANDLE_FDK_BITSTREAM bs, CLatmDemux *pLatmDemux);
diff --git a/libMpegTPDec/src/tpdec_lib.cpp b/libMpegTPDec/src/tpdec_lib.cpp
index 2dca5d5..1f9f29b 100644
--- a/libMpegTPDec/src/tpdec_lib.cpp
+++ b/libMpegTPDec/src/tpdec_lib.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -197,8 +197,7 @@ HANDLE_TRANSPORTDEC transportDec_Open( const TRANSPORT_TYPE transportFmt, const
 
   if (hInput != NULL) {
     /* Create bitstream */
-    if ( (transportFmt == TT_MP4_RAW)
-      || (transportFmt == TT_DRM) ){
+    if ( TT_IS_PACKET(transportFmt) ) {
       hInput->bsBuffer = NULL;
     } else {
       hInput->bsBuffer = GetRam_TransportDecoderBuffer(0);
@@ -222,7 +221,9 @@ TRANSPORTDEC_ERROR transportDec_OutOfBandConfig(HANDLE_TRANSPORTDEC hTp, UCHAR *
   FDK_BITSTREAM bs;
   HANDLE_FDK_BITSTREAM  hBs        = &bs;
 
-  FDKinitBitStream(hBs, conf, 0x80000000, length<<3, BS_READER);
+  FDKinitBitStream(hBs, conf, 0x10000000, length<<3, BS_READER);
+
+  int fConfigFound = 0;
 
   /* config transport decoder */
   switch (hTp->transportFmt) {
@@ -234,27 +235,27 @@ TRANSPORTDEC_ERROR transportDec_OutOfBandConfig(HANDLE_TRANSPORTDEC hTp, UCHAR *
           return TRANSPORTDEC_INVALID_PARAMETER;
         }
         CLatmDemux *pLatmDemux = &hTp->parser.latm;
-        err = CLatmDemux_ReadStreamMuxConfig(hBs, pLatmDemux, &hTp->callbacks, hTp->asc);
+        err = CLatmDemux_ReadStreamMuxConfig(hBs, pLatmDemux, &hTp->callbacks, hTp->asc, &fConfigFound);
         if (err != TRANSPORTDEC_OK) {
           return err;
         }
       }
       break;
     default:
-    case TT_MP4_RAW:
+      fConfigFound = 1;
       err = AudioSpecificConfig_Parse(&hTp->asc[layer], hBs, 1, &hTp->callbacks);
-      break;
-  }
-  if (err == TRANSPORTDEC_OK) {
-    int errC;
+      if (err == TRANSPORTDEC_OK) {
+        int errC;
 
-    errC = hTp->callbacks.cbUpdateConfig(hTp->callbacks.cbUpdateConfigData, &hTp->asc[layer]);
-    if (errC != 0) {
-      err = TRANSPORTDEC_PARSE_ERROR;
-    }
+        errC = hTp->callbacks.cbUpdateConfig(hTp->callbacks.cbUpdateConfigData, &hTp->asc[layer]);
+        if (errC != 0) {
+          err = TRANSPORTDEC_PARSE_ERROR;
+        }
+      }
+      break;
   }
 
-  if (err == TRANSPORTDEC_OK) {
+  if (err == TRANSPORTDEC_OK && fConfigFound) {
     hTp->flags |= TPDEC_CONFIG_FOUND;
   }
 
@@ -313,18 +314,16 @@ TRANSPORTDEC_ERROR transportDec_FillData(
   /* set bitbuffer shortcut */
   hBs = &hTp->bitStream[layer];
 
-  switch (hTp->transportFmt) {
-  case TT_MP4_RAW:
-  case TT_DRM:
+  if ( TT_IS_PACKET(hTp->transportFmt) ) {
+    if (hTp->numberOfRawDataBlocks == 0) {
     /* For packet based transport, pass input buffer to bitbuffer without copying the data.
        Unfortunately we do not know the actual buffer size. And the FDK bit buffer implementation
        needs a number 2^x. So we assume the maximum of 48 channels with 6144 bits per channel
        and round it up to the next power of 2 => 65536 bytes */
     FDKinitBitStream(hBs, pBuffer, 0x10000, (*pBytesValid)<<3, BS_READER);
     *pBytesValid = 0;
-    break;
-
-  default:
+    }
+  } else {
     /* ... else feed bitbuffer with new stream data (append). */
     if (hTp->numberOfRawDataBlocks <= 0) {
       FDKfeedBuffer (hBs, pBuffer, bufferSize, pBytesValid) ;
@@ -369,6 +368,82 @@ INT transportDec_GetBufferFullness( const HANDLE_TRANSPORTDEC hTp )
 }
 
 /**
+ * \brief adjust bit stream position and the end of an access unit.
+ * \param hTp transport decoder handle.
+ * \return error code.
+ */
+static
+TRANSPORTDEC_ERROR transportDec_AdjustEndOfAccessUnit(HANDLE_TRANSPORTDEC hTp)
+{
+  HANDLE_FDK_BITSTREAM hBs = &hTp->bitStream[0];
+  TRANSPORTDEC_ERROR err = TRANSPORTDEC_OK;
+
+  switch (hTp->transportFmt) {
+    case TT_MP4_LOAS:
+    case TT_MP4_LATM_MCP0:
+    case TT_MP4_LATM_MCP1:
+      if ( hTp->numberOfRawDataBlocks == 0 )
+      {
+        /* Do byte align at the end of AudioMuxElement. */
+        FDKbyteAlign(hBs, hTp->globalFramePos);
+
+        /* Check global frame length */
+        if (hTp->transportFmt == TT_MP4_LOAS && hTp->parser.latm.m_audioMuxLengthBytes > 0)
+        {
+          int loasOffset;
+
+          loasOffset = (hTp->parser.latm.m_audioMuxLengthBytes*8 + FDKgetValidBits(hBs)) - hTp->globalFramePos;
+          if (loasOffset != 0) {
+            FDKpushBiDirectional(hBs, loasOffset);
+            /* For ELD and other payloads there is an unknown amount of padding, so ignore unread bits, but
+               throw an error only if too many bits where read. */
+            if (loasOffset < 0) {
+              err = TRANSPORTDEC_PARSE_ERROR;
+            }
+          }
+        }
+      }
+      break;
+
+    case TT_MP4_ADTS:
+      if (hTp->parser.adts.bs.protection_absent == 0)
+      {
+        int offset;
+
+        /* Calculate offset to end of AU */
+        offset  = hTp->parser.adts.rawDataBlockDist[hTp->parser.adts.bs.num_raw_blocks-hTp->numberOfRawDataBlocks]<<3;
+        /* CAUTION: The PCE (if available) is declared to be a part of the header! */
+        offset -= hTp->accessUnitAnchor[0] - FDKgetValidBits(hBs) + 16 + hTp->parser.adts.bs.num_pce_bits;
+        FDKpushBiDirectional(hBs, offset);
+      }
+      if (hTp->parser.adts.bs.num_raw_blocks > 0 && hTp->parser.adts.bs.protection_absent == 0) {
+        /* Note this CRC read currently happens twice because of transportDec_CrcCheck() */
+        hTp->parser.adts.crcReadValue = FDKreadBits(hBs, 16);
+      }
+      if ( hTp->numberOfRawDataBlocks == 0 )
+      {
+        /* Check global frame length */
+        if (hTp->parser.adts.bs.protection_absent == 0)
+        {
+          int offset;
+
+          offset = (hTp->parser.adts.bs.frame_length*8 - ADTS_SYNCLENGTH + FDKgetValidBits(hBs)) - hTp->globalFramePos;
+          if (offset != 0) {
+            FDKpushBiDirectional(hBs, offset);
+          }
+        }
+      }
+      break;
+
+    default:
+      break;
+  }
+
+  return err;
+}
+
+
+/**
  * \brief Determine additional buffer fullness contraint due to burst data reception.
  *        The parameter TPDEC_PARAM_BURSTPERIOD must have been set as a precondition.
  * \param hTp transport decoder handle.
@@ -432,51 +507,152 @@ TRANSPORTDEC_ERROR additionalHoldOffNeeded(
   }
 }
 
-/**
- * \brief adjust bit stream position and the end of an access unit.
- * \param hTp transport decoder handle.
- * \return error code.
- */
-static
-TRANSPORTDEC_ERROR transportDec_AdjustEndOfAccessUnit(HANDLE_TRANSPORTDEC hTp)
+static TRANSPORTDEC_ERROR transportDec_readHeader(
+        HANDLE_TRANSPORTDEC hTp,
+        HANDLE_FDK_BITSTREAM hBs,
+        int syncLength,
+        int ignoreBufferFullness,
+        int *pRawDataBlockLength,
+        int *pfTraverseMoreFrames,
+        int *pSyncLayerFrameBits,
+        int *pfConfigFound,
+        int *pHeaderBits
+        )
 {
-  HANDLE_FDK_BITSTREAM hBs = &hTp->bitStream[0];
   TRANSPORTDEC_ERROR err = TRANSPORTDEC_OK;
+  int rawDataBlockLength = *pRawDataBlockLength;
+  int fTraverseMoreFrames = (pfTraverseMoreFrames != NULL) ? *pfTraverseMoreFrames : 0;
+  int syncLayerFrameBits = (pSyncLayerFrameBits != NULL) ? *pSyncLayerFrameBits : 0;
+  int fConfigFound = (pfConfigFound != NULL) ? *pfConfigFound : 0;
+  int startPos;
+
+  startPos = FDKgetValidBits(hBs);
 
   switch (hTp->transportFmt) {
-    case TT_MP4_LOAS:
-    case TT_MP4_LATM_MCP0:
-    case TT_MP4_LATM_MCP1:
-      if ( hTp->numberOfRawDataBlocks == 0 )
+    case TT_MP4_ADTS:
+      if (hTp->numberOfRawDataBlocks <= 0)
       {
-        /* Check global frame length */
-        if (hTp->transportFmt == TT_MP4_LOAS && hTp->parser.latm.m_audioMuxLengthBytes > 0)
-        {
-          int loasOffset;
+        int errC;
 
-          loasOffset = (hTp->parser.latm.m_audioMuxLengthBytes*8 + FDKgetValidBits(hBs)) - hTp->globalFramePos;
-          if (loasOffset != 0) {
-            FDKpushBiDirectional(hBs, loasOffset);
-            /* For ELD and other payloads there is an unknown amount of padding, so ignore unread bits, but
-               throw an error only if too many bits where read. */
-            if (loasOffset < 0) {
-              err = TRANSPORTDEC_PARSE_ERROR;
+        hTp->globalFramePos = FDKgetValidBits(hBs);
+
+        /* Parse ADTS header */
+        err = adtsRead_DecodeHeader( &hTp->parser.adts, &hTp->asc[0], hBs, ignoreBufferFullness );
+        if (err != TRANSPORTDEC_OK) {
+          if (err != TRANSPORTDEC_NOT_ENOUGH_BITS) {
+            err = TRANSPORTDEC_SYNC_ERROR;
+          }
+        } else {
+          errC = hTp->callbacks.cbUpdateConfig(hTp->callbacks.cbUpdateConfigData, &hTp->asc[0]);
+          if (errC != 0) {
+            if (errC == TRANSPORTDEC_NEED_TO_RESTART) {
+              err = TRANSPORTDEC_NEED_TO_RESTART;
+              goto bail;
+            } else {
+              err = TRANSPORTDEC_SYNC_ERROR;
             }
+          } else {
+            fConfigFound = 1;
+            hTp->numberOfRawDataBlocks = hTp->parser.adts.bs.num_raw_blocks+1;
           }
         }
+      }
+      else {
+        /* Reset CRC because the next bits are the beginning of a raw_data_block() */
+        FDKcrcReset(&hTp->parser.adts.crcInfo);
+        hTp->parser.adts.bs.num_pce_bits = 0;
+      }
+      if (err == TRANSPORTDEC_OK) {
+        hTp->numberOfRawDataBlocks--;
+        rawDataBlockLength = adtsRead_GetRawDataBlockLength(&hTp->parser.adts, (hTp->parser.adts.bs.num_raw_blocks-hTp->numberOfRawDataBlocks));
+        if (rawDataBlockLength <= 0) {
+          /* No further frame traversal possible. */
+          fTraverseMoreFrames = 0;
+        }
+        syncLayerFrameBits = (hTp->parser.adts.bs.frame_length<<3) - (startPos - FDKgetValidBits(hBs)) - syncLength;
+        if (syncLayerFrameBits <= 0) {
+          err = TRANSPORTDEC_SYNC_ERROR;
+        }
+      } else {
+        hTp->numberOfRawDataBlocks = 0;
+      }
+      break;
+    case TT_MP4_LOAS:
+      if (hTp->numberOfRawDataBlocks <= 0)
+      {
+        syncLayerFrameBits = FDKreadBits(hBs, 13);
+        hTp->parser.latm.m_audioMuxLengthBytes = syncLayerFrameBits;
+        syncLayerFrameBits <<= 3;
+      }
+    case TT_MP4_LATM_MCP1:
+    case TT_MP4_LATM_MCP0:
+      if (hTp->numberOfRawDataBlocks <= 0)
+      {
+        hTp->globalFramePos = FDKgetValidBits(hBs);
 
-        /* Do global LOAS/LATM audioMuxElement byte alignment */
-        FDKbyteAlign(hBs, hTp->globalFramePos);
+        err = CLatmDemux_Read(
+                hBs,
+               &hTp->parser.latm,
+                hTp->transportFmt,
+               &hTp->callbacks,
+                hTp->asc,
+               &fConfigFound,
+                ignoreBufferFullness);
+
+        if (err != TRANSPORTDEC_OK) {
+          if (err != TRANSPORTDEC_NOT_ENOUGH_BITS) {
+            err = TRANSPORTDEC_SYNC_ERROR;
+          }
+        } else {
+          hTp->numberOfRawDataBlocks = CLatmDemux_GetNrOfSubFrames(&hTp->parser.latm);
+          if (hTp->transportFmt == TT_MP4_LOAS) {
+            syncLayerFrameBits -= startPos - FDKgetValidBits(hBs) - (13);            
+          }
+        }
+      } else {
+        err = CLatmDemux_ReadPayloadLengthInfo(hBs, &hTp->parser.latm);
+        if (err != TRANSPORTDEC_OK) {
+          err = TRANSPORTDEC_SYNC_ERROR;
+        }
+      }
+      if (err == TRANSPORTDEC_OK) {
+        rawDataBlockLength = CLatmDemux_GetFrameLengthInBits(&hTp->parser.latm);
+        hTp->numberOfRawDataBlocks--;
+      } else {
+        hTp->numberOfRawDataBlocks = 0;
       }
       break;
     default:
+      {
+        syncLayerFrameBits = 0;
+      }
       break;
   }
 
+bail:
+
+  *pRawDataBlockLength = rawDataBlockLength;
+
+  if (pHeaderBits != NULL) {
+    *pHeaderBits += startPos - (INT)FDKgetValidBits(hBs);
+  }
+  if (pfConfigFound != NULL) {
+    *pfConfigFound = fConfigFound;
+  }
+
+  if (pfTraverseMoreFrames != NULL) {
+    *pfTraverseMoreFrames = fTraverseMoreFrames;
+  }
+  if  (pSyncLayerFrameBits != NULL) {
+    *pSyncLayerFrameBits = syncLayerFrameBits;
+  }
+  if (pfConfigFound != NULL) {
+    *pfConfigFound = fConfigFound;
+  }
+
   return err;
 }
 
-
 /* How many bits to advance for synchronization search. */
 #define TPDEC_SYNCSKIP 8
 
@@ -493,9 +669,9 @@ TRANSPORTDEC_ERROR synchronization(
   INT rawDataBlockLength = 0, rawDataBlockLengthPrevious;
   INT totalBits;
   INT headerBits = 0, headerBitsFirstFrame = 0, headerBitsPrevious;
-  INT numFramesTraversed = 0, fTraverseMoreFrames, fConfigFound = 0, startPos, startPosFirstFrame = -1;
+  INT numFramesTraversed = 0, fTraverseMoreFrames, fConfigFound = (hTp->flags & TPDEC_CONFIG_FOUND), startPosFirstFrame = -1;
   INT numRawDataBlocksFirstFrame = 0, numRawDataBlocksPrevious, globalFramePosFirstFrame = 0, rawDataBlockLengthFirstFrame = 0;
-  INT ignoreBufferFullness = hTp->flags & (TPDEC_IGNORE_BUFFERFULLNESS|TPDEC_SYNCOK);
+  INT ignoreBufferFullness = hTp->flags & (TPDEC_LOST_FRAMES_PENDING|TPDEC_IGNORE_BUFFERFULLNESS|TPDEC_SYNCOK);
 
   /* Synch parameters */
   INT syncLength;      /* Length of sync word in bits */
@@ -506,10 +682,7 @@ TRANSPORTDEC_ERROR synchronization(
   totalBits = (INT)FDKgetValidBits(hBs);
 
   if (totalBits <= 0) {
-    /* Return sync error, because this happens only in case of severly damaged bit streams.
-       Returning TRANSPORTDEC_NOT_ENOUGH_BITS here is very dangerous. */
-    /* numberOfRawDataBlocks must be always reset in case of sync errors. */
-    hTp->numberOfRawDataBlocks = 0;
+    err = TRANSPORTDEC_NOT_ENOUGH_BITS;
     goto bail;
   }
 
@@ -579,100 +752,22 @@ TRANSPORTDEC_ERROR synchronization(
     numRawDataBlocksPrevious = hTp->numberOfRawDataBlocks;
 
     /* Parse transport header (raw data block granularity) */
-    startPos = FDKgetValidBits(hBs);
 
     if (err == TRANSPORTDEC_OK )
     {
-      switch (hTp->transportFmt) {
-        case TT_MP4_ADTS:
-          if (hTp->numberOfRawDataBlocks <= 0)
-          {
-            int errC;
-
-            /* Parse ADTS header */
-            err = adtsRead_DecodeHeader( &hTp->parser.adts, &hTp->asc[0], hBs, ignoreBufferFullness );
-            if (err != TRANSPORTDEC_OK) {
-              if (err != TRANSPORTDEC_NOT_ENOUGH_BITS) {
-                err = TRANSPORTDEC_SYNC_ERROR;
-              }
-            } else {
-              errC = hTp->callbacks.cbUpdateConfig(hTp->callbacks.cbUpdateConfigData, &hTp->asc[0]);
-              if (errC != 0) {
-                err = TRANSPORTDEC_SYNC_ERROR;
-              } else {
-                hTp->numberOfRawDataBlocks = hTp->parser.adts.bs.num_raw_blocks+1;
-                /* CAUTION: The PCE (if available) is declared to be a part of the header! */
-                hTp->globalFramePos = FDKgetValidBits(hBs) + hTp->parser.adts.bs.num_pce_bits;
-              }
-            }
-          }
-          else {
-            /* Reset CRC because the next bits are the beginning of a raw_data_block() */
-            FDKcrcReset(&hTp->parser.adts.crcInfo);
-            hTp->globalFramePos = FDKgetValidBits(hBs);
-          }
-          if (err == TRANSPORTDEC_OK) {
-            hTp->numberOfRawDataBlocks--;
-            rawDataBlockLength = adtsRead_GetRawDataBlockLength(&hTp->parser.adts, (hTp->parser.adts.bs.num_raw_blocks-hTp->numberOfRawDataBlocks));
-            syncLayerFrameBits = (hTp->parser.adts.bs.frame_length<<3) - (startPos - FDKgetValidBits(hBs)) - syncLength;
-            if (syncLayerFrameBits <= 0) {
-              err = TRANSPORTDEC_SYNC_ERROR;
-            }
-          } else {
-            hTp->numberOfRawDataBlocks = 0;
-          }
-          break;
-        case TT_MP4_LOAS:
-          if (hTp->numberOfRawDataBlocks <= 0)
-          {
-            syncLayerFrameBits = FDKreadBits(hBs, 13);
-            hTp->parser.latm.m_audioMuxLengthBytes = syncLayerFrameBits;
-            syncLayerFrameBits <<= 3;
-          }
-        case TT_MP4_LATM_MCP1:
-        case TT_MP4_LATM_MCP0:
-          if (hTp->numberOfRawDataBlocks <= 0)
-          {
-            hTp->globalFramePos = FDKgetValidBits(hBs);
-
-            err = CLatmDemux_Read(
-                    hBs,
-                   &hTp->parser.latm,
-                    hTp->transportFmt,
-                   &hTp->callbacks,
-                    hTp->asc,
-                    ignoreBufferFullness);
-
-            if (err != TRANSPORTDEC_OK) {
-              if (err != TRANSPORTDEC_NOT_ENOUGH_BITS) {
-                err = TRANSPORTDEC_SYNC_ERROR;
-              }
-            } else {
-              hTp->numberOfRawDataBlocks = CLatmDemux_GetNrOfSubFrames(&hTp->parser.latm);
-              syncLayerFrameBits -= startPos - FDKgetValidBits(hBs) - (13);            
-            }
-          } else {
-            err = CLatmDemux_ReadPayloadLengthInfo(hBs, &hTp->parser.latm);
-            if (err != TRANSPORTDEC_OK) {
-              err = TRANSPORTDEC_SYNC_ERROR;
-            }
-          }
-          if (err == TRANSPORTDEC_OK) {
-            rawDataBlockLength = CLatmDemux_GetFrameLengthInBits(&hTp->parser.latm);
-            hTp->numberOfRawDataBlocks--;
-          } else {
-            hTp->numberOfRawDataBlocks = 0;
-          }
-          break;
-        default:
-          {
-            syncLayerFrameBits = 0;
-          }
-          break;
-      }
+      err = transportDec_readHeader(
+              hTp,
+              hBs,
+              syncLength,
+              ignoreBufferFullness,
+             &rawDataBlockLength,
+             &fTraverseMoreFrames,
+             &syncLayerFrameBits,
+             &fConfigFound,
+             &headerBits
+              );
     }
 
-    headerBits += startPos - (INT)FDKgetValidBits(hBs);
     bitsAvail -= headerBits;
 
     checkLengthBits  = syncLayerFrameBits;
@@ -702,12 +797,12 @@ TRANSPORTDEC_ERROR synchronization(
       /* Enforce re-sync of transport headers. */
       hTp->numberOfRawDataBlocks = 0;
 
-      /* Ensure that the bit amount lands and a multiple of TPDEC_SYNCSKIP */
+      /* Ensure that the bit amount lands at a multiple of TPDEC_SYNCSKIP */
       bits = (bitsAvail + headerBits) % TPDEC_SYNCSKIP;
       /* Rewind - TPDEC_SYNCSKIP, in order to look for a synch one bit ahead next time. */
       FDKpushBiDirectional(hBs, -(headerBits - TPDEC_SYNCSKIP) + bits);
       bitsAvail += headerBits - TPDEC_SYNCSKIP - bits;
-      headerBits = 0;      
+      headerBits = 0;
     }
 
     /* Frame traversal */
@@ -731,7 +826,15 @@ TRANSPORTDEC_ERROR synchronization(
         }
 
         /* Break when config was found or it is not possible anymore to find a config */
-        if (startPosFirstFrame != -1 && (fConfigFound || err != TRANSPORTDEC_OK)) {
+        if (startPosFirstFrame != -1 && (fConfigFound || err != TRANSPORTDEC_OK))
+        {
+          /* In case of ECD and sync error, do not rewind anywhere. */
+          if (err == TRANSPORTDEC_SYNC_ERROR)
+          {
+            startPosFirstFrame = -1;
+            fConfigFound = 0;
+            numFramesTraversed = 0;
+          }
           break;
         }
       }
@@ -759,7 +862,7 @@ TRANSPORTDEC_ERROR synchronization(
   } 
 
   /* Additional burst data mode buffer fullness check. */
-  if ( !(hTp->flags & (TPDEC_IGNORE_BUFFERFULLNESS|TPDEC_SYNCOK)) && err == TRANSPORTDEC_OK) {
+  if ( !(hTp->flags & (TPDEC_LOST_FRAMES_PENDING|TPDEC_IGNORE_BUFFERFULLNESS|TPDEC_SYNCOK)) && err == TRANSPORTDEC_OK) {
     err = additionalHoldOffNeeded(hTp, transportDec_GetBufferFullness(hTp), FDKgetValidBits(hBs) - syncLayerFrameBits);
     if (err == TRANSPORTDEC_NOT_ENOUGH_BITS) {
       hTp->holdOffFrames++;
@@ -789,10 +892,22 @@ TRANSPORTDEC_ERROR synchronization(
 bail:
   hTp->auLength[0] = rawDataBlockLength;
 
+  /* Detect pointless TRANSPORTDEC_NOT_ENOUGH_BITS error case, were the bit buffer is already full,
+     or no new burst packet fits. Recover by advancing the bit buffer. */
+  if ( (TRANSPORTDEC_NOT_ENOUGH_BITS == err) &&  (FDKgetValidBits(hBs) >= ((TRANSPORTDEC_INBUF_SIZE*8 - ((hTp->avgBitRate*hTp->burstPeriod)/1000)) - 7)) )
+  {
+    FDKpushFor(hBs, TPDEC_SYNCSKIP);
+    err = TRANSPORTDEC_SYNC_ERROR;
+  }
+
   if (err == TRANSPORTDEC_OK) {
     hTp->flags |= TPDEC_SYNCOK;
   }
 
+  if (fConfigFound) {
+    hTp->flags |= TPDEC_CONFIG_FOUND;
+  }
+
   if (pHeaderBits != NULL) {
     *pHeaderBits = headerBits;
   }
@@ -925,6 +1040,10 @@ TRANSPORTDEC_ERROR transportDec_ReadAccessUnit( const HANDLE_TRANSPORTDEC hTp, c
 
   hBs = &hTp->bitStream[layer];
 
+  if ((INT)FDKgetValidBits(hBs) <= 0) {
+    err = TRANSPORTDEC_NOT_ENOUGH_BITS;
+  }
+
   switch (hTp->transportFmt) {
 
     case TT_MP4_ADIF:
@@ -964,20 +1083,26 @@ TRANSPORTDEC_ERROR transportDec_ReadAccessUnit( const HANDLE_TRANSPORTDEC hTp, c
       break;
 
     case TT_MP4_RAW:
-      if ((INT)FDKgetValidBits(hBs) <= 0 && layer == 0) {
-        err = TRANSPORTDEC_NOT_ENOUGH_BITS;
-      }
       /* One Access Unit was filled into buffer.
          So get the length out of the buffer. */
       hTp->auLength[layer] = FDKgetValidBits(hBs);
       hTp->flags |= TPDEC_SYNCOK;
       break;
 
+    case TT_MP4_LATM_MCP0:
+    case TT_MP4_LATM_MCP1:
+      {
+        int fConfigFound = hTp->flags & TPDEC_CONFIG_FOUND;
+        err = transportDec_readHeader(hTp, hBs, 0, 1, &hTp->auLength[layer], NULL, NULL, &fConfigFound, NULL);
+        if (fConfigFound) {
+          hTp->flags |= TPDEC_CONFIG_FOUND;
+        }
+      }
+      break;
+
     case TT_RSVD50:
     case TT_MP4_ADTS:
     case TT_MP4_LOAS:
-    case TT_MP4_LATM_MCP0:
-    case TT_MP4_LATM_MCP1:
       err = transportDec_readStream(hTp, layer);
       break;
 
@@ -1026,13 +1151,10 @@ TRANSPORTDEC_ERROR transportDec_EndAccessUnit(HANDLE_TRANSPORTDEC hTp)
 {
   TRANSPORTDEC_ERROR err = TRANSPORTDEC_OK;
 
+
   err = transportDec_AdjustEndOfAccessUnit(hTp);
 
   switch (hTp->transportFmt) {
-    case TT_MP4_LOAS:
-    case TT_MP4_LATM_MCP0:
-    case TT_MP4_LATM_MCP1:
-      break;
     default:
       break;
   }
@@ -1082,8 +1204,11 @@ TRANSPORTDEC_ERROR transportDec_SetParam ( const HANDLE_TRANSPORTDEC hTp,
           FDKresetBitbuffer(&hTp->bitStream[i]);
           hTp->auLength[i] = 0;
           hTp->accessUnitAnchor[i] = 0;
-        }        
+        }
         hTp->flags &= ~(TPDEC_SYNCOK|TPDEC_LOST_FRAMES_PENDING);
+        if (hTp->transportFmt != TT_MP4_ADIF) {
+          hTp->flags &= ~TPDEC_CONFIG_FOUND;
+        }
         hTp->remainder = 0;
         hTp->avgBitRate = 0;
         hTp->missingAccessUnits = 0;
@@ -1117,7 +1242,7 @@ void transportDec_Close(HANDLE_TRANSPORTDEC *phTp)
   if (phTp != NULL)
   {
     if (*phTp != NULL) {
-      if ((*phTp)->transportFmt != TT_MP4_RAW && (*phTp)->transportFmt != TT_DRM) {
+      if ( ! TT_IS_PACKET((*phTp)->transportFmt) ) {
         FreeRam_TransportDecoderBuffer(&(*phTp)->bsBuffer);
       }
       if (*phTp != NULL) {
@@ -1188,13 +1313,8 @@ TRANSPORTDEC_ERROR transportDec_CrcCheck(HANDLE_TRANSPORTDEC pTp)
     if ( (pTp->parser.adts.bs.num_raw_blocks > 0) && (pTp->parser.adts.bs.protection_absent == 0) )
     {
       HANDLE_FDK_BITSTREAM hBs = &pTp->bitStream[0];
-      int bitDiff;
       
-      /* Calculate possible offset to CRC value. */
-      bitDiff  = pTp->parser.adts.rawDataBlockDist[pTp->parser.adts.bs.num_raw_blocks-pTp->numberOfRawDataBlocks]<<3;
-      bitDiff -= pTp->globalFramePos - FDKgetValidBits(hBs) + 16;
-      FDKpushBiDirectional(hBs, bitDiff);
-      pTp->parser.adts.crcReadValue = FDKreadBits(hBs, 16);
+      transportDec_AdjustEndOfAccessUnit(pTp);
     }
     return adtsRead_CrcCheck(&pTp->parser.adts);
   default:
diff --git a/libMpegTPDec/src/version b/libMpegTPDec/src/version
index 5b9a425..2803347 100644
--- a/libMpegTPDec/src/version
+++ b/libMpegTPDec/src/version
@@ -1,8 +1,8 @@
 
