AAC/SBR decoder improvements and bugfixes

* AAC-Decoder

   - Add support for AOT 20 (ER-AAC scalable) (base layer only)
   - Add support for AAC as used in Digital Radio Mondiale (DRM30/DRM+)
     Modified file(s):
        libAACdec/src/aacdecoder.cpp
        libAACdec/src/aacdecoder_lib.cpp
        libFDK/src/FDK_core.cpp
        libFDK/src/FDK_tools_rom.cpp
        libMpegTPDec/src/tpdec_asc.cpp
        libMpegTPDec/src/tpdec_lib.cpp
        libMpegTPDec/src/version
        libSBRdec/include/sbrdecoder.h
        libSBRdec/src/env_extr.h
        libSBRdec/src/sbrdecoder.cpp
     Added file(s):
        libMpegTPDec/src/tpdec_drm.cpp
        libMpegTPDec/src/tpdec_drm.h

   - Fix sanity check in HCR module that was performed at the wrong point in time.
     Modified file(s):
        libAACdec/src/aacdecoder_lib.cpp
        libAACdec/src/block.cpp

   - Extend core sampling rate support up to 96 kHz.
     Modified file(s):
        libAACdec/src/aac_rom.cpp
        libAACdec/src/aacdecoder.cpp
        libAACdec/src/aacdecoder_lib.cpp

   - Return correct audio output channel description according number of output
     channels.
     Modified file(s):
        libAACdec/src/aacdecoder_lib.cpp

   - Indroduce decoder intern output buffer. This change allows to use framework
     output buffer with the actual size of the deocder output channels.
     Modified file(s):
        libAACdec/include/aacdecoder_lib.h
        libAACdec/src/aacdecoder.h
        libAACdec/src/aacdecoder_lib.cpp

* SBR-Decoder

   - Increase robustness for erroneous input data.
   - Improve error concealment performance.
   - Fix handling of lowest sub-band for LD-SBR
     Modified file(s):
        libAACdec/src/aacdecoder.cpp
        libAACdec/src/aacdecoder_lib.cpp
        libSBRdec/src/env_calc.cpp
        libSBRdec/src/env_dec.cpp
        libSBRdec/src/env_extr.cpp
        libSBRdec/src/env_extr.h
        libSBRdec/src/sbr_dec.cpp
        libSBRdec/src/sbr_rom.cpp
        libSBRdec/src/sbr_rom.h
        libSBRdec/src/sbrdecoder.cpp

   - Add QMF delay compensation for ELD v2 streams decoded with the complex
     low delay filter-bank.
     Modified file(s):
        libSBRdec/src/sbr_dec.cpp
        libSBRdec/src/sbr_dec.h
        libSBRdec/src/sbrdecoder.cpp

   - Introduce a different handling of frames to be flushed
     dependent on whether there are delayed frames available or not.
     Modified file(s):
        libSBRdec/src/sbr_ram.h
        libSBRdec/src/sbrdecoder.cpp

   - Calculate the correct number of samples for dual-mono copy in case of no
     available PS data.
     Modified file(s):
        libSBRdec/src/sbrdecoder.cpp

* SYS-Library

   - Change include order of genericStds.h to prevent conflict with definitions
     which are also used in math.h.
     Modified file(s):
        libSYS/src/genericStds.cpp

Change-Id: I3ecffbad85f39b056213107955cfadbeb3f4b6e1

10 files changed, 389 insertions, 99 deletions

diff --git a/libSBRdec/src/env_calc.cpp b/libSBRdec/src/env_calc.cpp
index ade57fc..fa5330a 100644
--- a/libSBRdec/src/env_calc.cpp
+++ b/libSBRdec/src/env_calc.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 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
@@ -151,13 +151,13 @@ typedef struct
 }
 ENV_CALC_NRGS;
 
-/*static*/ void equalizeFiltBufferExp(FIXP_DBL *filtBuffer,
+static void equalizeFiltBufferExp(FIXP_DBL *filtBuffer,
                                   SCHAR   *filtBuffer_e,
                                   FIXP_DBL *NrgGain,
                                   SCHAR   *NrgGain_e,
                                   int    subbands);
 
-/*static*/ void calcNrgPerSubband(FIXP_DBL  **analysBufferReal,
+static void calcNrgPerSubband(FIXP_DBL  **analysBufferReal,
                               FIXP_DBL  **analysBufferImag,
                               int       lowSubband, int highSubband,
                               int       start_pos,  int next_pos,
@@ -165,7 +165,7 @@ ENV_CALC_NRGS;
                               FIXP_DBL *nrgEst,
                               SCHAR    *nrgEst_e );
 
-/*static*/ void calcNrgPerSfb(FIXP_DBL  **analysBufferReal,
+static void calcNrgPerSfb(FIXP_DBL  **analysBufferReal,
                           FIXP_DBL  **analysBufferImag,
                           int       nSfb,
                           UCHAR    *freqBandTable,
@@ -174,13 +174,13 @@ ENV_CALC_NRGS;
                           FIXP_DBL *nrg_est,
                           SCHAR    *nrg_est_e );
 
-/*static*/ void calcSubbandGain(FIXP_DBL  nrgRef, SCHAR nrgRef_e, ENV_CALC_NRGS* nrgs, int c,
+static void calcSubbandGain(FIXP_DBL  nrgRef, SCHAR nrgRef_e, ENV_CALC_NRGS* nrgs, int c,
                             FIXP_DBL  tmpNoise, SCHAR tmpNoise_e,
                             UCHAR     sinePresentFlag,
                             UCHAR     sineMapped,
                             int       noNoiseFlag);
 
-/*static*/ void calcAvgGain(ENV_CALC_NRGS* nrgs,
+static void calcAvgGain(ENV_CALC_NRGS* nrgs,
                         int        lowSubband,
                         int        highSubband,
                         FIXP_DBL  *sumRef_m,
@@ -188,7 +188,7 @@ ENV_CALC_NRGS;
                         FIXP_DBL  *ptrAvgGain_m,
                         SCHAR     *ptrAvgGain_e);
 
