SBR/AAC encoder updates, code clean up

* SBR-Encoder

   - Prevent noise level overflow in noise floor detection.
   - Saturate threshold calculation in transient detection.
     Modified file(s):
        libSBRenc/src/nf_est.cpp
        libSBRenc/src/sbr_encoder.cpp
        libSBRenc/src/tran_det.cpp

* AAC-Encoder

   - Expand input data range of GetInvInt() function. There was an encoder
     assert observed in non-default bitrate configuration.
     Modified file(s):
        libAACenc/src/aacenc_lib.cpp
        libAACenc/src/intensity.cpp
        libFDK/include/fixpoint_math.h
        libFDK/src/FDK_core.cpp
        libFDK/src/FDK_tools_rom.cpp

   - Make sure that the encoder is stable with regard to very low audio bandwidth
     confguration parameter value.
   - Fix lowdelay blending for low audio bandwidth.
     Modified file(s):
        libAACenc/src/aacenc.cpp
        libAACenc/src/aacenc_lib.cpp
        libAACenc/src/adj_thr.cpp
        libAACenc/src/psy_configuration.cpp
        libAACenc/src/psy_main.cpp

   - Disable pseudo surround flag in case metadata matrix mixdown index is
     present in program config element.
     Modified file(s):
        libAACenc/src/aacenc_lib.cpp

   - Enable variable bitrate mode in encoder api.
   - Add AACENC_PEAK_BITRATE parameter to encoder api.
   - Add AACENC_AUDIOMUXVER parameter to encoder api.
     Modified file(s):
        libAACenc/include/aacenc_lib.h
        libAACenc/src/aacenc.cpp
        libAACenc/src/aacenc.h
        libAACenc/src/aacenc_lib.cpp
        libAACenc/src/qc_main.cpp
        libMpegTPEnc/src/tpenc_latm.cpp
        libMpegTPEnc/src/version

* FDK-Sources

   - Code clean up. Remove unneeded pseudo audio object types and transport types.
     Modified file(s):
        libAACdec/src/aacdecoder.cpp
        libAACdec/src/aacdecoder_lib.cpp
        libAACenc/include/aacenc_lib.h
        libAACenc/src/aacenc.cpp
        libAACenc/src/aacenc_lib.cpp
        libFDK/src/FDK_tools_rom.cpp
        libMpegTPDec/src/tpdec_lib.cpp
        libMpegTPDec/src/version
        libMpegTPEnc/src/tpenc_latm.cpp
        libMpegTPEnc/src/version
        libSBRdec/src/sbrdecoder.cpp
        libSBRenc/src/sbr_encoder.cpp
        libSYS/include/FDK_audio.h
        libSYS/src/genericStds.cpp

Change-Id: I807a53cb7f48c9ee7563cb8da1d0c52221576ca6

25 files changed, 227 insertions, 215 deletions

diff --git a/documentation/aacEncoder.pdf b/documentation/aacEncoder.pdf
index e8420b1..efb1858 100644
--- a/documentation/aacEncoder.pdf
+++ b/documentation/aacEncoder.pdf
diff --git a/libAACdec/src/aacdecoder.cpp b/libAACdec/src/aacdecoder.cpp
index e19c501..8270f69 100644
--- a/libAACdec/src/aacdecoder.cpp
+++ b/libAACdec/src/aacdecoder.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -159,21 +159,6 @@ amm-info@iis.fraunhofer.de
 
 
 
-#define CAN_DO_PS(aot) \
-  ((aot) == AOT_AAC_LC \
-|| (aot) == AOT_SBR \
-|| (aot) == AOT_PS \
-|| (aot) == AOT_ER_BSAC \
-|| (aot) == AOT_DRM_AAC)
-
-#define IS_USAC(aot) \
-  ((aot) == AOT_USAC \
-|| (aot) == AOT_RSVD50)
-
-#define IS_LOWDELAY(aot) \
-  ((aot) == AOT_ER_AAC_LD \
-|| (aot) == AOT_ER_AAC_ELD)
-
 void CAacDecoder_SyncQmfMode(HANDLE_AACDECODER self)
 {
 
diff --git a/libAACdec/src/aacdecoder_lib.cpp b/libAACdec/src/aacdecoder_lib.cpp
index 94eb3c0..b501eb9 100644
--- a/libAACdec/src/aacdecoder_lib.cpp
+++ b/libAACdec/src/aacdecoder_lib.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -110,7 +110,7 @@ amm-info@iis.fraunhofer.de
 /* Decoder library info */
 #define AACDECODER_LIB_VL0 2
 #define AACDECODER_LIB_VL1 5
-#define AACDECODER_LIB_VL2 10
+#define AACDECODER_LIB_VL2 11
 #define AACDECODER_LIB_TITLE "AAC Decoder Lib"
 #ifdef __ANDROID__
 #define AACDECODER_LIB_BUILD_DATE ""
diff --git a/libAACenc/include/aacenc_lib.h b/libAACenc/include/aacenc_lib.h
index 8c0e1cd..828a917 100644
--- a/libAACenc/include/aacenc_lib.h
+++ b/libAACenc/include/aacenc_lib.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -897,11 +897,7 @@ typedef enum
                                                         This configuration can be used only with stereo input audio data.
                                                   - 23: MPEG-4 AAC Low-Delay.
                                                   - 39: MPEG-4 AAC Enhanced Low-Delay. Since there is no ::AUDIO_OBJECT_TYPE for ELD in
-                                                        combination with SBR defined, enable SBR explicitely by ::AACENC_SBR_MODE parameter.
-                                                  - 129: MPEG-2 AAC Low Complexity.
-                                                  - 132: MPEG-2 AAC Low Complexity with Spectral Band Replication (HE-AAC).
-                                                  - 156: MPEG-2 AAC Low Complexity with Spectral Band Replication and Parametric Stereo (HE-AAC v2).
-                                                         This configuration can be used only with stereo input audio data. */
+                                                        combination with SBR defined, enable SBR explicitely by ::AACENC_SBR_MODE parameter. */
 
   AACENC_BITRATE                  = 0x0101,  /*!< Total encoder bitrate. This parameter is mandatory and interacts with ::AACENC_BITRATEMODE.
                                                   - CBR: Bitrate in bits/second.
@@ -958,6 +954,16 @@ typedef enum
                                                   - 1 to fs/2: Frequency bandwidth in Hertz. (Experts only, better do not
                                                                touch this value to avoid degraded audio quality) */
 
+  AACENC_PEAK_BITRATE             = 0x0207,  /*!< Peak bitrate configuration parameter to adjust maximum bits per audio frame. Bitrate is in bits/second. 
+                                                  The peak bitrate will internally be limited to the chosen bitrate ::AACENC_BITRATE as lower limit
+                                                  and the number_of_effective_channels*6144 bit as upper limit.
+
+                                                  Setting the peak bitrate equal to ::AACENC_BITRATE does not necessarily mean that the audio frames
+                                                  will be of constant size. Since the peak bitate is in bits/second, the frame sizes can vary by
+                                                  one byte in one or the other direction over various frames. However, it is not recommended to reduce
+                                                  the peak pitrate to ::AACENC_BITRATE - it would disable the bitreservoir, which would affect the
+                                                  audio quality by a large amount. */
+
   AACENC_TRANSMUX                 = 0x0300,  /*!< Transport type to be used. See ::TRANSPORT_TYPE in FDK_audio.h. Following
                                                   types can be configured in encoder library:
                                                   - 0: raw access units
@@ -1023,6 +1029,11 @@ typedef enum
                                                   - ADTS: Maximum number of sub frames restricted to 4.
                                                   - LOAS/LATM: Maximum number of sub frames restricted to 2.*/
 