 /* library info */
 #define TP_LIB_VL0 2
-#define TP_LIB_VL1 2
-#define TP_LIB_VL2 1
+#define TP_LIB_VL1 3
+#define TP_LIB_VL2 3
 #define TP_LIB_TITLE "MPEG Transport"
 #define TP_LIB_BUILD_DATE __DATE__
 #define TP_LIB_BUILD_TIME __TIME__
diff --git a/libMpegTPEnc/include/mpegFileWrite.h b/libMpegTPEnc/include/mpegFileWrite.h
index 6716d8f..f886a0b 100644
--- a/libMpegTPEnc/include/mpegFileWrite.h
+++ b/libMpegTPEnc/include/mpegFileWrite.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libMpegTPEnc/include/tp_data.h b/libMpegTPEnc/include/tp_data.h
index 6e8218a..5269858 100644
--- a/libMpegTPEnc/include/tp_data.h
+++ b/libMpegTPEnc/include/tp_data.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libMpegTPEnc/include/tpenc_lib.h b/libMpegTPEnc/include/tpenc_lib.h
index a06e7a7..2833e82 100644
--- a/libMpegTPEnc/include/tpenc_lib.h
+++ b/libMpegTPEnc/include/tpenc_lib.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libMpegTPEnc/src/tpenc_adif.cpp b/libMpegTPEnc/src/tpenc_adif.cpp
index 06bd30b..b48a32e 100644
--- a/libMpegTPEnc/src/tpenc_adif.cpp
+++ b/libMpegTPEnc/src/tpenc_adif.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libMpegTPEnc/src/tpenc_adif.h b/libMpegTPEnc/src/tpenc_adif.h
index 0921712..d590354 100644
--- a/libMpegTPEnc/src/tpenc_adif.h
+++ b/libMpegTPEnc/src/tpenc_adif.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libMpegTPEnc/src/tpenc_adts.cpp b/libMpegTPEnc/src/tpenc_adts.cpp
index 39460f0..f4f3178 100644
--- a/libMpegTPEnc/src/tpenc_adts.cpp
+++ b/libMpegTPEnc/src/tpenc_adts.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libMpegTPEnc/src/tpenc_adts.h b/libMpegTPEnc/src/tpenc_adts.h
index 9123f1e..c12c7c7 100644
--- a/libMpegTPEnc/src/tpenc_adts.h
+++ b/libMpegTPEnc/src/tpenc_adts.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libMpegTPEnc/src/tpenc_asc.cpp b/libMpegTPEnc/src/tpenc_asc.cpp
index 6200c14..bc4302e 100644
--- a/libMpegTPEnc/src/tpenc_asc.cpp
+++ b/libMpegTPEnc/src/tpenc_asc.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -127,8 +127,7 @@ typedef struct {
  *
  * The number of channel element parameter describes the kind of consecutively elements.
  * E.g. MODE_1_2_2_2_1 means:
- *                          - First 2 elements (SCE,CPE) are front channel elements.
- *                          - Following element (CPE) is a side channel element.
+ *                          - First 3 elements (SCE,CPE,CPE) are front channel elements.
  *                          - Next element (CPE) is a back channel element.
  *                          - Last element (LFE) is a lfe channel element.
  */
@@ -140,7 +139,8 @@ static const CHANNEL_CONFIGURATION pceConfigTab[] =
   { MODE_1_2_1,                    {  2, 0, 1, 0, { ID_SCE,  ID_CPE,  ID_SCE,  ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE } } },
   { MODE_1_2_2,                    {  2, 0, 1, 0, { ID_SCE,  ID_CPE,  ID_CPE,  ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE } } },
   { MODE_1_2_2_1,                  {  2, 0, 1, 1, { ID_SCE,  ID_CPE,  ID_CPE,  ID_LFE, ID_NONE, ID_NONE, ID_NONE, ID_NONE } } },
-  { MODE_1_2_2_2_1,                {  2, 1, 1, 1, { ID_SCE,  ID_CPE,  ID_CPE,  ID_CPE,  ID_LFE, ID_NONE, ID_NONE, ID_NONE } } },
+  { MODE_1_2_2_2_1,                {  3, 0, 1, 1, { ID_SCE,  ID_CPE,  ID_CPE,  ID_CPE,  ID_LFE,  ID_NONE, ID_NONE, ID_NONE } } },
+
 
   { MODE_1_1,                      {  2, 0, 0, 0, { ID_SCE,  ID_SCE,  ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE } } },
   { MODE_1_1_1_1,                  {  2, 2, 0, 0, { ID_SCE,  ID_SCE,  ID_SCE,  ID_SCE,  ID_NONE, ID_NONE, ID_NONE, ID_NONE } } },
@@ -151,7 +151,11 @@ static const CHANNEL_CONFIGURATION pceConfigTab[] =
   { MODE_2_2_2,                    {  1, 1, 1, 0, { ID_CPE,  ID_CPE,  ID_CPE,  ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE } } },
   { MODE_2_2_2_2,                  {  4, 0, 0, 0, { ID_CPE,  ID_CPE,  ID_CPE,  ID_CPE,  ID_NONE, ID_NONE, ID_NONE, ID_NONE } } },
 
-  { MODE_2_1,                      {  1, 0, 1, 0, { ID_CPE,  ID_SCE,  ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE } } }
+  { MODE_2_1,                      {  1, 0, 1, 0, { ID_CPE,  ID_SCE,  ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE, ID_NONE } } },
+
+  { MODE_7_1_REAR_SURROUND,        {  2, 0, 2, 1, { ID_SCE,  ID_CPE,  ID_CPE,  ID_CPE,  ID_LFE,  ID_NONE, ID_NONE, ID_NONE } } },
+  { MODE_7_1_FRONT_CENTER,         {  3, 0, 1, 1, { ID_SCE,  ID_CPE,  ID_CPE,  ID_CPE,  ID_LFE,  ID_NONE, ID_NONE, ID_NONE } } },
+
 };
 
 
@@ -255,7 +259,7 @@ int transportEnc_writePCE(HANDLE_FDK_BITSTREAM hBs,
   if ( matrixMixdownA!=0 && ((channelMode==MODE_1_2_2)||(channelMode==MODE_1_2_2_1)) ) {
       FDKwriteBits(hBs, 1, 1);                                  /* Matrix mixdown present */
       FDKwriteBits(hBs, (matrixMixdownA-1)&0x3, 2);             /* matrix_mixdown_idx */
-      FDKwriteBits(hBs, pseudoSurroundEnable&0x1, 1);           /* pseudo_surround_enable */
+      FDKwriteBits(hBs, (pseudoSurroundEnable)?1:0, 1);         /* pseudo_surround_enable */
   }
   else {
       FDKwriteBits(hBs, 0, 1);                                  /* Matrix mixdown not present */
@@ -379,7 +383,7 @@ int transportEnc_writeGASpecificConfig(
 
   /* Write PCE if channel config is not 1-7 */
   if (getChannelConfig(config->channelMode) == 0) {
-      transportEnc_writePCE(asc, config->channelMode, config->samplingRate, 0, 1, 0, 0, alignAnchor);
+      transportEnc_writePCE(asc, config->channelMode, config->samplingRate, 0, 1, config->matrixMixdownA, (config->flags&CC_PSEUDO_SURROUND)?1:0, alignAnchor);
   }
   if (extFlg) {
     if (aot == AOT_ER_BSAC) {
@@ -474,7 +478,7 @@ int transportEnc_writeASC (
         break;
   }
 
-  if (config->extAOT == AOT_SBR || config->extAOT == AOT_PS)
+  if (config->sbrSignaling==SIG_EXPLICIT_HIERARCHICAL && config->sbrPresent)
     writeAot(asc, config->extAOT);
   else
     writeAot(asc, config->aot);
@@ -492,7 +496,7 @@ int transportEnc_writeASC (
 
   FDKwriteBits( asc, getChannelConfig(config->channelMode), 4 );
 
-  if (config->extAOT == AOT_SBR || config->extAOT == AOT_PS) {
+  if (config->sbrSignaling==SIG_EXPLICIT_HIERARCHICAL && config->sbrPresent) {
     writeSampleRate(asc, config->extSamplingRate);
     writeAot(asc, config->aot);
   }
@@ -545,6 +549,26 @@ int transportEnc_writeASC (
       break;
   }
 
+  /* backward compatible explicit signaling of extension AOT */
+  if (config->sbrSignaling==SIG_EXPLICIT_BW_COMPATIBLE)
+  {
+    TP_ASC_EXTENSION_ID ascExtId = ASCEXT_UNKOWN;
+
+    if (config->sbrPresent) {
+      ascExtId=ASCEXT_SBR;
+      FDKwriteBits( asc, ascExtId, 11 );
+      writeAot(asc, config->extAOT);
+      FDKwriteBits( asc, 1, 1 ); /* sbrPresentFlag=1 */
+      writeSampleRate(asc, config->extSamplingRate);
+      if (config->psPresent) {
+        ascExtId=ASCEXT_PS;
+        FDKwriteBits( asc, ascExtId, 11 );
+        FDKwriteBits( asc, 1, 1 ); /* psPresentFlag=1 */
+      }
+    }
+
+  }
+
   /* Make sure all bits are sync'ed */
   FDKsyncCache( asc );
 
diff --git a/libMpegTPEnc/src/tpenc_asc.h b/libMpegTPEnc/src/tpenc_asc.h
index 990f74f..47fe7a1 100644
--- a/libMpegTPEnc/src/tpenc_asc.h
+++ b/libMpegTPEnc/src/tpenc_asc.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libMpegTPEnc/src/tpenc_latm.cpp b/libMpegTPEnc/src/tpenc_latm.cpp
index 54fd717..58e51ef 100644
--- a/libMpegTPEnc/src/tpenc_latm.cpp
+++ b/libMpegTPEnc/src/tpenc_latm.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libMpegTPEnc/src/tpenc_latm.h b/libMpegTPEnc/src/tpenc_latm.h
index fd95a60..34eea58 100644
--- a/libMpegTPEnc/src/tpenc_latm.h
+++ b/libMpegTPEnc/src/tpenc_latm.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libMpegTPEnc/src/tpenc_lib.cpp b/libMpegTPEnc/src/tpenc_lib.cpp
index 4f3177d..d0a2fec 100644
--- a/libMpegTPEnc/src/tpenc_lib.cpp
+++ b/libMpegTPEnc/src/tpenc_lib.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -150,12 +150,18 @@ C_ALLOC_MEM(Ram_TransportEncoder, TRANSPORTENC, 1)
 
 TRANSPORTENC_ERROR transportEnc_Open( HANDLE_TRANSPORTENC *phTpEnc )
 {
-  HANDLE_TRANSPORTENC hTpEnc = GetRam_TransportEncoder(0);
+  HANDLE_TRANSPORTENC hTpEnc;
 
-  if ( hTpEnc == NULL ) {
+  if ( phTpEnc == NULL ){
     return TRANSPORTENC_INVALID_PARAMETER;
   }
 
+  hTpEnc = GetRam_TransportEncoder(0);
+
+  if ( hTpEnc == NULL ) {
+    return TRANSPORTENC_NO_MEM;
+  }
+
   *phTpEnc = hTpEnc;
   return TRANSPORTENC_OK;
 }
@@ -417,7 +423,7 @@ TRANSPORTENC_ERROR transportEnc_WriteAccessUnit(
     }
 
     /* Write PCE as first raw_data_block element */
-    transportEnc_writePCE(&hTp->bitStream, hTp->config.channelMode, hTp->config.samplingRate, 0, 1, hTp->config.matrixMixdownA, hTp->config.flags & CC_PSEUDO_SURROUND, alignAnchor);
+    transportEnc_writePCE(&hTp->bitStream, hTp->config.channelMode, hTp->config.samplingRate, 0, 1, hTp->config.matrixMixdownA, (hTp->config.flags&CC_PSEUDO_SURROUND)?1:0, alignAnchor);
 
     if ( (hTp->transportFmt==TT_MP4_ADTS) && !hTp->writer.adts.protection_absent) {
       adtsWrite_CrcEndReg(&hTp->writer.adts, &hTp->bitStream, crcIndex);
diff --git a/libMpegTPEnc/src/version b/libMpegTPEnc/src/version
index 5b9a425..2803347 100644
--- a/libMpegTPEnc/src/version
+++ b/libMpegTPEnc/src/version
@@ -1,8 +1,8 @@
 
 /* library info */
 #define TP_LIB_VL0 2
-#define TP_LIB_VL1 2
-#define TP_LIB_VL2 1
+#define TP_LIB_VL1 3
+#define TP_LIB_VL2 3
 #define TP_LIB_TITLE "MPEG Transport"
 #define TP_LIB_BUILD_DATE __DATE__
 #define TP_LIB_BUILD_TIME __TIME__
diff --git a/libPCMutils/include/pcmutils_lib.h b/libPCMutils/include/pcmutils_lib.h
index fd4a26d..5ba74be 100644
--- a/libPCMutils/include/pcmutils_lib.h
+++ b/libPCMutils/include/pcmutils_lib.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libPCMutils/src/pcmutils_lib.cpp b/libPCMutils/src/pcmutils_lib.cpp
index e0a9817..a84a050 100644
--- a/libPCMutils/src/pcmutils_lib.cpp
+++ b/libPCMutils/src/pcmutils_lib.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -96,8 +96,8 @@ amm-info at iis.fraunhofer.de
 
 /* Decoder library info */
 #define PCMDMX_LIB_VL0 2
-#define PCMDMX_LIB_VL1 3
-#define PCMDMX_LIB_VL2 1
+#define PCMDMX_LIB_VL1 4
+#define PCMDMX_LIB_VL2 0
 #define PCMDMX_LIB_TITLE "PCM Downmix Lib"
 #define PCMDMX_LIB_BUILD_DATE __DATE__
 #define PCMDMX_LIB_BUILD_TIME __TIME__
diff --git a/libSBRdec/include/sbrdecoder.h b/libSBRdec/include/sbrdecoder.h
index 300a6d9..13707e0 100644
--- a/libSBRdec/include/sbrdecoder.h
+++ b/libSBRdec/include/sbrdecoder.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRdec/src/arm/env_calc_arm.cpp b/libSBRdec/src/arm/env_calc_arm.cpp
index 4e3a6de..12b17d8 100644
--- a/libSBRdec/src/arm/env_calc_arm.cpp
+++ b/libSBRdec/src/arm/env_calc_arm.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRdec/src/arm/lpp_tran_arm.cpp b/libSBRdec/src/arm/lpp_tran_arm.cpp
index 541e7c7..78bbc2c 100644
--- a/libSBRdec/src/arm/lpp_tran_arm.cpp
+++ b/libSBRdec/src/arm/lpp_tran_arm.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRdec/src/env_calc.cpp b/libSBRdec/src/env_calc.cpp
index 11df761..ade57fc 100644
--- a/libSBRdec/src/env_calc.cpp
+++ b/libSBRdec/src/env_calc.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRdec/src/env_calc.h b/libSBRdec/src/env_calc.h
index d21e0b3..8154166 100644
--- a/libSBRdec/src/env_calc.h
+++ b/libSBRdec/src/env_calc.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRdec/src/env_dec.cpp b/libSBRdec/src/env_dec.cpp
index ac6c299..24b2d3b 100644
--- a/libSBRdec/src/env_dec.cpp
+++ b/libSBRdec/src/env_dec.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -624,8 +624,8 @@ timeCompensateFirstEnvelope (HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static con
 
   deltaExp = FDK_getNumOctavesDiv8(newLen, refLen);
 
-  /* Shift by -3 to rescale ld-table, 1-ampRes to enable coarser steps */
-  shift = (FRACT_BITS - 1 - ENV_EXP_FRACT + 1 - h_sbr_data->ampResolutionCurrentFrame - 3);
+  /* Shift by -3 to rescale ld-table, ampRes-1 to enable coarser steps */
+  shift = (FRACT_BITS - 1 - ENV_EXP_FRACT - 1 + h_sbr_data->ampResolutionCurrentFrame - 3);
   deltaExp = deltaExp >> shift;
   pFrameInfo->borders[0] = estimatedStartPos;
   pFrameInfo->bordersNoise[0] = estimatedStartPos;
diff --git a/libSBRdec/src/env_dec.h b/libSBRdec/src/env_dec.h
index 3e656ed..6f6dae3 100644
--- a/libSBRdec/src/env_dec.h
+++ b/libSBRdec/src/env_dec.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRdec/src/env_extr.cpp b/libSBRdec/src/env_extr.cpp
index 716fb91..6cb0b99 100644
--- a/libSBRdec/src/env_extr.cpp
+++ b/libSBRdec/src/env_extr.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRdec/src/env_extr.h b/libSBRdec/src/env_extr.h
index be46246..5db6d3d 100644
--- a/libSBRdec/src/env_extr.h
+++ b/libSBRdec/src/env_extr.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRdec/src/huff_dec.cpp b/libSBRdec/src/huff_dec.cpp
index 6e00b23..31d686d 100644
--- a/libSBRdec/src/huff_dec.cpp
+++ b/libSBRdec/src/huff_dec.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRdec/src/huff_dec.h b/libSBRdec/src/huff_dec.h
index 7b2b50b..5443658 100644
--- a/libSBRdec/src/huff_dec.h
+++ b/libSBRdec/src/huff_dec.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRdec/src/lpp_tran.cpp b/libSBRdec/src/lpp_tran.cpp
index 2d51831..117e739 100644
--- a/libSBRdec/src/lpp_tran.cpp
+++ b/libSBRdec/src/lpp_tran.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -875,22 +875,8 @@ resetLppTransposer (HANDLE_SBR_LPP_TRANS hLppTrans,  /*!< Handle of lpp transpos
   /*
    * Initialize the patching parameter
    */
- desiredBorder = 21;
- if (fs < 92017) {
-    desiredBorder = 23;
-  }
-  if (fs < 75132) {
-    desiredBorder = 32;
-  }
-  if (fs < 55426) {
-    desiredBorder = 43;
-  }
-  if (fs < 46009) {
-    desiredBorder = 46;
-  }
-  if (fs < 35777) {
-    desiredBorder = 64;
-  }
+  /* ISO/IEC 14496-3 (Figure 4.48): goalSb = round( 2.048e6 / fs ) */
+  desiredBorder    = (((2048000*2) / fs) + 1) >> 1;
 
   desiredBorder = findClosestEntry(desiredBorder, v_k_master, numMaster, 1); /* Adapt region to master-table */
 
diff --git a/libSBRdec/src/lpp_tran.h b/libSBRdec/src/lpp_tran.h
index 1838c06..003a547 100644
--- a/libSBRdec/src/lpp_tran.h
+++ b/libSBRdec/src/lpp_tran.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRdec/src/psbitdec.cpp b/libSBRdec/src/psbitdec.cpp
index dfd532f..29bddf7 100644
--- a/libSBRdec/src/psbitdec.cpp
+++ b/libSBRdec/src/psbitdec.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRdec/src/psbitdec.h b/libSBRdec/src/psbitdec.h
index 3b65468..a2d4d6c 100644
--- a/libSBRdec/src/psbitdec.h
+++ b/libSBRdec/src/psbitdec.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRdec/src/psdec.cpp b/libSBRdec/src/psdec.cpp
index d494c65..965917a 100644
--- a/libSBRdec/src/psdec.cpp
+++ b/libSBRdec/src/psdec.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRdec/src/psdec.h b/libSBRdec/src/psdec.h
index e3a0424..3dbc76d 100644
--- a/libSBRdec/src/psdec.h
+++ b/libSBRdec/src/psdec.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRdec/src/psdec_hybrid.cpp b/libSBRdec/src/psdec_hybrid.cpp
index 7fc2c0a..cbd0e92 100644
--- a/libSBRdec/src/psdec_hybrid.cpp
+++ b/libSBRdec/src/psdec_hybrid.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRdec/src/psdec_hybrid.h b/libSBRdec/src/psdec_hybrid.h
index 6503df9..fcf9e3e 100644
--- a/libSBRdec/src/psdec_hybrid.h
+++ b/libSBRdec/src/psdec_hybrid.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRdec/src/sbr_crc.cpp b/libSBRdec/src/sbr_crc.cpp
index 760bd1f..a495f10 100644
--- a/libSBRdec/src/sbr_crc.cpp
+++ b/libSBRdec/src/sbr_crc.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRdec/src/sbr_crc.h b/libSBRdec/src/sbr_crc.h
index 542843d..30b8329 100644
--- a/libSBRdec/src/sbr_crc.h
+++ b/libSBRdec/src/sbr_crc.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRdec/src/sbr_deb.cpp b/libSBRdec/src/sbr_deb.cpp
index aa37ffe..9baff2e 100644
--- a/libSBRdec/src/sbr_deb.cpp
+++ b/libSBRdec/src/sbr_deb.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRdec/src/sbr_deb.h b/libSBRdec/src/sbr_deb.h
index 324dea9..cb954ba 100644
--- a/libSBRdec/src/sbr_deb.h
+++ b/libSBRdec/src/sbr_deb.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRdec/src/sbr_dec.cpp b/libSBRdec/src/sbr_dec.cpp
index bd3cd10..1282338 100644
--- a/libSBRdec/src/sbr_dec.cpp
+++ b/libSBRdec/src/sbr_dec.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -760,6 +760,8 @@ createSbrDec (SBR_CHANNEL * hSbrChannel,
   */
   {
     int qmfErr;
+    /* Adapted QMF analysis post-twiddles for down-sampled HQ SBR */
+    const UINT downSampledFlag = (downsampleFac==2) ? QMF_FLAG_DOWNSAMPLED : 0;
 
     qmfErr = qmfInitAnalysisFilterBank (
                     &hs->AnalysiscQMF,
@@ -768,7 +770,7 @@ createSbrDec (SBR_CHANNEL * hSbrChannel,
                      hHeaderData->freqBandData.lowSubband,
                      hHeaderData->freqBandData.highSubband,
                      hHeaderData->numberOfAnalysisBands,
-                     qmfFlags & (~QMF_FLAG_KEEP_STATES)
+                     (qmfFlags & (~QMF_FLAG_KEEP_STATES)) | downSampledFlag
                      );
     if (qmfErr != 0) {
       return SBRDEC_UNSUPPORTED_CONFIG;
diff --git a/libSBRdec/src/sbr_dec.h b/libSBRdec/src/sbr_dec.h
index 309327f..175e7b2 100644
--- a/libSBRdec/src/sbr_dec.h
+++ b/libSBRdec/src/sbr_dec.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRdec/src/sbr_ram.cpp b/libSBRdec/src/sbr_ram.cpp
index ee95e01..6ae941f 100644
--- a/libSBRdec/src/sbr_ram.cpp
+++ b/libSBRdec/src/sbr_ram.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRdec/src/sbr_ram.h b/libSBRdec/src/sbr_ram.h
index 5469e51..8fc2dae 100644
--- a/libSBRdec/src/sbr_ram.h
+++ b/libSBRdec/src/sbr_ram.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRdec/src/sbr_rom.cpp b/libSBRdec/src/sbr_rom.cpp
index 8520b14..e84c3cd 100644
--- a/libSBRdec/src/sbr_rom.cpp
+++ b/libSBRdec/src/sbr_rom.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -110,6 +110,8 @@ const UCHAR FDK_sbrDecoder_sbr_start_freq_32[16] = {10, 12, 14, 15, 16, 17, 18,
 const UCHAR FDK_sbrDecoder_sbr_start_freq_40[16] = {12, 13, 14, 15, 16, 17, 18, 19, 20, 21, 22, 24, 26, 28, 30, 32};
 const UCHAR FDK_sbrDecoder_sbr_start_freq_44[16] = { 8, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19, 21, 23, 25, 28, 32};
 const UCHAR FDK_sbrDecoder_sbr_start_freq_48[16] = { 7,  9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 20, 22, 24, 27, 31};
+const UCHAR FDK_sbrDecoder_sbr_start_freq_64[16] = { 6,  8,  9, 10, 11, 12, 13, 14, 15, 16, 17, 19, 21, 23, 26, 30};
+const UCHAR FDK_sbrDecoder_sbr_start_freq_88[16] = { 5,  6,  7,  8,  9, 10, 11, 12, 13, 14, 16, 18, 20, 23, 27, 31};
 //@}
 
 
diff --git a/libSBRdec/src/sbr_rom.h b/libSBRdec/src/sbr_rom.h
index 912f7e4..c318870 100644
--- a/libSBRdec/src/sbr_rom.h
+++ b/libSBRdec/src/sbr_rom.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -106,6 +106,8 @@ extern const UCHAR FDK_sbrDecoder_sbr_start_freq_32[16];
 extern const UCHAR FDK_sbrDecoder_sbr_start_freq_40[16];
 extern const UCHAR FDK_sbrDecoder_sbr_start_freq_44[16];
 extern const UCHAR FDK_sbrDecoder_sbr_start_freq_48[16];
+extern const UCHAR FDK_sbrDecoder_sbr_start_freq_64[16];
+extern const UCHAR FDK_sbrDecoder_sbr_start_freq_88[16];
 
 /*
   Low-Power-Profile Transposer
diff --git a/libSBRdec/src/sbr_scale.h b/libSBRdec/src/sbr_scale.h
index a98fc05..5fccd71 100644
--- a/libSBRdec/src/sbr_scale.h
+++ b/libSBRdec/src/sbr_scale.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRdec/src/sbrdec_drc.cpp b/libSBRdec/src/sbrdec_drc.cpp
index 7497b66..a834c0b 100644
--- a/libSBRdec/src/sbrdec_drc.cpp
+++ b/libSBRdec/src/sbrdec_drc.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -127,8 +127,8 @@ void sbrDecoder_drcInitChannel (
   hDrcData->currFact_exp = 1;
   hDrcData->nextFact_exp = 1;
 
-  hDrcData->numBandsCurr = 0;
-  hDrcData->numBandsNext = 0;
+  hDrcData->numBandsCurr = 1;
+  hDrcData->numBandsNext = 1;
 
   hDrcData->winSequenceCurr = 0;
   hDrcData->winSequenceNext = 0;
@@ -490,9 +490,7 @@ void sbrDecoder_drcApply (
   if (hDrcData == NULL) {
     return;
   }
-  if ( (hDrcData->enable == 0)
-    || ((hDrcData->numBandsCurr == 0) && (hDrcData->numBandsNext == 0))
-     ) {
+  if (hDrcData->enable == 0) {
     return;  /* Avoid changing the scaleFactor even though the processing is disabled. */
   }
 
diff --git a/libSBRdec/src/sbrdec_drc.h b/libSBRdec/src/sbrdec_drc.h
index 14deff7..872c6a0 100644
--- a/libSBRdec/src/sbrdec_drc.h
+++ b/libSBRdec/src/sbrdec_drc.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRdec/src/sbrdec_freq_sca.cpp b/libSBRdec/src/sbrdec_freq_sca.cpp
index b877545..8adfbb1 100644
--- a/libSBRdec/src/sbrdec_freq_sca.cpp
+++ b/libSBRdec/src/sbrdec_freq_sca.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -124,6 +124,13 @@ getStartBand(UINT   fs,               /*!< Output sampling frequency */
     fsMapped = fs;
 
   switch (fsMapped) {
+    case 96000:
+    case 88200:
+      band = FDK_sbrDecoder_sbr_start_freq_88[startFreq];
+      break;
+    case 64000:
+      band = FDK_sbrDecoder_sbr_start_freq_64[startFreq];
+      break;
     case 48000:
       band = FDK_sbrDecoder_sbr_start_freq_48[startFreq];
       break;
diff --git a/libSBRdec/src/sbrdec_freq_sca.h b/libSBRdec/src/sbrdec_freq_sca.h
index eebdd52..cfe4f0e 100644
--- a/libSBRdec/src/sbrdec_freq_sca.h
+++ b/libSBRdec/src/sbrdec_freq_sca.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRdec/src/sbrdecoder.cpp b/libSBRdec/src/sbrdecoder.cpp
index 192bdd2..26b2ea2 100644
--- a/libSBRdec/src/sbrdecoder.cpp
+++ b/libSBRdec/src/sbrdecoder.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -83,7 +83,7 @@ amm-info at iis.fraunhofer.de
 
 /*!
   \file
-  \brief  SBR decoder frontend  
+  \brief  SBR decoder frontend
   This module provides a frontend to the SBR decoder. The function openSBR() is called for
   initialization. The function sbrDecoder_Apply() is called for each frame. sbr_Apply() will call the
   required functions to decode the raw SBR data (provided by env_extr.cpp), to decode the envelope data and noise floor levels [decodeSbrData()],
@@ -136,7 +136,7 @@ amm-info at iis.fraunhofer.de
 
 /* Decoder library info */
 #define SBRDECODER_LIB_VL0 2
-#define SBRDECODER_LIB_VL1 1
+#define SBRDECODER_LIB_VL1 2
 #define SBRDECODER_LIB_VL2 3
 #define SBRDECODER_LIB_TITLE "SBR Decoder"
 #define SBRDECODER_LIB_BUILD_DATE __DATE__
@@ -217,14 +217,14 @@ SBR_ERROR sbrDecoder_ResetElement (
 
   /* Check in/out samplerates */
   if ( sampleRateIn < 6400
-    || sampleRateIn > 24000
+    || sampleRateIn > 48000
      )
   {
     sbrError = SBRDEC_UNSUPPORTED_CONFIG;
     goto bail;
   }
 
-  if ( sampleRateOut > 48000 )
+  if ( sampleRateOut > 96000 )
   {
     sbrError = SBRDEC_UNSUPPORTED_CONFIG;
     goto bail;
@@ -552,7 +552,7 @@ bail:
       sbrDecoder_DestroyElement( self, elementIndex );
     } else if (self->pSbrElement[elementIndex] != NULL) {
       /* Set error flag to trigger concealment */
-      self->pSbrElement[elementIndex]->frameErrorFlag[self->pSbrElement[elementIndex]->useFrameSlot] = 1;;
+      self->pSbrElement[elementIndex]->frameErrorFlag[self->pSbrElement[elementIndex]->useFrameSlot] = 1;
     }
   }
 