-/*static*/ void adjustTimeSlotLC(FIXP_DBL  *ptrReal,
+static void adjustTimeSlot_EldGrid(FIXP_DBL  *ptrReal,
                            ENV_CALC_NRGS* nrgs,
                            UCHAR *ptrHarmIndex,
                            int    lowSubbands,
@@ -196,8 +196,17 @@ ENV_CALC_NRGS;
                            int    scale_change,
                            int    noNoiseFlag,
                            int   *ptrPhaseIndex,
-                           int    fCldfb);
-/*static*/ void adjustTimeSlotHQ(FIXP_DBL  *ptrReal,
+                           int    scale_diff_low);
+
+static void adjustTimeSlotLC(FIXP_DBL  *ptrReal,
+                           ENV_CALC_NRGS* nrgs,
+                           UCHAR *ptrHarmIndex,
+                           int    lowSubbands,
+                           int    noSubbands,
+                           int    scale_change,
+                           int    noNoiseFlag,
+                           int   *ptrPhaseIndex);
+static void adjustTimeSlotHQ(FIXP_DBL  *ptrReal,
                            FIXP_DBL  *ptrImag,
                            HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env,
                            ENV_CALC_NRGS* nrgs,
@@ -224,7 +233,7 @@ ENV_CALC_NRGS;
   Additionally, the flags in harmFlagsPrev are being updated by this function
   for the next frame.
 */
-/*static*/ void mapSineFlags(UCHAR *freqBandTable, /*!< Band borders (there's only 1 flag per band) */
+static void mapSineFlags(UCHAR *freqBandTable,         /*!< Band borders (there's only 1 flag per band) */
                          int nSfb,                     /*!< Number of bands in the table */
                          UCHAR *addHarmonics,           /*!< vector with 1 flag per sfb */
                          int *harmFlagsPrev,           /*!< Packed 'addHarmonics' */
@@ -990,7 +999,6 @@ calculateSbrEnvelope (QMF_SCALE_FACTOR  *sbrScaleFactor,           /*!< Scaling
           /* Prevent the smoothing filter from running on constant levels */
           if (j-start_pos < smooth_length)
             smooth_ratio = FDK_sbrDecoder_sbr_smoothFilter[j-start_pos];
-
           else
             smooth_ratio = FL2FXCONST_SGL(0.0f);
 
@@ -1007,7 +1015,8 @@ calculateSbrEnvelope (QMF_SCALE_FACTOR  *sbrScaleFactor,           /*!< Scaling
         }
         else
         {
-          adjustTimeSlotLC(&analysBufferReal[j][lowSubband],
+          if (flags & SBRDEC_ELD_GRID) {
+            adjustTimeSlot_EldGrid(&analysBufferReal[j][lowSubband],
                            pNrgs,
                           &h_sbr_cal_env->harmIndex,
                            lowSubband,
@@ -1015,7 +1024,18 @@ calculateSbrEnvelope (QMF_SCALE_FACTOR  *sbrScaleFactor,           /*!< Scaling
                            scale_change,
                            noNoiseFlag,
                           &h_sbr_cal_env->phaseIndex,
-                           (flags & SBRDEC_ELD_GRID));
+                           EXP2SCALE(adj_e) - sbrScaleFactor->lb_scale);
+          } else
+          {
+            adjustTimeSlotLC(&analysBufferReal[j][lowSubband],
+                           pNrgs,
+                          &h_sbr_cal_env->harmIndex,
+                           lowSubband,
+                           noSubbands,
+                           scale_change,
+                           noNoiseFlag,
+                          &h_sbr_cal_env->phaseIndex);
+          }
         }
       } // for
 
@@ -1176,7 +1196,7 @@ resetSbrEnvelopeCalc (HANDLE_SBR_CALCULATE_ENVELOPE hCalEnv) /*!< pointer to env
   can be performed.
   This function is called once for each envelope before adjusting.
 */
-/*static*/ void equalizeFiltBufferExp(FIXP_DBL *filtBuffer,     /*!< bufferd gains */
+static void equalizeFiltBufferExp(FIXP_DBL *filtBuffer,     /*!< bufferd gains */
                                   SCHAR    *filtBuffer_e,   /*!< exponents of bufferd gains */
                                   FIXP_DBL *nrgGain,        /*!< gains for current envelope */
                                   SCHAR    *nrgGain_e,      /*!< exponents of gains for current envelope */
@@ -1331,7 +1351,7 @@ FIXP_DBL maxSubbandSample( FIXP_DBL ** re,   /*!< Real part of input and output
 
   This function is used when interpolFreq is true.
 */
-/*static*/ void calcNrgPerSubband(FIXP_DBL  **analysBufferReal, /*!< Real part of subband samples */
+static void calcNrgPerSubband(FIXP_DBL  **analysBufferReal, /*!< Real part of subband samples */
                               FIXP_DBL  **analysBufferImag, /*!< Imaginary part of subband samples */
                               int       lowSubband,           /*!< Begin of the SBR frequency range */
                               int       highSubband,          /*!< High end of the SBR frequency range */
@@ -1452,7 +1472,7 @@ FIXP_DBL maxSubbandSample( FIXP_DBL ** re,   /*!< Real part of input and output
 
   This function is used when interpolFreq is false.
 */
-/*static*/ void calcNrgPerSfb(FIXP_DBL  **analysBufferReal,  /*!< Real part of subband samples */
+static void calcNrgPerSfb(FIXP_DBL  **analysBufferReal,  /*!< Real part of subband samples */
                           FIXP_DBL  **analysBufferImag,  /*!< Imaginary part of subband samples */
                           int       nSfb,                /*!< Number of scale factor bands */
                           UCHAR    *freqBandTable,       /*!< First Subband for each Sfb */
@@ -1585,7 +1605,7 @@ FIXP_DBL maxSubbandSample( FIXP_DBL ** re,   /*!< Real part of input and output
 
   The resulting energy gain is given by mantissa and exponent.
 */
-/*static*/ void calcSubbandGain(FIXP_DBL  nrgRef,            /*!< Reference Energy according to envelope data */
+static void calcSubbandGain(FIXP_DBL  nrgRef,            /*!< Reference Energy according to envelope data */
                             SCHAR     nrgRef_e,          /*!< Reference Energy according to envelope data (exponent) */
                             ENV_CALC_NRGS* nrgs,
                             int       i,
@@ -1689,7 +1709,7 @@ FIXP_DBL maxSubbandSample( FIXP_DBL ** re,   /*!< Real part of input and output
   The result is used as a relative limit for all gains within the
   current "limiter band" (a certain frequency range).
 */
-/*static*/ void calcAvgGain(ENV_CALC_NRGS* nrgs,
+static void calcAvgGain(ENV_CALC_NRGS* nrgs,
                         int        lowSubband,    /*!< Begin of the limiter band */
                         int        highSubband,   /*!< High end of the limiter band */
                         FIXP_DBL  *ptrSumRef,
@@ -1728,21 +1748,101 @@ FIXP_DBL maxSubbandSample( FIXP_DBL ** re,   /*!< Real part of input and output
   *ptrSumRef_e = sumRef_e;
 }
 