+  AACENC_AUDIOMUXVER              = 0x0304,  /*!< AudioMuxVersion to be used for LATM. (AudioMuxVersionA, currently not implemented):
+                                                  - 0: Default, no transmission of tara Buffer fullness, no ASC length and including actual latm Buffer fullnes.
+                                                  - 1: Transmission of tara Buffer fullness, ASC length and actual latm Buffer fullness.
+                                                  - 2: Transmission of tara Buffer fullness, ASC length and maximum level of latm Buffer fullness. */
+
   AACENC_PROTECTION               = 0x0306,  /*!< Configure protection in tranpsort layer:
                                                   - 0: No protection. (default)
                                                   - 1: CRC active for ADTS bitstream format. */
diff --git a/libAACenc/src/aacenc.cpp b/libAACenc/src/aacenc.cpp
index c47e882..5e8c08d 100644
--- a/libAACenc/src/aacenc.cpp
+++ b/libAACenc/src/aacenc.cpp
@@ -107,6 +107,39 @@ amm-info@iis.fraunhofer.de
 
 #define MIN_BUFSIZE_PER_EFF_CHAN 6144
 
+INT FDKaacEnc_CalcBitsPerFrame(
+        const INT bitRate,
+        const INT frameLength,
+        const INT samplingRate
+        )
+{
+    int shift = 0;
+    while ((frameLength & ~((1 << (shift + 1)) - 1)) == frameLength
+        && (samplingRate & ~((1 << (shift + 1)) - 1)) == samplingRate)
+    {
+      shift++;
+    }
+
+    return (bitRate*(frameLength>>shift)) / (samplingRate>>shift);
+}
+
+INT FDKaacEnc_CalcBitrate(
+        const INT bitsPerFrame,
+        const INT frameLength,
+        const INT samplingRate
+        )
+{
+    int shift = 0;
+    while ((frameLength & ~((1 << (shift + 1)) - 1)) == frameLength
+        && (samplingRate & ~((1 << (shift + 1)) - 1)) == samplingRate)
+    {
+      shift++;
+    }
+
+    return (bitsPerFrame * (samplingRate>>shift)) / ( frameLength>>shift) ;
+
+}
+
 static AAC_ENCODER_ERROR FDKaacEnc_InitCheckAncillary(INT bitRate,
                                                       INT framelength,
                                                       INT ancillaryRate,
@@ -220,21 +253,19 @@ INT FDKaacEnc_GetVBRBitrate(INT bitrateMode, CHANNEL_MODE channelMode)
 /**
  * \brief  Convert encoder bitreservoir value for transport library.
  *
- * \param bitrateMode           Bitratemode used in current encoder instance. Se ::AACENC_BITRATE_MODE
- * \param bitresTotal           Encoder bitreservoir level in bits.
+ * \param hAacEnc               Encoder handle
  *
  * \return  Corrected bitreservoir level used in transport library.
  */
 static INT FDKaacEnc_EncBitresToTpBitres(
-        const AACENC_BITRATE_MODE bitrateMode,
-        const INT                 bitresTotal
+        const HANDLE_AAC_ENC      hAacEnc
         )
 {
   INT transporBitreservoir = 0;
 
-  switch (bitrateMode) {
+  switch (hAacEnc->bitrateMode) {
     case AACENC_BR_MODE_CBR:
-      transporBitreservoir = bitresTotal; /* encoder bitreservoir level */
+      transporBitreservoir = hAacEnc->qcKernel->bitResTot; /* encoder bitreservoir level */
       break;
     case AACENC_BR_MODE_VBR_1:
     case AACENC_BR_MODE_VBR_2:
@@ -253,6 +284,10 @@ static INT FDKaacEnc_EncBitresToTpBitres(
       FDK_ASSERT(0);
   }
 
+  if (hAacEnc->config->audioMuxVersion==2) {
+    transporBitreservoir = MIN_BUFSIZE_PER_EFF_CHAN * hAacEnc->channelMapping.nChannelsEff;
+  }
+
   return transporBitreservoir;
 }
 
@@ -289,6 +324,7 @@ void FDKaacEnc_AacInitDefaultConfig(AACENC_CONFIG *config)
     config->minBitsPerFrame = -1;                   /* minum number of bits in each AU */
     config->maxBitsPerFrame = -1;                   /* minum number of bits in each AU */
     config->bitreservoir    = -1;                   /* default, uninitialized value */
+    config->audioMuxVersion = -1;                   /* audio mux version not configured */
 
     /* init tabs in fixpoint_math */
     InitLdInt();
@@ -564,7 +600,10 @@ AAC_ENCODER_ERROR FDKaacEnc_Initialize(HANDLE_AAC_ENC      hAacEnc,
       qcInit.averageBits     = (averageBitsPerFrame+7)&~7;
       qcInit.bitRes          = MIN_BUFSIZE_PER_EFF_CHAN*cm->nChannelsEff;
       qcInit.maxBits         = MIN_BUFSIZE_PER_EFF_CHAN*cm->nChannelsEff;
-      qcInit.minBits         = 0;
+      qcInit.maxBits         = (config->maxBitsPerFrame!=-1) ? fixMin(qcInit.maxBits, config->maxBitsPerFrame) : qcInit.maxBits;
+      qcInit.maxBits         = fixMax(qcInit.maxBits, (averageBitsPerFrame+7)&~7);
+      qcInit.minBits         = (config->minBitsPerFrame!=-1) ? config->minBitsPerFrame : 0;
+      qcInit.minBits         = fixMin(qcInit.minBits, averageBitsPerFrame&~7);
   }
   else
   {
@@ -575,9 +614,11 @@ AAC_ENCODER_ERROR FDKaacEnc_Initialize(HANDLE_AAC_ENC      hAacEnc,
 
       qcInit.maxBits         = fixMin(MIN_BUFSIZE_PER_EFF_CHAN*cm->nChannelsEff, ((averageBitsPerFrame+7)&~7)+qcInit.bitRes);
       qcInit.maxBits         = (config->maxBitsPerFrame!=-1) ? fixMin(qcInit.maxBits, config->maxBitsPerFrame) : qcInit.maxBits;
+      qcInit.maxBits         = fixMin(MIN_BUFSIZE_PER_EFF_CHAN*cm->nChannelsEff, fixMax(qcInit.maxBits, (averageBitsPerFrame+7+8)&~7));
 
       qcInit.minBits         = fixMax(0, ((averageBitsPerFrame-1)&~7)-qcInit.bitRes-transportEnc_GetStaticBits(hTpEnc, ((averageBitsPerFrame+7)&~7)+qcInit.bitRes));
       qcInit.minBits         = (config->minBitsPerFrame!=-1) ? fixMax(qcInit.minBits, config->minBitsPerFrame) : qcInit.minBits;
+      qcInit.minBits         = fixMin(qcInit.minBits, (averageBitsPerFrame - transportEnc_GetStaticBits(hTpEnc, qcInit.maxBits))&~7);
   }
 
   qcInit.sampleRate          = config->sampleRate;
@@ -585,11 +626,9 @@ AAC_ENCODER_ERROR FDKaacEnc_Initialize(HANDLE_AAC_ENC      hAacEnc,
   qcInit.nSubFrames          = config->nSubFrames;
   qcInit.padding.paddingRest = config->sampleRate;
 