@@ -731,6 +731,12 @@ SBR_ERROR sbrDecoder_SetParam (HANDLE_SBRDECODER   self,
   case SBR_BS_INTERRUPTION:
     {
       int elementIndex;
+
+      if (self == NULL) {
+        errorStatus = SBRDEC_NOT_INITIALIZED;
+        break;
+      }
+
       /* Loop over SBR elements */
       for (elementIndex = 0; elementIndex < self->numSbrElements; elementIndex++)
       {
@@ -878,7 +884,7 @@ void sbrDecoder_drcDisable ( HANDLE_SBRDECODER  self,
   pSbrDrcChannelData = sbrDecoder_drcGetChannel( self, ch );
 
   if ( pSbrDrcChannelData != NULL ) {
-    pSbrDrcChannelData->enable = 0;
+    sbrDecoder_drcInitChannel( pSbrDrcChannelData );
   }
 }
 
diff --git a/libSBRdec/src/transcendent.h b/libSBRdec/src/transcendent.h
index f0ee21e..ad88bc9 100644
--- a/libSBRdec/src/transcendent.h
+++ b/libSBRdec/src/transcendent.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRenc/include/sbr_encoder.h b/libSBRenc/include/sbr_encoder.h
index 992c20c..93dc46d 100644
--- a/libSBRenc/include/sbr_encoder.h
+++ b/libSBRenc/include/sbr_encoder.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -101,6 +101,14 @@ amm-info at iis.fraunhofer.de
 #define MAX_CODEC_FRAME_RATIO 2
 #define MAX_PAYLOAD_SIZE    256
 
+typedef enum codecType
+{
+  CODEC_AAC=0,
+  CODEC_AACLD=1,
+  CODEC_UNSPECIFIED=99
+} CODEC_TYPE;
+
+
 typedef struct
 {
   INT bitRate;
@@ -129,10 +137,11 @@ enum
 
 typedef struct
 {
+  CODEC_TYPE       coreCoder;        /*!< LC or ELD */
   UINT             bitrateFrom;      /*!< inclusive */
   UINT             bitrateTo;        /*!< exclusive */
 
-  USHORT           sampleRate;       /*!<   */
+  UINT             sampleRate;       /*!<   */
   UCHAR            numChannels;      /*!<   */
 
   UCHAR            startFreq;        /*!< bs_start_freq */
@@ -158,6 +167,7 @@ typedef struct sbrConfiguration
   INT crcSbr;                 /*!< Flag: usage of SBR-CRC. */
   INT dynBwSupported;         /*!< Flag: support for dynamic bandwidth in this combination. */
   INT parametricCoding;       /*!< Flag: usage of parametric coding tool. */
+  INT downSampleFactor;       /*!< Sampling rate relation between the SBR and the core encoder. */
   int freq_res_fixfix[3];     /*!< Frequency resolution of envelopes in frame class FIXFIX
                                  0=1 Env; 1=2 Env; 2=4 Env; */
   /*
@@ -194,7 +204,6 @@ typedef struct sbrConfiguration
   INT useSaPan;               /*!< Flag: usage of SAPAN stereo. */
   INT dynBwEnabled;           /*!< Flag: usage of dynamic bandwidth. */
   INT bParametricStereo;      /*!< Flag: usage of parametric stereo coding tool. */
-  INT bDownSampledSbr;        /*!< Signal downsampled SBR is used. */
 
   /*
      header_extra1 configuration
@@ -214,7 +223,7 @@ typedef struct sbrConfiguration
   UCHAR init_amp_res_FF;
 } sbrConfiguration, *sbrConfigurationPtr ;
 
-typedef struct
+typedef struct SBR_CONFIG_DATA
 {
   UINT sbrSyntaxFlags;                  /**< SBR syntax flags derived from AOT. */
   INT nChannels;                        /**< Number of channels.  */
@@ -240,9 +249,7 @@ typedef struct
   INT xposCtrlSwitch;                   /**< Flag indicates whether to switch xpos ctrl on the fly. */
   INT switchTransposers;                /**< Flag indicates whether to switch xpos on the fly .     */
   UCHAR initAmpResFF;
-} SBR_CONFIG_DATA;
-
-typedef SBR_CONFIG_DATA *HANDLE_SBR_CONFIG_DATA;
+} SBR_CONFIG_DATA, *HANDLE_SBR_CONFIG_DATA;
 
 typedef struct {
   MP4_ELEMENT_ID elType;
@@ -275,15 +282,26 @@ INT sbrEncoder_Open(
         );
 
 /**
- * \brief get closest working bit rate to specified desired bit rate for a single SBR element
- * \param bitRate the desired target bit rate
- * \param numChannels the amount of audio channels
- * \param coreSampleRate the sample rate of the core coder
- * \param the current Audio Object Type
- * \return closest working bit rate to bitRate value
+ * \brief                 Get closest working bitrate to specified desired
+ *                        bitrate for a single SBR element.
+ * \param bitRate         The desired target bit rate
+ * \param numChannels     The amount of audio channels
+ * \param coreSampleRate  The sample rate of the core coder
+ * \param aot             The current Audio Object Type
+ * \return                Closest working bit rate to bitRate value
  */
 UINT sbrEncoder_LimitBitRate(UINT bitRate, UINT numChannels, UINT coreSampleRate, AUDIO_OBJECT_TYPE aot);
 
+
+/**
+ * \brief                Check whether downsampled SBR single rate is possible
+ *                       with given audio object type.
+ * \param aot            The Audio object type.
+ * \return               0 when downsampled SBR is not possible,
+ *                       1 when downsampled SBR is possible.
+ */
+UINT sbrEncoder_IsSingleRatePossible(AUDIO_OBJECT_TYPE aot);
+
 /**
  * \brief                  Initialize SBR Encoder instance.
  * \param phSbrEncoder     Pointer to a SBR Encoder instance.
@@ -294,26 +312,33 @@ UINT sbrEncoder_LimitBitRate(UINT bitRate, UINT numChannels, UINT coreSampleRate
  * \param bufferOffset     Returns the offset for the audio input data in order to do delay balancing.
  * \param numChannels      Input: Encoder input channels. output: core encoder channels.
  * \param sampleRate       Input: Encoder samplerate. output core encoder samplerate.
+ * \param downSampleFactor Input: Relation between SBR and core coder sampling rate;
  * \param frameLength      Input: Encoder frameLength. output core encoder frameLength.
  * \param aot              Input: Desired AOT. output AOT to be used after parameter checking.
  * \param delay            Input: core encoder delay. Output: total delay because of SBR.
  * \param transformFactor  The core encoder transform factor (blockswitching).
+ * \param headerPeriod     Repetition rate of the SBR header:
+ *                           - (-1) means intern configuration.
+ *                           - (1-10) corresponds to header repetition rate in frames.
  * \return                 0 on success, and non-zero if failed.
  */
-INT sbrEncoder_Init( HANDLE_SBR_ENCODER hSbrEncoder,
-                     SBR_ELEMENT_INFO elInfo[(6)],
-                     int              noElements,
-                     INT_PCM *inputBuffer,
-                     INT   *bandwidth,
-                     INT   *bufferOffset,
-                     INT   *numChannels,
-                     INT   *sampleRate,
-                     INT   *frameLength,
-                     AUDIO_OBJECT_TYPE *aot,
-                     int *delay,
-                     int  transformFactor,
-                     ULONG statesInitFlag
-                    );
+INT sbrEncoder_Init(
+        HANDLE_SBR_ENCODER   hSbrEncoder,
+        SBR_ELEMENT_INFO     elInfo[(8)],
+        int                  noElements,
+        INT_PCM             *inputBuffer,
+        INT                 *coreBandwidth,
+        INT                 *inputBufferOffset,
+        INT                 *numChannels,
+        INT                 *sampleRate,
+        UINT                *downSampleFactor,
+        INT                 *frameLength,
+        AUDIO_OBJECT_TYPE    aot,
+        int                 *delay,
+        int                  transformFactor,
+        const int            headerPeriod,
+        ULONG                statesInitFlag
+        );
 
 /**
  * \brief             Do delay line buffers housekeeping. To be called after each encoded audio frame.
@@ -344,8 +369,8 @@ void sbrEncoder_Close(HANDLE_SBR_ENCODER *phEbrEncoder);
 INT sbrEncoder_EncodeFrame(HANDLE_SBR_ENCODER  hEnvEncoder,
                            INT_PCM            *samples,
                            UINT                timeInStride,
-                           UINT                sbrDataBits[(6)],
-                           UCHAR               sbrData[(6)][MAX_PAYLOAD_SIZE]
+                           UINT                sbrDataBits[(8)],
+                           UCHAR               sbrData[(8)][MAX_PAYLOAD_SIZE]
                           );
 
 /**
@@ -356,7 +381,7 @@ INT sbrEncoder_EncodeFrame(HANDLE_SBR_ENCODER  hEnvEncoder,
  * \param fSendHeaders  Flag indicating that the SBR encoder should send more headers in the SBR payload or not.
  * \return              void
  */
-void sbrEncoder_GetHeader(SBR_ENCODER   *sbrEncoder,
+void sbrEncoder_GetHeader(HANDLE_SBR_ENCODER   sbrEncoder,
                           HANDLE_FDK_BITSTREAM hBs,
                           INT            element_index,
                           int            fSendHeaders);
diff --git a/libSBRenc/src/bit_sbr.cpp b/libSBRenc/src/bit_sbr.cpp
index 734a8aa..963aeff 100644
--- a/libSBRenc/src/bit_sbr.cpp
+++ b/libSBRenc/src/bit_sbr.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRenc/src/bit_sbr.h b/libSBRenc/src/bit_sbr.h
index bf170c1..1ce2c1e 100644
--- a/libSBRenc/src/bit_sbr.h
+++ b/libSBRenc/src/bit_sbr.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -125,11 +125,6 @@ struct SBR_HEADER_DATA
   INT freqScale;
 
   /*
-    element of sbrdata
-  */
-  SR_MODE sampleRateMode;
-
-  /*
     element of channelpairelement
   */
   INT coupling;
diff --git a/libSBRenc/src/cmondata.h b/libSBRenc/src/cmondata.h
index c3be1d7..32e6993 100644
--- a/libSBRenc/src/cmondata.h
+++ b/libSBRenc/src/cmondata.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRenc/src/code_env.cpp b/libSBRenc/src/code_env.cpp
index 7c169e6..e1a28d5 100644
--- a/libSBRenc/src/code_env.cpp
+++ b/libSBRenc/src/code_env.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRenc/src/code_env.h b/libSBRenc/src/code_env.h
index dd2b9ae..50a365e 100644
--- a/libSBRenc/src/code_env.h
+++ b/libSBRenc/src/code_env.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRenc/src/env_bit.cpp b/libSBRenc/src/env_bit.cpp
index 55c6967..ea31183 100644
--- a/libSBRenc/src/env_bit.cpp
+++ b/libSBRenc/src/env_bit.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRenc/src/env_bit.h b/libSBRenc/src/env_bit.h
index 38578f8..038a32a 100644
--- a/libSBRenc/src/env_bit.h
+++ b/libSBRenc/src/env_bit.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRenc/src/env_est.cpp b/libSBRenc/src/env_est.cpp
index a9a7881..929f229 100644
--- a/libSBRenc/src/env_est.cpp
+++ b/libSBRenc/src/env_est.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -129,9 +129,6 @@ FDKsbrEnc_getEnergyFromCplxQmfData(FIXP_DBL **RESTRICT energyValues,/*!< the res
   /* Get Scratch buffer */
   C_ALLOC_SCRATCH_START(tmpNrg, FIXP_DBL, QMF_CHANNELS*QMF_MAX_TIME_SLOTS/2);
 
-  FDK_ASSERT(numberBands <= QMF_CHANNELS);
-  FDK_ASSERT(numberCols <= QMF_MAX_TIME_SLOTS);
-
   /* Get max possible scaling of QMF data */
   scale = DFRACT_BITS;
   for (k=0; k<numberCols; k++) {
@@ -817,22 +814,22 @@ calculateSbrEnvelope (FIXP_DBL **RESTRICT YBufferLeft,  /*! energy buffer left *
       }
 
       /* ld64 to integer conversion */
-      nrgLeft = fixMin(fixMax(nrgLeft,FL2FXCONST_DBL(0.0f)),FL2FXCONST_DBL(0.5f));
+      nrgLeft = fixMin(fixMax(nrgLeft,FL2FXCONST_DBL(0.0f)),(FL2FXCONST_DBL(0.5f)>>oneBitLess));
       nrgLeft = (FIXP_DBL)(LONG)nrgLeft >> (DFRACT_BITS-1-LD_DATA_SHIFT-1-oneBitLess-1);
       sfb_nrgLeft[m] = ((INT)nrgLeft+1)>>1; /* rounding */
 
       if (stereoMode == SBR_COUPLING) {
         FIXP_DBL scaleFract;
+        int sc0, sc1;
 
-        if (nrgRight != FL2FXCONST_DBL(0.0f)) {
-          int sc0 = CountLeadingBits(nrgLeft2);
-          int sc1 = CountLeadingBits(nrgRight);
+        nrgLeft2 = fixMax((FIXP_DBL)0x1, nrgLeft2);
+        nrgRight = fixMax((FIXP_DBL)0x1, nrgRight);
 
-          scaleFract = ((FIXP_DBL)(sc0-sc1)) << (DFRACT_BITS-1-LD_DATA_SHIFT); /* scale value in ld64 representation */
-          nrgRight = CalcLdData(nrgLeft2<<sc0) - CalcLdData(nrgRight<<sc1) - scaleFract;
-        }
-        else
-          nrgRight =  FL2FXCONST_DBL(0.5f);   /* ld64(4294967296.0f) */
+        sc0 = CountLeadingBits(nrgLeft2);
+        sc1 = CountLeadingBits(nrgRight);
+
+        scaleFract = ((FIXP_DBL)(sc0-sc1)) << (DFRACT_BITS-1-LD_DATA_SHIFT); /* scale value in ld64 representation */
+        nrgRight = CalcLdData(nrgLeft2<<sc0) - CalcLdData(nrgRight<<sc1) - scaleFract;
 
         /* ld64 to integer conversion */
         nrgRight = (FIXP_DBL)(LONG)(nrgRight) >> (DFRACT_BITS-1-LD_DATA_SHIFT-1-oneBitLess);
diff --git a/libSBRenc/src/env_est.h b/libSBRenc/src/env_est.h
index 4c30a50..5e632a4 100644
--- a/libSBRenc/src/env_est.h
+++ b/libSBRenc/src/env_est.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRenc/src/fram_gen.cpp b/libSBRenc/src/fram_gen.cpp
index afef6e4..86c3c81 100644
--- a/libSBRenc/src/fram_gen.cpp
+++ b/libSBRenc/src/fram_gen.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRenc/src/fram_gen.h b/libSBRenc/src/fram_gen.h
index fe4b262..3769266 100644
--- a/libSBRenc/src/fram_gen.h
+++ b/libSBRenc/src/fram_gen.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRenc/src/invf_est.cpp b/libSBRenc/src/invf_est.cpp
index 788ab7c..32df6d9 100644
--- a/libSBRenc/src/invf_est.cpp
+++ b/libSBRenc/src/invf_est.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRenc/src/invf_est.h b/libSBRenc/src/invf_est.h
index 7c66cf6..2bd2a78 100644
--- a/libSBRenc/src/invf_est.h
+++ b/libSBRenc/src/invf_est.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRenc/src/mh_det.cpp b/libSBRenc/src/mh_det.cpp
index a7fa208..73d1b8b 100644
--- a/libSBRenc/src/mh_det.cpp
+++ b/libSBRenc/src/mh_det.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -1244,25 +1244,38 @@ FDKsbrEnc_InitSbrMissingHarmonicsDetector (
 
   FDK_ASSERT(totNoEst <= MAX_NO_OF_ESTIMATES);
 
-  switch(frameSize){
-  case 2048:
-      hs->transientPosOffset = FRAME_MIDDLE_SLOT_2048;
-      hs->timeSlots          = NUMBER_TIME_SLOTS_2048;
-      break;
-  case 1920:
-      hs->transientPosOffset = FRAME_MIDDLE_SLOT_1920;
-      hs->timeSlots          = NUMBER_TIME_SLOTS_1920;
-      break;
-  case 1024:
-      hs->transientPosOffset = FRAME_MIDDLE_SLOT_512LD;
-      hs->timeSlots          = 16;
-      break;
-  case 960:
-      hs->transientPosOffset = FRAME_MIDDLE_SLOT_512LD;
-      hs->timeSlots          = 15;
-      break;
-  default:
-      return -1;
+  if (sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY)
+  {
+    switch(frameSize){
+    case 1024:
+    case 512:
+        hs->transientPosOffset = FRAME_MIDDLE_SLOT_512LD;
+        hs->timeSlots          = 16;
+        break;
+    case 960:
+    case 480:
+        hs->transientPosOffset = FRAME_MIDDLE_SLOT_512LD;
+        hs->timeSlots          = 15;
+        break;
+    default:
+        return -1;
+    }
+  } else
+  {
+    switch(frameSize){
+    case 2048:
+    case 1024:
+        hs->transientPosOffset = FRAME_MIDDLE_SLOT_2048;
+        hs->timeSlots          = NUMBER_TIME_SLOTS_2048;
+        break;
+    case 1920:
+    case 960:
+        hs->transientPosOffset = FRAME_MIDDLE_SLOT_1920;
+        hs->timeSlots          = NUMBER_TIME_SLOTS_1920;
+        break;
+    default:
+        return -1;
+    }
   }
 
   if (sbrSyntaxFlags & SBR_SYNTAX_LOW_DELAY) {
diff --git a/libSBRenc/src/mh_det.h b/libSBRenc/src/mh_det.h
index ac62532..74c2a99 100644
--- a/libSBRenc/src/mh_det.h
+++ b/libSBRenc/src/mh_det.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRenc/src/nf_est.cpp b/libSBRenc/src/nf_est.cpp
index 62bcc79..7a3c022 100644
--- a/libSBRenc/src/nf_est.cpp
+++ b/libSBRenc/src/nf_est.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -102,7 +102,7 @@ static const FIXP_DBL QuantOffset = (INT)0xfc000000;  /* ld64(0.25) */
 #define max(a,b) ( a > b ? a:b)
 #endif
 
-#define NOISE_FLOOR_OFFSET_SCALING  (3)
+#define NOISE_FLOOR_OFFSET_SCALING  (4)
 
 
 
@@ -484,11 +484,13 @@ FDKsbrEnc_InitSbrNoiseFloorEstimate (HANDLE_SBR_NOISE_FLOOR_ESTIMATE  h_sbrNoise
     tmp = ((FIXP_DBL)MAXVAL_DBL)>>NOISE_FLOOR_OFFSET_SCALING;
   }
   else {
-    FDK_ASSERT(noiseFloorOffset<=8); /* because of NOISE_FLOOR_OFFSET_SCALING */
+    /* noiseFloorOffset has to be smaller than 12, because
+       the result of the calculation below must be smaller than 1:
+       (2^(noiseFloorOffset/3))*2^4<1                                        */
+    FDK_ASSERT(noiseFloorOffset<12);
 
-      /* Assumes the noise floor offset in tuning table are in q31    */
-      /* Currently the table contains only 0 for noise floor offset   */
-      /* Change the qformat here when non-zero values would be filled */
+    /* Assumes the noise floor offset in tuning table are in q31    */
+    /* Change the qformat here when non-zero values would be filled */
     exp = fDivNorm((FIXP_DBL)noiseFloorOffset, 3, &qexp);
     tmp = fPow(2, DFRACT_BITS-1, exp, qexp, &qtmp);
     tmp = scaleValue(tmp, qtmp-NOISE_FLOOR_OFFSET_SCALING);
@@ -527,24 +529,30 @@ FDKsbrEnc_resetSbrNoiseFloorEstimate (HANDLE_SBR_NOISE_FLOOR_ESTIMATE h_sbrNoise
         h_sbrNoiseFloorEstimate->noNoiseBands = 1;
     }
     else{
-    /*
-    * Calculate number of noise bands 1,2 or 3 bands/octave
+        /*
+        * Calculate number of noise bands 1,2 or 3 bands/octave
         ********************************************************/
         FIXP_DBL tmp, ratio, lg2;
-        INT ratio_e, qlg2;
+        INT ratio_e, qlg2, nNoiseBands;
 
         ratio = fDivNorm(k2, kx, &ratio_e);
         lg2 = fLog2(ratio, ratio_e, &qlg2);
         tmp = fMult((FIXP_DBL)(h_sbrNoiseFloorEstimate->noiseBands<<24), lg2);
         tmp = scaleValue(tmp, qlg2-23);
 
-        h_sbrNoiseFloorEstimate->noNoiseBands = (INT)((tmp + (FIXP_DBL)1) >> 1);
+        nNoiseBands = (INT)((tmp + (FIXP_DBL)1) >> 1);
+
+
+        if (nNoiseBands > MAX_NUM_NOISE_COEFFS ) {
+          nNoiseBands = MAX_NUM_NOISE_COEFFS;
+        }
+
+        if( nNoiseBands == 0 ) {
+          nNoiseBands = 1;
+        }
 
-        if (h_sbrNoiseFloorEstimate->noNoiseBands > MAX_NUM_NOISE_COEFFS)
-          h_sbrNoiseFloorEstimate->noNoiseBands = MAX_NUM_NOISE_COEFFS;
+        h_sbrNoiseFloorEstimate->noNoiseBands = nNoiseBands;
 
-        if( h_sbrNoiseFloorEstimate->noNoiseBands==0)
-            h_sbrNoiseFloorEstimate->noNoiseBands=1;
     }
 
 
diff --git a/libSBRenc/src/nf_est.h b/libSBRenc/src/nf_est.h
index 084899a..d407274 100644
--- a/libSBRenc/src/nf_est.h
+++ b/libSBRenc/src/nf_est.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRenc/src/ps_bitenc.cpp b/libSBRenc/src/ps_bitenc.cpp
index 419d989..8a42a20 100644
--- a/libSBRenc/src/ps_bitenc.cpp
+++ b/libSBRenc/src/ps_bitenc.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -267,13 +267,13 @@ static const INT psBands[] =
   PS_BANDS_MID
 };
 
-static INT getNoBands(PS_RESOLUTION mode)
+static INT getNoBands(UINT mode)
 {
   if(mode>=6)
     return 0;
 
   if(mode>=3)
-    mode = (PS_RESOLUTION)(mode-3);
+    mode = mode-3;
 
   return psBands[mode];
 }
@@ -524,7 +524,7 @@ static INT encodeIpdOpd(HANDLE_PS_OUT        psOut,
       bitCnt += FDKsbrEnc_EncodeIpd( hBitBuf,
                            psOut->ipd[env],
                            ipdLast,
-                           getNoBands((PS_RESOLUTION)psOut->iidMode),
+                           getNoBands((UINT)psOut->iidMode),
                            psOut->deltaIPD[env],
                            &error);
 
@@ -532,7 +532,7 @@ static INT encodeIpdOpd(HANDLE_PS_OUT        psOut,
       bitCnt += FDKsbrEnc_EncodeOpd( hBitBuf,
                            psOut->opd[env],
                            opdLast,
-                           getNoBands((PS_RESOLUTION)psOut->iidMode),
+                           getNoBands((UINT)psOut->iidMode),
                            psOut->deltaOPD[env],
                            &error );
     }
@@ -661,7 +661,7 @@ INT FDKsbrEnc_WritePSBitstream(const HANDLE_PS_OUT   psOut,
         bitCnt += FDKsbrEnc_EncodeIid( hBitBuf,
                              psOut->iid[env],
                              iidLast,
-                             getNoBands((PS_RESOLUTION)psOut->iidMode),
+                             getNoBands((UINT)psOut->iidMode),
                              (PS_IID_RESOLUTION)getIIDRes(psOut->iidMode),
                              psOut->deltaIID[env],
                              &error );
@@ -677,7 +677,7 @@ INT FDKsbrEnc_WritePSBitstream(const HANDLE_PS_OUT   psOut,
         bitCnt += FDKsbrEnc_EncodeIcc( hBitBuf,
                              psOut->icc[env],
                              iccLast,
-                             getNoBands((PS_RESOLUTION)psOut->iccMode),
+                             getNoBands((UINT)psOut->iccMode),
                              psOut->deltaICC[env],
                              &error);
 
diff --git a/libSBRenc/src/ps_bitenc.h b/libSBRenc/src/ps_bitenc.h
index cfc5af7..e98fe58 100644
--- a/libSBRenc/src/ps_bitenc.h
+++ b/libSBRenc/src/ps_bitenc.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRenc/src/ps_const.h b/libSBRenc/src/ps_const.h
index 08101e2..633d210 100644
--- a/libSBRenc/src/ps_const.h
+++ b/libSBRenc/src/ps_const.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRenc/src/ps_encode.cpp b/libSBRenc/src/ps_encode.cpp
index e60f83d..2ae2788 100644
--- a/libSBRenc/src/ps_encode.cpp
+++ b/libSBRenc/src/ps_encode.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRenc/src/ps_encode.h b/libSBRenc/src/ps_encode.h
index a12f7c6..f728d47 100644
--- a/libSBRenc/src/ps_encode.h
+++ b/libSBRenc/src/ps_encode.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRenc/src/ps_main.cpp b/libSBRenc/src/ps_main.cpp
index bbab25a..ab183e2 100644
--- a/libSBRenc/src/ps_main.cpp
+++ b/libSBRenc/src/ps_main.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -227,6 +227,8 @@ FDK_PSENC_ERROR PSEnc_Init(
     /* clear bs buffer */
     FDKmemclear(hParametricStereo->psOut, sizeof(hParametricStereo->psOut));
 
+    hParametricStereo->psOut[0].enablePSHeader = 1; /* write ps header in first frame */
+
     /* clear scaling buffer */
     FDKmemclear(hParametricStereo->dynBandScale, sizeof(UCHAR)*PS_MAX_BANDS);
     FDKmemclear(hParametricStereo->maxBandValue, sizeof(FIXP_QMF)*PS_MAX_BANDS);
@@ -313,7 +315,7 @@ static FDK_PSENC_ERROR DownmixPSQmfData(
   }
   else {
     int n, k;
-    C_ALLOC_SCRATCH_START(pWorkBuffer, FIXP_QMF, QMF_CHANNELS*2);    
+    C_AALLOC_SCRATCH_START(pWorkBuffer, FIXP_QMF, 2*QMF_CHANNELS)
 
     /* define scalings */
     int dynQmfScale = fixMax(0, hParametricStereo->dmxScale-1); /* scale one bit more for addition of left and right */
@@ -398,8 +400,7 @@ static FDK_PSENC_ERROR DownmixPSQmfData(
 
     *qmfScale = -downmixScale + 7;
 
-    C_ALLOC_SCRATCH_END(pWorkBuffer, FIXP_QMF, QMF_CHANNELS*2);
-
+    C_AALLOC_SCRATCH_END(pWorkBuffer, FIXP_QMF, 2*QMF_CHANNELS)
 
   {
     const INT noQmfSlots2 = hParametricStereo->noQmfSlots>>1;
@@ -473,10 +474,9 @@ FDK_PSENC_ERROR FDKsbrEnc_PSEnc_ParametricStereoProcessing(
         )
 {
   FDK_PSENC_ERROR error = PSENC_OK;
-  INT noQmfBands  = hParametricStereo->noQmfBands;
   INT psQmfScale[MAX_PS_CHANNELS] = {0};
   int psCh, i;
-  C_ALLOC_SCRATCH_START(pWorkBuffer, FIXP_DBL, QMF_CHANNELS*4);    
+  C_AALLOC_SCRATCH_START(pWorkBuffer, FIXP_QMF, 4*QMF_CHANNELS)
 
   for (psCh = 0; psCh<MAX_PS_CHANNELS; psCh ++) {
 
@@ -505,7 +505,7 @@ FDK_PSENC_ERROR FDKsbrEnc_PSEnc_ParametricStereoProcessing(
 
   } /* for psCh */
 
-  C_ALLOC_SCRATCH_END(pWorkBuffer, FIXP_DBL, QMF_CHANNELS*4);
+  C_AALLOC_SCRATCH_END(pWorkBuffer, FIXP_QMF, 4*QMF_CHANNELS)
 
   /* find best scaling in new QMF and Hybrid data */
   psFindBestScaling( hParametricStereo,
diff --git a/libSBRenc/src/ps_main.h b/libSBRenc/src/ps_main.h
index 6180299..21b32ff 100644
--- a/libSBRenc/src/ps_main.h
+++ b/libSBRenc/src/ps_main.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRenc/src/resampler.cpp b/libSBRenc/src/resampler.cpp
index e8ab263..4adb243 100644
--- a/libSBRenc/src/resampler.cpp
+++ b/libSBRenc/src/resampler.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRenc/src/resampler.h b/libSBRenc/src/resampler.h
index 29e170c..0192970 100644
--- a/libSBRenc/src/resampler.h
+++ b/libSBRenc/src/resampler.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRenc/src/sbr.h b/libSBRenc/src/sbr.h
index 13caadd..c74ad2a 100644
--- a/libSBRenc/src/sbr.h
+++ b/libSBRenc/src/sbr.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -105,7 +105,8 @@ amm-info at iis.fraunhofer.de
 /* SBR bitstream delay */
   #define DELAY_FRAMES 2
 
-typedef struct {
+
+typedef struct SBR_CHANNEL {
     struct ENV_CHANNEL        hEnvChannel;
     //INT_PCM                  *pDSOutBuffer;            /**< Pointer to downsampled audio output of SBR encoder */
     DOWNSAMPLER               downSampler;
@@ -113,7 +114,7 @@ typedef struct {
 } SBR_CHANNEL;
 typedef SBR_CHANNEL* HANDLE_SBR_CHANNEL;
 
-typedef struct {
+typedef struct SBR_ELEMENT {
     HANDLE_SBR_CHANNEL        sbrChannel[2];
     QMF_FILTER_BANK          *hQmfAnalysis[2];
     SBR_CONFIG_DATA           sbrConfigData;
@@ -126,14 +127,13 @@ typedef struct {
     UCHAR                     payloadDelayLine[1+DELAY_FRAMES][MAX_PAYLOAD_SIZE];
     UINT                      payloadDelayLineSize[1+DELAY_FRAMES];                 /* Sizes in bits */
 
-} SBR_ELEMENT;
-typedef SBR_ELEMENT* HANDLE_SBR_ELEMENT;
+} SBR_ELEMENT, *HANDLE_SBR_ELEMENT;
 
-struct SBR_ENCODER
+typedef struct SBR_ENCODER
 {
-  HANDLE_SBR_ELEMENT   sbrElement[(6)];
-  HANDLE_SBR_CHANNEL   pSbrChannel[(6)];
-  QMF_FILTER_BANK      QmfAnalysis[(6)];
+  HANDLE_SBR_ELEMENT   sbrElement[(8)];
+  HANDLE_SBR_CHANNEL   pSbrChannel[(8)];
+  QMF_FILTER_BANK      QmfAnalysis[(8)];
   DOWNSAMPLER          lfeDownSampler;
   int                  lfeChIdx;                 /* -1 default for no lfe, else assign channel index */
   int                  noElements;               /* Number of elements */
@@ -142,6 +142,7 @@ struct SBR_ENCODER
   int                  bufferOffset;             /* Offset for SBR parameter extraction in time domain input buffer. */
   int                  downsampledOffset;        /* Offset of downsampled/mixed output for core encoder. */
   int                  downmixSize;              /* Size in samples of downsampled/mixed output for core encoder. */
+  INT                  downSampleFactor;         /* Sampling rate relation between the SBR and the core encoder. */
   int                  fTimeDomainDownsampling;  /* Flag signalling time domain downsampling instead of QMF downsampling. */
   int                  nBitstrDelay;             /* Amount of SBR frames to be delayed in bitstream domain. */
   INT                  estimateBitrate;          /* estimate bitrate of SBR encoder */
@@ -158,7 +159,8 @@ struct SBR_ENCODER
   INT                  maxChannels;
   INT                  supportPS;
 
-} ;
+
+} SBR_ENCODER;
 
 
 #endif /* __SBR_H */
diff --git a/libSBRenc/src/sbr_def.h b/libSBRenc/src/sbr_def.h
index 1d99f7f..8b7cfc6 100644
--- a/libSBRenc/src/sbr_def.h
+++ b/libSBRenc/src/sbr_def.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -270,13 +270,6 @@ INVF_MODE;
 
 typedef enum
 {
-  SINGLE_RATE,
-  DUAL_RATE
-}
-SR_MODE;
-
-typedef enum
-{
   FREQ_RES_LOW = 0,
   FREQ_RES_HIGH
 }
diff --git a/libSBRenc/src/sbr_encoder.cpp b/libSBRenc/src/sbr_encoder.cpp
index e991199..3e95d6b 100644
--- a/libSBRenc/src/sbr_encoder.cpp
+++ b/libSBRenc/src/sbr_encoder.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -83,7 +83,7 @@ amm-info at iis.fraunhofer.de
 
 /***************************  Fraunhofer IIS FDK Tools  ***********************
 
-   Author(s): Andreas Ehret
+   Author(s): Andreas Ehret, Tobias Chalupka
    Description: SBR encoder top level processing.
 