+static void adjustTimeSlot_EldGrid(
+                              FIXP_DBL *ptrReal,        /*!< Subband samples to be adjusted, real part */
+                              ENV_CALC_NRGS* nrgs,
+                              UCHAR    *ptrHarmIndex,   /*!< Harmonic index */
+                              int       lowSubband,     /*!< Lowest QMF-channel in the currently used SBR range. */
+                              int       noSubbands,     /*!< Number of QMF subbands */
+                              int       scale_change,   /*!< Number of bits to shift adjusted samples */
+                              int       noNoiseFlag,    /*!< Flag to suppress noise addition */
+                              int      *ptrPhaseIndex,  /*!< Start index to random number array */
+                              int       scale_diff_low) /*!<  */
+{
+  int k;
+  FIXP_DBL  signalReal, sbNoise;
+  int tone_count = 0;
+
+  FIXP_DBL *pGain       = nrgs->nrgGain;     /*!< Gains of current envelope */
+  FIXP_DBL *pNoiseLevel = nrgs->noiseLevel;  /*!< Noise levels of current envelope */
+  FIXP_DBL *pSineLevel  = nrgs->nrgSine;     /*!< Sine levels */
+
+  int    phaseIndex = *ptrPhaseIndex;
+  UCHAR  harmIndex  = *ptrHarmIndex;
+
+  static const INT harmonicPhase [2][4] = {
+    { 1, 0, -1,  0},
+    { 0, 1,  0, -1}
+  };
+
+  static const FIXP_DBL harmonicPhaseX [2][4] = {
+    { FL2FXCONST_DBL(2.0*1.245183154539139e-001),  FL2FXCONST_DBL(2.0*-1.123767859325028e-001),  FL2FXCONST_DBL(2.0*-1.245183154539139e-001), FL2FXCONST_DBL(2.0* 1.123767859325028e-001) },
+    { FL2FXCONST_DBL(2.0*1.245183154539139e-001),  FL2FXCONST_DBL(2.0* 1.123767859325028e-001),  FL2FXCONST_DBL(2.0*-1.245183154539139e-001), FL2FXCONST_DBL(2.0*-1.123767859325028e-001) }
+  };
+
+  for (k=0; k < noSubbands; k++) {
+
+    phaseIndex = (phaseIndex + 1) & (SBR_NF_NO_RANDOM_VAL - 1);
+
+    if( (pSineLevel[0] != FL2FXCONST_DBL(0.0f)) || (noNoiseFlag == 1) ){
+      sbNoise = FL2FXCONST_DBL(0.0f);
+    } else {
+      sbNoise = pNoiseLevel[0];
+    }
+
+    signalReal = fMultDiv2(*ptrReal,*pGain) << ((int)scale_change);
+
+    signalReal += (fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[phaseIndex][0], sbNoise)<<4);
+
+    signalReal += pSineLevel[0] * harmonicPhase[0][harmIndex];
+
+    *ptrReal = signalReal;
+
+    if (k == 0) {
+      *(ptrReal-1) += scaleValue(fMultDiv2(harmonicPhaseX[lowSubband&1][harmIndex], pSineLevel[0]), -scale_diff_low)  ;
+      if (k < noSubbands - 1) {
+        *(ptrReal)   += fMultDiv2(pSineLevel[1], harmonicPhaseX[(lowSubband+1)&1][harmIndex]);
+      }
+    }
+    if (k > 0 && k < noSubbands - 1 && tone_count < 16) {
+      *(ptrReal)   += fMultDiv2(pSineLevel[- 1], harmonicPhaseX [(lowSubband+k)&1]  [harmIndex]);
+      *(ptrReal)   += fMultDiv2(pSineLevel[+ 1], harmonicPhaseX [(lowSubband+k+1)&1][harmIndex]);
+    }
+    if (k == noSubbands - 1 && tone_count < 16) {
+      if (k > 0) {
+        *(ptrReal)   += fMultDiv2(pSineLevel[- 1], harmonicPhaseX [(lowSubband+k)&1][harmIndex]);
+      }
+      if (k + lowSubband + 1< 63) {
+        *(ptrReal+1) += fMultDiv2(pSineLevel[0], harmonicPhaseX[(lowSubband+k+1)&1][harmIndex]);
+      }
+    }
+
+    if(pSineLevel[0] != FL2FXCONST_DBL(0.0f)){
+      tone_count++;
+    }
+    ptrReal++;
+    pNoiseLevel++;
+    pGain++;
+    pSineLevel++;
+  }
+
+  *ptrHarmIndex = (harmIndex + 1) & 3;
+  *ptrPhaseIndex = phaseIndex & (SBR_NF_NO_RANDOM_VAL - 1);
+}
 
 /*!
   \brief   Amplify one timeslot of the signal with the calculated gains
            and add the noisefloor.
 */
 
-/*static*/ void adjustTimeSlotLC(FIXP_DBL *ptrReal,       /*!< Subband samples to be adjusted, real part */
+static void adjustTimeSlotLC(FIXP_DBL *ptrReal,       /*!< Subband samples to be adjusted, real part */
                              ENV_CALC_NRGS* nrgs,
                              UCHAR    *ptrHarmIndex,  /*!< Harmonic index */
                              int       lowSubband,    /*!< Lowest QMF-channel in the currently used SBR range. */
                              int       noSubbands,    /*!< Number of QMF subbands */
                              int       scale_change,  /*!< Number of bits to shift adjusted samples */
                              int       noNoiseFlag,   /*!< Flag to suppress noise addition */
-                             int      *ptrPhaseIndex, /*!< Start index to random number array */
-                             int       fCldfb)        /*!< CLDFB 80 flag */
+                             int      *ptrPhaseIndex) /*!< Start index to random number array */
 {
   FIXP_DBL *pGain       = nrgs->nrgGain;     /*!< Gains of current envelope */
   FIXP_DBL *pNoiseLevel = nrgs->noiseLevel;  /*!< Noise levels of current envelope */
@@ -1775,41 +1875,10 @@ FIXP_DBL maxSubbandSample( FIXP_DBL ** re,   /*!< Real part of input and output
   sineLevelNext = (noSubbands > 1) ? pSineLevel[0] : FL2FXCONST_DBL(0.0f);
 
   if (sineLevel!=FL2FXCONST_DBL(0.0f)) tone_count++;
-
   else if (!noNoiseFlag)
         /* Add noisefloor to the amplified signal */
         signalReal += (fMultDiv2(FDK_sbrDecoder_sbr_randomPhase[index][0], pNoiseLevel[0])<<4);
 
-  if (fCldfb) {
-
-    if (!(harmIndex&0x1)) {
-      /* harmIndex 0,2 */
-      signalReal += (harmIndex&0x2) ? -sineLevel : sineLevel;
-      *ptrReal++ = signalReal;
-    }
-    else {
-      /* harmIndex 1,3 in combination with freqInvFlag */
-      int shift = (int) (scale_change+1);
-      shift = (shift>=0) ? fixMin(DFRACT_BITS-1,shift) : fixMax(-(DFRACT_BITS-1),shift);
-
-      FIXP_DBL tmp1 = scaleValue( fMultDiv2(C1_CLDFB, sineLevel), -shift );
-
-      FIXP_DBL tmp2 = fMultDiv2(C1_CLDFB, sineLevelNext);
-
-
-      /* save switch and compare operations and reduce to XOR statement */
-      if ( ((harmIndex>>1)&0x1)^freqInvFlag) {
-          *(ptrReal-1) += tmp1;
-          signalReal   -= tmp2;
-      } else {
-          *(ptrReal-1) -= tmp1;
-          signalReal   += tmp2;
-      }
-      *ptrReal++ = signalReal;
-      freqInvFlag = !freqInvFlag;
-    }
-
-  } else
   {
     if (!(harmIndex&0x1)) {
       /* harmIndex 0,2 */
@@ -1933,8 +2002,9 @@ FIXP_DBL maxSubbandSample( FIXP_DBL ** re,   /*!< Real part of input and output
   *ptrHarmIndex = (harmIndex + 1) & 3;
   *ptrPhaseIndex = index & (SBR_NF_NO_RANDOM_VAL - 1);
 }
-void adjustTimeSlotHQ(FIXP_DBL *RESTRICT ptrReal,           /*!< Subband samples to be adjusted, real part */
-                      FIXP_DBL *RESTRICT ptrImag,               /*!< Subband samples to be adjusted, imag part */
+static void adjustTimeSlotHQ(
+                      FIXP_DBL *RESTRICT ptrReal,      /*!< Subband samples to be adjusted, real part */
+                      FIXP_DBL *RESTRICT ptrImag,      /*!< Subband samples to be adjusted, imag part */
                       HANDLE_SBR_CALCULATE_ENVELOPE h_sbr_cal_env,
                       ENV_CALC_NRGS* nrgs,
                       int       lowSubband,            /*!< Lowest QMF-channel in the currently used SBR range. */
@@ -2137,7 +2207,6 @@ ResetLimiterBands ( UCHAR *limiterBandTable,   /*!< Resulting band borders in QM
   UCHAR workLimiterBandTable[MAX_FREQ_COEFFS / 2 + MAX_NUM_PATCHES + 1];
   int patchBorders[MAX_NUM_PATCHES + 1];
   int kx, k2;
-  FIXP_DBL temp;
 