-  /* Calc meanPe */
-  bw_ratio = fDivNorm((FIXP_DBL)hAacEnc->bandwidth90dB, (FIXP_DBL)(config->sampleRate>>1), &qbw);
-  qbw = DFRACT_BITS-1-qbw;
-  /* qcInit.meanPe = 10.0f * FRAME_LEN_LONG * hAacEnc->bandwidth90dB/(config->sampleRate/2.0f); */
-  qcInit.meanPe = fMult(bw_ratio, (FIXP_DBL)((10*config->framelength)<<16)) >> (qbw-15);
+  /* Calc meanPe: qcInit.meanPe = 10.0f * FRAME_LEN_LONG * hAacEnc->bandwidth90dB/(config->sampleRate/2.0f); */
+  bw_ratio = fDivNorm((FIXP_DBL)(10*config->framelength*hAacEnc->bandwidth90dB), (FIXP_DBL)(config->sampleRate), &qbw);
+  qcInit.meanPe = FDKmax((INT)scaleValue(bw_ratio, qbw+1-(DFRACT_BITS-1)), 1);
 
   /* Calc maxBitFac */
   mbfac = fDivNorm((MIN_BUFSIZE_PER_EFF_CHAN-744)*cm->nChannelsEff, qcInit.averageBits/qcInit.nSubFrames, &qmbfac);
@@ -651,23 +690,7 @@ AAC_ENCODER_ERROR FDKaacEnc_Initialize(HANDLE_AAC_ENC      hAacEnc,
   if (ErrorStatus != AAC_ENC_OK)
     goto bail;
 
-  /* Map virtual aot's to intern aot used in bitstream writer. */
-  switch (hAacEnc->config->audioObjectType) {
-    case AOT_MP2_AAC_LC:
-    case AOT_DABPLUS_AAC_LC:
-      hAacEnc->aot = AOT_AAC_LC;
-      break;
-    case AOT_MP2_SBR:
-    case AOT_DABPLUS_SBR:
-      hAacEnc->aot = AOT_SBR;
-      break;
-    case AOT_MP2_PS:
-    case AOT_DABPLUS_PS:
-      hAacEnc->aot = AOT_PS;
-      break;
-    default:
-      hAacEnc->aot = hAacEnc->config->audioObjectType;
-  }
+  hAacEnc->aot = hAacEnc->config->audioObjectType;
 
   /* common things */
 
@@ -932,7 +955,7 @@ AAC_ENCODER_ERROR FDKaacEnc_EncodeFrame( HANDLE_AAC_ENC       hAacEnc,
               transportEnc_WriteAccessUnit(
                     hTpEnc,
                     totalBits,
-                    FDKaacEnc_EncBitresToTpBitres(hAacEnc->bitrateMode, hAacEnc->qcKernel->bitResTot),
+                    FDKaacEnc_EncBitresToTpBitres(hAacEnc),
                     cm->nChannelsEff);
 
               /* write bitstream */
diff --git a/libAACenc/src/aacenc.h b/libAACenc/src/aacenc.h
index 87fc3e8..61520de 100644
--- a/libAACenc/src/aacenc.h
+++ b/libAACenc/src/aacenc.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -204,6 +204,8 @@ struct AACENC_CONFIG {
   INT   maxBitsPerFrame;        /* maximum number of bits in AU */
   INT   bitreservoir;           /* size of bitreservoir */
 
+  INT   audioMuxVersion;        /* audio mux version in loas/latm transport format */
+
   UINT  sbrRatio;               /* sbr sampling rate ratio: dual- or single-rate */
 
   UCHAR useTns;                 /* flag: use temporal noise shaping */
@@ -223,6 +225,36 @@ typedef struct {
 typedef struct AAC_ENC *HANDLE_AAC_ENC;
 
 /**
+ * \brief Calculate framesize in bits for given bit rate, frame length and sampling rate.
+ *
+ * \param bitRate               Ttarget bitrate in bits per second.
+ * \param frameLength           Number of audio samples in one frame.
+ * \param samplingRate          Sampling rate in Hz.
+ *
+ * \return                      Framesize in bits per frame.
+*/
+INT FDKaacEnc_CalcBitsPerFrame(
+        const INT bitRate,
+        const INT frameLength,
+        const INT samplingRate
+        );
+
+/**
+ * \brief Calculate bitrate in bits per second for given framesize, frame length and sampling rate.
+ *
+ * \param bitsPerFrame          Framesize in bits per frame.
+ * \param frameLength           Number of audio samples in one frame.
+ * \param samplingRate          Sampling rate in Hz.
+ *
+ * \return                      Bitrate in bits per second.
+*/
+INT FDKaacEnc_CalcBitrate(
+        const INT bitsPerFrame,
+        const INT frameLength,
+        const INT samplingRate
+        );
+
+/**
  * \brief Limit given bit rate to a valid value
  * \param hTpEnc transport encoder handle
  * \param coreSamplingRate the sample rate to be used for the AAC encoder
diff --git a/libAACenc/src/aacenc_lib.cpp b/libAACenc/src/aacenc_lib.cpp
index 02f83bf..9a3b214 100644
--- a/libAACenc/src/aacenc_lib.cpp
+++ b/libAACenc/src/aacenc_lib.cpp
@@ -98,7 +98,7 @@ amm-info@iis.fraunhofer.de
 /* Encoder library info */
 #define AACENCODER_LIB_VL0 3
 #define AACENCODER_LIB_VL1 4
-#define AACENCODER_LIB_VL2 14
+#define AACENCODER_LIB_VL2 19
 #define AACENCODER_LIB_TITLE "AAC Encoder"
 #ifdef __ANDROID__
 #define AACENCODER_LIB_BUILD_DATE ""
@@ -153,6 +153,7 @@ typedef struct {
     UINT              userAfterburner;
     UINT              userFramelength;
     UINT              userAncDataRate;
+    UINT              userPeakBitrate;
 
     UCHAR             userTns;               /*!< Use TNS coding. */
     UCHAR             userPns;               /*!< Use PNS coding. */
@@ -326,10 +327,7 @@ static inline INT isSbrActive(const HANDLE_AACENC_CONFIG hAacConfig)
 {
     INT sbrUsed = 0;
 
-    if ( (hAacConfig->audioObjectType==AOT_SBR)         || (hAacConfig->audioObjectType==AOT_PS)
-      || (hAacConfig->audioObjectType==AOT_MP2_SBR)     || (hAacConfig->audioObjectType==AOT_MP2_PS)
-      || (hAacConfig->audioObjectType==AOT_DABPLUS_SBR) || (hAacConfig->audioObjectType==AOT_DABPLUS_PS)
-      || (hAacConfig->audioObjectType==AOT_DRM_SBR)     || (hAacConfig->audioObjectType==AOT_DRM_MPEG_PS) )
+    if ( (hAacConfig->audioObjectType==AOT_SBR) || (hAacConfig->audioObjectType==AOT_PS) )
     {
         sbrUsed = 1;
     }
@@ -345,10 +343,7 @@ static inline INT isPsActive(const AUDIO_OBJECT_TYPE audioObjectType)
 {
     INT psUsed = 0;
 
-    if ( (audioObjectType==AOT_PS)
-      || (audioObjectType==AOT_MP2_PS)
-      || (audioObjectType==AOT_DABPLUS_PS)
-      || (audioObjectType==AOT_DRM_MPEG_PS) )
+    if ( (audioObjectType==AOT_PS) )
     {
         psUsed = 1;
     }
@@ -373,8 +368,7 @@ static SBR_PS_SIGNALING getSbrSignalingMode(
     sbrSignaling = SIG_IMPLICIT; /* default: implicit signaling */
   }
 