 ******************************************************************************/
@@ -102,8 +102,8 @@ amm-info at iis.fraunhofer.de
 #include "ps_main.h"
 
 #define SBRENCODER_LIB_VL0 3
-#define SBRENCODER_LIB_VL1 2
-#define SBRENCODER_LIB_VL2 2
+#define SBRENCODER_LIB_VL1 3
+#define SBRENCODER_LIB_VL2 4
 
 
 
@@ -119,34 +119,30 @@ amm-info at iis.fraunhofer.de
            (core2sbr delay     )          ds     (read, core and ds area)
 */
 
-#define DOWN_SMPL_FAC        (2)
+#define SFB(dwnsmp)        (32 << (dwnsmp-1))     /* SBR Frequency bands: 64 for dual-rate, 32 for single-rate */
+#define STS(fl)            (((fl)==1024)?32:30)   /* SBR Time Slots: 32 for core frame length 1024, 30 for core frame length 960 */
 
-#define SFL(fl)            (fl*DOWN_SMPL_FAC)   /* SBR frame length (hardcoded to downsample factor of 2) */
-#define STS(fl)            (SFL(fl)/64)  /* SBR Time Slots */
-
-#define DELAY_QMF_ANA      (640 - 64)    /* Full bandwidth */
-#define DELAY_QMF_ANAELD   (32)
-#define DELAY_HYB_ANA      (10*64)       /* + 0.5 */
-#define DELAY_HYB_SYN      (6*64 - 32)
-#define DELAY_QMF_SYNELD   (32)
-#define DELAY_DEC_QMF      (6*64)        /* Decoder QMF overlap */
-#define DELAY_QMF_SYN      (2)           /* NO_POLY/2 */
-#define DELAY_QMF_DS       (32)          /* QMF synthesis for downsampled time signal */
+#define DELAY_QMF_ANA(dwnsmp) ((320<<((dwnsmp)-1)) - (32<<((dwnsmp)-1))) /* Full bandwidth */
+#define DELAY_HYB_ANA         (10*64)       /* + 0.5 */              /*  */
+#define DELAY_HYB_SYN         (6*64 - 32)                            /*  */
+#define DELAY_QMF_POSTPROC(dwnsmp) (32*(dwnsmp))                     /* QMF postprocessing delay */
+#define DELAY_DEC_QMF(dwnsmp) (6 * SFB(dwnsmp) )                     /* Decoder QMF overlap */
+#define DELAY_QMF_SYN         (2)                                    /* NO_POLY/2=2.5, rounded down to 2 */
+#define DELAY_QMF_DS          (32)                                   /* QMF synthesis for downsampled time signal */
 
 /* Delay in QMF paths */
-#define DELAY_SBR(fl)      (DELAY_QMF_ANA + (64*STS(fl)-1) + DELAY_QMF_SYN)
-#define DELAY_PS(fl)       (DELAY_QMF_ANA + DELAY_HYB_ANA + DELAY_DEC_QMF + (64*STS(fl)-1) + DELAY_HYB_SYN + DELAY_QMF_SYN)
-#define DELAY_ELDSBR(fl)   (DELAY_QMF_ANAELD + (((fl)+((fl)/2))*2 - 1) + DELAY_QMF_SYNELD)
+#define DELAY_SBR(fl,dwnsmp)     (DELAY_QMF_ANA(dwnsmp) + (SFB(dwnsmp)*STS(fl) - 1) + DELAY_QMF_SYN)
+#define DELAY_PS(fl,dwnsmp)      (DELAY_QMF_ANA(dwnsmp) + DELAY_HYB_ANA + DELAY_DEC_QMF(dwnsmp) + (SFB(dwnsmp)*STS(fl)-1) + DELAY_HYB_SYN + DELAY_QMF_SYN)
+#define DELAY_ELDSBR(fl,dwnsmp)  ( ( ((fl)/2)*(dwnsmp) ) - 1 + DELAY_QMF_POSTPROC(dwnsmp) )
 
 /* Delay differences for SBR and SBR+PS */
-    #define MAX_DS_FILTER_DELAY (34)                                      /* the additional max downsampler filter delay (source fs) */
-#define DELAY_AAC2SBR(fl)      ((/*RESAMPLER +*/ /*(DELAY_AAC(fl)*2) + */ DELAY_QMF_ANA + DELAY_DEC_QMF + DELAY_QMF_SYN) - DELAY_SBR(fl)) /* 1537 */
-#define DELAY_ELD2SBR(fl)      ((/*RESAMPLER +*/ /*(DELAY_ELD(fl)*2) + */ DELAY_QMF_ANAELD + DELAY_QMF_SYNELD) - DELAY_ELDSBR(fl))
-
-#define DELAY_AAC2PS(fl)       ((DELAY_QMF_ANA + DELAY_QMF_DS + /*(DELAY_AAC(fl)*2)*/ + DELAY_QMF_ANA + DELAY_DEC_QMF + DELAY_HYB_SYN + DELAY_QMF_SYN) - DELAY_PS(fl)) /* 2048 - 463*2 */
+#define MAX_DS_FILTER_DELAY (5)                                           /* the additional max downsampler filter delay (source fs) */
+#define DELAY_AAC2SBR(fl,dwnsmp) ((DELAY_QMF_ANA(dwnsmp) + DELAY_DEC_QMF(dwnsmp) + DELAY_QMF_SYN) - DELAY_SBR((fl),(dwnsmp)))
+#define DELAY_ELD2SBR(fl,dwnsmp) ((DELAY_QMF_POSTPROC(dwnsmp)) - DELAY_ELDSBR(fl,dwnsmp))
+#define DELAY_AAC2PS(fl,dwnsmp) ((DELAY_QMF_ANA(dwnsmp) + DELAY_QMF_DS + /*(DELAY_AAC(fl)*2) + */ DELAY_QMF_ANA(dwnsmp) + DELAY_DEC_QMF(dwnsmp) + DELAY_HYB_SYN + DELAY_QMF_SYN) - DELAY_PS(fl,dwnsmp)) /* 2048 - 463*2 */
 
-/* Assumption: that the sample delay resulting of of DELAY_AAC2PS is always smaller than the sample delay implied by DELAY_AAC2SBR */
-#define MAX_SAMPLE_DELAY       (DELAY_AAC2SBR(1024) + MAX_DS_FILTER_DELAY)
+/* Assumption: The sample delay resulting of of DELAY_AAC2PS is always smaller than the sample delay implied by DELAY_AAC2SBR */
+#define MAX_SAMPLE_DELAY       (DELAY_AAC2SBR(1024,2) + MAX_DS_FILTER_DELAY) /* maximum delay: frame length of 1024 and dual-rate sbr */
 
 /***************************************************************************/
 
@@ -172,41 +168,39 @@ getSbrTuningTableIndex(UINT bitrate,    /*! the total bitrate in bits/sec */
                        UINT *pBitRateClosest
                        )
 {
-  int i, paramSetTop, bitRateClosestLowerIndex=-1, bitRateClosestUpperIndex=-1, found = 0;
+  int i, bitRateClosestLowerIndex=-1, bitRateClosestUpperIndex=-1, found = 0;
   UINT bitRateClosestUpper = 0, bitRateClosestLower=DISTANCE_CEIL_VALUE;
+  int isforThisCodec=0;
 
-  FDK_ASSERT(SBRENC_TUNING_SIZE == sizeof(sbrTuningTable)/sizeof(sbrTuningTable[0]));
-
-  if (core == AOT_ER_AAC_ELD) {
-    paramSetTop = SBRENC_TUNING_SIZE;
-    i = SBRENC_AACLC_TUNING_SIZE;
-  } else {
-    paramSetTop = SBRENC_AACLC_TUNING_SIZE;
-    i = 0;
-  }
+  #define isForThisCore(i) \
+    ( ( sbrTuningTable[i].coreCoder == CODEC_AACLD && core == AOT_ER_AAC_ELD ) || \
+      ( sbrTuningTable[i].coreCoder == CODEC_AAC   && core != AOT_ER_AAC_ELD ) )
 
-  for (; i < paramSetTop ; i++) {
-    if ( numChannels == sbrTuningTable [i].numChannels
-      && sampleRate == sbrTuningTable [i].sampleRate )
+  for (i=0; i < sbrTuningTableSize ; i++) {
+    if ( isForThisCore(i) ) /* tuning table is for this core codec */
     {
-      found = 1;
-      if ((bitrate >= sbrTuningTable [i].bitrateFrom) &&
-          (bitrate < sbrTuningTable [i].bitrateTo)) {
-            bitRateClosestLower = bitrate;
-            bitRateClosestUpper = bitrate;
-            //FDKprintf("entry %d\n", i);
-        return i ;
-      } else {
-        if ( sbrTuningTable [i].bitrateFrom > bitrate ) {
-          if (sbrTuningTable [i].bitrateFrom < bitRateClosestLower) {
-            bitRateClosestLower = sbrTuningTable [i].bitrateFrom;
-            bitRateClosestLowerIndex = i;
+      if ( numChannels == sbrTuningTable [i].numChannels
+        && sampleRate == sbrTuningTable [i].sampleRate )
+      {
+        found = 1;
+        if ((bitrate >= sbrTuningTable [i].bitrateFrom) &&
+            (bitrate < sbrTuningTable [i].bitrateTo)) {
+              bitRateClosestLower = bitrate;
+              bitRateClosestUpper = bitrate;
+              //FDKprintf("entry %d\n", i);
+          return i ;
+        } else {
+          if ( sbrTuningTable [i].bitrateFrom > bitrate ) {
+            if (sbrTuningTable [i].bitrateFrom < bitRateClosestLower) {
+              bitRateClosestLower = sbrTuningTable [i].bitrateFrom;
+              bitRateClosestLowerIndex = i;
+            }
           }
-        }
-        if ( sbrTuningTable [i].bitrateTo <= bitrate ) {
-          if (sbrTuningTable [i].bitrateTo > bitRateClosestUpper) {
-            bitRateClosestUpper = sbrTuningTable [i].bitrateTo-1;
-            bitRateClosestUpperIndex = i;
+          if ( sbrTuningTable [i].bitrateTo <= bitrate ) {
+            if (sbrTuningTable [i].bitrateTo > bitRateClosestUpper) {
+              bitRateClosestUpper = sbrTuningTable [i].bitrateTo-1;
+              bitRateClosestUpperIndex = i;
+            }
           }
         }
       }
@@ -215,7 +209,7 @@ getSbrTuningTableIndex(UINT bitrate,    /*! the total bitrate in bits/sec */
 
   if (pBitRateClosest != NULL)
   {
-    /* Is there was at least one matching tuning entry found then pick the least distance bit rate */
+    /* If there was at least one matching tuning entry found then pick the least distance bit rate */
     if (found)
     {
       int distanceUpper=DISTANCE_CEIL_VALUE, distanceLower=DISTANCE_CEIL_VALUE;
@@ -295,6 +289,52 @@ getPsTuningTableIndex(UINT bitrate, UINT *pBitRateClosest){
   return INVALID_TABLE_IDX;
 }
 
+/***************************************************************************/
+/*!
+
+  \brief  In case of downsampled SBR we may need to lower the stop freq
+          of a tuning setting to fit into the lower half of the
+          spectrum ( which is sampleRate/4 )
+
+  \return the adapted stop frequency index (-1 -> error)
+
+  \ingroup SbrEncCfg
+
+****************************************************************************/
+static INT
+FDKsbrEnc_GetDownsampledStopFreq (
+        const INT sampleRateCore,
+        const INT startFreq,
+              INT stopFreq,
+        const INT downSampleFactor
+        )
+{
+  INT maxStopFreqRaw = sampleRateCore / 2;
+  INT startBand, stopBand;
+  HANDLE_ERROR_INFO err;
+
+  while (stopFreq > 0 && FDKsbrEnc_getSbrStopFreqRAW(stopFreq, sampleRateCore) > maxStopFreqRaw) {
+    stopFreq--;
+  }
+
+  if (FDKsbrEnc_getSbrStopFreqRAW( stopFreq, sampleRateCore) > maxStopFreqRaw)
+    return -1;
+
+  err = FDKsbrEnc_FindStartAndStopBand (
+                sampleRateCore<<(downSampleFactor-1),
+                sampleRateCore,
+                32<<(downSampleFactor-1),
+                startFreq,
+                stopFreq,
+               &startBand,
+               &stopBand
+                );
+  if (err)
+    return -1;
+
+  return stopFreq;
+}
+
 
 /***************************************************************************/
 /*!
@@ -307,15 +347,16 @@ getPsTuningTableIndex(UINT bitrate, UINT *pBitRateClosest){
 
 ****************************************************************************/
 static UINT
-FDKsbrEnc_IsSbrSettingAvail (UINT bitrate,          /*! the total bitrate in bits/sec */
-                   UINT vbrMode,          /*! the vbr paramter, 0 means constant bitrate */
-                   UINT numOutputChannels,/*! the number of channels for the core coder */
-                   UINT sampleRateInput,   /*! the input sample rate [in Hz] */
-                   AUDIO_OBJECT_TYPE core
-                   )
+FDKsbrEnc_IsSbrSettingAvail (
+        UINT bitrate,           /*! the total bitrate in bits/sec */
+        UINT vbrMode,           /*! the vbr paramter, 0 means constant bitrate */
+        UINT numOutputChannels, /*! the number of channels for the core coder */
+        UINT sampleRateInput,   /*! the input sample rate [in Hz] */
+        UINT sampleRateCore,    /*! the core's sampling rate */
+        AUDIO_OBJECT_TYPE core
+        )
 {
   INT idx = INVALID_TABLE_IDX;
-  UINT sampleRateCore;
 
   if (sampleRateInput < 16000)
     return 0;
@@ -335,8 +376,6 @@ FDKsbrEnc_IsSbrSettingAvail (UINT bitrate,          /*! the total bitrate in bit
     bitrate *= numOutputChannels;
   }
 
-  /* try DOWN_SMPL_FAC of the input sampling rate */
-  sampleRateCore = sampleRateInput/DOWN_SMPL_FAC;
   idx = getSbrTuningTableIndex(bitrate, numOutputChannels, sampleRateCore, core, NULL);
 
   return (idx == INVALID_TABLE_IDX ? 0 : 1);
@@ -356,7 +395,8 @@ static UINT
 FDKsbrEnc_AdjustSbrSettings (const sbrConfigurationPtr config, /*! output, modified */
                    UINT bitRate,             /*! the total bitrate in bits/sec */
                    UINT numChannels,         /*! the core coder number of channels */
-                   UINT fsCore,              /*! the core coder sampling rate in Hz */
+                   UINT sampleRateCore,      /*! the core coder sampling rate in Hz */
+                   UINT sampleRateSbr,       /*! the sbr coder sampling rate in Hz */
                    UINT transFac,            /*! the short block to long block ratio */
                    UINT standardBitrate,     /*! the standard bitrate per channel in bits/sec */
                    UINT vbrMode,             /*! the vbr paramter, 0 poor quality .. 100 high quality*/
@@ -366,15 +406,12 @@ FDKsbrEnc_AdjustSbrSettings (const sbrConfigurationPtr config, /*! output, modif
                    AUDIO_OBJECT_TYPE core)   /* Core audio codec object type */
 {
   INT idx = INVALID_TABLE_IDX;
-  UINT sampleRate;
-
-  /* set the codec settings */
+  /* set the core codec settings */
   config->codecSettings.bitRate         = bitRate;
   config->codecSettings.nChannels       = numChannels;
-  config->codecSettings.sampleFreq      = fsCore;
+  config->codecSettings.sampleFreq      = sampleRateCore;
   config->codecSettings.transFac        = transFac;
   config->codecSettings.standardBitrate = standardBitrate;
-  sampleRate  = fsCore * DOWN_SMPL_FAC;
 
   if (bitRate==0) {
     /* map vbr quality to bitrate */
@@ -391,13 +428,13 @@ FDKsbrEnc_AdjustSbrSettings (const sbrConfigurationPtr config, /*! output, modif
     bitRate *= numChannels;
     /* fix to enable mono vbrMode<40 @ 44.1 of 48kHz */
     if (numChannels==1) {
-      if (sampleRate==44100 || sampleRate==48000) {
+      if (sampleRateSbr==44100 || sampleRateSbr==48000) {
         if (vbrMode<40) bitRate = 32000;
       }
     }
   }
 
-  idx = getSbrTuningTableIndex(bitRate,numChannels,fsCore, core, NULL);
+  idx = getSbrTuningTableIndex(bitRate,numChannels,sampleRateCore, core, NULL);
 
   if (idx != INVALID_TABLE_IDX) {
     config->startFreq       = sbrTuningTable[idx].startFreq ;
@@ -407,6 +444,21 @@ FDKsbrEnc_AdjustSbrSettings (const sbrConfigurationPtr config, /*! output, modif
       config->stopFreq      = sbrTuningTable[idx].stopFreqSpeech;
     }
 
+    /* Adapt stop frequency in case of downsampled SBR - only 32 bands then */
+    if (1 == config->downSampleFactor) {
+      INT dsStopFreq = FDKsbrEnc_GetDownsampledStopFreq(
+                               sampleRateCore,
+                               config->startFreq,
+                               config->stopFreq,
+                               config->downSampleFactor
+                               );
+      if (dsStopFreq < 0) {
+        return 0;
+      }
+
+      config->stopFreq = dsStopFreq;
+    }
+
     config->sbr_noise_bands = sbrTuningTable[idx].numNoiseBands ;
     if (core == AOT_ER_AAC_ELD)
       config->init_amp_res_FF = SBR_AMP_RES_1_5;
@@ -455,19 +507,20 @@ FDKsbrEnc_AdjustSbrSettings (const sbrConfigurationPtr config, /*! output, modif
  description:  initializes the SBR confifuration
  returns:      error status
  input:        - core codec type,
-               - fac of SBR to core frame length,
+               - factor of SBR to core frame length,
                - core frame length
  output:       initialized SBR configuration
 
 *****************************************************************************/
 static UINT
 FDKsbrEnc_InitializeSbrDefaults (sbrConfigurationPtr config,
-                                 INT                 coreSbrFrameLenFac,
-                                 UINT                codecGranuleLen)
+                                 INT                 downSampleFactor,
+                                 UINT                codecGranuleLen
+                                 )
 {
-    if ( (coreSbrFrameLenFac != 2) ||
-         (codecGranuleLen*coreSbrFrameLenFac > QMF_CHANNELS*QMF_MAX_TIME_SLOTS) )
-       return(1);
+    if ( (downSampleFactor < 1 || downSampleFactor > 2) ||
+         (codecGranuleLen*downSampleFactor > QMF_CHANNELS*QMF_MAX_TIME_SLOTS) )
+       return(0); /* error */
 
     config->SendHeaderDataTime     = 1000;
     config->useWaveCoding          = 0;
@@ -476,8 +529,8 @@ FDKsbrEnc_InitializeSbrDefaults (sbrConfigurationPtr config,
     config->tran_thr               = 13000;
     config->parametricCoding       = 1;
 
-    config->sbrFrameSize  = codecGranuleLen * coreSbrFrameLenFac;
-
+    config->sbrFrameSize           = codecGranuleLen * downSampleFactor;
+    config->downSampleFactor       = downSampleFactor;
 
     /* sbr default parameters */
     config->sbr_data_extra         = 0;
@@ -497,7 +550,6 @@ FDKsbrEnc_InitializeSbrDefaults (sbrConfigurationPtr config,
     config->sbr_xpos_level  = 0;
     config->useSaPan        = 0;
     config->dynBwEnabled    = 0;
-    config->bDownSampledSbr = 0;
 
 
     /* the following parameters are overwritten by the FDKsbrEnc_AdjustSbrSettings() function since
@@ -601,7 +653,7 @@ void sbrEncoder_Close (HANDLE_SBR_ENCODER *phSbrEncoder)
   {
     int el, ch;
 
-    for (el=0; el<(6); el++)
+    for (el=0; el<(8); el++)
     {
       if (hSbrEncoder->sbrElement[el]!=NULL) {
         sbrEncoder_ElementClose(&hSbrEncoder->sbrElement[el]);
@@ -609,7 +661,7 @@ void sbrEncoder_Close (HANDLE_SBR_ENCODER *phSbrEncoder)
     }
 
     /* Close sbr Channels */
-    for (ch=0; ch<(6); ch++)
+    for (ch=0; ch<(8); ch++)
     {
       if (hSbrEncoder->pSbrChannel[ch]) {
         sbrEncoder_ChannelClose(hSbrEncoder->pSbrChannel[ch]);
@@ -645,46 +697,62 @@ void sbrEncoder_Close (HANDLE_SBR_ENCODER *phSbrEncoder)
  output:       error info
 
 *****************************************************************************/
-static INT updateFreqBandTable(HANDLE_SBR_CONFIG_DATA  sbrConfigData,
-                               HANDLE_SBR_HEADER_DATA  sbrHeaderData,
-                               INT noQmfChannels)
+static INT updateFreqBandTable(
+        HANDLE_SBR_CONFIG_DATA  sbrConfigData,
+        HANDLE_SBR_HEADER_DATA  sbrHeaderData,
+        const INT               downSampleFactor
+        )
 {
   INT k0, k2;
 
-  if(FDKsbrEnc_FindStartAndStopBand(sbrConfigData->sampleFreq,
-                                    noQmfChannels,
-                                    sbrHeaderData->sbr_start_frequency,
-                                    sbrHeaderData->sbr_stop_frequency,
-                                    sbrHeaderData->sampleRateMode,
-                                    &k0, &k2))
+  if( FDKsbrEnc_FindStartAndStopBand (
+              sbrConfigData->sampleFreq,
+              sbrConfigData->sampleFreq >> (downSampleFactor-1),
+              sbrConfigData->noQmfBands,
+              sbrHeaderData->sbr_start_frequency,
+              sbrHeaderData->sbr_stop_frequency,
+             &k0,
+             &k2
+              )
+    )
     return(1);
 
 
-  if(FDKsbrEnc_UpdateFreqScale(sbrConfigData->v_k_master, &sbrConfigData->num_Master,
-                     k0, k2, sbrHeaderData->freqScale,
-                     sbrHeaderData->alterScale))
+  if( FDKsbrEnc_UpdateFreqScale(
+              sbrConfigData->v_k_master,
+             &sbrConfigData->num_Master,
+              k0,
+              k2,
+              sbrHeaderData->freqScale,
+              sbrHeaderData->alterScale
+              )
+    )
     return(1);
 
 
   sbrHeaderData->sbr_xover_band=0;
 
 
-  if(FDKsbrEnc_UpdateHiRes(sbrConfigData->freqBandTable[HI],
-                           &sbrConfigData->nSfb[HI],
-                           sbrConfigData->v_k_master,
-                           sbrConfigData->num_Master ,
-                           &sbrHeaderData->sbr_xover_band,
-                           sbrHeaderData->sampleRateMode,
-                           noQmfChannels))
+  if( FDKsbrEnc_UpdateHiRes(
+              sbrConfigData->freqBandTable[HI],
+              &sbrConfigData->nSfb[HI],
+              sbrConfigData->v_k_master,
+              sbrConfigData->num_Master,
+              &sbrHeaderData->sbr_xover_band
+              )
+    )
     return(1);
 
 
-  FDKsbrEnc_UpdateLoRes(sbrConfigData->freqBandTable[LO],
-                        &sbrConfigData->nSfb[LO],
-                        sbrConfigData->freqBandTable[HI],
-                        sbrConfigData->nSfb[HI]);
+  FDKsbrEnc_UpdateLoRes(
+          sbrConfigData->freqBandTable[LO],
+          &sbrConfigData->nSfb[LO],
+          sbrConfigData->freqBandTable[HI],
+          sbrConfigData->nSfb[HI]
+          );
+
 
-  sbrConfigData->xOverFreq = (sbrConfigData->freqBandTable[LOW_RES][0] * sbrConfigData->sampleFreq / noQmfChannels+1)>>1;
+  sbrConfigData->xOverFreq = (sbrConfigData->freqBandTable[LOW_RES][0] * sbrConfigData->sampleFreq / sbrConfigData->noQmfBands+1)>>1;
 
   return (0);
 }
@@ -866,7 +934,8 @@ FDKsbrEnc_EnvEncodeFrame(HANDLE_SBR_ENCODER   hEnvEncoder,
       */
       if(updateFreqBandTable(&hSbrElement->sbrConfigData,
                              &hSbrElement->sbrHeaderData,
-                              hSbrElement->sbrConfigData.noQmfBands))
+                             hEnvEncoder->downSampleFactor
+                             ))
         return(1);
 
 
@@ -891,8 +960,6 @@ FDKsbrEnc_EnvEncodeFrame(HANDLE_SBR_ENCODER   hEnvEncoder,
                                       &crcInfo,
                                        hSbrElement->sbrConfigData.sbrSyntaxFlags);
 
-  INT error = noError;
-
   /* Temporal Envelope Data */
   SBR_FRAME_TEMP_DATA _fData;
   SBR_FRAME_TEMP_DATA *fData = &_fData;
@@ -923,9 +990,9 @@ FDKsbrEnc_EnvEncodeFrame(HANDLE_SBR_ENCODER   hEnvEncoder,
 
       if(hSbrElement->elInfo.fParametricStereo == 0)
       {
-        C_ALLOC_SCRATCH_START(qmfWorkBuffer, FIXP_DBL, QMF_CHANNELS*2);
         QMF_SCALE_FACTOR tmpScale;
         FIXP_DBL **pQmfReal, **pQmfImag;
+        C_AALLOC_SCRATCH_START(qmfWorkBuffer, FIXP_DBL, QMF_CHANNELS*2)
 
 
         /* Obtain pointers to QMF buffers. */
@@ -940,10 +1007,11 @@ FDKsbrEnc_EnvEncodeFrame(HANDLE_SBR_ENCODER   hEnvEncoder,
                                timeInStride,
                                qmfWorkBuffer );
 
-        C_ALLOC_SCRATCH_END(qmfWorkBuffer, FIXP_DBL, QMF_CHANNELS*2);
-
         h_envChan->qmfScale = tmpScale.lb_scale + 7;
 
+
+        C_AALLOC_SCRATCH_END(qmfWorkBuffer, FIXP_DBL, QMF_CHANNELS*2)
+
       } /* fParametricStereo == 0 */
 
 
@@ -952,6 +1020,8 @@ FDKsbrEnc_EnvEncodeFrame(HANDLE_SBR_ENCODER   hEnvEncoder,
       */
       if (hSbrElement->elInfo.fParametricStereo)
       {
+        INT error = noError;
+
 
         /* Limit Parametric Stereo to one instance */
         FDK_ASSERT(ch == 0);
@@ -1177,10 +1247,12 @@ initEnvChannel (  HANDLE_SBR_CONFIG_DATA sbrConfigData,
     break;
   case 2048:
   case 1024:
+  case 512:
     timeSlots = 16;
     break;
   case 1920:
   case 960:
+  case 480:
     timeSlots = 15;
     break;
   case 1152:
@@ -1221,9 +1293,9 @@ initEnvChannel (  HANDLE_SBR_CONFIG_DATA sbrConfigData,
 
   tran_fc  = params->tran_fc;
 