   int lowSubband = freqBandTable[0];
   int highSubband = freqBandTable[noFreqBands];
@@ -2169,13 +2238,32 @@ ResetLimiterBands ( UCHAR *limiterBandTable,   /*!< Resulting band borders in QM
 
 
     while (hiLimIndex <= tempNoLim) {
+      FIXP_DBL div_m, oct_m, temp;
+      INT div_e  = 0, oct_e  = 0, temp_e = 0;
+
       k2 = workLimiterBandTable[hiLimIndex] + lowSubband;
       kx = workLimiterBandTable[loLimIndex] + lowSubband;
 
-      temp = FX_SGL2FX_DBL(FDK_getNumOctavesDiv8(kx,k2)); /* Number of octaves */
-      temp = fMult(temp, FDK_sbrDecoder_sbr_limiterBandsPerOctaveDiv4[limiterBands]);
+      div_m  = fDivNorm(k2, kx, &div_e);
+
+      /* calculate number of octaves */
+      oct_m  = fLog2(div_m, div_e, &oct_e);
+
+      /* multiply with limiterbands per octave    */
+      /* values 1, 1.2, 2, 3 -> scale factor of 2 */
+      temp = fMultNorm(oct_m, FDK_sbrDecoder_sbr_limiterBandsPerOctaveDiv4_DBL[limiterBands], &temp_e);
+
+      /* overall scale factor of temp ist addition of scalefactors from log2 calculation,
+         limiter bands scalefactor (2) and limiter bands multiplication */
+      temp_e += oct_e + 2;
+
+      /*    div can be a maximum of 64 (k2 = 64 and kx = 1)
+         -> oct can be a maximum of 6
+         -> temp can be a maximum of 18 (as limiterBandsPerOctoave is a maximum factor of 3)
+         -> we need a scale factor of 5 for comparisson
+      */
+      if (temp >> (5 - temp_e) < FL2FXCONST_DBL (0.49f) >> 5) {
 
-      if (temp < FL2FXCONST_DBL (0.49f)>>5) {
         if (workLimiterBandTable[hiLimIndex] == workLimiterBandTable[loLimIndex]) {
           workLimiterBandTable[hiLimIndex] = highSubband;
           nBands--;
diff --git a/libSBRdec/src/env_dec.cpp b/libSBRdec/src/env_dec.cpp
index 24b2d3b..c65c169 100644
--- a/libSBRdec/src/env_dec.cpp
+++ b/libSBRdec/src/env_dec.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 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
@@ -369,7 +369,7 @@ leanSbrConcealment(HANDLE_SBR_HEADER_DATA hHeaderData,     /*!< Static control d
   FIXP_SGL step;    /* speed of fade */
   int i;
 
-  int currentStartPos = h_prev_data->stopPos - hHeaderData->numberTimeSlots;
+  int currentStartPos = FDKmax(0, h_prev_data->stopPos - hHeaderData->numberTimeSlots);
   int currentStopPos = hHeaderData->numberTimeSlots;
 
 
diff --git a/libSBRdec/src/env_extr.cpp b/libSBRdec/src/env_extr.cpp
index 6cb0b99..dacd951 100644
--- a/libSBRdec/src/env_extr.cpp
+++ b/libSBRdec/src/env_extr.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 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
@@ -327,7 +327,7 @@ sbrGetHeaderData (HANDLE_SBR_HEADER_DATA hHeaderData,
   }
 
   /* Look for new settings. IEC 14496-3, 4.6.18.3.1 */
-  if(hHeaderData->syncState != SBR_ACTIVE ||
+  if(hHeaderData->syncState < SBR_HEADER ||
      lastHeader.startFreq   != pBsData->startFreq   ||
      lastHeader.stopFreq    != pBsData->stopFreq    ||
      lastHeader.freqScale   != pBsData->freqScale   ||
diff --git a/libSBRdec/src/env_extr.h b/libSBRdec/src/env_extr.h
index ab6b704..0518ea9 100644
--- a/libSBRdec/src/env_extr.h
+++ b/libSBRdec/src/env_extr.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 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
@@ -125,6 +125,7 @@ amm-info@iis.fraunhofer.de
 typedef enum
 {
   HEADER_NOT_PRESENT,
+  HEADER_ERROR,
   HEADER_OK,
   HEADER_RESET
 }
@@ -132,10 +133,10 @@ SBR_HEADER_STATUS;
 
 typedef enum
 {
-  SBR_NOT_INITIALIZED,
-  UPSAMPLING,
-  SBR_HEADER,
-  SBR_ACTIVE
+  SBR_NOT_INITIALIZED = 0,
+  UPSAMPLING = 1,
+  SBR_HEADER = 2,
+  SBR_ACTIVE = 3
 }
 SBR_SYNC_STATE;
 