-  if ((audioObjectType==AOT_AAC_LC)     || (audioObjectType==AOT_SBR)     || (audioObjectType==AOT_PS)    ||
-      (audioObjectType==AOT_MP2_AAC_LC) || (audioObjectType==AOT_MP2_SBR) || (audioObjectType==AOT_MP2_PS) ) {
+  if ( (audioObjectType==AOT_AAC_LC) || (audioObjectType==AOT_SBR) || (audioObjectType==AOT_PS) ) {
     switch (transportType) {
       case TT_MP4_ADIF:
       case TT_MP4_ADTS:
@@ -430,22 +424,7 @@ static void FDKaacEnc_MapConfig(
 
   cc->flags = 0;
 
-  /* Map virtual aot to transport aot. */
-  switch (hAacConfig->audioObjectType) {
-    case AOT_MP2_AAC_LC:
-      transport_AOT = AOT_AAC_LC;
-      break;
-    case AOT_MP2_SBR:
-      transport_AOT = AOT_SBR;
-      cc->flags |= CC_SBR;
-     break;
-    case AOT_MP2_PS:
-      transport_AOT = AOT_PS;
-      cc->flags |= CC_SBR;
-      break;
-    default:
-      transport_AOT = hAacConfig->audioObjectType;
-  }
+  transport_AOT = hAacConfig->audioObjectType;
 
   if (hAacConfig->audioObjectType == AOT_ER_AAC_ELD) {
     cc->flags |= (hAacConfig->syntaxFlags & AC_SBR_PRESENT) ? CC_SBR : 0;
@@ -511,16 +490,7 @@ static void FDKaacEnc_MapConfig(
   cc->samplingRate    = hAacConfig->sampleRate;
 
   /* Mpeg-4 signaling for transport library. */
-  switch ( hAacConfig->audioObjectType ) {
-    case AOT_MP2_AAC_LC:
-    case AOT_MP2_SBR:
-    case AOT_MP2_PS:
-      cc->flags &= ~CC_MPEG_ID; /* Required for ADTS. */
-      cc->extAOT = AOT_NULL_OBJECT;
-      break;
-    default:
-      cc->flags |= CC_MPEG_ID;
-  }
+  cc->flags |= CC_MPEG_ID;
 
   /* ER-tools signaling. */
   cc->flags     |= (hAacConfig->syntaxFlags & AC_ER_VCB11) ? CC_VCB11 : 0;
@@ -585,6 +555,7 @@ AAC_ENCODER_ERROR aacEncDefaultConfig(HANDLE_AACENC_CONFIG hAacConfig,
     config->userChannelMode = hAacConfig->channelMode;
     config->userBitrate     = hAacConfig->bitRate;
     config->userBitrateMode = hAacConfig->bitrateMode;
+    config->userPeakBitrate = (UINT)-1;
     config->userBandwidth   = hAacConfig->bandWidth;
     config->userTns         = hAacConfig->useTns;
     config->userPns         = hAacConfig->usePns;
@@ -792,12 +763,15 @@ AACENC_ERROR FDKaacEnc_AdjustEncSettings(HANDLE_AACENCODER hAacEncoder,
     hAacConfig->syntaxFlags     = 0;
     hAacConfig->epConfig        = -1;
 
+    if (config->userTpType==TT_MP4_LATM_MCP1 || config->userTpType==TT_MP4_LATM_MCP0 || config->userTpType==TT_MP4_LOAS) {
+      hAacConfig->audioMuxVersion = config->userTpAmxv;
+    }
+    else {
+      hAacConfig->audioMuxVersion = -1;
+    }
+
     /* Adapt internal AOT when necessary. */
     switch ( hAacConfig->audioObjectType ) {
-      case AOT_MP2_AAC_LC:
-      case AOT_MP2_SBR:
-      case AOT_MP2_PS:
-          hAacConfig->usePns = 0;
       case AOT_AAC_LC:
       case AOT_SBR:
       case AOT_PS:
@@ -884,6 +858,18 @@ AACENC_ERROR FDKaacEnc_AdjustEncSettings(HANDLE_AACENCODER hAacEncoder,
         }
     }
 
+    if ((hAacConfig->bitrateMode >= 0) && (hAacConfig->bitrateMode <= 5)) {
+      if ((INT)config->userPeakBitrate != -1) {
+        hAacConfig->maxBitsPerFrame = (FDKaacEnc_CalcBitsPerFrame(fMax(hAacConfig->bitRate, (INT)config->userPeakBitrate), hAacConfig->framelength, hAacConfig->sampleRate) + 7)&~7;
+      }
+      else {
+        hAacConfig->maxBitsPerFrame = -1;
+      }
+      if (hAacConfig->audioMuxVersion==2) {
+        hAacConfig->minBitsPerFrame = fMin(32*8, FDKaacEnc_CalcBitsPerFrame(hAacConfig->bitRate, hAacConfig->framelength, hAacConfig->sampleRate))&~7;
+      }
+    }
+
     /* Initialize SBR parameters */
     if ( (hAacConfig->audioObjectType==AOT_ER_AAC_ELD)
       && (config->userSbrEnabled == (UCHAR)-1) && (config->userSbrRatio==0) )
@@ -1139,7 +1125,7 @@ static AACENC_ERROR aacEncInit(HANDLE_AACENCODER  hAacEncoder,
                 hAacConfig);
 