-  if (tran_fc == 0)
-    tran_fc = fixMin (5000, FDKsbrEnc_getSbrStartFreqRAW (sbrHeaderData->sbr_start_frequency,64,sbrConfigData->sampleFreq));
-
+  if (tran_fc == 0) {
+    tran_fc = fixMin (5000, FDKsbrEnc_getSbrStartFreqRAW (sbrHeaderData->sbr_start_frequency,params->codecSettings.sampleFreq));
+  }
 
   tran_fc = (tran_fc*4*sbrConfigData->noQmfBands/sbrConfigData->sampleFreq + 1)>>1;
 
@@ -1233,11 +1305,11 @@ initEnvChannel (  HANDLE_SBR_CONFIG_DATA sbrConfigData,
   } else
   {
     frameShift = 0;
-    switch (params->sbrFrameSize) {
+    switch (timeSlots) {
       /* The factor of 2 is by definition. */
-      case 2048: tran_off = 8 + FRAME_MIDDLE_SLOT_2048 * timeStep; break;
-      case 1920: tran_off = 7 + FRAME_MIDDLE_SLOT_1920 * timeStep; break;
-      default: return 1; break;
+      case NUMBER_TIME_SLOTS_2048: tran_off = 8 + FRAME_MIDDLE_SLOT_2048 * timeStep; break;
+      case NUMBER_TIME_SLOTS_1920: tran_off = 7 + FRAME_MIDDLE_SLOT_1920 * timeStep; break;
+      default: return 1;
     }
   }
   if ( FDKsbrEnc_InitExtractSbrEnvelope (&hEnv->sbrExtractEnvelope,
@@ -1330,7 +1402,6 @@ INT sbrEncoder_Open(
       hSbrEncoder->pSBRdynamic_RAM = (UCHAR*)GetRam_SbrDynamic_RAM();
       hSbrEncoder->dynamicRam = hSbrEncoder->pSBRdynamic_RAM;
 
-
   for (i=0; i<nElements; i++) {
     hSbrEncoder->sbrElement[i] = GetRam_SbrElement(i);
     if (hSbrEncoder->sbrElement[i]==NULL) {
@@ -1397,7 +1468,7 @@ bail:
 static
 INT FDKsbrEnc_Reallocate(
                     HANDLE_SBR_ENCODER   hSbrEncoder,
-                    SBR_ELEMENT_INFO     elInfo[(6)],
+                    SBR_ELEMENT_INFO     elInfo[(8)],
                     const INT            noElements)
 {
   INT totalCh = 0;
@@ -1462,7 +1533,9 @@ INT FDKsbrEnc_EnvInit (
                        AUDIO_OBJECT_TYPE aot,
                        int       nBitstrDelay,
                        int       nElement,
-                       ULONG     statesInitFlag
+                       const int headerPeriod,
+                       ULONG     statesInitFlag,
+                       int       fTimeDomainDownsampling
                       ,UCHAR    *dynamic_RAM
                       )
 {
@@ -1496,8 +1569,16 @@ INT FDKsbrEnc_EnvInit (
     hSbrElement->sbrConfigData.sbrSyntaxFlags |= SBR_SYNTAX_CRC;
   }
 
-  hSbrElement->sbrConfigData.noQmfBands = QMF_CHANNELS;
-  hSbrElement->sbrConfigData.noQmfSlots = params->sbrFrameSize/hSbrElement->sbrConfigData.noQmfBands;
+  hSbrElement->sbrConfigData.noQmfBands = QMF_CHANNELS>>(2-params->downSampleFactor);
+  switch (hSbrElement->sbrConfigData.noQmfBands)
+  {
+    case 64: hSbrElement->sbrConfigData.noQmfSlots = params->sbrFrameSize>>6;
+             break;
+    case 32: hSbrElement->sbrConfigData.noQmfSlots = params->sbrFrameSize>>5;
+             break;
+    default: hSbrElement->sbrConfigData.noQmfSlots = params->sbrFrameSize>>6;
+             return(2);
+  }
 
   FDKinitBitStream(&hSbrElement->CmonData.sbrBitbuf, bitstreamBuffer, MAX_PAYLOAD_SIZE*sizeof(UCHAR), 0, BS_WRITER);
 
@@ -1513,17 +1594,21 @@ INT FDKsbrEnc_EnvInit (
 
   hSbrElement->sbrConfigData.frameSize   = params->sbrFrameSize;
 
-  /* implicit rule for sampleRateMode */
-  /* run in "multirate" mode where sbr fs is 2 * codec fs */
-  hSbrElement->sbrHeaderData.sampleRateMode = DUAL_RATE;
-  hSbrElement->sbrConfigData.sampleFreq = 2 * params->codecSettings.sampleFreq;
+  hSbrElement->sbrConfigData.sampleFreq = params->downSampleFactor * params->codecSettings.sampleFreq;
 
   hSbrElement->sbrBitstreamData.CountSendHeaderData = 0;
   if (params->SendHeaderDataTime > 0 ) {
 
-    hSbrElement->sbrBitstreamData.NrSendHeaderData = (INT)(params->SendHeaderDataTime * hSbrElement->sbrConfigData.sampleFreq
+    if (headerPeriod==-1) {
+
+      hSbrElement->sbrBitstreamData.NrSendHeaderData = (INT)(params->SendHeaderDataTime * hSbrElement->sbrConfigData.sampleFreq
                                                / (1000 * hSbrElement->sbrConfigData.frameSize));
-    hSbrElement->sbrBitstreamData.NrSendHeaderData = fixMax(hSbrElement->sbrBitstreamData.NrSendHeaderData,1);
+      hSbrElement->sbrBitstreamData.NrSendHeaderData = fixMax(hSbrElement->sbrBitstreamData.NrSendHeaderData,1);
+    }
+    else {
+      /* assure header period at least once per second */
+      hSbrElement->sbrBitstreamData.NrSendHeaderData = fixMin(fixMax(headerPeriod,1),(hSbrElement->sbrConfigData.sampleFreq/hSbrElement->sbrConfigData.frameSize));
+    }
   }
   else {
    hSbrElement->sbrBitstreamData.NrSendHeaderData = 0;
@@ -1584,7 +1669,8 @@ INT FDKsbrEnc_EnvInit (
   /* init freq band table */
   if(updateFreqBandTable(&hSbrElement->sbrConfigData,
                          &hSbrElement->sbrHeaderData,
-                         hSbrElement->sbrConfigData.noQmfBands))
+                         params->downSampleFactor
+                         ))
   {
     return(1);
   }
@@ -1624,6 +1710,9 @@ INT FDKsbrEnc_EnvInit (
                                      hSbrElement->sbrConfigData.noQmfBands,
                                      hSbrElement->sbrConfigData.noQmfBands,
                                      qmfFlags );
+    if (0!=err) {
+      return err;
+    }
   }
 
   /*  */
@@ -1645,7 +1734,7 @@ INT sbrEncoder_GetInBufferSize(int noChannels)
 {
   INT temp;
 
-  temp = (1024*DOWN_SMPL_FAC);
+  temp = (2048);
   temp += 1024 + MAX_SAMPLE_DELAY;
   temp *= noChannels;
   temp *= sizeof(INT_PCM);
@@ -1677,8 +1766,8 @@ INT FDKsbrEnc_DelayCompensation (
                                      1
                                     ))
           return -1;
-          sbrEncoder_UpdateBuffers(hEnvEnc, timeBuffer);
       }
+      sbrEncoder_UpdateBuffers(hEnvEnc, timeBuffer);
     }
 	  return 0;
 }
@@ -1709,29 +1798,36 @@ UINT sbrEncoder_LimitBitRate(UINT bitRate, UINT numChannels, UINT coreSampleRate
   return newBitRate;
 }
 
+UINT sbrEncoder_IsSingleRatePossible(AUDIO_OBJECT_TYPE aot)
+{
+  UINT isPossible=(AOT_PS==aot)?0:1;
+  return isPossible;
+}
 
 INT sbrEncoder_Init(
-                     HANDLE_SBR_ENCODER hSbrEncoder,
-                     SBR_ELEMENT_INFO elInfo[(6)],
-                     int              noElements,
-                     INT_PCM *inputBuffer,
-                     INT   *coreBandwidth,
-                     INT   *inputBufferOffset,
-                     INT   *numChannels,
-                     INT   *sampleRate,
-                     INT   *frameLength,
-                     AUDIO_OBJECT_TYPE *aot,
-                     int *delay,
-                     int  transformFactor,
-                     ULONG  statesInitFlag
-                    )
+        HANDLE_SBR_ENCODER   hSbrEncoder,
+        SBR_ELEMENT_INFO     elInfo[(8)],
+        int                  noElements,
+        INT_PCM             *inputBuffer,
+        INT                 *coreBandwidth,
+        INT                 *inputBufferOffset,
+        INT                 *numChannels,
+        INT                 *coreSampleRate,
+        UINT                *downSampleFactor,
+        INT                 *frameLength,
+        AUDIO_OBJECT_TYPE    aot,
+        int                 *delay,
+        int                  transformFactor,
+        const int            headerPeriod,
+        ULONG                statesInitFlag
+        )
 {
     HANDLE_ERROR_INFO errorInfo = noError;
-    sbrConfiguration sbrConfig[(6)];
+    sbrConfiguration sbrConfig[(8)];
     INT error = 0;
     INT lowestBandwidth;
     /* Save input parameters */
-    INT inputSampleRate = *sampleRate;
+    INT inputSampleRate = *coreSampleRate;
     int coreFrameLength = *frameLength;
     int inputBandWidth = *coreBandwidth;
     int inputChannels = *numChannels;
@@ -1739,20 +1835,26 @@ INT sbrEncoder_Init(
     int downsampledOffset = 0;
     int sbrOffset = 0;
     int downsamplerDelay = 0;
-    int downsample = 0;
+    int timeDomainDownsample = 0;
     int nBitstrDelay = 0;
-    int lowestSbrStartFreq, lowestSbrStopFreq;
+    int highestSbrStartFreq, highestSbrStopFreq;
     int lowDelay = 0;
     int usePs = 0;
 
     /* check whether SBR setting is available for the current encoder configuration (bitrate, samplerate) */
-    if ( (*aot==AOT_PS) || (*aot==AOT_MP2_PS) || (*aot==AOT_DABPLUS_PS) || (*aot==AOT_DRM_MPEG_PS) ) {
+    if (!sbrEncoder_IsSingleRatePossible(aot)) {
+      *downSampleFactor = 2;
+    }
+
+
+
+    if ( (aot==AOT_PS) || (aot==AOT_MP2_PS) || (aot==AOT_DABPLUS_PS) || (aot==AOT_DRM_MPEG_PS) ) {
         usePs = 1;
     }
-    if ( (*aot==AOT_ER_AAC_ELD) ) {
+    if ( (aot==AOT_ER_AAC_ELD) ) {
         lowDelay = 1;
     }
-    else if ( (*aot==AOT_ER_AAC_LD) ) {
+    else if ( (aot==AOT_ER_AAC_LD) ) {
         error = 1;
         goto bail;
     }
@@ -1767,25 +1869,25 @@ INT sbrEncoder_Init(
         /* core encoder gets downmixed mono signal */
         *numChannels  = 1;
       } else {
-        switch (*aot) {
-          case AOT_MP2_PS:
-            *aot = AOT_MP2_SBR;
-            break;
-          case AOT_DABPLUS_PS:
-            *aot = AOT_DABPLUS_SBR;
-            break;
-          case AOT_DRM_MPEG_PS:
-            *aot = AOT_DRM_SBR;
-            break;
-          case AOT_PS:
-          default:
-            *aot = AOT_SBR;
-        }
-        usePs = 0;
+        error = 1;
+        goto bail;
       }
     } /* usePs */
 
-    /* check whether SBR setting is available for the current encoder configuration (bitrate, samplerate) */
+    /* set the core's sample rate */
+    switch (*downSampleFactor) {
+    case 1:
+      *coreSampleRate = inputSampleRate;
+      break;
+    case 2:
+      *coreSampleRate = inputSampleRate>>1;
+      break;
+    default:
+      *coreSampleRate = inputSampleRate>>1;
+      return 0; /* return error */
+    }
+
+    /* check whether SBR setting is available for the current encoder configuration (bitrate, coreSampleRate) */
     {
       int delayDiff = 0;
       int el, coreEl;
@@ -1798,54 +1900,37 @@ INT sbrEncoder_Init(
           continue;
         }
         /* check if desired configuration is available */
-        if ( !FDKsbrEnc_IsSbrSettingAvail (elInfo[coreEl].bitRate, 0, elInfo[coreEl].nChannelsInEl, inputSampleRate, *aot) )
+        if ( !FDKsbrEnc_IsSbrSettingAvail (elInfo[coreEl].bitRate, 0, elInfo[coreEl].nChannelsInEl, inputSampleRate, *coreSampleRate, aot) )
         {
-            /* otherwise - change to AAC-LC */
-            switch (*aot) {
-              case AOT_MP2_SBR:
-              case AOT_MP2_PS:
-                *aot = AOT_MP2_AAC_LC;
-                break;
-              case AOT_DABPLUS_SBR:
-              case AOT_DABPLUS_PS:
-                *aot = AOT_DABPLUS_AAC_LC;
-                break;
-              case AOT_DRM_SBR:
-              case AOT_DRM_MPEG_PS:
-                *aot = AOT_DRM_AAC;
-                break;
-              case AOT_ER_AAC_ELD:
-                break;
-              case AOT_SBR:
-              case AOT_PS:
-              default:
-                *aot = AOT_AAC_LC;
-            }
             error = 1;
             goto bail;
         }
       }
 
-      *sampleRate /= DOWN_SMPL_FAC;
-
       /* Determine Delay balancing and new encoder delay */
       if (lowDelay) {
-        downsample = 1;  /* activate downsampler */
-        delayDiff = (*delay*DOWN_SMPL_FAC) + DELAY_ELD2SBR(coreFrameLength);
-        *delay = DELAY_ELDSBR(coreFrameLength);
+        {
+        delayDiff = (*delay * *downSampleFactor) + DELAY_ELD2SBR(coreFrameLength,*downSampleFactor);
+        *delay = DELAY_ELDSBR(coreFrameLength,*downSampleFactor);
+        }
       }
       else if (usePs) {
-        delayDiff = (*delay*DOWN_SMPL_FAC) + DELAY_AAC2PS(coreFrameLength);
-        *delay = DELAY_PS(coreFrameLength);
+        delayDiff = (*delay * *downSampleFactor) + DELAY_AAC2PS(coreFrameLength,*downSampleFactor);
+        *delay = DELAY_PS(coreFrameLength,*downSampleFactor);
       }
       else {
-        downsample = 1;  /* activate downsampler */
-        delayDiff = (*delay*DOWN_SMPL_FAC) + DELAY_AAC2SBR(coreFrameLength);
-        *delay = DELAY_SBR(coreFrameLength);
+        delayDiff = DELAY_AAC2SBR(coreFrameLength,*downSampleFactor);
+        delayDiff += (*delay * *downSampleFactor);
+        *delay = DELAY_SBR(coreFrameLength,*downSampleFactor);
       }
 
+      if (!usePs) {
+          timeDomainDownsample = *downSampleFactor-1;  /* activate time domain downsampler when downSampleFactor is != 1 */
+      }
+
+
       /* Take care about downsampled data bound to the SBR path */
-      if (!downsample && delayDiff > 0) {
+      if (!timeDomainDownsample && delayDiff > 0) {
         /*
          * We must tweak the balancing into a situation where the downsampled path
          * is the one to be delayed, because delaying the QMF domain input, also delays
@@ -1854,12 +1939,15 @@ INT sbrEncoder_Init(
         while ( delayDiff > 0 )
         {
           /* Encoder delay increases */
-          *delay += coreFrameLength*DOWN_SMPL_FAC;
-          /* Add one frame delay to SBR path */
-          delayDiff -= coreFrameLength*DOWN_SMPL_FAC;
+          {
+            *delay += coreFrameLength * *downSampleFactor;
+            /* Add one frame delay to SBR path */
+            delayDiff -= coreFrameLength * *downSampleFactor;
+          }
           nBitstrDelay += 1;
         }
-      } else {
+      } else
+      {
         *delay += fixp_abs(delayDiff);
       }
 
@@ -1867,32 +1955,33 @@ INT sbrEncoder_Init(
         /* Delay AAC data */
         delayDiff = -delayDiff;
         /* Multiply downsampled offset by AAC core channels. Divide by 2 because of half samplerate of downsampled data. */
-        downsampledOffset = (delayDiff*(*numChannels))/DOWN_SMPL_FAC;
+        FDK_ASSERT(*downSampleFactor>0 && *downSampleFactor<=2);
+        downsampledOffset = (delayDiff*(*numChannels))>>(*downSampleFactor-1);
         sbrOffset = 0;
       } else {
         /* Delay SBR input */
-        if ( delayDiff > (int)coreFrameLength*DOWN_SMPL_FAC )
+        if ( delayDiff > (int)coreFrameLength * (int)*downSampleFactor )
         {
           /* Do bitstream frame-wise delay balancing if we have more than SBR framelength samples delay difference */
-          delayDiff -= coreFrameLength*DOWN_SMPL_FAC;
+          delayDiff -= coreFrameLength * *downSampleFactor;
           nBitstrDelay = 1;
         }
         /* Multiply input offset by input channels */
         sbrOffset = delayDiff*(*numChannels);
         downsampledOffset = 0;
       }
-
-      hSbrEncoder->nBitstrDelay = nBitstrDelay;
-      hSbrEncoder->nChannels = *numChannels;
-      hSbrEncoder->frameSize = *frameLength*DOWN_SMPL_FAC;
-      hSbrEncoder->fTimeDomainDownsampling = downsample;
-      hSbrEncoder->estimateBitrate = 0;
-      hSbrEncoder->inputDataDelay = 0;
+      hSbrEncoder->nBitstrDelay            = nBitstrDelay;
+      hSbrEncoder->nChannels               = *numChannels;
+      hSbrEncoder->frameSize               = coreFrameLength * *downSampleFactor;
+      hSbrEncoder->fTimeDomainDownsampling = timeDomainDownsample;
+      hSbrEncoder->downSampleFactor        = *downSampleFactor;
+      hSbrEncoder->estimateBitrate         = 0;
+      hSbrEncoder->inputDataDelay          = 0;
 
 
       /* Open SBR elements */
       el = -1;
-      lowestSbrStartFreq = lowestSbrStopFreq = 9999;
+      highestSbrStartFreq = highestSbrStopFreq = 0;
       lowestBandwidth = 99999;
 
       /* Loop through each core encoder element and get a matching SBR element config */
@@ -1915,28 +2004,38 @@ INT sbrEncoder_Init(
         /*
          * Init sbrConfig structure
          */
-        FDKsbrEnc_InitializeSbrDefaults ( &sbrConfig[el],
-                                           DOWN_SMPL_FAC,
-                                           coreFrameLength);
+        if ( ! FDKsbrEnc_InitializeSbrDefaults ( &sbrConfig[el],
+                                                 *downSampleFactor,
+                                                  coreFrameLength
+                                                  ) )
+        {
+          error = 1;
+          goto bail;
+        }
+
         /*
          * Modify sbrConfig structure according to Element parameters
          */
-        FDKsbrEnc_AdjustSbrSettings ( &sbrConfig[el],
-                                       elInfo[coreEl].bitRate,
-                                       elInfo[coreEl].nChannelsInEl,
-                                      *sampleRate,
-                                       transformFactor,
-                                       24000,
-                                       0,
-                                       0,                                            /* useSpeechConfig */
-                                       0,                                            /* lcsMode */
-                                       usePs, /* bParametricStereo */
-                                       *aot);
+        if ( ! FDKsbrEnc_AdjustSbrSettings (&sbrConfig[el],
+                                             elInfo[coreEl].bitRate,
+                                             elInfo[coreEl].nChannelsInEl,
+                                            *coreSampleRate,
+                                             inputSampleRate,
+                                             transformFactor,
+                                             24000,
+                                             0,
+                                             0,     /* useSpeechConfig */
+                                             0,     /* lcsMode */
+                                             usePs, /* bParametricStereo */
+                                             aot) )
+        {
+          error = 1;
+          goto bail;
+        }
 
         /* Find common frequency border for all SBR elements */
-        lowestSbrStartFreq = fixMin(lowestSbrStartFreq, sbrConfig[el].startFreq);
-        lowestSbrStopFreq = fixMin(lowestSbrStopFreq, sbrConfig[el].stopFreq);
-
+        highestSbrStartFreq = fixMax(highestSbrStartFreq, sbrConfig[el].startFreq);
+        highestSbrStopFreq = fixMax(highestSbrStopFreq, sbrConfig[el].stopFreq);
 
       } /* first element loop */
 
@@ -1952,21 +2051,24 @@ INT sbrEncoder_Init(
         int bandwidth = *coreBandwidth;
 
         /* Use lowest common bandwidth */
-        sbrConfig[el].startFreq = lowestSbrStartFreq;
-        sbrConfig[el].stopFreq = lowestSbrStopFreq;
+        sbrConfig[el].startFreq = highestSbrStartFreq;
+        sbrConfig[el].stopFreq = highestSbrStopFreq;
 
         /* initialize SBR element, and get core bandwidth */
         error = FDKsbrEnc_EnvInit(hSbrEncoder->sbrElement[el],
                                   &sbrConfig[el],
                                   &bandwidth,
-                                  *aot,
+                                   aot,
                                    nBitstrDelay,
                                    el,
-                                   statesInitFlag
+                                   headerPeriod,
+                                   statesInitFlag,
+                                   hSbrEncoder->fTimeDomainDownsampling
                                   ,hSbrEncoder->dynamicRam
                                   );
 
         if (error != 0) {
+          error = 2;
           goto bail;
         }
 
@@ -1988,30 +2090,29 @@ INT sbrEncoder_Init(
 
           for (ch=0; ch<hSbrEl->elInfo.nChannelsInEl; ch++)
           {
-            FDKaacEnc_InitDownsampler (&hSbrEl->sbrChannel[ch]->downSampler, Wc, DOWN_SMPL_FAC);
+            FDKaacEnc_InitDownsampler (&hSbrEl->sbrChannel[ch]->downSampler, Wc, *downSampleFactor);
+            FDK_ASSERT (hSbrEl->sbrChannel[ch]->downSampler.delay <=MAX_DS_FILTER_DELAY);
           }
 
-          FDK_ASSERT (hSbrEl->sbrChannel[0]->downSampler.delay <=MAX_DS_FILTER_DELAY && hSbrEl->sbrChannel[0]->downSampler.delay <=MAX_DS_FILTER_DELAY);
           downsamplerDelay = hSbrEl->sbrChannel[0]->downSampler.delay;
         } /* third element loop */
 
         /* lfe */
-        FDKaacEnc_InitDownsampler (&hSbrEncoder->lfeDownSampler, 0, DOWN_SMPL_FAC);
+        FDKaacEnc_InitDownsampler (&hSbrEncoder->lfeDownSampler, 0, *downSampleFactor);
 
         /* Add the resampler additional delay to get the final delay and buffer offset values. */
-        if (sbrOffset > 0 || downsampledOffset <= ((downsamplerDelay * (*numChannels))/DOWN_SMPL_FAC)) {
+        if (sbrOffset > 0 || downsampledOffset <= ((downsamplerDelay * (*numChannels))>>(*downSampleFactor-1))) {
           sbrOffset += (downsamplerDelay - downsampledOffset) * (*numChannels) ;
           *delay += downsamplerDelay - downsampledOffset;
           downsampledOffset = 0;
         } else {
-          downsampledOffset -= (downsamplerDelay * (*numChannels))/DOWN_SMPL_FAC;
+          downsampledOffset -= (downsamplerDelay * (*numChannels))>>(*downSampleFactor-1);
           sbrOffset = 0;
         }
 
         hSbrEncoder->inputDataDelay = downsamplerDelay;
       }
 
-
       /* Assign core encoder Bandwidth */
       *coreBandwidth = lowestBandwidth;
 
@@ -2025,7 +2126,7 @@ INT sbrEncoder_Init(
         FDK_ASSERT(hSbrEncoder->noElements == 1);
         INT psTuningTableIdx = getPsTuningTableIndex(elInfo[0].bitRate, NULL);
 
-        psEncConfig.frameSize           = *frameLength; //sbrConfig.sbrFrameSize;
+        psEncConfig.frameSize           = coreFrameLength; //sbrConfig.sbrFrameSize;
         psEncConfig.qmfFilterMode       = 0;
         psEncConfig.sbrPsDelay          = 0;
 
@@ -2037,7 +2138,7 @@ INT sbrEncoder_Init(
 
           /* calculation is not quite linear, increased number of envelopes causes more bits */
           /* assume avg. 50 bits per frame for 10 stereo bands / 1 envelope configuration */
-          hSbrEncoder->estimateBitrate += ( (((*sampleRate) * 5 * psEncConfig.nStereoBands * psEncConfig.maxEnvelopes) / hSbrEncoder->frameSize));
+          hSbrEncoder->estimateBitrate += ( (((*coreSampleRate) * 5 * psEncConfig.nStereoBands * psEncConfig.maxEnvelopes) / hSbrEncoder->frameSize));
 
         } else {
           error = ERROR(CDI, "Invalid ps tuning table index.");
@@ -2066,10 +2167,16 @@ INT sbrEncoder_Init(
             errorInfo = handBack(errorInfo);
           }
         }
+
+        /* QMF analysis + Hybrid analysis + Hybrid synthesis + QMF synthesis + downsampled input buffer delay */
+        hSbrEncoder->inputDataDelay = (64*10/2) + (6*64) +  (0) + (64*10/2-64+1) + ((*downSampleFactor)*downsampledOffset);
       }
 
       hSbrEncoder->downsampledOffset = downsampledOffset;
-      hSbrEncoder->downmixSize = coreFrameLength*(*numChannels);
+      {
+        hSbrEncoder->downmixSize = coreFrameLength*(*numChannels);
+      }
+
       hSbrEncoder->bufferOffset = sbrOffset;
       /* Delay Compensation: fill bitstream delay buffer with zero input signal */
       if ( hSbrEncoder->nBitstrDelay > 0 )
@@ -2080,7 +2187,7 @@ INT sbrEncoder_Init(
       }
 
       /* Set Output frame length */
-      *frameLength = coreFrameLength*DOWN_SMPL_FAC;
+      *frameLength = coreFrameLength * *downSampleFactor;
       /* Input buffer offset */
       *inputBufferOffset = fixMax(sbrOffset, downsampledOffset);
 
@@ -2091,7 +2198,7 @@ INT sbrEncoder_Init(
 
 bail:
     /* Restore input settings */
-    *sampleRate = inputSampleRate;
+    *coreSampleRate = inputSampleRate;
     *frameLength = coreFrameLength;
     *numChannels = inputChannels;
     *coreBandwidth = inputBandWidth;
@@ -2104,8 +2211,8 @@ INT
 sbrEncoder_EncodeFrame(  HANDLE_SBR_ENCODER   hSbrEncoder,
                          INT_PCM             *samples,
                          UINT                 timeInStride,
-                         UINT                 sbrDataBits[(6)],
-                         UCHAR                sbrData[(6)][MAX_PAYLOAD_SIZE]
+                         UINT                 sbrDataBits[(8)],
+                         UCHAR                sbrData[(8)][MAX_PAYLOAD_SIZE]
                         )
 {
   INT error;
@@ -2129,8 +2236,8 @@ sbrEncoder_EncodeFrame(  HANDLE_SBR_ENCODER   hSbrEncoder,
     }
   }
 
-  if ( (hSbrEncoder->lfeChIdx!=-1) && (hSbrEncoder->fTimeDomainDownsampling) )
-  {
+  if ( ( hSbrEncoder->lfeChIdx!=-1) && (hSbrEncoder->downSampleFactor > 1) )
+  {   /* lfe downsampler */
       INT nOutSamples;
 
       FDKaacEnc_Downsample(&hSbrEncoder->lfeDownSampler,
@@ -2140,7 +2247,9 @@ sbrEncoder_EncodeFrame(  HANDLE_SBR_ENCODER   hSbrEncoder,
                             samples + hSbrEncoder->downsampledOffset + hSbrEncoder->lfeChIdx,
                            &nOutSamples,
                             hSbrEncoder->nChannels);
-  } /* lfe downsampler */
+
+
+  }
 
   return 0;
 }
diff --git a/libSBRenc/src/sbr_misc.cpp b/libSBRenc/src/sbr_misc.cpp
index c44be22..c673b81 100644
--- a/libSBRenc/src/sbr_misc.cpp
+++ b/libSBRenc/src/sbr_misc.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRenc/src/sbr_misc.h b/libSBRenc/src/sbr_misc.h
index 33b9cf9..f471974 100644
--- a/libSBRenc/src/sbr_misc.h
+++ b/libSBRenc/src/sbr_misc.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRenc/src/sbr_ram.cpp b/libSBRenc/src/sbr_ram.cpp
index e304c39..ee6c37f 100644
--- a/libSBRenc/src/sbr_ram.cpp
+++ b/libSBRenc/src/sbr_ram.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -107,39 +107,39 @@ C_ALLOC_MEM (Ram_SbrDynamic_RAM, FIXP_DBL, ((SBR_ENC_DYN_RAM_SIZE)/sizeof(FIXP_D
   in module sbr_ram and sbr rom
 */
 C_ALLOC_MEM (Ram_SbrEncoder, SBR_ENCODER, 1)
-C_ALLOC_MEM2(Ram_SbrChannel, SBR_CHANNEL, 1, (6))
-C_ALLOC_MEM2(Ram_SbrElement, SBR_ELEMENT, 1, (6))
+C_ALLOC_MEM2(Ram_SbrChannel, SBR_CHANNEL, 1, (8))
+C_ALLOC_MEM2(Ram_SbrElement, SBR_ELEMENT, 1, (8))
 