@@ -179,6 +180,7 @@ typedef FREQ_BAND_DATA *HANDLE_FREQ_BAND_DATA;
 #define SBRDEC_LOW_POWER      16  /* Flag indicating that Low Power QMF mode shall be used. */
 #define SBRDEC_PS_DECODED     32  /* Flag indicating that PS was decoded and rendered. */
 #define SBRDEC_LD_MPS_QMF    512  /* Flag indicating that the LD-MPS QMF shall be used. */
+#define SBRDEC_SYNTAX_DRM   2048  /* Flag indicating that DRM30/DRM+ reverse syntax is being used. */
 #define SBRDEC_DOWNSAMPLE   8192  /* Flag indicating that the downsampling mode is used. */
 #define SBRDEC_FLUSH       16384  /* Flag is used to flush all elements in use. */
 #define SBRDEC_FORCE_RESET 32768  /* Flag is used to force a reset of all elements in use. */
diff --git a/libSBRdec/src/sbr_dec.cpp b/libSBRdec/src/sbr_dec.cpp
index 1282338..0864348 100644
--- a/libSBRdec/src/sbr_dec.cpp
+++ b/libSBRdec/src/sbr_dec.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 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
@@ -225,7 +225,14 @@ static void changeQmfType( HANDLE_SBR_DEC hSbrDec,                     /*!< hand
   }
 
   if (resetAnaQmf) {
-    int qmfErr = qmfInitAnalysisFilterBank (
+    QMF_FILTER_BANK prvAnaQmf;
+    int  qmfErr;
+
+    /* Store current configuration */
+    FDKmemcpy(&prvAnaQmf, &hSbrDec->AnalysiscQMF, sizeof(QMF_FILTER_BANK));
+
+    /* Reset analysis QMF */
+    qmfErr = qmfInitAnalysisFilterBank (
            &hSbrDec->AnalysiscQMF,
             hSbrDec->anaQmfStates,
             hSbrDec->AnalysiscQMF.no_col,
@@ -234,13 +241,22 @@ static void changeQmfType( HANDLE_SBR_DEC hSbrDec,                     /*!< hand
             hSbrDec->AnalysiscQMF.no_channels,
             anaQmfFlags | QMF_FLAG_KEEP_STATES
             );
+
     if (qmfErr != 0) {
-      FDK_ASSERT(0);
+      /* Restore old configuration of analysis QMF */
+      FDKmemcpy(&hSbrDec->AnalysiscQMF, &prvAnaQmf, sizeof(QMF_FILTER_BANK));
     }
   }
 
   if (resetSynQmf) {
-    int qmfErr = qmfInitSynthesisFilterBank (
+    QMF_FILTER_BANK prvSynQmf;
+    int  qmfErr;
+
+    /* Store current configuration */
+    FDKmemcpy(&prvSynQmf, &hSbrDec->SynthesisQMF, sizeof(QMF_FILTER_BANK));
+
+    /* Reset synthesis QMF */
+    qmfErr = qmfInitSynthesisFilterBank (
            &hSbrDec->SynthesisQMF,
             hSbrDec->pSynQmfStates,
             hSbrDec->SynthesisQMF.no_col,
@@ -251,7 +267,8 @@ static void changeQmfType( HANDLE_SBR_DEC hSbrDec,                     /*!< hand
             );
 
     if (qmfErr != 0) {
-      FDK_ASSERT(0);
+      /* Restore old configuration of synthesis QMF */
+      FDKmemcpy(&hSbrDec->SynthesisQMF, &prvSynQmf, sizeof(QMF_FILTER_BANK));
     }
   }
 }
@@ -321,7 +338,8 @@ sbr_dec ( HANDLE_SBR_DEC hSbrDec,            /*!< handle to Decoder channel */
           HANDLE_SBR_PREV_FRAME_DATA hPrevFrameData,  /*!< Some control data of last frame */
           const int applyProcessing,         /*!< Flag for SBR operation */
           HANDLE_PS_DEC h_ps_d,
-          const UINT flags
+          const UINT flags,
+          const int codecFrameSize
          )
 {
   int i, slot, reserve;
@@ -348,6 +366,33 @@ sbr_dec ( HANDLE_SBR_DEC hSbrDec,            /*!< handle to Decoder channel */
   if (flags & SBRDEC_ELD_GRID) {
     /* Choose the right low delay filter bank */
     changeQmfType( hSbrDec, (flags & SBRDEC_LD_MPS_QMF) ? 1 : 0 );
+
+    /* If the LD-MPS QMF is not available delay the signal by (96-48*ldSbrSamplingRate)
+     * samples according to ISO/IEC 14496-3:2009/FDAM 2:2010(E) chapter 4.5.2.13. */
+    if ( (flags & SBRDEC_LD_MPS_QMF)
+      && (hSbrDec->AnalysiscQMF.flags & QMF_FLAG_CLDFB) )
+    {
+      INT_PCM *pDlyBuf = hSbrDec->coreDelayBuf;  /* DLYBUF */
+      int smpl, delay = 96 >> (!(flags & SBRDEC_DOWNSAMPLE) ? 1 : 0);
+      /* Create TMPBUF */
+      C_AALLOC_SCRATCH_START(pcmTemp, INT_PCM, (96));
+      /* Copy delay samples from INBUF to TMPBUF */
+      for (smpl = 0; smpl < delay; smpl += 1) {
+        pcmTemp[smpl] = timeIn[(codecFrameSize-delay+smpl)*strideIn];
+      }
+      /* Move input signal remainder to the very end of INBUF */
+      for (smpl = (codecFrameSize-delay-1)*strideIn; smpl >= 0; smpl -= strideIn) {
+        timeIn[smpl+delay] = timeIn[smpl];
+      }
+      /* Copy delayed samples from last frame from DLYBUF to the very beginning of INBUF */
+      for (smpl = 0; smpl < delay; smpl += 1) {
+        timeIn[smpl*strideIn] = pDlyBuf[smpl];
+      }
+      /* Copy TMPBUF to DLYBUF */
+      FDKmemcpy(pDlyBuf, pcmTemp, delay*sizeof(INT_PCM));
+      /* Destory TMPBUF */
+      C_AALLOC_SCRATCH_END(pcmTemp, INT_PCM, (96));
+    }
   }
 
   /*
@@ -761,7 +806,7 @@ createSbrDec (SBR_CHANNEL * hSbrChannel,
   {
     int qmfErr;
     /* Adapted QMF analysis post-twiddles for down-sampled HQ SBR */
-    const UINT downSampledFlag = (downsampleFac==2) ? QMF_FLAG_DOWNSAMPLED : 0;
+    const UINT downSampledFlag = (flags & SBRDEC_DOWNSAMPLE) ? QMF_FLAG_DOWNSAMPLED : 0;
 
     qmfErr = qmfInitAnalysisFilterBank (
                     &hs->AnalysiscQMF,
@@ -836,6 +881,9 @@ createSbrDec (SBR_CHANNEL * hSbrChannel,
     }
   }
 
+  /* Clear input delay line */
+  FDKmemclear(hs->coreDelayBuf, (96)*sizeof(INT_PCM));
+
   /* assign qmf time slots */
   assignTimeSlots( &hSbrChannel->SbrDec, hHeaderData->numberTimeSlots * hHeaderData->timeStep, qmfFlags & QMF_FLAG_LP);
 
diff --git a/libSBRdec/src/sbr_dec.h b/libSBRdec/src/sbr_dec.h
index 175e7b2..edde637 100644
--- a/libSBRdec/src/sbr_dec.h
+++ b/libSBRdec/src/sbr_dec.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 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
@@ -118,6 +118,9 @@ typedef struct
   FIXP_DBL * WorkBuffer1;
   FIXP_DBL * WorkBuffer2;
 