         /* create flags for transport encoder */
-        if (config->userTpAmxv == 1) {
+        if (config->userTpAmxv != 0) {
             flags |= TP_FLAG_LATM_AMV;
         }
         /* Clear output buffer */
@@ -1569,7 +1555,7 @@ AACENC_ERROR aacEncEncode(
           && ((hAacEncoder->extParam.userChannelMode==MODE_1_2_2)||(hAacEncoder->extParam.userChannelMode==MODE_1_2_2_1)) )
         {
           /* Set matrix mixdown coefficient. */
-          UINT pceValue = (UINT)( (1<<3) | ((matrix_mixdown_idx&0x3)<<1) | 1 );
+          UINT pceValue = (UINT)( (0<<3) | ((matrix_mixdown_idx&0x3)<<1) | 1 );
           if (hAacEncoder->extParam.userPceAdditions != pceValue) {
             hAacEncoder->extParam.userPceAdditions = pceValue;
             hAacEncoder->InitFlags |= AACENC_INIT_TRANSPORT;
@@ -1785,19 +1771,16 @@ AACENC_ERROR aacEncoder_SetParam(
             /* check if AOT matches the allocated modules */
             switch ( value ) {
               case AOT_PS:
-              case AOT_MP2_PS:
                 if (!(hAacEncoder->encoder_modis & (ENC_MODE_FLAG_PS))) {
                   err = AACENC_INVALID_CONFIG;
                   goto bail;
                 }
               case AOT_SBR:
-              case AOT_MP2_SBR:
                 if (!(hAacEncoder->encoder_modis & (ENC_MODE_FLAG_SBR))) {
                   err = AACENC_INVALID_CONFIG;
                   goto bail;
                 }
               case AOT_AAC_LC:
-              case AOT_MP2_AAC_LC:
               case AOT_ER_AAC_LD:
               case AOT_ER_AAC_ELD:
                 if (!(hAacEncoder->encoder_modis & (ENC_MODE_FLAG_AAC))) {
@@ -1823,6 +1806,7 @@ AACENC_ERROR aacEncoder_SetParam(
         if (settings->userBitrateMode != value) {
             switch ( value ) {
               case 0:
+              case 1: case 2: case 3: case 4: case 5:
               case 8:
                 settings->userBitrateMode = value;
                 hAacEncoder->InitFlags |= AACENC_INIT_CONFIG | AACENC_INIT_TRANSPORT;
@@ -1973,6 +1957,16 @@ AACENC_ERROR aacEncoder_SetParam(
             hAacEncoder->InitFlags |= AACENC_INIT_TRANSPORT;
         }
         break;
+    case AACENC_AUDIOMUXVER:
+        if (settings->userTpAmxv != value) {
+            if ( !((value==0) || (value==1) || (value==2)) ) {
+                err = AACENC_INVALID_CONFIG;
+                break;
+            }
+            settings->userTpAmxv = value;
+            hAacEncoder->InitFlags |= AACENC_INIT_TRANSPORT;
+        }
+        break;
     case AACENC_TPSUBFRAMES:
         if (settings->userTpNsubFrames != value) {
             if (! ( (value>=1) && (value<=4) ) ) {
@@ -2006,6 +2000,12 @@ AACENC_ERROR aacEncoder_SetParam(
             hAacEncoder->InitFlags |= AACENC_INIT_CONFIG;
         }
         break;
+    case AACENC_PEAK_BITRATE:
+        if (settings->userPeakBitrate != value) {
+            settings->userPeakBitrate = value;
+            hAacEncoder->InitFlags |= AACENC_INIT_CONFIG | AACENC_INIT_TRANSPORT;
+        }
+        break;
     default:
       err = AACENC_UNSUPPORTED_PARAMETER;
       break;
@@ -2076,6 +2076,9 @@ UINT aacEncoder_GetParam(
     case AACENC_HEADER_PERIOD:
         value = (UINT)hAacEncoder->coderConfig.headerPeriod;
         break;
+    case AACENC_AUDIOMUXVER:
+        value = (UINT)hAacEncoder->aacConfig.audioMuxVersion;
+        break;
     case AACENC_TPSUBFRAMES:
         value = (UINT)settings->userTpNsubFrames;
         break;
@@ -2088,6 +2091,12 @@ UINT aacEncoder_GetParam(
     case AACENC_METADATA_MODE:
         value = (hAacEncoder->metaDataAllowed==0) ? 0 : (UINT)settings->userMetaDataMode;
         break;
+    case AACENC_PEAK_BITRATE:
+        value = (UINT)-1; /* peak bitrate parameter is meaningless */
+        if ( ((INT)hAacEncoder->extParam.userPeakBitrate!=-1) ) {
+          value = (UINT)(fMax((INT)hAacEncoder->extParam.userPeakBitrate, hAacEncoder->aacConfig.bitRate)); /* peak bitrate parameter is in use */
+        }
+        break;
     default:
       //err = MPS_INVALID_PARAMETER;
       break;
diff --git a/libAACenc/src/adj_thr.cpp b/libAACenc/src/adj_thr.cpp
index 69a70b4..c318843 100644
--- a/libAACenc/src/adj_thr.cpp
+++ b/libAACenc/src/adj_thr.cpp
@@ -2138,7 +2138,7 @@ static FIXP_DBL FDKaacEnc_bitresCalcBitFac(const INT       bitresBits,
         bresParam->clipSpendLow, bresParam->clipSpendHigh,
         bresParam->minBitSpend, bresParam->maxBitSpend, bitspend_slope);
 
-    pe_pers = fDivNorm(pex - adjThrChan->peMin, adjThrChan->peMax - adjThrChan->peMin);
+    pe_pers = (pex > adjThrChan->peMin) ? fDivNorm(pex - adjThrChan->peMin, adjThrChan->peMax - adjThrChan->peMin) : 0;
     tmp_fix = fMult(((FIXP_DBL)bitSpend + (FIXP_DBL)bitSave), pe_pers);
     bitresFac_fix = (UNITY>>1) - ((FIXP_DBL)bitSave>>1) + (tmp_fix>>1); qbres = (DFRACT_BITS-2);
 
diff --git a/libAACenc/src/intensity.cpp b/libAACenc/src/intensity.cpp
index 6d807f7..b45b27b 100644
--- a/libAACenc/src/intensity.cpp
+++ b/libAACenc/src/intensity.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -325,7 +325,6 @@ FDKaacEnc_prepareIntensityDecision(const FIXP_DBL    *sfbEnergyLeft,
 
       channelCorr[sfb + sfboffs] = FL2FXCONST_DBL(0.0f);
 
-      FDK_ASSERT(50 >= 49);
       /* max width of scalefactorband is 96; width's are always even */
       /* inv_n is scaled with factor 2 to compensate fMultDiv2() in subsequent loops */
       inv_n = GetInvInt((sfbOffset[sfb + sfboffs + 1] - sfbOffset[sfb + sfboffs])>>1);
diff --git a/libAACenc/src/psy_configuration.cpp b/libAACenc/src/psy_configuration.cpp
index 4393fa1..9a72c68 100644
--- a/libAACenc/src/psy_configuration.cpp
+++ b/libAACenc/src/psy_configuration.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -634,13 +634,14 @@ AAC_ENCODER_ERROR FDKaacEnc_InitPsyConfiguration(INT   bitrate,
         if (psyConf->sfbOffset[sfb] >= psyConf->lowpassLine)
             break;
     }
-    psyConf->sfbActive = sfb;
+    psyConf->sfbActive = FDKmax(sfb, 1);
 
     for (sfb = 0; sfb < psyConf->sfbCnt; sfb++){
         if (psyConf->sfbOffset[sfb] >= psyConf->lowpassLineLFE)
             break;
     }
     psyConf->sfbActiveLFE = sfb;
+    psyConf->sfbActive = FDKmax(psyConf->sfbActive, psyConf->sfbActiveLFE);
 
     /* calculate minSnr */
     FDKaacEnc_initMinSnr(bitrate,
diff --git a/libAACenc/src/psy_main.cpp b/libAACenc/src/psy_main.cpp
index 1c593f8..3cc9438 100644
--- a/libAACenc/src/psy_main.cpp
+++ b/libAACenc/src/psy_main.cpp
@@ -621,7 +621,7 @@ AAC_ENCODER_ERROR FDKaacEnc_psyMain(INT                 channels,
           FDKmemclear(&psyData[ch]->mdctSpectrum[psyData[ch]->lowpassLine+wOffset],
                       (windowLength[ch]-psyData[ch]->lowpassLine)*sizeof(FIXP_DBL));
 
-          if (hPsyConfLong->filterbank != FB_LC) {
+          if ( (hPsyConfLong->filterbank != FB_LC) && (psyData[ch]->lowpassLine >= FADE_OUT_LEN) ) {
             /* Do blending to reduce gibbs artifacts */
             for (int i=0; i<FADE_OUT_LEN; i++) {
               psyData[ch]->mdctSpectrum[psyData[ch]->lowpassLine+wOffset - FADE_OUT_LEN + i] = fMult(psyData[ch]->mdctSpectrum[psyData[ch]->lowpassLine+wOffset - FADE_OUT_LEN + i], fadeOutFactor[i]);
diff --git a/libAACenc/src/qc_main.cpp b/libAACenc/src/qc_main.cpp
index 278bcb2..7503309 100644
--- a/libAACenc/src/qc_main.cpp
+++ b/libAACenc/src/qc_main.cpp
@@ -1248,6 +1248,8 @@ AAC_ENCODER_ERROR FDKaacEnc_updateFillBits(CHANNEL_MAPPING*          cm,
     case QCDATA_BR_MODE_VBR_4:
     case QCDATA_BR_MODE_VBR_5:
       qcOut[0]->totFillBits = (qcOut[0]->grantedDynBits - qcOut[0]->usedDynBits)&7; /* precalculate alignment bits */
+      qcOut[0]->totalBits = qcOut[0]->staticBits + qcOut[0]->usedDynBits + qcOut[0]->totFillBits + qcOut[0]->elementExtBits + qcOut[0]->globalExtBits;
+      qcOut[0]->totFillBits += ( fixMax(0, qcKernel->minBitsPerFrame - qcOut[0]->totalBits) + 7) & ~7;
       break;
 