 /*! Filter states for QMF-analysis. <br>
   Dimension: #MAXNRSBRCHANNELS * #SBR_QMF_FILTER_LENGTH
 */
-C_AALLOC_MEM2_L (Ram_Sbr_QmfStatesAnalysis, FIXP_QAS, QMF_FILTER_LENGTH, (6), SECT_DATA_L1)
+C_AALLOC_MEM2_L (Ram_Sbr_QmfStatesAnalysis, FIXP_QAS, QMF_FILTER_LENGTH, (8), SECT_DATA_L1)
 
 
 /*! Matrix holding the quota values for all estimates, all channels
   Dimension #MAXNRSBRCHANNELS * +#SBR_QMF_CHANNELS* #MAX_NO_OF_ESTIMATES
 */
-C_ALLOC_MEM2_L (Ram_Sbr_quotaMatrix, FIXP_DBL, (MAX_NO_OF_ESTIMATES*QMF_CHANNELS), (6), SECT_DATA_L1)
+C_ALLOC_MEM2_L (Ram_Sbr_quotaMatrix, FIXP_DBL, (MAX_NO_OF_ESTIMATES*QMF_CHANNELS), (8), SECT_DATA_L1)
 
 /*! Matrix holding the sign values for all estimates, all channels
   Dimension #MAXNRSBRCHANNELS * +#SBR_QMF_CHANNELS* #MAX_NO_OF_ESTIMATES
 */
-C_ALLOC_MEM2 (Ram_Sbr_signMatrix, INT, (MAX_NO_OF_ESTIMATES*QMF_CHANNELS), (6))
+C_ALLOC_MEM2 (Ram_Sbr_signMatrix, INT, (MAX_NO_OF_ESTIMATES*QMF_CHANNELS), (8))
 
 /*! Frequency band table (low res) <br>
   Dimension #MAX_FREQ_COEFFS/2+1
 */
-C_ALLOC_MEM2 (Ram_Sbr_freqBandTableLO, UCHAR, (MAX_FREQ_COEFFS/2+1), (6))
+C_ALLOC_MEM2 (Ram_Sbr_freqBandTableLO, UCHAR, (MAX_FREQ_COEFFS/2+1), (8))
 
 /*! Frequency band table (high res) <br>
   Dimension #MAX_FREQ_COEFFS +1
 */
-C_ALLOC_MEM2 (Ram_Sbr_freqBandTableHI, UCHAR, (MAX_FREQ_COEFFS+1), (6))
+C_ALLOC_MEM2 (Ram_Sbr_freqBandTableHI, UCHAR, (MAX_FREQ_COEFFS+1), (8))
 
 /*! vk matser table <br>
   Dimension #MAX_FREQ_COEFFS +1
 */
-C_ALLOC_MEM2 (Ram_Sbr_v_k_master, UCHAR, (MAX_FREQ_COEFFS+1), (6))
+C_ALLOC_MEM2 (Ram_Sbr_v_k_master, UCHAR, (MAX_FREQ_COEFFS+1), (8))
 
 
 /*
@@ -149,23 +149,23 @@ C_ALLOC_MEM2 (Ram_Sbr_v_k_master, UCHAR, (MAX_FREQ_COEFFS+1), (6))
 /*! sbr_detectionVectors <br>
   Dimension #MAX_NUM_CHANNELS*#MAX_NO_OF_ESTIMATES*#MAX_FREQ_COEFFS]
 */
-C_ALLOC_MEM2 (Ram_Sbr_detectionVectors, UCHAR, (MAX_NO_OF_ESTIMATES*MAX_FREQ_COEFFS), (6))
+C_ALLOC_MEM2 (Ram_Sbr_detectionVectors, UCHAR, (MAX_NO_OF_ESTIMATES*MAX_FREQ_COEFFS), (8))
 
 /*! sbr_prevCompVec[ <br>
   Dimension #MAX_NUM_CHANNELS*#MAX_FREQ_COEFFS]
 */
-C_ALLOC_MEM2 (Ram_Sbr_prevEnvelopeCompensation, UCHAR, MAX_FREQ_COEFFS, (6))
+C_ALLOC_MEM2 (Ram_Sbr_prevEnvelopeCompensation, UCHAR, MAX_FREQ_COEFFS, (8))
 /*! sbr_guideScfb[ <br>
   Dimension #MAX_NUM_CHANNELS*#MAX_FREQ_COEFFS]
 */
-C_ALLOC_MEM2 (Ram_Sbr_guideScfb, UCHAR, MAX_FREQ_COEFFS, (6))
+C_ALLOC_MEM2 (Ram_Sbr_guideScfb, UCHAR, MAX_FREQ_COEFFS, (8))
 
 /*! sbr_guideVectorDetected <br>
   Dimension #MAX_NUM_CHANNELS*#MAX_NO_OF_ESTIMATES*#MAX_FREQ_COEFFS]
 */
-C_ALLOC_MEM2 (Ram_Sbr_guideVectorDetected, UCHAR, (MAX_NO_OF_ESTIMATES*MAX_FREQ_COEFFS), (6))
-C_ALLOC_MEM2 (Ram_Sbr_guideVectorDiff, FIXP_DBL, (MAX_NO_OF_ESTIMATES*MAX_FREQ_COEFFS), (6))
-C_ALLOC_MEM2 (Ram_Sbr_guideVectorOrig, FIXP_DBL, (MAX_NO_OF_ESTIMATES*MAX_FREQ_COEFFS), (6))
+C_ALLOC_MEM2 (Ram_Sbr_guideVectorDetected, UCHAR, (MAX_NO_OF_ESTIMATES*MAX_FREQ_COEFFS), (8))
+C_ALLOC_MEM2 (Ram_Sbr_guideVectorDiff, FIXP_DBL, (MAX_NO_OF_ESTIMATES*MAX_FREQ_COEFFS), (8))
+C_ALLOC_MEM2 (Ram_Sbr_guideVectorOrig, FIXP_DBL, (MAX_NO_OF_ESTIMATES*MAX_FREQ_COEFFS), (8))
 
 /*
   Static Parametric Stereo memory
@@ -191,7 +191,7 @@ C_ALLOC_MEM   (Ram_ParamStereo, PARAMETRIC_STEREO, 1)
   /*! Energy buffer for envelope extraction <br>
     Dimension #MAXNRSBRCHANNELS * +#SBR_QMF_SLOTS *  #SBR_QMF_CHANNELS
   */
-  C_ALLOC_MEM2 (Ram_Sbr_envYBuffer, FIXP_DBL, (QMF_MAX_TIME_SLOTS/2 * QMF_CHANNELS), (6))
+  C_ALLOC_MEM2 (Ram_Sbr_envYBuffer, FIXP_DBL, (QMF_MAX_TIME_SLOTS/2 * QMF_CHANNELS), (8))
 
   FIXP_DBL* GetRam_Sbr_envYBuffer (int n, UCHAR* dynamic_RAM) {
     FDK_ASSERT(dynamic_RAM!=0);
diff --git a/libSBRenc/src/sbr_ram.h b/libSBRenc/src/sbr_ram.h
index 0f9e9cc..7e3d0c8 100644
--- a/libSBRenc/src/sbr_ram.h
+++ b/libSBRenc/src/sbr_ram.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRenc/src/sbr_rom.cpp b/libSBRenc/src/sbr_rom.cpp
index c8b945f..a2b6527 100644
--- a/libSBRenc/src/sbr_rom.cpp
+++ b/libSBRenc/src/sbr_rom.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -506,216 +506,277 @@ const UCHAR bookSbrNoiseBalanceL11T[25] =
 /*
    tuningTable
 */
-const sbrTuningTable_t sbrTuningTable[SBRENC_TUNING_SIZE] =
+const sbrTuningTable_t sbrTuningTable[] =
 {
+  /* Some of the low bitrates are commented out here, this is because the
+     encoder could lose frames at those bitrates and throw an error because
+     it has insufficient bits to encode for some test items.
+  */
 
-  /*** AAC ***/
+  /*** HE-AAC section ***/
   /*                        sf,sfsp,sf,sfsp,nnb,nfo,saml,SM,FS*/
 
   /*** mono ***/
 
   /* 8/16 kHz dual rate */
-  { 8000,  10000,   8000, 1,  7, 6, 11,10, 1, 0, 6, SBR_MONO, 3 },
-  { 10000, 12000,   8000, 1, 11, 7, 13,12, 1, 0, 6, SBR_MONO, 3 },
-  { 12000, 16001,   8000, 1, 14,10, 13,13, 1, 0, 6, SBR_MONO, 3 },
-  { 16000, 24000,   8000, 1, 14,10, 14,14, 2, 0, 3, SBR_MONO, 2 }, /* placebo */
-  { 24000, 32000,   8000, 1, 14,10, 14,14, 2, 0, 3, SBR_MONO, 2 }, /* placebo */
-  { 32000, 48001,   8000, 1, 14,11, 15,15, 2, 0, 3, SBR_MONO, 2 }, /* placebo */ /* bitrates higher than 48000 not supported by AAC core */
+  { CODEC_AAC,   8000, 10000,   8000, 1,  7, 6, 11,10, 1, 0, 6, SBR_MONO, 3 },
+  { CODEC_AAC,  10000, 12000,   8000, 1, 11, 7, 13,12, 1, 0, 6, SBR_MONO, 3 },
+  { CODEC_AAC,  12000, 16001,   8000, 1, 14,10, 13,13, 1, 0, 6, SBR_MONO, 3 },
+  { CODEC_AAC,  16000, 24000,   8000, 1, 14,10, 14,14, 2, 0, 3, SBR_MONO, 2 }, /* placebo */
+  { CODEC_AAC,  24000, 32000,   8000, 1, 14,10, 14,14, 2, 0, 3, SBR_MONO, 2 }, /* placebo */
+  { CODEC_AAC,  32000, 48001,   8000, 1, 14,11, 15,15, 2, 0, 3, SBR_MONO, 2 }, /* placebo */ /* bitrates higher than 48000 not supported by AAC core */
 
   /* 11/22 kHz dual rate */
-  {  8000, 10000,  11025, 1,  5, 4,  6, 6, 1, 0, 6, SBR_MONO, 3 },
-  { 10000, 12000,  11025, 1,  8, 5, 12, 9, 1, 0, 6, SBR_MONO, 3 },
-  { 12000, 16000,  11025, 1, 12, 8, 13, 8, 1, 0, 6, SBR_MONO, 3 },
-  { 16000, 20000,  11025, 1, 12, 8, 13, 8, 1, 0, 6, SBR_MONO, 3 }, /* at such "high" bitrates it's better to upsample the input */
-  { 20000, 24001,  11025, 1, 13, 9, 13, 8, 1, 0, 6, SBR_MONO, 3 }, /* signal by a factor of 2 before sending it into the encoder */
-  { 24000, 32000,  11025, 1, 14,10, 14, 9, 2, 0, 3, SBR_MONO, 2 }, /* placebo */
-  { 32000, 48000,  11025, 1, 15,11, 15,10, 2, 0, 3, SBR_MONO, 2 }, /* placebo */
-  { 48000, 64001,  11025, 1, 15,11, 15,10, 2, 0, 3, SBR_MONO, 1 }, /* placebo */
+  { CODEC_AAC,   8000, 10000,  11025, 1,  5, 4,  6, 6, 1, 0, 6, SBR_MONO, 3 },
+  { CODEC_AAC,  10000, 12000,  11025, 1,  8, 5, 12, 9, 1, 0, 6, SBR_MONO, 3 },
+  { CODEC_AAC,  12000, 16000,  11025, 1, 12, 8, 13, 8, 1, 0, 6, SBR_MONO, 3 },
+  { CODEC_AAC,  16000, 20000,  11025, 1, 12, 8, 13, 8, 1, 0, 6, SBR_MONO, 3 }, /* at such "high" bitrates it's better to upsample the input */
+  { CODEC_AAC,  20000, 24001,  11025, 1, 13, 9, 13, 8, 1, 0, 6, SBR_MONO, 3 }, /* signal by a factor of 2 before sending it into the encoder */
+  { CODEC_AAC,  24000, 32000,  11025, 1, 14,10, 14, 9, 2, 0, 3, SBR_MONO, 2 }, /* placebo */
+  { CODEC_AAC,  32000, 48000,  11025, 1, 15,11, 15,10, 2, 0, 3, SBR_MONO, 2 }, /* placebo */
+  { CODEC_AAC,  48000, 64001,  11025, 1, 15,11, 15,10, 2, 0, 3, SBR_MONO, 1 }, /* placebo */
 
   /* 12/24 kHz dual rate */
-  {  8000, 10000,  12000, 1,  4, 3,  6, 6, 1, 0, 6, SBR_MONO, 3 }, /* nominal:  8 kbit/s */
-  { 10000, 12000,  12000, 1,  7, 4, 11, 8, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 10 kbit/s */
-  { 12000, 16000,  12000, 1, 11, 7, 12, 8, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 12 kbit/s */
-  { 16000, 20000,  12000, 1, 11, 7, 12, 8, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 16 kbit/s */ /* at such "high" bitrates it's better to upsample the input */
-  { 20000, 24001,  12000, 1, 12, 8, 12, 8, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 20 kbit/s */ /* signal by a factor of 2 before sending it into the encoder */
-  { 24000, 32000,  12000, 1, 13, 9, 13, 9, 2, 0, 3, SBR_MONO, 2 }, /* placebo */
-  { 32000, 48000,  12000, 1, 14,10, 14,10, 2, 0, 3, SBR_MONO, 2 }, /* placebo */
-  { 48000, 64001,  12000, 1, 15,11, 15,11, 2, 0, 3, SBR_MONO, 1 }, /* placebo */
+  { CODEC_AAC,   8000, 10000,  12000, 1,  4, 3,  6, 6, 1, 0, 6, SBR_MONO, 3 }, /* nominal:  8 kbit/s */
+  { CODEC_AAC,  10000, 12000,  12000, 1,  7, 4, 11, 8, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 10 kbit/s */
+  { CODEC_AAC,  12000, 16000,  12000, 1, 11, 7, 12, 8, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 12 kbit/s */
+  { CODEC_AAC,  16000, 20000,  12000, 1, 11, 7, 12, 8, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 16 kbit/s */ /* at such "high" bitrates it's better to upsample the input */
+  { CODEC_AAC,  20000, 24001,  12000, 1, 12, 8, 12, 8, 1, 0, 6, SBR_MONO, 3 }, /* nominal: 20 kbit/s */ /* signal by a factor of 2 before sending it into the encoder */
+  { CODEC_AAC,  24000, 32000,  12000, 1, 13, 9, 13, 9, 2, 0, 3, SBR_MONO, 2 }, /* placebo */
+  { CODEC_AAC,  32000, 48000,  12000, 1, 14,10, 14,10, 2, 0, 3, SBR_MONO, 2 }, /* placebo */
+  { CODEC_AAC,  48000, 64001,  12000, 1, 14,11, 15,11, 2, 0, 3, SBR_MONO, 1 }, /* placebo */
 
   /* 16/32 kHz dual rate */
-  {  8000, 10000,  16000, 1,  1, 1, 0, 0,  1, 0, 6, SBR_MONO, 3 }, /* nominal:  8 kbit/s */
-  { 10000, 12000,  16000, 1,  2, 1, 6, 0,  1, 0, 6, SBR_MONO, 3 }, /* nominal: 10 kbit/s */
-  { 12000, 16000,  16000, 1,  4, 2, 6, 0,  1, 0, 6, SBR_MONO, 3 }, /* nominal: 12 kbit/s */
-  { 16000, 18000,  16000, 1,  4, 2, 8, 3,  1, 0, 6, SBR_MONO, 3 }, /* nominal: 16 kbit/s */
-  { 18000, 22000,  16000, 1,  6, 5,11, 7,  2, 0, 6, SBR_MONO, 2 }, /* nominal: 20 kbit/s */
-  { 22000, 28000,  16000, 1, 10, 9,12, 8,  2, 0, 6, SBR_MONO, 2 }, /* nominal: 24 kbit/s */
-  { 28000, 36000,  16000, 1, 12,12,13,13,  2, 0, 3, SBR_MONO, 2 }, /* nominal: 32 kbit/s */
-  { 36000, 44000,  16000, 1, 14,14,13,13,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 40 kbit/s */
-  { 44000, 64001,  16000, 1, 15,15,13,13,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 48 kbit/s */
+  { CODEC_AAC,   8000, 10000,  16000, 1,  1, 1, 0, 0,  1, 0, 6, SBR_MONO, 3 }, /* nominal:  8 kbit/s */
+  { CODEC_AAC,  10000, 12000,  16000, 1,  2, 1, 6, 0,  1, 0, 6, SBR_MONO, 3 }, /* nominal: 10 kbit/s */
+  { CODEC_AAC,  12000, 16000,  16000, 1,  4, 2, 6, 0,  1, 0, 6, SBR_MONO, 3 }, /* nominal: 12 kbit/s */
+  { CODEC_AAC,  16000, 18000,  16000, 1,  4, 2, 8, 3,  1, 0, 6, SBR_MONO, 3 }, /* nominal: 16 kbit/s */
+  { CODEC_AAC,  18000, 22000,  16000, 1,  6, 5,11, 7,  2, 0, 6, SBR_MONO, 2 }, /* nominal: 20 kbit/s */
+  { CODEC_AAC,  22000, 28000,  16000, 1, 10, 9,12, 8,  2, 0, 6, SBR_MONO, 2 }, /* nominal: 24 kbit/s */
+  { CODEC_AAC,  28000, 36000,  16000, 1, 12,12,13,13,  2, 0, 3, SBR_MONO, 2 }, /* nominal: 32 kbit/s */
+  { CODEC_AAC,  36000, 44000,  16000, 1, 14,14,13,13,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 40 kbit/s */
+  { CODEC_AAC,  44000, 64001,  16000, 1, 14,14,13,13,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 48 kbit/s */
 
   /* 22.05/44.1 kHz dual rate */
-  /* {  8000, 11369,  22050, 1,  1, 1, 1, 1,  1, 0, 6, SBR_MONO, 3 }, */ /* nominal:  8 kbit/s */   /* encoder can not work stable at this extremely low bitrate */
-  { 11369, 16000,  22050, 1,  3, 1, 4, 4,  1, 0, 6, SBR_MONO, 3 }, /* nominal: 12 kbit/s */
-  { 16000, 18000,  22050, 1,  3, 1, 5, 4,  1, 0, 6, SBR_MONO, 3 }, /* nominal: 16 kbit/s */
-  { 18000, 22000,  22050, 1,  4, 4, 8, 5,  2, 0, 6, SBR_MONO, 2 }, /* nominal: 20 kbit/s */
-  { 22000, 28000,  22050, 1,  7, 6, 8, 6,  2, 0, 6, SBR_MONO, 2 }, /* nominal: 24 kbit/s */
-  { 28000, 36000,  22050, 1, 10,10, 9, 9,  2, 0, 3, SBR_MONO, 2 }, /* nominal: 32 kbit/s */
-  { 36000, 44000,  22050, 1, 11,11,10,10,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 40 kbit/s */
-  { 44000, 64001,  22050, 1, 13,13,12,12,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 48 kbit/s */
+  /* { CODEC_AAC,   8000, 11369,  22050, 1,  1, 1, 1, 1,  1, 0, 6, SBR_MONO, 3 }, */ /* nominal:  8 kbit/s */   /* encoder can not work stable at this extremely low bitrate */
+  { CODEC_AAC,  11369, 16000,  22050, 1,  3, 1, 4, 4,  1, 0, 6, SBR_MONO, 3 }, /* nominal: 12 kbit/s */
+  { CODEC_AAC,  16000, 18000,  22050, 1,  3, 1, 5, 4,  1, 0, 6, SBR_MONO, 3 }, /* nominal: 16 kbit/s */
+  { CODEC_AAC,  18000, 22000,  22050, 1,  4, 4, 8, 5,  2, 0, 6, SBR_MONO, 2 }, /* nominal: 20 kbit/s */
+  { CODEC_AAC,  22000, 28000,  22050, 1,  7, 6, 8, 6,  2, 0, 6, SBR_MONO, 2 }, /* nominal: 24 kbit/s */
+  { CODEC_AAC,  28000, 36000,  22050, 1, 10,10, 9, 9,  2, 0, 3, SBR_MONO, 2 }, /* nominal: 32 kbit/s */
+  { CODEC_AAC,  36000, 44000,  22050, 1, 11,11,10,10,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 40 kbit/s */
+  { CODEC_AAC,  44000, 64001,  22050, 1, 13,13,12,12,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 48 kbit/s */
 
   /* 24/48 kHz dual rate */
-  /* {  8000, 12000,  24000, 1,  1, 1, 1, 1,  1, 0, 6, SBR_MONO, 3 }, */ /* nominal:  8 kbit/s */   /* encoder can not work stable at this extremely low bitrate */
-  { 12000, 16000,  24000, 1,  3, 1, 4, 4,  1, 0, 6, SBR_MONO, 3 }, /* nominal: 12 kbit/s */
-  { 16000, 18000,  24000, 1,  3, 1, 5, 4,  1, 0, 6, SBR_MONO, 3 }, /* nominal: 16 kbit/s */
-  { 18000, 22000,  24000, 1,  4, 3, 8, 5,  2, 0, 6, SBR_MONO, 2 }, /* nominal: 20 kbit/s */
-  { 22000, 28000,  24000, 1,  7, 6, 8, 6,  2, 0, 6, SBR_MONO, 2 }, /* nominal: 24 kbit/s */
-  { 28000, 36000,  24000, 1, 10,10, 9, 9,  2, 0, 3, SBR_MONO, 2 }, /* nominal: 32 kbit/s */
-  { 36000, 44000,  24000, 1, 11,11,10,10,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 40 kbit/s */
-  { 44000, 64001,  24000, 1, 13,13,11,11,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 48 kbit/s */
+  /* { CODEC_AAC,   8000, 12000,  24000, 1,  1, 1, 1, 1,  1, 0, 6, SBR_MONO, 3 }, */ /* nominal:  8 kbit/s */   /* encoder can not work stable at this extremely low bitrate */
+  { CODEC_AAC,  12000, 16000,  24000, 1,  3, 1, 4, 4,  1, 0, 6, SBR_MONO, 3 }, /* nominal: 12 kbit/s */
+  { CODEC_AAC,  16000, 18000,  24000, 1,  3, 1, 5, 4,  1, 0, 6, SBR_MONO, 3 }, /* nominal: 16 kbit/s */
+  { CODEC_AAC,  18000, 22000,  24000, 1,  4, 3, 8, 5,  2, 0, 6, SBR_MONO, 2 }, /* nominal: 20 kbit/s */
+  { CODEC_AAC,  22000, 28000,  24000, 1,  7, 6, 8, 6,  2, 0, 6, SBR_MONO, 2 }, /* nominal: 24 kbit/s */
+  { CODEC_AAC,  28000, 36000,  24000, 1, 10,10, 9, 9,  2, 0, 3, SBR_MONO, 2 }, /* nominal: 32 kbit/s */
+  { CODEC_AAC,  36000, 44000,  24000, 1, 11,11,10,10,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 40 kbit/s */
+  { CODEC_AAC,  44000, 64001,  24000, 1, 13,13,11,11,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 48 kbit/s */
 
   /* 32/64 kHz dual rate */  /* placebo settings */
-  {  24000, 36000, 32000, 1,  4, 4, 4, 4,  2, 0, 3, SBR_MONO, 3 }, /* lowest range  */
-  {  36000, 60000, 32000, 1,  7, 7, 6, 6,  2, 0, 3, SBR_MONO, 2 }, /* lowest range */
-  {  60000, 72000, 32000, 1,  9, 9, 8, 8,  2, 0, 3, SBR_MONO, 1 }, /* low range */
-  {  72000,100000, 32000, 1, 11,11,10,10,  2, 0, 3, SBR_MONO, 1 }, /* SBR sweet spot */
-  { 100000,160001, 32000, 1, 13,13,11,11,  2, 0, 3, SBR_MONO, 1 }, /* backwards compatible */
+  { CODEC_AAC,   24000, 36000, 32000, 1,  4, 4, 4, 4,  2, 0, 3, SBR_MONO, 3 }, /* lowest range  */
+  { CODEC_AAC,   36000, 60000, 32000, 1,  7, 7, 6, 6,  2, 0, 3, SBR_MONO, 2 }, /* lowest range */
+  { CODEC_AAC,   60000, 72000, 32000, 1,  9, 9, 8, 8,  2, 0, 3, SBR_MONO, 1 }, /* low range */
+  { CODEC_AAC,   72000,100000, 32000, 1, 11,11,10,10,  2, 0, 3, SBR_MONO, 1 }, /* SBR sweet spot */
+  { CODEC_AAC,  100000,160001, 32000, 1, 13,13,11,11,  2, 0, 3, SBR_MONO, 1 }, /* backwards compatible */
 
   /* 44.1/88.2 kHz dual rate */  /* placebo settings */
-  {  24000, 36000, 44100, 1,  4, 4, 4, 4,  2, 0, 3, SBR_MONO, 3 }, /* lowest range (multichannel rear) */
-  {  36000, 60000, 44100, 1,  7, 7, 6, 6,  2, 0, 3, SBR_MONO, 2 }, /* lowest range (multichannel rear) */
-  {  60000, 72000, 44100, 1,  9, 9, 8, 8,  2, 0, 3, SBR_MONO, 1 }, /* low range */
-  {  72000,100000, 44100, 1, 11,11,10,10,  2, 0, 3, SBR_MONO, 1 }, /* SBR sweet spot */
-  { 100000,160001, 44100, 1, 13,13,11,11,  2, 0, 3, SBR_MONO, 1 }, /* backwards compatible */
+  { CODEC_AAC,   24000, 36000, 44100, 1,  4, 4, 4, 4,  2, 0, 3, SBR_MONO, 3 }, /* lowest range (multichannel rear) */
+  { CODEC_AAC,   36000, 60000, 44100, 1,  7, 7, 6, 6,  2, 0, 3, SBR_MONO, 2 }, /* lowest range (multichannel rear) */
+  { CODEC_AAC,   60000, 72000, 44100, 1,  9, 9, 8, 8,  2, 0, 3, SBR_MONO, 1 }, /* low range */
+  { CODEC_AAC,   72000,100000, 44100, 1, 11,11,10,10,  2, 0, 3, SBR_MONO, 1 }, /* SBR sweet spot */
+  { CODEC_AAC,  100000,160001, 44100, 1, 13,13,11,11,  2, 0, 3, SBR_MONO, 1 }, /* backwards compatible */
 
   /* 48/96 kHz dual rate */  /* not yet finally tuned */
-  {  32000, 36000, 48000, 1,  4, 4, 9, 9,  2, 0, 3, SBR_MONO, 3 }, /* lowest range (multichannel rear) */
-  {  36000, 60000, 48000, 1,  7, 7,10,10,  2, 0, 3, SBR_MONO, 2 }, /* nominal: 40 */
-  {  60000, 72000, 48000, 1,  9, 9,10,10,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 64 */
-  {  72000,100000, 48000, 1, 11,11,11,11,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 80 */
-  { 100000,160001, 48000, 1, 13,13,11,11,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 128 */
+  { CODEC_AAC,   32000, 36000, 48000, 1,  4, 4, 9, 9,  2, 0, 3, SBR_MONO, 3 }, /* lowest range (multichannel rear) */
+  { CODEC_AAC,   36000, 60000, 48000, 1,  7, 7,10,10,  2, 0, 3, SBR_MONO, 2 }, /* nominal: 40 */
+  { CODEC_AAC,   60000, 72000, 48000, 1,  9, 9,10,10,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 64 */
+  { CODEC_AAC,   72000,100000, 48000, 1, 11,11,11,11,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 80 */
+  { CODEC_AAC,  100000,160001, 48000, 1, 13,13,11,11,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 128 */
 
   /*** stereo ***/
   /* 08/16 kHz dual rate */
-  { 16000, 24000,   8000, 2,  6, 6,  9, 7, 1, 0,-3, SBR_SWITCH_LRC, 3 }, /* nominal: 20 kbit/s */ /* placebo */
-  { 24000, 28000,   8000, 2,  9, 9, 11, 9, 1, 0,-3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */
-  { 28000, 36000,   8000, 2, 11, 9, 11, 9, 2, 0,-3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */
-  { 36000, 44000,   8000, 2, 13,11, 13,11, 2, 0,-3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */
-  { 44000, 52000,   8000, 2, 14,12, 13,12, 2, 0,-3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */
-  { 52000, 60000,   8000, 2, 15,15, 13,13, 3, 0,-3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */
-  { 60000, 76000,   8000, 2, 15,15, 13,13, 3, 0,-3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */
-  { 76000,128001,   8000, 2, 15,15, 13,13, 3, 0,-3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
+  { CODEC_AAC,  16000, 24000,   8000, 2,  6, 6,  9, 7, 1, 0,-3, SBR_SWITCH_LRC, 3 }, /* nominal: 20 kbit/s */ /* placebo */
+  { CODEC_AAC,  24000, 28000,   8000, 2,  9, 9, 11, 9, 1, 0,-3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */
+  { CODEC_AAC,  28000, 36000,   8000, 2, 11, 9, 11, 9, 2, 0,-3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */
+  { CODEC_AAC,  36000, 44000,   8000, 2, 13,11, 13,11, 2, 0,-3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */
+  { CODEC_AAC,  44000, 52000,   8000, 2, 14,12, 13,12, 2, 0,-3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */
+  { CODEC_AAC,  52000, 60000,   8000, 2, 14,14, 13,13, 3, 0,-3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */
+  { CODEC_AAC,  60000, 76000,   8000, 2, 14,14, 13,13, 3, 0,-3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */
+  { CODEC_AAC,  76000,128001,   8000, 2, 14,14, 13,13, 3, 0,-3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
 