+  /* Delayed time input signal needed to align CLDFD with LD-MPS QMF. */
+  INT_PCM    coreDelayBuf[(96)];
+
   /* QMF filter states */
   FIXP_QAS   anaQmfStates[(320)];
   FIXP_QSS * pSynQmfStates;
@@ -182,7 +185,8 @@ sbr_dec (HANDLE_SBR_DEC hSbrDec,            /*!< handle to Decoder channel */
          HANDLE_SBR_PREV_FRAME_DATA hPrevFrameData,  /*!< Some control data of last frame */
          const int applyProcessing,         /*!< Flag for SBR operation */
          HANDLE_PS_DEC h_ps_d,
-         const UINT flags
+         const UINT flags,
+         const int codecFrameSize
         );
 
 
diff --git a/libSBRdec/src/sbr_ram.h b/libSBRdec/src/sbr_ram.h
index f12631d..7ab5044 100644
--- a/libSBRdec/src/sbr_ram.h
+++ b/libSBRdec/src/sbr_ram.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 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
@@ -135,6 +135,7 @@ struct SBR_DECODER_INSTANCE
   USHORT codecFrameSize;
   UCHAR synDownsampleFac;
   UCHAR numDelayFrames;        /* The current number of additional delay frames used for processing. */
+  UCHAR numFlushedFrames;      /* The variable counts the number of frames which are flushed consecutively. */
 
   UINT flags;
 
diff --git a/libSBRdec/src/sbr_rom.cpp b/libSBRdec/src/sbr_rom.cpp
index e84c3cd..c48ce35 100644
--- a/libSBRdec/src/sbr_rom.cpp
+++ b/libSBRdec/src/sbr_rom.cpp
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 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
@@ -189,6 +189,15 @@ const FIXP_SGL FDK_sbrDecoder_sbr_limiterBandsPerOctaveDiv4[4] =
   FL2FXCONST_SGL(3.0f / 4.0f)
 };
 
+/*! Constants for calculating the number of limiter bands */
+const FIXP_DBL FDK_sbrDecoder_sbr_limiterBandsPerOctaveDiv4_DBL[4] =
+{
+  FL2FXCONST_DBL(1.0f / 4.0f),
+  FL2FXCONST_DBL(1.2f / 4.0f),
+  FL2FXCONST_DBL(2.0f / 4.0f),
+  FL2FXCONST_DBL(3.0f / 4.0f)
+};
+
 /*! Ratio of old gains and noise levels for the first 4 timeslots of an envelope */
 const FIXP_SGL FDK_sbrDecoder_sbr_smoothFilter[4] = {
   FL2FXCONST_SGL(0.66666666666666f),
diff --git a/libSBRdec/src/sbr_rom.h b/libSBRdec/src/sbr_rom.h
index c318870..1f800bc 100644
--- a/libSBRdec/src/sbr_rom.h
+++ b/libSBRdec/src/sbr_rom.h
@@ -2,7 +2,7 @@
 /* -----------------------------------------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 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
@@ -124,6 +124,7 @@ extern const FIXP_DBL FDK_sbrDecoder_sbr_whFactorsTable[NUM_WHFACTOR_TABLE_ENTRI
 extern const FIXP_SGL FDK_sbrDecoder_sbr_limGains_m[4];
 extern const UCHAR    FDK_sbrDecoder_sbr_limGains_e[4];
 extern const FIXP_SGL FDK_sbrDecoder_sbr_limiterBandsPerOctaveDiv4[4];
+extern const FIXP_DBL FDK_sbrDecoder_sbr_limiterBandsPerOctaveDiv4_DBL[4];
 extern const FIXP_SGL FDK_sbrDecoder_sbr_smoothFilter[4];
 extern const FIXP_SGL FDK_sbrDecoder_sbr_randomPhase[SBR_NF_NO_RANDOM_VAL][2];
 extern const FIXP_SGL harmonicPhaseX [2][4];
diff --git a/libSBRdec/src/sbrdecoder.cpp b/libSBRdec/src/sbrdecoder.cpp
index e15145a..f9ded54 100644
--- a/libSBRdec/src/sbrdecoder.cpp
+++ b/libSBRdec/src/sbrdecoder.cpp
@@ -128,6 +128,7 @@ amm-info@iis.fraunhofer.de
 #include "lpp_tran.h"
 #include "transcendent.h"
 
+#include "FDK_crc.h"
 
 #include "sbrdec_drc.h"
 
@@ -137,7 +138,7 @@ amm-info@iis.fraunhofer.de
 /* Decoder library info */
 #define SBRDECODER_LIB_VL0 2
 #define SBRDECODER_LIB_VL1 2
-#define SBRDECODER_LIB_VL2 7
+#define SBRDECODER_LIB_VL2 12
 #define SBRDECODER_LIB_TITLE "SBR Decoder"
 #ifdef __ANDROID__
 #define SBRDECODER_LIB_BUILD_DATE ""
@@ -194,6 +195,33 @@ static void copySbrHeader( HANDLE_SBR_HEADER_DATA hDst, const HANDLE_SBR_HEADER_
   hDst->freqBandData.freqBandTable[1] = hDst->freqBandData.freqBandTableHi;
 }
 
+static int compareSbrHeader( const HANDLE_SBR_HEADER_DATA hHdr1, const HANDLE_SBR_HEADER_DATA hHdr2 )
+{
+  int result = 0;
+
+  /* compare basic data */
+  result |= (hHdr1->syncState != hHdr2->syncState) ? 1 : 0;
+  result |= (hHdr1->status != hHdr2->status) ? 1 : 0;
+  result |= (hHdr1->frameErrorFlag != hHdr2->frameErrorFlag) ? 1 : 0;
+  result |= (hHdr1->numberTimeSlots != hHdr2->numberTimeSlots) ? 1 : 0;
+  result |= (hHdr1->numberOfAnalysisBands != hHdr2->numberOfAnalysisBands) ? 1 : 0;
+  result |= (hHdr1->timeStep != hHdr2->timeStep) ? 1 : 0;
+  result |= (hHdr1->sbrProcSmplRate != hHdr2->sbrProcSmplRate) ? 1 : 0;
+
+  /* compare bitstream data */
+  result |= FDKmemcmp( &hHdr1->bs_data, &hHdr2->bs_data, sizeof(SBR_HEADER_DATA_BS) );
+  result |= FDKmemcmp( &hHdr1->bs_info, &hHdr2->bs_info, sizeof(SBR_HEADER_DATA_BS_INFO) );
+
+  /* compare frequency band data */
+  result |= FDKmemcmp( &hHdr1->freqBandData, &hHdr2->freqBandData, (8+MAX_NUM_LIMITERS+1)*sizeof(UCHAR) );
+  result |= FDKmemcmp( hHdr1->freqBandData.freqBandTableLo, hHdr2->freqBandData.freqBandTableLo, (MAX_FREQ_COEFFS/2+1)*sizeof(UCHAR) );
+  result |= FDKmemcmp( hHdr1->freqBandData.freqBandTableHi, hHdr2->freqBandData.freqBandTableHi, (MAX_FREQ_COEFFS+1)*sizeof(UCHAR) );
+  result |= FDKmemcmp( hHdr1->freqBandData.freqBandTableNoise, hHdr2->freqBandData.freqBandTableNoise, (MAX_NOISE_COEFFS+1)*sizeof(UCHAR) );
+  result |= FDKmemcmp( hHdr1->freqBandData.v_k_master, hHdr2->freqBandData.v_k_master, (MAX_FREQ_COEFFS+1)*sizeof(UCHAR) );
+
+  return result;
+}
+
 