     case QCDATA_BR_MODE_CBR:
@@ -1257,6 +1259,8 @@ AAC_ENCODER_ERROR FDKaacEnc_updateFillBits(CHANNEL_MAPPING*          cm,
       /* processing fill-bits */
       INT deltaBitRes = qcOut[0]->grantedDynBits - qcOut[0]->usedDynBits ;
       qcOut[0]->totFillBits = fixMax((deltaBitRes&7), (deltaBitRes - (fixMax(0,bitResSpace-7)&~7)));
+      qcOut[0]->totalBits = qcOut[0]->staticBits + qcOut[0]->usedDynBits + qcOut[0]->totFillBits + qcOut[0]->elementExtBits + qcOut[0]->globalExtBits;
+      qcOut[0]->totFillBits += ( fixMax(0, qcKernel->minBitsPerFrame - qcOut[0]->totalBits) + 7) & ~7;
       break;
   } /* switch (qcKernel->bitrateMode) */
 
diff --git a/libFDK/include/fixpoint_math.h b/libFDK/include/fixpoint_math.h
index df141d3..88e2f09 100644
--- a/libFDK/include/fixpoint_math.h
+++ b/libFDK/include/fixpoint_math.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -438,11 +438,11 @@ inline FIXP_DBL fAddSaturate(const FIXP_DBL a, const FIXP_DBL b)
 
 /*****************************************************************************
 
- array for 1/n, n=1..50
+ array for 1/n, n=1..80
 
 ****************************************************************************/
 
-  extern const FIXP_DBL invCount[50];
+  extern const FIXP_DBL invCount[80];
 
   LNK_SECTION_INITCODE
   inline void InitInvInt(void) {}
@@ -450,14 +450,14 @@ inline FIXP_DBL fAddSaturate(const FIXP_DBL a, const FIXP_DBL b)
 
 /**
  * \brief Calculate the value of 1/i where i is a integer value. It supports
- *        input values from 1 upto 50.
+ *        input values from 1 upto 80.
  * \param intValue Integer input value.
  * \param FIXP_DBL representation of 1/intValue
  */
 inline FIXP_DBL GetInvInt(int intValue)
 {
-  FDK_ASSERT((intValue > 0) && (intValue < 50));
-  FDK_ASSERT(intValue<50);
+  FDK_ASSERT((intValue > 0) && (intValue < 80));
+  FDK_ASSERT(intValue<80);
 	return invCount[intValue];
 }
 
diff --git a/libFDK/src/FDK_core.cpp b/libFDK/src/FDK_core.cpp
index cccb925..8fa5596 100644
--- a/libFDK/src/FDK_core.cpp
+++ b/libFDK/src/FDK_core.cpp
@@ -93,7 +93,7 @@ amm-info@iis.fraunhofer.de
 /* FDK tools library info */
 #define FDK_TOOLS_LIB_VL0 2
 #define FDK_TOOLS_LIB_VL1 3
-#define FDK_TOOLS_LIB_VL2 3
+#define FDK_TOOLS_LIB_VL2 4
 #define FDK_TOOLS_LIB_TITLE "FDK Tools"
 #ifdef __ANDROID__
 #define FDK_TOOLS_LIB_BUILD_DATE ""
diff --git a/libFDK/src/FDK_tools_rom.cpp b/libFDK/src/FDK_tools_rom.cpp
index c6e517e..49f0ee1 100644
--- a/libFDK/src/FDK_tools_rom.cpp
+++ b/libFDK/src/FDK_tools_rom.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -1902,7 +1902,7 @@ const USHORT sqrt_tab[49]={
 0xb504};
 
 LNK_SECTION_CONSTDATA_L1
-const FIXP_DBL invCount[50]=  /* This could be 16-bit wide */
+const FIXP_DBL invCount[80]=  /* This could be 16-bit wide */
 {
     0x00000000, 0x7fffffff, 0x40000000, 0x2aaaaaab, 0x20000000,
     0x1999999a, 0x15555555, 0x12492492, 0x10000000, 0x0e38e38e,
@@ -1913,7 +1913,13 @@ const FIXP_DBL invCount[50]=  /* This could be 16-bit wide */
     0x04444444, 0x04210842, 0x04000000, 0x03e0f83e, 0x03c3c3c4,
     0x03a83a84, 0x038e38e4, 0x03759f23, 0x035e50d8, 0x03483483,
     0x03333333, 0x031f3832, 0x030c30c3, 0x02fa0be8, 0x02e8ba2f,
-    0x02d82d83, 0x02c8590b, 0x02b93105, 0x02aaaaab, 0x029cbc15
+    0x02d82d83, 0x02c8590b, 0x02b93105, 0x02aaaaab, 0x029cbc15,
+    0x028f5c29, 0x02828283, 0x02762762, 0x026a439f, 0x025ed098,
+    0x0253c825, 0x02492492, 0x023ee090, 0x0234f72c, 0x022b63cc,
+    0x02222222, 0x02192e2a, 0x02108421, 0x02082082, 0x02000000,
+    0x01f81f82, 0x01f07c1f, 0x01e9131b, 0x01e1e1e2, 0x01dae607,
+    0x01d41d42, 0x01cd8569, 0x01c71c72, 0x01c0e070, 0x01bacf91,
+    0x01b4e81b, 0x01af286c, 0x01a98ef6, 0x01a41a42, 0x019ec8e9
 };
 
 
@@ -2033,19 +2039,6 @@ static const element_list_t node_aac_cpe = {
   { &node_aac_cpe0, &node_aac_cpe1 }
 };
 
-#define el_mpegsres_sce &el_aac_sce[2]
-
-static const element_list_t node_mpegsres_sce = {
-  el_mpegsres_sce,
-  { NULL, NULL }
-};
-
-static const element_list_t node_mpegsres_cpe = {
-  el_aac_cpe1,
-  { NULL, NULL }
-};
-
-
 /*
  * AOT C- {17,23}
  * epConfig = 0,1
@@ -2424,13 +2417,6 @@ const element_list_t * getBitstreamElementList(AUDIO_OBJECT_TYPE aot, SCHAR epCo
         else
           return &node_eld_cpe_epc1;
       }
-    case AOT_MPEGS_RESIDUALS:
-      if (nChannels == 1) {
-        return &node_mpegsres_sce;
-      } else {
-        return &node_mpegsres_cpe;
-      }
-      break;
     default:
       break;
   }
diff --git a/libMpegTPDec/src/tpdec_lib.cpp b/libMpegTPDec/src/tpdec_lib.cpp
index 445615d..950af64 100644
--- a/libMpegTPDec/src/tpdec_lib.cpp
+++ b/libMpegTPDec/src/tpdec_lib.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -1100,7 +1100,6 @@ TRANSPORTDEC_ERROR transportDec_ReadAccessUnit( const HANDLE_TRANSPORTDEC hTp, c
       }
       break;
 