   /* 11/22 kHz dual rate */
-  { 16000, 24000,  11025, 2,  7, 5, 9, 7,  1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 20 kbit/s */ /* placebo */
-  { 24000, 28000,  11025, 2, 10, 8,10, 8,  1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */
-  { 28000, 36000,  11025, 2, 12, 8,12, 8,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */
-  { 36000, 44000,  11025, 2, 13, 9,13, 9,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */
-  { 44000, 52000,  11025, 2, 14,11,13,11,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */
-  { 52000, 60000,  11025, 2, 15,15,13,13,  3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */
-  { 60000, 76000,  11025, 2, 15,15,13,13,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */
-  { 76000,128001,  11025, 2, 15,15,13,13,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
+  { CODEC_AAC,  16000, 24000,  11025, 2,  7, 5, 9, 7,  1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 20 kbit/s */ /* placebo */
+  { CODEC_AAC,  24000, 28000,  11025, 2, 10, 8,10, 8,  1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */
+  { CODEC_AAC,  28000, 36000,  11025, 2, 12, 8,12, 8,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */
+  { CODEC_AAC,  36000, 44000,  11025, 2, 13, 9,13, 9,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */
+  { CODEC_AAC,  44000, 52000,  11025, 2, 14,11,13,11,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */
+  { CODEC_AAC,  52000, 60000,  11025, 2, 15,15,13,13,  3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */
+  { CODEC_AAC,  60000, 76000,  11025, 2, 15,15,13,13,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */
+  { CODEC_AAC,  76000,128001,  11025, 2, 15,15,13,13,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
 
   /* 12/24 kHz dual rate */
-  { 16000, 24000,  12000, 2,  6, 4, 9, 7,  1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 20 kbit/s */ /* placebo */
-  { 24000, 28000,  12000, 2,  9, 7,10, 8,  1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */
-  { 28000, 36000,  12000, 2, 11, 7,12, 8,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */
-  { 36000, 44000,  12000, 2, 12, 9,12, 9,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */
-  { 44000, 52000,  12000, 2, 13,12,13,12,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */
-  { 52000, 60000,  12000, 2, 14,14,13,13,  3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */
-  { 60000, 76000,  12000, 2, 15,15,13,13,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */
-  { 76000,128001,  12000, 2, 15,15,13,13,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
+  { CODEC_AAC,  16000, 24000,  12000, 2,  6, 4, 9, 7,  1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 20 kbit/s */ /* placebo */
+  { CODEC_AAC,  24000, 28000,  12000, 2,  9, 7,10, 8,  1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */
+  { CODEC_AAC,  28000, 36000,  12000, 2, 11, 7,12, 8,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */
+  { CODEC_AAC,  36000, 44000,  12000, 2, 12, 9,12, 9,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */
+  { CODEC_AAC,  44000, 52000,  12000, 2, 13,12,13,12,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */
+  { CODEC_AAC,  52000, 60000,  12000, 2, 14,14,13,13,  3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */
+  { CODEC_AAC,  60000, 76000,  12000, 2, 14,14,13,13,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */
+  { CODEC_AAC,  76000,128001,  12000, 2, 14,14,13,13,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
 
   /* 16/32 kHz dual rate */
-  { 16000, 24000,  16000, 2,  4, 2, 1, 0,  1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 20 kbit/s */
-  { 24000, 28000,  16000, 2,  8, 7,10, 8,  1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */
-  { 28000, 36000,  16000, 2, 10, 9,12,11,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */
-  { 36000, 44000,  16000, 2, 13,13,13,13,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */
-  { 44000, 52000,  16000, 2, 15,15,13,13,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */
-  { 52000, 60000,  16000, 2, 15,15,13,13,  3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */
-  { 60000, 76000,  16000, 2, 15,15,13,13,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */
-  { 76000,128001,  16000, 2, 15,15,13,13,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
+  { CODEC_AAC,  16000, 24000,  16000, 2,  4, 2, 1, 0,  1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 20 kbit/s */
+  { CODEC_AAC,  24000, 28000,  16000, 2,  8, 7,10, 8,  1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */
+  { CODEC_AAC,  28000, 36000,  16000, 2, 10, 9,12,11,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */
+  { CODEC_AAC,  36000, 44000,  16000, 2, 13,13,13,13,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */
+  { CODEC_AAC,  44000, 52000,  16000, 2, 14,14,13,13,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */
+  { CODEC_AAC,  52000, 60000,  16000, 2, 14,14,13,13,  3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */
+  { CODEC_AAC,  60000, 76000,  16000, 2, 14,14,13,13,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */
+  { CODEC_AAC,  76000,128001,  16000, 2, 14,14,13,13,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
 
   /* 22.05/44.1 kHz dual rate */
-  { 16000, 24000,  22050, 2,  2, 1, 1, 0,  1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 20 kbit/s */
-  { 24000, 28000,  22050, 2,  5, 4, 6, 5,  1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */
-  { 28000, 32000,  22050, 2,  5, 4, 8, 7,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 28 kbit/s */
-  { 32000, 36000,  22050, 2,  7, 6, 8, 7,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */
-  { 36000, 44000,  22050, 2, 10,10, 9, 9,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */
-  { 44000, 52000,  22050, 2, 12,12, 9, 9,  3, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */
-  { 52000, 60000,  22050, 2, 13,13,10,10,  3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */
-  { 60000, 76000,  22050, 2, 14,14,12,12,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */
-  { 76000,128001,  22050, 2, 14,14,12,12,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
+  { CODEC_AAC,  16000, 24000,  22050, 2,  2, 1, 1, 0,  1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 20 kbit/s */
+  { CODEC_AAC,  24000, 28000,  22050, 2,  5, 4, 6, 5,  1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */
+  { CODEC_AAC,  28000, 32000,  22050, 2,  5, 4, 8, 7,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 28 kbit/s */
+  { CODEC_AAC,  32000, 36000,  22050, 2,  7, 6, 8, 7,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */
+  { CODEC_AAC,  36000, 44000,  22050, 2, 10,10, 9, 9,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */
+  { CODEC_AAC,  44000, 52000,  22050, 2, 12,12, 9, 9,  3, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */
+  { CODEC_AAC,  52000, 60000,  22050, 2, 13,13,10,10,  3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */
+  { CODEC_AAC,  60000, 76000,  22050, 2, 14,14,12,12,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */
+  { CODEC_AAC,  76000,128001,  22050, 2, 14,14,12,12,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
 
   /* 24/48 kHz dual rate */
-  { 16000, 24000,  24000, 2,  2, 1, 1, 0,  1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 20 kbit/s */
-  { 24000, 28000,  24000, 2,  5, 5, 6, 6,  1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */
-  { 28000, 36000,  24000, 2,  7, 6, 8, 7,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */
-  { 36000, 44000,  24000, 2, 10,10, 9, 9,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */
-  { 44000, 52000,  24000, 2, 12,12, 9, 9,  3, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */
-  { 52000, 60000,  24000, 2, 13,13,10,10,  3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */
-  { 60000, 76000,  24000, 2, 14,14,12,12,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */
-  { 76000,128001,  24000, 2, 15,15,12,12,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
+  { CODEC_AAC,  16000, 24000,  24000, 2,  2, 1, 1, 0,  1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 20 kbit/s */
+  { CODEC_AAC,  24000, 28000,  24000, 2,  5, 5, 6, 6,  1, 0, -3, SBR_SWITCH_LRC, 3 }, /* nominal: 24 kbit/s */
+  { CODEC_AAC,  28000, 36000,  24000, 2,  7, 6, 8, 7,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */
+  { CODEC_AAC,  36000, 44000,  24000, 2, 10,10, 9, 9,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */
+  { CODEC_AAC,  44000, 52000,  24000, 2, 12,12, 9, 9,  3, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */
+  { CODEC_AAC,  52000, 60000,  24000, 2, 13,13,10,10,  3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */
+  { CODEC_AAC,  60000, 76000,  24000, 2, 14,14,12,12,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */
+  { CODEC_AAC,  76000,128001,  24000, 2, 14,14,12,12,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
 
   /* 32/64 kHz dual rate */  /* placebo settings */
-  {  32000, 60000, 32000, 2,  4, 4, 4, 4,  2, 0, -3, SBR_SWITCH_LRC, 3 }, /* lowest range (multichannel rear) */
-  {  60000, 80000, 32000, 2,  7, 7, 6, 6,  3, 0, -3, SBR_SWITCH_LRC, 2 }, /* lowest range (multichannel rear) */
-  {  80000,112000, 32000, 2,  9, 9, 8, 8,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* low range */
-  { 112000,144000, 32000, 2, 11,11,10,10,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* SBR sweet spot */
-  { 144000,256001, 32000, 2, 13,13,11,11,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* backwards compatible */
+  { CODEC_AAC,   32000, 60000, 32000, 2,  4, 4, 4, 4,  2, 0, -3, SBR_SWITCH_LRC, 3 }, /* lowest range (multichannel rear) */
+  { CODEC_AAC,   60000, 80000, 32000, 2,  7, 7, 6, 6,  3, 0, -3, SBR_SWITCH_LRC, 2 }, /* lowest range (multichannel rear) */
+  { CODEC_AAC,   80000,112000, 32000, 2,  9, 9, 8, 8,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* low range */
+  { CODEC_AAC,  112000,144000, 32000, 2, 11,11,10,10,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* SBR sweet spot */
+  { CODEC_AAC,  144000,256001, 32000, 2, 13,13,11,11,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* backwards compatible */
 
   /* 44.1/88.2 kHz dual rate */  /* placebo settings */
-  {  32000, 60000, 44100, 2,  4, 4, 4, 4,  2, 0, -3, SBR_SWITCH_LRC, 3 }, /* lowest range (multichannel rear) */
-  {  60000, 80000, 44100, 2,  7, 7, 6, 6,  3, 0, -3, SBR_SWITCH_LRC, 2 }, /* lowest range (multichannel rear) */
-  {  80000,112000, 44100, 2,  9, 9, 8, 8,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* low range */
-  { 112000,144000, 44100, 2, 11,11,10,10,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* SBR sweet spot */
-  { 144000,256001, 44100, 2, 13,13,11,11,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* backwards compatible */
+  { CODEC_AAC,   32000, 60000, 44100, 2,  4, 4, 4, 4,  2, 0, -3, SBR_SWITCH_LRC, 3 }, /* lowest range (multichannel rear) */
+  { CODEC_AAC,   60000, 80000, 44100, 2,  7, 7, 6, 6,  3, 0, -3, SBR_SWITCH_LRC, 2 }, /* lowest range (multichannel rear) */
+  { CODEC_AAC,   80000,112000, 44100, 2,  9, 9, 8, 8,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* low range */
+  { CODEC_AAC,  112000,144000, 44100, 2, 11,11,10,10,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* SBR sweet spot */
+  { CODEC_AAC,  144000,256001, 44100, 2, 13,13,11,11,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* backwards compatible */
 
   /* 48/96 kHz dual rate */  /* not yet finally tuned */
-  {  36000, 60000, 48000, 2,  4, 4, 9, 9,  2, 0, -3, SBR_SWITCH_LRC, 3 }, /* lowest range (multichannel rear) */
-  {  60000, 80000, 48000, 2,  7, 7, 9, 9,  3, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 64 */
-  {  80000,112000, 48000, 2,  9, 9,10,10,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 96 */
-  { 112000,144000, 48000, 2, 11,11,11,11,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 128 */
-  { 144000,256001, 48000, 2, 13,13,11,11,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 192 */
+  { CODEC_AAC,   36000, 60000, 48000, 2,  4, 4, 9, 9,  2, 0, -3, SBR_SWITCH_LRC, 3 }, /* lowest range (multichannel rear) */
+  { CODEC_AAC,   60000, 80000, 48000, 2,  7, 7, 9, 9,  3, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 64 */
+  { CODEC_AAC,   80000,112000, 48000, 2,  9, 9,10,10,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 96 */
+  { CODEC_AAC,  112000,144000, 48000, 2, 11,11,11,11,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 128 */
+  { CODEC_AAC,  144000,256001, 48000, 2, 13,13,11,11,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 192 */
+
 
   /** AAC LOW DELAY SECTION **/
 
+  /*** mono ***/
+  /* 16/32 kHz dual rate not yet tuned ->alb copied from non LD tables*/
+  { CODEC_AACLD,  16000, 18000,  16000, 1,  4, 5, 9, 7,  1, 0, 6, SBR_MONO, 3 }, /* nominal: 16 kbit/s wrr: tuned */
+  { CODEC_AACLD,  18000, 22000,  16000, 1,  7, 7,12,12,  1, 6, 9, SBR_MONO, 3 }, /* nominal: 20 kbit/s wrr: tuned */
+  { CODEC_AACLD,  22000, 28000,  16000, 1,  6, 6, 9, 9,  2, 3, 6, SBR_MONO, 3 }, /* nominal: 24 kbit/s wrr: tuned */
+  { CODEC_AACLD,  28000, 36000,  16000, 1,  8, 8,12, 7,  2, 9,12, SBR_MONO, 3 }, /* jgr: special */ /* wrr: tuned */
+  { CODEC_AACLD,  36000, 44000,  16000, 1, 10,14,12,13,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 40 kbit/s */
+  { CODEC_AACLD,  44000, 64001,  16000, 1, 11,14,13,13,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 48 kbit/s */
+
   /* 22.05/44.1 kHz dual rate */
-  { 18000, 22000,  22050, 1,  4, 4, 5, 5,  2, 0, 6, SBR_MONO, 2 }, /* nominal: 20 kbit/s */
-  { 22000, 28000,  22050, 1,  4, 4, 6, 5,  2, 0, 6, SBR_MONO, 2 }, /* nominal: 24 kbit/s */
-  { 28000, 36000,  22050, 1,  7, 8, 8, 8,  2, 0, 3, SBR_MONO, 2 }, /* nominal: 32 kbit/s */
-  { 36000, 44000,  22050, 1,  9, 9, 9, 9,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 40 kbit/s */
-  { 44000, 52000,  22050, 1, 11,11,11,11,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 48 kbit/s */
-  { 52000, 64001,  22050, 1, 12,11,11,11,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 56 kbit/s */
+  { CODEC_AACLD,  18000, 22000,  22050, 1,  4, 4, 5, 5,  2, 0, 6, SBR_MONO, 3 }, /* nominal: 20 kbit/s */
+  { CODEC_AACLD,  22000, 28000,  22050, 1,  5, 5, 6, 6,  2, 0, 6, SBR_MONO, 2 }, /* nominal: 24 kbit/s */
+  { CODEC_AACLD,  28000, 36000,  22050, 1,  7, 8, 8, 8,  2, 0, 3, SBR_MONO, 2 }, /* nominal: 32 kbit/s */
+  { CODEC_AACLD,  36000, 44000,  22050, 1,  9, 9, 9, 9,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 40 kbit/s */
+  { CODEC_AACLD,  44000, 52000,  22050, 1, 12,11,11,11,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 48 kbit/s */
+  { CODEC_AACLD,  52000, 64001,  22050, 1, 13,11,11,10,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 56 kbit/s */
 
   /* 24/48 kHz dual rate */
-  { 20000, 22000,  24000, 1,  4, 4, 5, 5,  2, 0, 6, SBR_MONO, 2 }, /* nominal: 20 kbit/s */
-  { 22000, 28000,  24000, 1,  4, 4, 6, 5,  2, 0, 6, SBR_MONO, 2 }, /* nominal: 24 kbit/s */
-  { 28000, 36000,  24000, 1,  6, 8, 8, 8,  2, 0, 3, SBR_MONO, 2 }, /* nominal: 32 kbit/s */
-  { 36000, 44000,  24000, 1,  8, 9, 9, 9,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 40 kbit/s */
-  { 44000, 52000,  24000, 1, 12,11,11,10,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 48 kbit/s */
-  { 52000, 64001,  24000, 1, 13,11,11,10,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 48 kbit/s */
+  { CODEC_AACLD,  20000, 22000,  24000, 1,  4, 1, 8, 4,  2, 3, 6, SBR_MONO, 2 }, /* nominal: 20 kbit/s */
+  { CODEC_AACLD,  22000, 28000,  24000, 1,  3, 8, 8, 7,  2, 0, 3, SBR_MONO, 2 }, /* nominal: 24 kbit/s */
+  { CODEC_AACLD,  28000, 36000,  24000, 1,  4, 8, 8, 7,  2, 0, 3, SBR_MONO, 2 }, /* nominal: 32 kbit/s */
+  { CODEC_AACLD,  36000, 56000,  24000, 1,  8, 9, 9, 9,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 40 kbit/s */
+  { CODEC_AACLD,  56000, 64001,  24000, 1, 13,11,11,10,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 64 kbit/s */
+
+  /* 32/64 kHz dual rate */  /* placebo settings */  /*jgr: new, copy from CODEC_AAC */
+  { CODEC_AACLD,  24000, 36000,  32000, 1,  4, 4, 4, 4,  2, 0, 3, SBR_MONO, 3 }, /* lowest range  */
+  { CODEC_AACLD,  36000, 60000,  32000, 1,  7, 7, 6, 6,  2, 0, 3, SBR_MONO, 2 }, /* lowest range */
+  { CODEC_AACLD,  60000, 72000,  32000, 1,  9, 9, 8, 8,  2, 0, 3, SBR_MONO, 1 }, /* low range */
+  { CODEC_AACLD,  72000,100000,  32000, 1, 11,11,10,10,  2, 0, 3, SBR_MONO, 1 }, /* SBR sweet spot */
+  { CODEC_AACLD, 100000,160001,  32000, 1, 13,13,11,11,  2, 0, 3, SBR_MONO, 1 }, /* backwards compatible */
+
+  /* 44/88 kHz dual rate */  /* not yet finally tuned */
+  { CODEC_AACLD,  36000, 60000,  44100, 1,  8, 7, 6, 9,  2, 0, 3, SBR_MONO, 2 }, /* nominal: 40 */
+  { CODEC_AACLD,  60000, 72000,  44100, 1,  9, 9,10,10,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 64 */
+  { CODEC_AACLD,  72000,100000,  44100, 1, 11,11,11,11,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 80 */
+  { CODEC_AACLD, 100000,160001,  44100, 1, 13,13,11,11,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 128 */
+
+  /* 48/96 kHz dual rate */  /* 32 and 40kbps line tuned for dual-rate SBR */
+  { CODEC_AACLD,  36000, 60000,  48000, 1,  8, 7, 6, 9,  2, 0, 3, SBR_MONO, 2 }, /* nominal: 40 */
+  { CODEC_AACLD,  60000, 72000,  48000, 1,  9, 9,10,10,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 64 */
+  { CODEC_AACLD,  72000,100000,  48000, 1, 11,11,11,11,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 80 */
+  { CODEC_AACLD, 100000,160001,  48000, 1, 13,13,11,11,  2, 0, 3, SBR_MONO, 1 }, /* nominal: 128 */
+
+  /*** stereo ***/
+  /* 16/32 kHz dual rate not yet tuned ->alb copied from non LD tables*/
+  { CODEC_AACLD,  32000, 36000,  16000, 2, 10, 9,12,11,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */
+  { CODEC_AACLD,  36000, 44000,  16000, 2, 13,13,13,13,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */
+  { CODEC_AACLD,  44000, 52000,  16000, 2, 10, 9,11, 9,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* tune12 nominal: 48 kbit/s */
+  { CODEC_AACLD,  52000, 60000,  16000, 2, 14,14,13,13,  3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */
+  { CODEC_AACLD,  60000, 76000,  16000, 2, 14,14,13,13,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */
+  { CODEC_AACLD,  76000,128001,  16000, 2, 14,14,13,13,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
 
   /* 22.05/44.1 kHz dual rate */
-  { 32000, 36000,  22050, 2,  5, 4, 7, 6,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */
-  { 36000, 44000,  22050, 2,  5, 8, 8, 8,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */
-  { 44000, 52000,  22050, 2,  7,10, 8, 8,  3, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */
-  { 52000, 60000,  22050, 2,  9,11, 9, 9,  3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */
-  { 60000, 76000,  22050, 2, 10,12,10,11,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */
-  { 76000, 82000,  22050, 2, 12,12,11,11,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
-  { 82000,128001,  22050, 2, 13,12,11,11,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
+  { CODEC_AACLD,  32000, 36000,  22050, 2,  5, 4, 7, 6,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */
+  { CODEC_AACLD,  36000, 44000,  22050, 2,  5, 8, 8, 8,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */
+  { CODEC_AACLD,  44000, 52000,  22050, 2,  7,10, 8, 8,  3, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */
+  { CODEC_AACLD,  52000, 60000,  22050, 2,  9,11, 9, 9,  3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */
+  { CODEC_AACLD,  60000, 76000,  22050, 2, 10,12,10,11,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */
+  { CODEC_AACLD,  76000, 82000,  22050, 2, 12,12,11,11,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
+  { CODEC_AACLD,  82000,128001,  22050, 2, 13,12,11,11,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
 
   /* 24/48 kHz dual rate */
-  { 32000, 36000,  24000, 2,  5, 4, 7, 6,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */
-  { 36000, 44000,  24000, 2,  4, 8, 8, 8,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */
-  { 44000, 52000,  24000, 2,  6,10, 8, 8,  3, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */
-  { 52000, 60000,  24000, 2,  9,11, 9, 9,  3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */
-  { 60000, 76000,  24000, 2, 11,12,10,11,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */
-  { 76000, 88000,  24000, 2, 12,13,11,11,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
-  { 88000,128001,  24000, 2, 13,13,11,11,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 92 kbit/s */
-
+  { CODEC_AACLD,  32000, 36000,  24000, 2,  5, 4, 7, 6,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 32 kbit/s */
+  { CODEC_AACLD,  36000, 44000,  24000, 2,  4, 8, 8, 8,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 40 kbit/s */
+  { CODEC_AACLD,  44000, 52000,  24000, 2,  6,10, 8, 8,  3, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 48 kbit/s */
+  { CODEC_AACLD,  52000, 60000,  24000, 2,  9,11, 9, 9,  3, 0, -3, SBR_SWITCH_LRC, 1 }, /* nominal: 56 kbit/s */
+  { CODEC_AACLD,  60000, 76000,  24000, 2, 11,12,10,11,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 64 kbit/s */
+  { CODEC_AACLD,  76000, 88000,  24000, 2, 12,13,11,11,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 80 kbit/s */
+  { CODEC_AACLD,  88000,128001,  24000, 2, 13,13,11,11,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 92 kbit/s */
+
+  /* 32/64 kHz dual rate */  /* placebo settings */ /*jgr: new, copy from CODEC_AAC */
+  { CODEC_AACLD,  60000, 80000,  32000, 2,  7, 7, 6, 6,  3, 0, -3, SBR_SWITCH_LRC, 2 }, /* lowest range (multichannel rear) */
+  { CODEC_AACLD,  80000,112000,  32000, 2,  9, 9, 8, 8,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* low range */
+  { CODEC_AACLD, 112000,144000,  32000, 2, 11,11,10,10,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* SBR sweet spot */
+  { CODEC_AACLD, 144000,256001,  32000, 2, 13,13,11,11,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* backwards compatible */
+
+  /* 44.1/88.2 kHz dual rate */  /* placebo settings */ /*wrr: new, copy from CODEC_AAC */
+  { CODEC_AACLD,  60000, 80000,  44100, 2,  7, 7, 6, 6,  3, 0, -3, SBR_SWITCH_LRC, 2 }, /* lowest range (multichannel rear) */
+  { CODEC_AACLD,  80000,112000,  44100, 2, 10,10, 8, 8,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* hlm 11-08-29 */
+  { CODEC_AACLD, 112000,144000,  44100, 2, 12,12,10,10,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* hlm 11-08-29 */
+  { CODEC_AACLD, 144000,256001,  44100, 2, 13,13,11,11,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* backwards compatible */
+
+  /* 48/96 kHz dual rate */  /* not yet finally tuned */ /*wrr: new, copy from CODEC_AAC */
+  { CODEC_AACLD,  60000, 80000,  48000, 2,  7, 7,10,10,  2, 0, -3, SBR_SWITCH_LRC, 2 }, /* nominal: 64 */
+  { CODEC_AACLD,  80000,112000,  48000, 2,  9, 9,10,10,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 96 */
+  { CODEC_AACLD, 112000,144000,  48000, 2, 11,11,11,11,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* nominal: 128 */
+  { CODEC_AACLD, 144000,176000,  48000, 2, 12,12,11,11,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* hlm 09-10-19 */
+  { CODEC_AACLD, 176000,256001,  48000, 2, 13,13,11,11,  3, 0, -3, SBR_LEFT_RIGHT, 1 }, /* hlm 09-10-19 */
 
 };
 
+const int sbrTuningTableSize = sizeof(sbrTuningTable)/sizeof(sbrTuningTable[0]);
+
 const psTuningTable_t psTuningTable[4] =
 {
   {  8000,  22000, PSENC_STEREO_BANDS_10, PSENC_NENV_1, FL2FXCONST_DBL(3.0f/4.0f) },
diff --git a/libSBRenc/src/sbr_rom.h b/libSBRenc/src/sbr_rom.h
index e79a730..afa924e 100644
--- a/libSBRenc/src/sbr_rom.h
+++ b/libSBRenc/src/sbr_rom.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -118,13 +118,8 @@ extern const UCHAR         v_Huff_NoiseLevelL11T[63];
 extern const INT           bookSbrNoiseBalanceC11T[25];
 extern const UCHAR         bookSbrNoiseBalanceL11T[25];
 
-#define SBRENC_AACLC_TUNING_SIZE   124
-#define SBRENC_AACELD_TUNING_SIZE   (26)
-#define SBRENC_AACELD2_TUNING_SIZE  (26)
-
-#define SBRENC_TUNING_SIZE (SBRENC_AACLC_TUNING_SIZE + SBRENC_AACELD_TUNING_SIZE)
-
-extern const sbrTuningTable_t sbrTuningTable[SBRENC_TUNING_SIZE];
+extern const sbrTuningTable_t sbrTuningTable[];
+extern const int sbrTuningTableSize;
 
 extern const psTuningTable_t psTuningTable[4];
 
diff --git a/libSBRenc/src/sbrenc_freq_sca.cpp b/libSBRenc/src/sbrenc_freq_sca.cpp
index bbcb29e..30bc5ca 100644
--- a/libSBRenc/src/sbrenc_freq_sca.cpp
+++ b/libSBRenc/src/sbrenc_freq_sca.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -84,6 +84,7 @@ amm-info at iis.fraunhofer.de
 /*!
   \file
   \brief  frequency scale  
+  \author Tobias Chalupka
 */
 
 #include "sbrenc_freq_sca.h"
@@ -92,10 +93,10 @@ amm-info at iis.fraunhofer.de
 #include "genericStds.h"
 
 /*  StartFreq */
-static INT getStartFreq(INT fs, const INT start_freq);
+static INT getStartFreq(INT fsCore, const INT start_freq);
 
 /* StopFreq */
-static INT getStopFreq(INT fs, const INT stop_freq, const INT noChannels);
+static INT getStopFreq(INT fsCore, const INT stop_freq);
 
 static INT  numberOfBands(INT b_p_o, INT start, INT stop, FIXP_DBL warp_factor);
 static void CalcBands(INT * diff, INT start , INT stop , INT num_bands);
@@ -115,7 +116,7 @@ static void cumSum(INT start_value, INT* diff, INT length, UCHAR  *start_adress)
  *******************************************************************************/
 
 INT
-FDKsbrEnc_getSbrStartFreqRAW (INT startFreq, INT QMFbands, INT fs)
+FDKsbrEnc_getSbrStartFreqRAW (INT startFreq, INT fsCore)
 {
   INT result;
 
@@ -123,9 +124,9 @@ FDKsbrEnc_getSbrStartFreqRAW (INT startFreq, INT QMFbands, INT fs)
     return -1;
   }
   /* Update startFreq struct */
-  result = getStartFreq(fs, startFreq);
+  result = getStartFreq(fsCore, startFreq);
 
-  result = (result*fs/QMFbands+1)>>1;
+  result = (result*(fsCore>>5)+1)>>1; /* (result*fsSBR/QMFbands+1)>>1; */
 
   return (result);
 
@@ -141,17 +142,16 @@ FDKsbrEnc_getSbrStartFreqRAW (INT startFreq, INT QMFbands, INT fs)
 
  Return:
  *******************************************************************************/
-INT FDKsbrEnc_getSbrStopFreqRAW  (INT stopFreq, INT QMFbands, INT fs)
+INT FDKsbrEnc_getSbrStopFreqRAW  (INT stopFreq, INT fsCore)
 {
   INT result;
 
   if ( stopFreq < 0 || stopFreq > 13)
     return -1;
 
-
   /* Uppdate stopFreq struct */
-  result = getStopFreq( fs, stopFreq, QMFbands);
-  result =   (result*fs/QMFbands+1)>>1;
+  result = getStopFreq(fsCore, stopFreq);
+  result = (result*(fsCore>>5)+1)>>1; /* (result*fsSBR/QMFbands+1)>>1; */
 
   return (result);
 } /* End getSbrStopFreq */
@@ -162,69 +162,73 @@ INT FDKsbrEnc_getSbrStopFreqRAW  (INT stopFreq, INT QMFbands, INT fs)
  *******************************************************************************
  Description:
 
- Arguments:
+ Arguments:  fsCore - core sampling rate
+
 
  Return:
  *******************************************************************************/
 static INT
-getStartFreq(INT fs, const INT start_freq)
+getStartFreq(INT fsCore, const INT start_freq)
 {
   INT k0_min;
 
-  switch(fs){
-  case 16000: k0_min = 24;
+  switch(fsCore){
+  case  8000: k0_min = 24; /* (3000 * nQmfChannels / fsSBR ) + 0.5 */
     break;
-  case 22050: k0_min = 17;
+  case 11025: k0_min = 17; /* (3000 * nQmfChannels / fsSBR ) + 0.5 */
     break;
-  case 24000: k0_min = 16;
+  case 12000: k0_min = 16; /* (3000 * nQmfChannels / fsSBR ) + 0.5 */
     break;
-  case 32000: k0_min = 16;
+  case 16000: k0_min = 16; /* (4000 * nQmfChannels / fsSBR ) + 0.5 */
     break;
-  case 44100: k0_min = 12;
+  case 22050: k0_min = 12; /* (4000 * nQmfChannels / fsSBR ) + 0.5 */
     break;
-  case 48000: k0_min = 11;
+  case 24000: k0_min = 11; /* (4000 * nQmfChannels / fsSBR ) + 0.5 */
     break;
-  case 64000: k0_min = 10;
+  case 32000: k0_min = 10; /* (5000 * nQmfChannels / fsSBR ) + 0.5 */
     break;
-  case 88200: k0_min = 7;
+  case 44100: k0_min = 7;  /* (5000 * nQmfChannels / fsSBR ) + 0.5 */
     break;
-  case 96000: k0_min = 7;
+  case 48000: k0_min = 7;  /* (5000 * nQmfChannels / fsSBR ) + 0.5 */
+    break;
+  case 96000: k0_min = 3;  /* (5000 * nQmfChannels / fsSBR ) + 0.5 */
     break;
   default:
     k0_min=11; /* illegal fs */
   }
 