 /*!
   \brief Reset SBR decoder.
@@ -391,6 +419,7 @@ int sbrDecoder_isCoreCodecValid(AUDIO_OBJECT_TYPE coreCodec)
     case AOT_PS:
     case AOT_ER_AAC_SCAL:
     case AOT_ER_AAC_ELD:
+    case AOT_DRM_AAC:
       return 1;
     default:
       return 0;
@@ -463,6 +492,8 @@ SBR_ERROR sbrDecoder_InitElement (
 
   self->flags = 0;
   self->flags |= (coreCodec == AOT_ER_AAC_ELD) ? SBRDEC_ELD_GRID : 0;
+  self->flags |= (coreCodec == AOT_ER_AAC_SCAL) ? SBRDEC_SYNTAX_SCAL : 0;
+  self->flags |= (coreCodec == AOT_DRM_AAC)     ? SBRDEC_SYNTAX_SCAL|SBRDEC_SYNTAX_DRM : 0;
 
   /* Init SBR elements */
   {
@@ -928,24 +959,73 @@ SBR_ERROR sbrDecoder_Parse(
         )
 {
   SBR_DECODER_ELEMENT   *hSbrElement;
-  HANDLE_SBR_HEADER_DATA hSbrHeader;
+  HANDLE_SBR_HEADER_DATA hSbrHeader = NULL;
   HANDLE_SBR_CHANNEL    *pSbrChannel;
 
   SBR_FRAME_DATA *hFrameDataLeft;
   SBR_FRAME_DATA *hFrameDataRight;
 
   SBR_ERROR errorStatus = SBRDEC_OK;
-  SBR_SYNC_STATE initialSyncState;
   SBR_HEADER_STATUS headerStatus = HEADER_NOT_PRESENT;
 
   INT  startPos;
   INT  CRCLen = 0;
+  HANDLE_FDK_BITSTREAM hBsOriginal = hBs;
+  FDK_CRCINFO  crcInfo;         /* shall be used for all other CRCs in the future (TBD) */
+  INT          crcReg = 0;
+  USHORT       drmSbrCrc = 0;
 
   int  stereo;
   int  fDoDecodeSbrData = 1;
 
   int lastSlot, lastHdrSlot = 0, thisHdrSlot;
 
+  /* Reverse bits of DRM SBR payload */
+  if ( (self->flags & SBRDEC_SYNTAX_DRM) && *count > 0 )
+  {
+    UCHAR *bsBufferDrm = (UCHAR*)self->workBuffer1;
+    HANDLE_FDK_BITSTREAM hBsBwd = (HANDLE_FDK_BITSTREAM) (bsBufferDrm + (512));
+    int dataBytes, dataBits;
+
+    dataBits = *count;
+
+    if (dataBits > ((512)*8)) {
+      /* do not flip more data than needed */
+      dataBits = (512)*8;
+    }
+
+    dataBytes = (dataBits+7)>>3;
+
+    int j;
+
+    if ((j = (int)FDKgetValidBits(hBs)) != 8) {
+      FDKpushBiDirectional(hBs, (j-8));
+    }
+
+    j = 0;
+    for ( ; dataBytes > 0; dataBytes--)
+    {
+      int i;
+      UCHAR tmpByte;
+      UCHAR buffer = 0x00;
+
+      tmpByte = (UCHAR) FDKreadBits(hBs, 8);
+      for (i = 0; i < 4; i++) {
+        int shift = 2 * i + 1;
+        buffer |= (tmpByte & (0x08>>i)) << shift;
+        buffer |= (tmpByte & (0x10<<i)) >> shift;
+      }
+      bsBufferDrm[j++] = buffer;
+      FDKpushBack(hBs, 16);
+    }
+
+    FDKinitBitStream(hBsBwd, bsBufferDrm, (512), dataBits, BS_READER);
+
+    /* Use reversed data */
+    hBs = hBsBwd;
+    bsPayLen = *count;
+  }
+
   /* Remember start position of  SBR element */
   startPos = FDKgetValidBits(hBs);
 
@@ -970,7 +1050,6 @@ SBR_ERROR sbrDecoder_Parse(
   hFrameDataLeft  = &self->pSbrElement[elementIndex]->pSbrChannel[0]->frameData[hSbrElement->useFrameSlot];
   hFrameDataRight = &self->pSbrElement[elementIndex]->pSbrChannel[1]->frameData[hSbrElement->useFrameSlot];
 
-  initialSyncState = hSbrHeader->syncState;
 
   /* reset PS flag; will be set after PS was found */
   self->flags &= ~SBRDEC_PS_DECODED;
@@ -1006,12 +1085,19 @@ SBR_ERROR sbrDecoder_Parse(
   */
   if (fDoDecodeSbrData)
   {
-    if (crcFlag == 1) {
+    if (crcFlag) {
       switch (self->coreCodec) {
       case AOT_ER_AAC_ELD:
         FDKpushFor (hBs, 10);
         /* check sbrcrc later: we don't know the payload length now */
         break;
+      case AOT_DRM_AAC:
+        drmSbrCrc = (USHORT)FDKreadBits(hBs, 8);
+        /* Setup CRC decoder */
+        FDKcrcInit(&crcInfo, 0x001d, 0xFFFF, 8);
+        /* Start CRC region */
+        crcReg = FDKcrcStartReg(&crcInfo, hBs, 0);
+        break;
       default:
         CRCLen = bsPayLen - 10;                     /* change: 0 => i */
         if (CRCLen < 0) {
@@ -1056,6 +1142,7 @@ SBR_ERROR sbrDecoder_Parse(
         hSbrHeader->syncState = SBR_HEADER;
       } else {
         hSbrHeader->syncState = SBR_NOT_INITIALIZED;
+        headerStatus = HEADER_ERROR;
       }
     }
 
@@ -1105,7 +1192,7 @@ SBR_ERROR sbrDecoder_Parse(
         valBits = (INT)FDKgetValidBits(hBs);
       }
 