-    case TT_RSVD50:
     case TT_MP4_ADTS:
     case TT_MP4_LOAS:
       err = transportDec_readStream(hTp, layer);
diff --git a/libMpegTPDec/src/version b/libMpegTPDec/src/version
index fc7e5f0..18fc9ba 100644
--- a/libMpegTPDec/src/version
+++ b/libMpegTPDec/src/version
@@ -2,7 +2,7 @@
 /* library info */
 #define TP_LIB_VL0 2
 #define TP_LIB_VL1 3
-#define TP_LIB_VL2 4
+#define TP_LIB_VL2 5
 #define TP_LIB_TITLE "MPEG Transport"
 #ifdef __ANDROID__
 #define TP_LIB_BUILD_DATE ""
diff --git a/libMpegTPEnc/src/tpenc_latm.cpp b/libMpegTPEnc/src/tpenc_latm.cpp
index 58e51ef..f292019 100644
--- a/libMpegTPEnc/src/tpenc_latm.cpp
+++ b/libMpegTPEnc/src/tpenc_latm.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -296,6 +296,7 @@ CreateStreamMuxConfig(
 
   USHORT coreFrameOffset=0;
 
+  hAss->taraBufferFullness  = 0xFF;
   hAss->audioMuxVersionA    = 0; /* for future extensions */
   hAss->streamMuxConfigBits = 0;
 
@@ -339,13 +340,7 @@ CreateStreamMuxConfig(
             hAss->streamMuxConfigBits+=1;
           }
           if( (useSameConfig == 0) || (transLayer==0) ) {
-            UINT bits;
-
-            if ( hAss->audioMuxVersion == 1 ) {
-              FDKpushFor(hBs, 2); /* align to ASC, even if we do not know the length of the "ascLen" field yet */
-            }
-
-            bits = FDKgetValidBits( hBs );
+            const UINT alignAnchor = FDKgetValidBits(hBs);
 
             transportEnc_writeASC(
                     hBs,
@@ -353,19 +348,24 @@ CreateStreamMuxConfig(
                     cb
                     );
 
-            bits = FDKgetValidBits( hBs ) - bits;
-
             if ( hAss->audioMuxVersion == 1 ) {
-              FDKpushBack(hBs, bits+2);
-              hAss->streamMuxConfigBits += transportEnc_LatmWriteValue( hBs, bits );
+              UINT ascLen = transportEnc_LatmWriteValue(hBs, 0);
+              FDKbyteAlign(hBs, alignAnchor);
+              ascLen = FDKgetValidBits(hBs) - alignAnchor - ascLen;
+              FDKpushBack(hBs, FDKgetValidBits(hBs) - alignAnchor);
+
+              transportEnc_LatmWriteValue(hBs, ascLen);
+
               transportEnc_writeASC(
                       hBs,
                       hAss->config[prog][layer],
                       cb
                       );
+
+              FDKbyteAlign(hBs, alignAnchor); /* asc length fillbits */
             }
 
-            hAss->streamMuxConfigBits += bits; /* add asc length to smc summary */
+            hAss->streamMuxConfigBits += FDKgetValidBits(hBs) - alignAnchor; /* add asc length to smc summary */
           }
           transLayer++;
 
@@ -384,7 +384,6 @@ CreateStreamMuxConfig(
           case AOT_ER_AAC_LD     :
           case AOT_ER_AAC_ELD    :
           case AOT_USAC:
-          case AOT_RSVD50:
             p_linfo->frameLengthType = 0;
 
             FDKwriteBits( hBs, p_linfo->frameLengthType, 3 );                        /* frameLengthType */
diff --git a/libMpegTPEnc/src/version b/libMpegTPEnc/src/version
index fc7e5f0..8742568 100644
--- a/libMpegTPEnc/src/version
+++ b/libMpegTPEnc/src/version
@@ -2,7 +2,7 @@
 /* library info */
 #define TP_LIB_VL0 2
 #define TP_LIB_VL1 3
-#define TP_LIB_VL2 4
+#define TP_LIB_VL2 6
 #define TP_LIB_TITLE "MPEG Transport"
 #ifdef __ANDROID__
 #define TP_LIB_BUILD_DATE ""
diff --git a/libSBRdec/src/sbrdecoder.cpp b/libSBRdec/src/sbrdecoder.cpp
index 619e4fd..e15145a 100644
--- a/libSBRdec/src/sbrdecoder.cpp
+++ b/libSBRdec/src/sbrdecoder.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -137,7 +137,7 @@ amm-info@iis.fraunhofer.de
 /* Decoder library info */
 #define SBRDECODER_LIB_VL0 2
 #define SBRDECODER_LIB_VL1 2
-#define SBRDECODER_LIB_VL2 6
+#define SBRDECODER_LIB_VL2 7
 #define SBRDECODER_LIB_TITLE "SBR Decoder"
 #ifdef __ANDROID__
 #define SBRDECODER_LIB_BUILD_DATE ""
@@ -318,7 +318,6 @@ SBR_ERROR sbrDecoder_ResetElement (
     case AOT_PS:
     case AOT_ER_AAC_SCAL:
     case AOT_DRM_AAC:
-    case AOT_DRM_SURROUND:
       if (CreatePsDec ( &self->hParametricStereoDec, samplesPerFrame )) {
         sbrError = SBRDEC_CREATE_ERROR;
         goto bail;
@@ -503,7 +502,6 @@ SBR_ERROR sbrDecoder_InitElement (
         case AOT_PS:
         case AOT_ER_AAC_SCAL:
         case AOT_DRM_AAC:
-        case AOT_DRM_SURROUND:
           elChannels = 2;
           break;
         default:
diff --git a/libSBRenc/src/nf_est.cpp b/libSBRenc/src/nf_est.cpp
index 7a3c022..385a043 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 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -268,8 +268,9 @@ qmfBasedNoiseFloorDetection(FIXP_DBL *noiseLevel,              /*!< Pointer to v
   /*
    * Add a noise floor offset to compensate for bias in the detector
    *****************************************************************/
-  if(!missingHarmonicFlag)
-    *noiseLevel = fMult(*noiseLevel, noiseFloorOffset)<<(NOISE_FLOOR_OFFSET_SCALING);
+  if(!missingHarmonicFlag) {
+    *noiseLevel = fixMin(fMult(*noiseLevel, noiseFloorOffset), (FIXP_DBL)MAXVAL_DBL>>NOISE_FLOOR_OFFSET_SCALING) << NOISE_FLOOR_OFFSET_SCALING;
+  }
 
   /*
    * check to see that we don't exceed the maximum allowed level
diff --git a/libSBRenc/src/sbr_encoder.cpp b/libSBRenc/src/sbr_encoder.cpp
index 221aecd..9bb98c8 100644
--- a/libSBRenc/src/sbr_encoder.cpp
+++ b/libSBRenc/src/sbr_encoder.cpp
@@ -103,7 +103,7 @@ amm-info@iis.fraunhofer.de
 
 #define SBRENCODER_LIB_VL0 3
 #define SBRENCODER_LIB_VL1 3
-#define SBRENCODER_LIB_VL2 6
+#define SBRENCODER_LIB_VL2 8
 
 
 
@@ -1552,12 +1552,6 @@ INT FDKsbrEnc_EnvInit (
   hSbrElement->sbrConfigData.sbrSyntaxFlags = 0;
 
   switch (aot) {
-  case AOT_DRM_MPEG_PS:
-  case AOT_DRM_SBR:
-    hSbrElement->sbrConfigData.sbrSyntaxFlags |= SBR_SYNTAX_SCALABLE;
-    hSbrElement->sbrConfigData.sbrSyntaxFlags |= SBR_SYNTAX_DRM_CRC;
-    hSbrElement->sbrConfigData.sbrSyntaxFlags |= SBR_SYNTAX_CRC;
-    break;
   case AOT_ER_AAC_ELD:
     hSbrElement->sbrConfigData.sbrSyntaxFlags |= SBR_SYNTAX_LOW_DELAY;
     break;
@@ -1847,7 +1841,7 @@ INT sbrEncoder_Init(
 