 
-  switch (fs) {
+  switch (fsCore) {
 
-  case 16000:
+  case  8000:
     {
       INT v_offset[]= {-8, -7, -6, -5, -4, -3, -2, -1, 0, 1, 2, 3, 4, 5, 6, 7};
       return (k0_min + v_offset[start_freq]);
     }
-  case 22050:
+  case 11025:
     {
       INT v_offset[]= {-5, -4, -3, -2, -1, 0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13};
       return (k0_min + v_offset[start_freq]);
     }
-  case 24000:
+  case 12000:
     {
       INT v_offset[]= {-5, -3, -2, -1, 0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13, 16};
       return (k0_min + v_offset[start_freq]);
     }
-  case 32000:
+  case 16000:
     {
       INT v_offset[]= {-6, -4, -2, -1, 0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13, 16};
       return (k0_min + v_offset[start_freq]);
     }
-  case 44100:
-  case 48000:
-  case 64000:
+  case 22050:
+  case 24000:
+  case 32000:
     {
       INT v_offset[]= {-4, -2, -1, 0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13, 16, 20};
       return (k0_min + v_offset[start_freq]);
     }
-  case 88200:
+  case 44100:
+  case 48000:
   case 96000:
     {
       INT v_offset[]= {-2, -1, 0, 1, 2, 3, 4, 5, 6, 7, 9, 11, 13, 16, 20, 24};
@@ -249,13 +253,12 @@ getStartFreq(INT fs, const INT start_freq)
  Return:
  *******************************************************************************/
  static INT
-getStopFreq(INT fs, const INT stop_freq, const INT noChannels)
+getStopFreq(INT fsCore, const INT stop_freq)
 {
   INT result,i;
   INT k1_min;
   INT v_dstop[13];
 
-
   INT *v_stop_freq = NULL;
   INT v_stop_freq_16[14] = {48,49,50,51,52,54,55,56,57,59,60,61,63,64};
   INT v_stop_freq_22[14] = {35,37,38,40,42,44,46,48,51,53,56,58,61,64};
@@ -266,40 +269,45 @@ getStopFreq(INT fs, const INT stop_freq, const INT noChannels)
   INT v_stop_freq_64[14] = {20,22,24,26,29,31,34,37,41,45,49,54,59,64};
   INT v_stop_freq_88[14] = {15,17,19,21,23,26,29,33,37,41,46,51,57,64};
   INT v_stop_freq_96[14] = {13,15,17,19,21,24,27,31,35,39,44,50,57,64};
+  INT v_stop_freq_192[14] = {7, 8,10,12,14,16,19,23,27,32,38,46,54,64};
 
-  switch(fs){
-  case 16000: k1_min = 48;
+  switch(fsCore){
+  case  8000: k1_min = 48;
               v_stop_freq =v_stop_freq_16;
     break;
-  case 22050: k1_min = 35;
+  case 11025: k1_min = 35;
               v_stop_freq =v_stop_freq_22;
     break;
-  case 24000: k1_min = 32;
+  case 12000: k1_min = 32;
               v_stop_freq =v_stop_freq_24;
     break;
-  case 32000: k1_min = 32;
+  case 16000: k1_min = 32;
               v_stop_freq =v_stop_freq_32;
     break;
-  case 44100: k1_min = 23;
+  case 22050: k1_min = 23;
               v_stop_freq =v_stop_freq_44;
     break;
-  case 48000: k1_min = 21;
+  case 24000: k1_min = 21;
               v_stop_freq =v_stop_freq_48;
     break;
-  case 64000: k1_min = 20;
+  case 32000: k1_min = 20;
               v_stop_freq =v_stop_freq_64;
     break;
-  case 88200: k1_min = 15;
+  case 44100: k1_min = 15;
               v_stop_freq =v_stop_freq_88;
     break;
-  case 96000: k1_min = 13;
+  case 48000: k1_min = 13;
               v_stop_freq =v_stop_freq_96;
     break;
+  case 96000: k1_min =  7;
+              v_stop_freq =v_stop_freq_192;
+    break;
   default:
     k1_min = 21; /* illegal fs  */
   }
 
-
+  /* if no valid core samplingrate is used this loop produces
+     a segfault, because v_stop_freq is not initialized */
   /* Ensure increasing bandwidth */
   for(i = 0; i <= 12; i++) {
     v_dstop[i] = v_stop_freq[i+1] - v_stop_freq[i];
@@ -322,34 +330,41 @@ getStopFreq(INT fs, const INT stop_freq, const INT noChannels)
  *******************************************************************************
  Description:
 
- Arguments:
+ Arguments:     srSbr            SBR sampling freqency
+                srCore           AAC core sampling freqency
+                noChannels       Number of QMF channels
+                startFreq        SBR start frequency in QMF bands
+                stopFreq         SBR start frequency in QMF bands
 
- Return:
+               *k0               Output parameter
+               *k2               Output parameter
+
+ Return:       Error code (0 is OK)
  *******************************************************************************/
 INT
-FDKsbrEnc_FindStartAndStopBand(const INT samplingFreq,
-                     const INT noChannels,
-                     const INT startFreq,
-                     const INT stopFreq,
-                     const SR_MODE sampleRateMode,
-                     INT *k0,
-                     INT *k2)
+FDKsbrEnc_FindStartAndStopBand(
+        const INT srSbr,
+        const INT srCore,
+        const INT noChannels,
+        const INT startFreq,
+        const INT stopFreq,
+        INT *k0,
+        INT *k2
+        )
 {
 
   /* Update startFreq struct */
-  *k0 = getStartFreq(samplingFreq, startFreq);
+  *k0 = getStartFreq(srCore, startFreq);
 
   /* Test if start freq is outside corecoder range */
-  if( ( sampleRateMode == 1 ) &&
-      ( samplingFreq*noChannels  <
-        2**k0 * samplingFreq) ) {
+  if( srSbr*noChannels < *k0 * srCore ) {
     return (1); /* raise the cross-over frequency and/or lower the number
                    of target bands per octave (or lower the sampling frequency) */
   }
 
   /*Update stopFreq struct */
   if ( stopFreq < 14 ) {
-    *k2 = getStopFreq(samplingFreq, stopFreq, noChannels);
+    *k2 = getStopFreq(srCore, stopFreq);
   } else if( stopFreq == 14 ) {
     *k2 = 2 * *k0;
   } else {
@@ -364,10 +379,10 @@ FDKsbrEnc_FindStartAndStopBand(const INT samplingFreq,
 
 
   /* Test for invalid  k0 k2 combinations */
-  if ( (samplingFreq == 44100) && ( (*k2 - *k0) > MAX_FREQ_COEFFS_FS44100 ) )
+  if ( (srCore == 22050) && ( (*k2 - *k0) > MAX_FREQ_COEFFS_FS44100 ) )
     return (1); /* Number of bands exceeds valid range of MAX_FREQ_COEFFS for fs=44.1kHz */
 
-  if ( (samplingFreq >= 48000) && ( (*k2 - *k0) > MAX_FREQ_COEFFS_FS48000 ) )
+  if ( (srCore >= 24000) && ( (*k2 - *k0) > MAX_FREQ_COEFFS_FS48000 ) )
     return (1); /* Number of bands exceeds valid range of MAX_FREQ_COEFFS for fs>=48kHz */
 
   if ((*k2 - *k0) > MAX_FREQ_COEFFS)
@@ -390,15 +405,19 @@ FDKsbrEnc_FindStartAndStopBand(const INT samplingFreq,
  Return:
  *******************************************************************************/
 INT
-FDKsbrEnc_UpdateFreqScale(UCHAR  *v_k_master, INT *h_num_bands,
-                const INT k0, const INT k2,
-                const INT freqScale,
-                const INT alterScale)
+FDKsbrEnc_UpdateFreqScale(
+        UCHAR     *v_k_master,
+        INT       *h_num_bands,
+        const INT  k0,
+        const INT  k2,
+        const INT  freqScale,
+        const INT  alterScale
+        )
 
 {
 
   INT     b_p_o = 0;        /* bands_per_octave */
-  FIXP_DBL   warp = FL2FXCONST_DBL(0.0f);
+  FIXP_DBL warp = FL2FXCONST_DBL(0.0f);
   INT     dk = 0;
 
   /* Internal variables */
@@ -426,7 +445,7 @@ FDKsbrEnc_UpdateFreqScale(UCHAR  *v_k_master, INT *h_num_bands,
         warp = FL2FXCONST_DBL(1.0f/2.6f);   /* 1.0/(1.3*2.0); */
 
 
-      if(4*k2 >= 9*k0)  /*two or more regions*/
+      if(4*k2 >= 9*k0)  /*two or more regions (how many times the basis band is copied)*/
         {
           k1=2*k0;
 
@@ -592,30 +611,31 @@ modifyBands(INT max_band_previous, INT * diff, INT length)
  *******************************************************************************
  Description:
 
+
  Arguments:
 
  Return:
  *******************************************************************************/
 INT
-FDKsbrEnc_UpdateHiRes(UCHAR *h_hires, INT *num_hires,UCHAR * v_k_master,
-            INT num_master , INT *xover_band, SR_MODE drOrSr,
-            INT noQMFChannels)
+FDKsbrEnc_UpdateHiRes(
+        UCHAR   *h_hires,
+        INT     *num_hires,
+        UCHAR   *v_k_master,
+        INT      num_master,
+        INT     *xover_band
+        )
 {
   INT i;
-  INT divider;
   INT max1,max2;
 
-  /* Check if we use a Dual rate => diver=2 else 1 */
-  divider = (drOrSr == DUAL_RATE) ? 2 : 1;
-
-  if( (v_k_master[*xover_band] > (noQMFChannels/divider) ) ||
+  if( (v_k_master[*xover_band] > 32 ) || /* v_k_master[*xover_band] > noQMFChannels(dualRate)/divider */
       ( *xover_band > num_master ) )  {
       /* xover_band error, too big for this startFreq. Will be clipped */
 
     /* Calculate maximum value for xover_band */
     max1=0;
     max2=num_master;
-    while( (v_k_master[max1+1] < (noQMFChannels/divider)) &&
+    while( (v_k_master[max1+1] < 32 ) && /* noQMFChannels(dualRate)/divider */
            ( (max1+1) < max2) )
       {
         max1++;
diff --git a/libSBRenc/src/sbrenc_freq_sca.h b/libSBRenc/src/sbrenc_freq_sca.h
index 613694a..6f2bb84 100644
--- a/libSBRenc/src/sbrenc_freq_sca.h
+++ b/libSBRenc/src/sbrenc_freq_sca.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -96,34 +96,42 @@ amm-info at iis.fraunhofer.de
 
 
 INT
-FDKsbrEnc_UpdateFreqScale(UCHAR *v_k_master, INT *h_num_bands,
-                const INT k0, const INT k2,
-                const INT freq_scale,
-                const INT alter_scale);
+FDKsbrEnc_UpdateFreqScale(
+        UCHAR *v_k_master,
+        INT   *h_num_bands,
+        const  INT k0,
+        const  INT k2,
+        const  INT freq_scale,
+        const  INT alter_scale
+        );
 
 INT
-FDKsbrEnc_UpdateHiRes(UCHAR *h_hires,
-            INT *num_hires,
-            UCHAR *v_k_master,
-            INT num_master ,
-            INT *xover_band,
-            SR_MODE drOrSr,
-            INT noQMFChannels);
-
-void  FDKsbrEnc_UpdateLoRes(UCHAR * v_lores,
-                  INT *num_lores,
-                  UCHAR * v_hires,
-                  INT num_hires);
+FDKsbrEnc_UpdateHiRes(
+        UCHAR *h_hires,
+        INT   *num_hires,
+        UCHAR *v_k_master,
+        INT    num_master,
+        INT   *xover_band
+        );
+
+void  FDKsbrEnc_UpdateLoRes(
+        UCHAR *v_lores,
+        INT   *num_lores,
+        UCHAR *v_hires,
+        INT    num_hires
+        );
 
 INT
-FDKsbrEnc_FindStartAndStopBand(const INT samplingFreq,
-                     const INT noChannels,
-                     const INT startFreq,
-                     const INT stop_freq,
-                     const SR_MODE sampleRateMode,
-                     INT *k0,
-                     INT *k2);
-
-INT FDKsbrEnc_getSbrStartFreqRAW (INT startFreq, INT QMFbands, INT fs );
-INT FDKsbrEnc_getSbrStopFreqRAW  (INT stopFreq, INT QMFbands, INT fs);
+FDKsbrEnc_FindStartAndStopBand(
+        const INT srSbr,
+        const INT srCore,
+        const INT noChannels,
+        const INT startFreq,
+        const INT stop_freq,
+        INT *k0,
+        INT *k2
+        );
+
+INT FDKsbrEnc_getSbrStartFreqRAW (INT startFreq, INT fsCore);
+INT FDKsbrEnc_getSbrStopFreqRAW  (INT stopFreq, INT fsCore);
 #endif
diff --git a/libSBRenc/src/ton_corr.cpp b/libSBRenc/src/ton_corr.cpp
index 3142870..224da11 100644
--- a/libSBRenc/src/ton_corr.cpp
+++ b/libSBRenc/src/ton_corr.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -303,8 +303,6 @@ FDKsbrEnc_CalculateTonalityQuotas( HANDLE_SBR_TON_CORR_EST hTonCorr,      /*!< H
     }
   }
 
-  FDK_ASSERT(noEstPerFrame == 2);
-
 
   C_ALLOC_SCRATCH_END(realBuf, FIXP_DBL, 2*BAND_V_SIZE*NUM_V_COMBINE);
   C_ALLOC_SCRATCH_END(ac, ACORR_COEFS, 1);
diff --git a/libSBRenc/src/ton_corr.h b/libSBRenc/src/ton_corr.h
index a37eca5..8c8425c 100644
--- a/libSBRenc/src/ton_corr.h
+++ b/libSBRenc/src/ton_corr.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRenc/src/tran_det.cpp b/libSBRenc/src/tran_det.cpp
index b6cde99..1e0a59f 100644
--- a/libSBRenc/src/tran_det.cpp
+++ b/libSBRenc/src/tran_det.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSBRenc/src/tran_det.h b/libSBRenc/src/tran_det.h
index 7e9a93c..95b5d2e 100644
--- a/libSBRenc/src/tran_det.h
+++ b/libSBRenc/src/tran_det.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSYS/include/FDK_audio.h b/libSYS/include/FDK_audio.h
index 8e7041d..c8d9c19 100644
--- a/libSYS/include/FDK_audio.h
+++ b/libSYS/include/FDK_audio.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -144,6 +144,12 @@ typedef enum
 
 } TRANSPORT_TYPE;
 
+#define TT_IS_PACKET(x) \
+       ( ((x) == TT_MP4_RAW) \
+      || ((x) == TT_DRM) \
+      || ((x) == TT_MP4_LATM_MCP0) \
+      || ((x) == TT_MP4_LATM_MCP1) )
+
 /**
  * Audio Object Type definitions.
  */
@@ -228,28 +234,32 @@ typedef enum
 
 /** Channel Mode ( 1-7 equals MPEG channel configurations, others are arbitrary). */
 typedef enum {
-  MODE_INVALID    = -1,
-  MODE_UNKNOWN    = 0,
-  MODE_1          = 1,           /**< SCE                 */
-  MODE_2          = 2,           /**< CPE                 */
-  MODE_1_2        = 3,           /**< SCE,CPE             */
-  MODE_1_2_1      = 4,           /**< SCE,CPE,SCE         */
-  MODE_1_2_2      = 5,           /**< SCE,CPE,CPE         */
-  MODE_1_2_2_1    = 6,           /**< SCE,CPE,CPE,LFE     */
-  MODE_1_2_2_2_1  = 7,           /**< SCE,CPE,CPE,CPE,LFE */
-
-  MODE_1_1                      = 16,          /**< 2 SCEs (dual mono) */
-  MODE_1_1_1_1                  = 17,          /**< 4 SCEs */
-  MODE_1_1_1_1_1_1              = 18,          /**< 6 SCEs */
-  MODE_1_1_1_1_1_1_1_1          = 19,          /**< 8 SCEs */
-  MODE_1_1_1_1_1_1_1_1_1_1_1_1  = 20,          /**< 12 SCEs */
-
-  MODE_2_2                      = 21,          /**< 2 CPEs */
-  MODE_2_2_2                    = 22,          /**< 3 CPEs */
-  MODE_2_2_2_2                  = 23,          /**< 4 CPEs */
-  MODE_2_2_2_2_2_2              = 24,          /**< 6 CPEs */
-
-  MODE_2_1                      = 30           /**< CPE,SCE (ARIB standard) */
+  MODE_INVALID                  = -1,
+  MODE_UNKNOWN                  =  0,
+  MODE_1                        =  1,       /**< C */
+  MODE_2                        =  2,       /**< L+R */
+  MODE_1_2                      =  3,       /**< C, L+R */
+  MODE_1_2_1                    =  4,       /**< C, L+R, Rear */
+  MODE_1_2_2                    =  5,       /**< C, L+R, LS+RS */
+  MODE_1_2_2_1                  =  6,       /**< C, L+R, LS+RS, LFE */
+  MODE_1_2_2_2_1                =  7,       /**< C, LC+RC, L+R, LS+RS, LFE */
+
+
+  MODE_1_1                      = 16,       /**< 2 SCEs (dual mono) */
+  MODE_1_1_1_1                  = 17,       /**< 4 SCEs */
+  MODE_1_1_1_1_1_1              = 18,       /**< 6 SCEs */
+  MODE_1_1_1_1_1_1_1_1          = 19,       /**< 8 SCEs */
+  MODE_1_1_1_1_1_1_1_1_1_1_1_1  = 20,       /**< 12 SCEs */
+
+  MODE_2_2                      = 21,       /**< 2 CPEs */
+  MODE_2_2_2                    = 22,       /**< 3 CPEs */
+  MODE_2_2_2_2                  = 23,       /**< 4 CPEs */
+  MODE_2_2_2_2_2_2              = 24,       /**< 6 CPEs */
+
+  MODE_2_1                      = 30,       /**< CPE,SCE (ARIB standard B32) */
+
+  MODE_7_1_REAR_SURROUND        = 33,       /**< C, L+R, LS+RS, Lrear+Rrear, LFE */
+  MODE_7_1_FRONT_CENTER         = 34        /**< C, LC+RC, L+R, LS+RS, LFE */
 
 } CHANNEL_MODE;
 
@@ -266,6 +276,15 @@ typedef enum {
   ACT_TOP /* Ts */
 } AUDIO_CHANNEL_TYPE;
 
+typedef enum
+{
+  SIG_UNKNOWN                 = -1,
+  SIG_IMPLICIT                =  0,
+  SIG_EXPLICIT_BW_COMPATIBLE  =  1,
+  SIG_EXPLICIT_HIERARCHICAL   =  2
+
+} SBR_PS_SIGNALING;
+
 /**
  * Audio Codec flags.
  */
@@ -328,6 +347,10 @@ typedef struct {
 
   UCHAR      stereoConfigIndex; /**< USAC MPS stereo mode */
   UCHAR      sbrMode;           /**< USAC SBR mode */
+  SBR_PS_SIGNALING sbrSignaling;/**< 0: implicit signaling, 1: backwards compatible explicit signaling, 2: hierarcical explicit signaling */
+
+  UCHAR      sbrPresent;
+  UCHAR      psPresent;
 } CODER_CONFIG;
 
 /** MP4 Element IDs. */
diff --git a/libSYS/include/audio.h b/libSYS/include/audio.h
index 7a827cb..6ef83ae 100644
--- a/libSYS/include/audio.h
+++ b/libSYS/include/audio.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSYS/include/cmdl_parser.h b/libSYS/include/cmdl_parser.h
index 86f20df..266e41d 100644
--- a/libSYS/include/cmdl_parser.h
+++ b/libSYS/include/cmdl_parser.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSYS/include/conv_string.h b/libSYS/include/conv_string.h
index 1280ae6..0409dc0 100644
--- a/libSYS/include/conv_string.h
+++ b/libSYS/include/conv_string.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSYS/include/genericStds.h b/libSYS/include/genericStds.h
index 6adf689..fe4dc36 100644
--- a/libSYS/include/genericStds.h
+++ b/libSYS/include/genericStds.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSYS/include/machine_type.h b/libSYS/include/machine_type.h
index d04a8e8..4745e43 100644
--- a/libSYS/include/machine_type.h
+++ b/libSYS/include/machine_type.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSYS/include/wav_file.h b/libSYS/include/wav_file.h
index e9c4e70..2bc43b2 100644
--- a/libSYS/include/wav_file.h
+++ b/libSYS/include/wav_file.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -111,16 +111,36 @@ amm-info at iis.fraunhofer.de
 extern "C" {
 #endif
 
+#define SPEAKER_FRONT_LEFT             0x1
+#define SPEAKER_FRONT_RIGHT            0x2
+#define SPEAKER_FRONT_CENTER           0x4
+#define SPEAKER_LOW_FREQUENCY          0x8
+#define SPEAKER_BACK_LEFT              0x10
+#define SPEAKER_BACK_RIGHT             0x20
+#define SPEAKER_FRONT_LEFT_OF_CENTER   0x40
+#define SPEAKER_FRONT_RIGHT_OF_CENTER  0x80
+#define SPEAKER_BACK_CENTER            0x100
+#define SPEAKER_SIDE_LEFT              0x200
+#define SPEAKER_SIDE_RIGHT             0x400
+#define SPEAKER_TOP_CENTER             0x800
+#define SPEAKER_TOP_FRONT_LEFT         0x1000
+#define SPEAKER_TOP_FRONT_CENTER       0x2000
+#define SPEAKER_TOP_FRONT_RIGHT        0x4000
+#define SPEAKER_TOP_BACK_LEFT          0x8000
+#define SPEAKER_TOP_BACK_CENTER        0x10000
+#define SPEAKER_TOP_BACK_RIGHT         0x20000
+#define SPEAKER_RESERVED               0x80000000
+
 /*!
  * RIFF WAVE file struct.
  * For details see WAVE file format documentation (for example at http://www.wotsit.org).
  */
 typedef struct WAV_HEADER
 {
-  char  riffType[4];
+  char   riffType[4];
   UINT   riffSize;
-  char  waveType[4];
-  char  formatType[4];
+  char   waveType[4];
+  char   formatType[4];
   UINT   formatSize;
   USHORT compressionCode;
   USHORT numChannels;
@@ -128,7 +148,7 @@ typedef struct WAV_HEADER
   UINT   bytesPerSecond;
   USHORT blockAlign;
   USHORT bitsPerSample;
-  char  dataType[4];
+  char   dataType[4];
   UINT   dataSize;
 } WAV_HEADER;
 
@@ -136,6 +156,7 @@ struct WAV
 {
   WAV_HEADER header;
   FDKFILE *fp;
+  UINT channelMask;
 };
 
 typedef struct WAV *HANDLE_WAV;
diff --git a/libSYS/src/cmdl_parser.cpp b/libSYS/src/cmdl_parser.cpp
index bf020ee..24ec0d5 100644
--- a/libSYS/src/cmdl_parser.cpp
+++ b/libSYS/src/cmdl_parser.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -465,7 +465,7 @@ INT CheckArg(TEXTCHAR* arg, TEXTCHAR* str, UINT numArgs, TEXTCHAR type, TEXTCHAR
 
   for ( i = 0; i < _tcslen(arg); ++i )
     {
-      if ( (type == 'd') && !_istdigit(arg[i]) && arg[i] != 'x' )
+      if ( (type == 'd') && !_istdigit(arg[i]) && arg[i] != 'x' && arg[i] != '-')
         {
 #ifdef _UNICODE
           _ftprintf(stderr, _TEXT("\n\nError: Argument '%ls' for switch '%ls' is not a valid number.\n" ), arg, cur_str);
@@ -549,7 +549,7 @@ int IIS_ProcessCmdlList(const char* param_filename, int (*pFunction)(int, TEXTCH
       /* Skip consecutive blanks. */
       while (*line_ptr == ' ' && line_ptr < line+CMDL_MAX_STRLEN)
         line_ptr++;
-      /* Assign argument. TODO: maybe handle quotes */
+      /* Assign argument. */
       argv_ptr[argc] = line_ptr;
       /* Get pointer to next blank. */
       line_ptr = (char*)FDKstrchr(line_ptr, ' ');
diff --git a/libSYS/src/conv_string.cpp b/libSYS/src/conv_string.cpp
index f730337..28cbc83 100644
--- a/libSYS/src/conv_string.cpp
+++ b/libSYS/src/conv_string.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSYS/src/genericStds.cpp b/libSYS/src/genericStds.cpp
index 6bc4bd2..f0199fb 100644
--- a/libSYS/src/genericStds.cpp
+++ b/libSYS/src/genericStds.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -98,8 +98,8 @@ amm-info at iis.fraunhofer.de
 
 /* library info */
 #define SYS_LIB_VL0 1
-#define SYS_LIB_VL1 2
-#define SYS_LIB_VL2 2
+#define SYS_LIB_VL1 3
+#define SYS_LIB_VL2 4
 #define SYS_LIB_TITLE "System Integration Library"
 #define SYS_LIB_BUILD_DATE __DATE__
 #define SYS_LIB_BUILD_TIME __TIME__
diff --git a/libSYS/src/linux/FDK_stackload_linux.cpp b/libSYS/src/linux/FDK_stackload_linux.cpp
index 549d99d..82fa9ab 100644
--- a/libSYS/src/linux/FDK_stackload_linux.cpp
+++ b/libSYS/src/linux/FDK_stackload_linux.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSYS/src/linux/audio_linux.cpp b/libSYS/src/linux/audio_linux.cpp
index d99f876..835b27b 100644
--- a/libSYS/src/linux/audio_linux.cpp
+++ b/libSYS/src/linux/audio_linux.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSYS/src/linux/coresup_linux.cpp b/libSYS/src/linux/coresup_linux.cpp
index 49a042b..9e6fa85 100644
--- a/libSYS/src/linux/coresup_linux.cpp
+++ b/libSYS/src/linux/coresup_linux.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSYS/src/linux/genericStds_linux.cpp b/libSYS/src/linux/genericStds_linux.cpp
index f28def7..da375c0 100644
--- a/libSYS/src/linux/genericStds_linux.cpp
+++ b/libSYS/src/linux/genericStds_linux.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSYS/src/linux/uart_linux.cpp b/libSYS/src/linux/uart_linux.cpp
index 31e0419..dee5694 100644
--- a/libSYS/src/linux/uart_linux.cpp
+++ b/libSYS/src/linux/uart_linux.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSYS/src/mips/genericStds_mips.cpp b/libSYS/src/mips/genericStds_mips.cpp
index 98a30c3..e08ec48 100644
--- a/libSYS/src/mips/genericStds_mips.cpp
+++ b/libSYS/src/mips/genericStds_mips.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
diff --git a/libSYS/src/wav_file.cpp b/libSYS/src/wav_file.cpp
index 8764be9..58bcfaa 100644
--- a/libSYS/src/wav_file.cpp
+++ b/libSYS/src/wav_file.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2012 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -174,7 +174,6 @@ INT WAV_InputOpen (HANDLE_WAV *pWav, const char *filename)
           0x80, 0x00, 0x00, 0xaa, 0x00, 0x38, 0x9b, 0x71
       };
       USHORT extraFormatBytes, validBitsPerSample;
-      UINT channelMask;
       UCHAR guid[16];
       INT i;
 
@@ -184,7 +183,7 @@ INT WAV_InputOpen (HANDLE_WAV *pWav, const char *filename)
 
       if (extraFormatBytes >= 22) {
         FDKfread_EL(&(validBitsPerSample), 2, 1, wav->fp);
-        FDKfread_EL(&(channelMask), 4, 1, wav->fp);
+        FDKfread_EL(&(wav->channelMask), 4, 1, wav->fp);
         FDKfread_EL(&(guid), 16, 1, wav->fp);
 
         /* check for PCM GUID */
@@ -228,12 +227,12 @@ INT WAV_InputOpen (HANDLE_WAV *pWav, const char *filename)
 
     /* Error path */
 error:
-    if (wav->fp) {
-      FDKfclose(wav->fp);
-      wav->fp = NULL;
-    }
 
     if (wav) {
+      if (wav->fp) {
+        FDKfclose(wav->fp);
+        wav->fp = NULL;
+      }
       FDKfree(wav);
     }
 
@@ -388,6 +387,11 @@ INT WAV_OutputOpen(HANDLE_WAV *pWav, const char *outputFilename, INT sampleRate,
   HANDLE_WAV wav = (HANDLE_WAV)FDKcalloc(1, sizeof(struct WAV));
   UINT size = 0;
 
+  if (wav == NULL) {
+    FDKprintfErr("WAV_OutputOpen(): Unable to allocate WAV struct.\n");
+    goto bail;
+  }
+
   if (bitsPerSample != 16 && bitsPerSample != 24 && bitsPerSample != 32)
   {
       FDKprintfErr("WAV_OutputOpen(): Invalid argument (bitsPerSample).\n");
@@ -433,10 +437,12 @@ INT WAV_OutputOpen(HANDLE_WAV *pWav, const char *outputFilename, INT sampleRate,
   return 0;
 
 bail:
-  if (wav->fp)
-    FDKfclose(wav->fp);
-  if (wav)
+  if (wav) {
+    if (wav->fp) {
+      FDKfclose(wav->fp);
+    }
     FDKfree(wav);
+  }
 
   pWav = NULL;
 

-- 
fdk-aac packaging

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