Merge remote-tracking branch 'aosp/kitkat-release' into kitkat-merge

Conflicts:
	libAACenc/src/quantize.cpp

42 files changed, 902 insertions, 678 deletions

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@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..b1fe12e 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
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@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@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@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@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@iis.fraunhofer.de
 /*!
   \file
   \brief  frequency scale  
+  \author Tobias Chalupka
 */
 
 #include "sbrenc_freq_sca.h"
@@ -92,10 +93,10 @@ amm-info@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@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