 
 
-    if ( (aot==AOT_PS) || (aot==AOT_MP2_PS) || (aot==AOT_DABPLUS_PS) || (aot==AOT_DRM_MPEG_PS) ) {
+    if ( (aot==AOT_PS) ) {
         usePs = 1;
     }
     if ( (aot==AOT_ER_AAC_ELD) ) {
diff --git a/libSBRenc/src/tran_det.cpp b/libSBRenc/src/tran_det.cpp
index 1e0a59f..6c62b4c 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 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -676,7 +676,7 @@ FDKsbrEnc_InitSbrTransientDetector(HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTransientD
     tmp = fixMax(tmp, FL2FXCONST_DBL(0.0001));
     tmp = fDivNorm(FL2FXCONST_DBL(0.000075), fPow2(tmp), &scale_1);
 
-    scale_1 = -(scale_1 + scale_0 + 2);
+    scale_1 = (scale_1 + scale_0 + 2);
 
     FDK_ASSERT(no_cols <= QMF_MAX_TIME_SLOTS);
     FDK_ASSERT(no_rows <= QMF_CHANNELS);
@@ -684,14 +684,7 @@ FDKsbrEnc_InitSbrTransientDetector(HANDLE_SBR_TRANSIENT_DETECTOR h_sbrTransientD
     h_sbrTransientDetector->no_cols = no_cols;
     h_sbrTransientDetector->tran_thr = (FIXP_DBL)((params->tran_thr << (32-24-1)) / no_rows);
     h_sbrTransientDetector->tran_fc = tran_fc;
-
-    if (scale_1>=0) {
-      h_sbrTransientDetector->split_thr = fMult(tmp, bitrateFactor_fix) >> scale_1;
-    }
-    else {
-      h_sbrTransientDetector->split_thr = fMult(tmp, bitrateFactor_fix) << (-scale_1);
-    }
-
+    h_sbrTransientDetector->split_thr = scaleValueSaturate(fMult(tmp, bitrateFactor_fix), scale_1);     
     h_sbrTransientDetector->no_rows = no_rows;
     h_sbrTransientDetector->mode = params->tran_det_mode;
     h_sbrTransientDetector->prevLowBandEnergy = FL2FXCONST_DBL(0.0f);
diff --git a/libSYS/include/FDK_audio.h b/libSYS/include/FDK_audio.h
index 0660f4c..98ded3b 100644
--- a/libSYS/include/FDK_audio.h
+++ b/libSYS/include/FDK_audio.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -134,13 +134,7 @@ typedef enum
 
   TT_MP4_LOAS          = 10, /**< Audio Sync Stream.         */
 
-  TT_DRM               = 12, /**< Digital Radio Mondial (DRM30/DRM+) bitstream format. */
-
-  TT_MP1_L1            = 16, /**< MPEG 1 Audio Layer 1 audio bitstream. */
-  TT_MP1_L2            = 17, /**< MPEG 1 Audio Layer 2 audio bitstream. */
-  TT_MP1_L3            = 18, /**< MPEG 1 Audio Layer 3 audio bitstream. */
-
-  TT_RSVD50            = 50 /**< */
+  TT_DRM               = 12  /**< Digital Radio Mondial (DRM30/DRM+) bitstream format. */
 
 } TRANSPORT_TYPE;
 
@@ -203,38 +197,22 @@ typedef enum
   AOT_SAOC             = 43, /**< SAOC                                      */
   AOT_LD_MPEGS         = 44, /**< Low Delay MPEG Surround                   */
 
-  AOT_RSVD50           = 50,  /**< Interim AOT for Rsvd50                   */
-
   /* Pseudo AOTs */
-  AOT_MP2_AAC_MAIN     = 128, /**< Virtual AOT MP2 Main profile                           */
-  AOT_MP2_AAC_LC       = 129, /**< Virtual AOT MP2 Low Complexity profile                 */
-  AOT_MP2_AAC_SSR      = 130, /**< Virtual AOT MP2 Scalable Sampling Rate profile         */
-
-  AOT_MP2_SBR          = 132, /**< Virtual AOT MP2 Low Complexity Profile with SBR        */
-
-  AOT_DAB              = 134, /**< Virtual AOT for DAB (Layer2 with scalefactor CRC)      */
-  AOT_DABPLUS_AAC_LC   = 135, /**< Virtual AOT for DAB plus AAC-LC                        */
-  AOT_DABPLUS_SBR      = 136, /**< Virtual AOT for DAB plus HE-AAC                        */
-  AOT_DABPLUS_PS       = 137, /**< Virtual AOT for DAB plus HE-AAC v2                     */
-
-  AOT_PLAIN_MP1        = 140, /**< Virtual AOT for plain mp1                              */
-  AOT_PLAIN_MP2        = 141, /**< Virtual AOT for plain mp2                              */
-  AOT_PLAIN_MP3        = 142, /**< Virtual AOT for plain mp3                              */
-
   AOT_DRM_AAC          = 143, /**< Virtual AOT for DRM (ER-AAC-SCAL without SBR)          */
   AOT_DRM_SBR          = 144, /**< Virtual AOT for DRM (ER-AAC-SCAL with SBR)             */
-  AOT_DRM_MPEG_PS      = 145, /**< Virtual AOT for DRM (ER-AAC-SCAL with SBR and MPEG-PS) */
-  AOT_DRM_SURROUND     = 146, /**< Virtual AOT for DRM Surround (ER-AAC-SCAL (+SBR) +MPS) */
-
-  AOT_MP2_PS           = 156, /**< Virtual AOT MP2 Low Complexity Profile with SBR and PS */
-
-  AOT_MPEGS_RESIDUALS  = 256  /**< Virtual AOT for MPEG Surround residuals                */
+  AOT_DRM_MPEG_PS      = 145  /**< Virtual AOT for DRM (ER-AAC-SCAL with SBR and MPEG-PS) */
 
 } AUDIO_OBJECT_TYPE;
 
+#define CAN_DO_PS(aot) \
+  ((aot) == AOT_AAC_LC \
+|| (aot) == AOT_SBR \
+|| (aot) == AOT_PS \
+|| (aot) == AOT_ER_BSAC \
+|| (aot) == AOT_DRM_AAC)
+
 #define IS_USAC(aot) \
-  ((aot) == AOT_USAC \
-|| (aot) == AOT_RSVD50)
+  ((aot) == AOT_USAC)
 
 #define IS_LOWDELAY(aot) \
   ((aot) == AOT_ER_AAC_LD \
diff --git a/libSYS/src/genericStds.cpp b/libSYS/src/genericStds.cpp
index 47d62fc..53abe06 100644
--- a/libSYS/src/genericStds.cpp
+++ b/libSYS/src/genericStds.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2013 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
+© Copyright  1995 - 2015 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V.
   All rights reserved.
 
  1.    INTRODUCTION
@@ -99,7 +99,7 @@ amm-info@iis.fraunhofer.de
 /* library info */
 #define SYS_LIB_VL0 1
 #define SYS_LIB_VL1 3
-#define SYS_LIB_VL2 6
+#define SYS_LIB_VL2 7
 #define SYS_LIB_TITLE "System Integration Library"
 #ifdef __ANDROID__
 #define SYS_LIB_BUILD_DATE ""