-      if ( crcFlag == 1 ) {
+      if ( crcFlag ) {
         switch (self->coreCodec) {
         case AOT_ER_AAC_ELD:
           {
@@ -1117,6 +1204,14 @@ SBR_ERROR sbrDecoder_Parse(
             FDKpushFor(hBs, crcLen);
           }
           break;
+        case AOT_DRM_AAC:
+          /* End CRC region */
+          FDKcrcEndReg(&crcInfo, hBs, crcReg);
+          /* Check CRC */
+          if ((FDKcrcGetCRC(&crcInfo)^0xFF) != drmSbrCrc) {
+            fDoDecodeSbrData = 0;
+          }
+          break;
         default:
           break;
         }
@@ -1167,8 +1262,25 @@ SBR_ERROR sbrDecoder_Parse(
   }
 
 bail:
-  if (errorStatus == SBRDEC_OK) {
-    if (headerStatus == HEADER_NOT_PRESENT) {
+
+  if ( self->flags & SBRDEC_SYNTAX_DRM )
+  {
+    hBs = hBsOriginal;
+  }
+
+  if ( (errorStatus == SBRDEC_OK)
+    || ( (errorStatus == SBRDEC_PARSE_ERROR)
+      && (headerStatus != HEADER_ERROR) ) )
+  {
+    int useOldHdr = ( (headerStatus == HEADER_NOT_PRESENT)
+                   || (headerStatus == HEADER_ERROR) ) ? 1 : 0;
+
+    if (!useOldHdr && (thisHdrSlot != lastHdrSlot)) {
+      useOldHdr |= ( compareSbrHeader( hSbrHeader,
+                                      &self->sbrHeader[elementIndex][lastHdrSlot] ) == 0 ) ? 1 : 0;
+    }
+
+    if (useOldHdr != 0) {
       /* Use the old header for this frame */
       hSbrElement->useHeaderSlot[hSbrElement->useFrameSlot] = lastHdrSlot;
     } else {
@@ -1229,12 +1341,21 @@ sbrDecoder_DecodeElement (
   int  numElementChannels = hSbrElement->nChannels; /* Number of channels of the current SBR element */
 
   if (self->flags & SBRDEC_FLUSH) {
-    /* Move frame pointer to the next slot which is up to be decoded/applied next */
-    hSbrElement->useFrameSlot = (hSbrElement->useFrameSlot+1) % (self->numDelayFrames+1);
-    /* Update header and frame data pointer because they have already been set */
-    hSbrHeader = &self->sbrHeader[elementIndex][hSbrElement->useHeaderSlot[hSbrElement->useFrameSlot]];
-    hFrameDataLeft  = &hSbrElement->pSbrChannel[0]->frameData[hSbrElement->useFrameSlot];
-    hFrameDataRight = &hSbrElement->pSbrChannel[1]->frameData[hSbrElement->useFrameSlot];
+    if ( self->numFlushedFrames > self->numDelayFrames ) {
+      int hdrIdx;
+      /* No valid SBR payload available, hence switch to upsampling (in all headers) */
+      for (hdrIdx = 0; hdrIdx < ((1)+1); hdrIdx += 1) {
+        self->sbrHeader[elementIndex][hdrIdx].syncState = UPSAMPLING;
+      }
+    }
+    else {
+      /* Move frame pointer to the next slot which is up to be decoded/applied next */
+      hSbrElement->useFrameSlot = (hSbrElement->useFrameSlot+1) % (self->numDelayFrames+1);
+      /* Update header and frame data pointer because they have already been set */
+      hSbrHeader = &self->sbrHeader[elementIndex][hSbrElement->useHeaderSlot[hSbrElement->useFrameSlot]];
+      hFrameDataLeft  = &hSbrElement->pSbrChannel[0]->frameData[hSbrElement->useFrameSlot];
+      hFrameDataRight = &hSbrElement->pSbrChannel[1]->frameData[hSbrElement->useFrameSlot];
+    }
   }
 
   /* Update the header error flag */
@@ -1354,7 +1475,8 @@ sbrDecoder_DecodeElement (
            &pSbrChannel[0]->prevFrameData,
             (hSbrHeader->syncState == SBR_ACTIVE),
             h_ps_d,
-            self->flags
+            self->flags,
+            codecFrameSize
           );
 
   if (stereo) {
@@ -1371,7 +1493,8 @@ sbrDecoder_DecodeElement (
              &pSbrChannel[1]->prevFrameData,
               (hSbrHeader->syncState == SBR_ACTIVE),
               NULL,
-              self->flags
+              self->flags,
+              codecFrameSize
             );
   }
 
@@ -1387,20 +1510,21 @@ sbrDecoder_DecodeElement (
     if ( !(self->flags & SBRDEC_PS_DECODED) ) {
       /* A decoder which is able to decode PS has to produce a stereo output even if no PS data is availble. */
       /* So copy left channel to right channel.                                                              */
+      int copyFrameSize = codecFrameSize * 2 / self->synDownsampleFac;
       if (interleaved) {
         INT_PCM *ptr;
         INT i;
         FDK_ASSERT(strideOut == 2);
 
         ptr = timeData;
-        for (i = codecFrameSize; i--; ) 
+        for (i = copyFrameSize>>1; i--; )
         {
           INT_PCM tmp; /* This temporal variable is required because some compilers can't do *ptr++ = *ptr++ correctly. */
           tmp = *ptr++; *ptr++ = tmp;
           tmp = *ptr++; *ptr++ = tmp;
         }
       } else {
-        FDKmemcpy( timeData+2*codecFrameSize, timeData, 2*codecFrameSize*sizeof(INT_PCM) );
+        FDKmemcpy( timeData+copyFrameSize, timeData, copyFrameSize*sizeof(INT_PCM) );
       }
     }
     *numOutChannels = 2;  /* Output minimum two channels when PS is enabled. */
@@ -1464,14 +1588,23 @@ SBR_ERROR sbrDecoder_Apply ( HANDLE_SBRDECODER   self,
     self->flags &= ~SBRDEC_PS_DECODED;
   }
 
+  if ( self->flags & SBRDEC_FLUSH ) {
+    /* flushing is signalized, hence increment the flush frame counter */
+    self->numFlushedFrames++;
+  }
+  else {
+    /* no flushing is signalized, hence reset the flush frame counter */
+    self->numFlushedFrames = 0;
+  }
+
   /* Loop over SBR elements */
   for (sbrElementNum = 0; sbrElementNum<self->numSbrElements; sbrElementNum++)
   {
     int numElementChan;
 
     if (psPossible && self->pSbrElement[sbrElementNum]->pSbrChannel[1] == NULL) {
-      errorStatus = SBRDEC_UNSUPPORTED_CONFIG;
-      goto bail;
+      /* Disable PS and try decoding SBR mono. */
+      psPossible = 0;
     }
 
     numElementChan = (self->pSbrElement[sbrElementNum]->elementID == ID_CPE) ? 2 : 1;
@@ -1579,6 +1712,7 @@ INT sbrDecoder_GetLibInfo( LIB_INFO *info )
     | CAPF_SBR_HQ
     | CAPF_SBR_LP
     | CAPF_SBR_PS_MPEG
+    | CAPF_SBR_DRM_BS
     | CAPF_SBR_CONCEALMENT
     | CAPF_SBR_DRC
       ;
@@ -1607,6 +1741,9 @@ UINT sbrDecoder_GetDelay( const HANDLE_SBRDECODER self )
         /* Low delay SBR: */
         {
           outputDelay += (flags & SBRDEC_DOWNSAMPLE) ? 32 : 64;   /* QMF synthesis */
+          if (flags & SBRDEC_LD_MPS_QMF) {
+            outputDelay += 32;
+          }
         }
       }
       else if (!IS_USAC(self->coreCodec)) {