Update fdk to v2.0.2

15 files changed, 234 insertions, 511 deletions

diff --git a/fdk-aac/libSBRdec/include/sbrdecoder.h b/fdk-aac/libSBRdec/include/sbrdecoder.h
index cc55572..c09c985 100644
--- a/fdk-aac/libSBRdec/include/sbrdecoder.h
+++ b/fdk-aac/libSBRdec/include/sbrdecoder.h
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -361,15 +361,20 @@ SBR_ERROR sbrDecoder_Parse(HANDLE_SBRDECODER self, HANDLE_FDK_BITSTREAM hBs,
  * error (0: core decoder found errors, 1: no errors).
  * \param psDecoded       Pointer to a buffer holding a flag. Input: PS is
  * possible, Output: PS has been rendered.
+ * \param inDataHeadroom  Headroom of the SBR input time signal to prevent
+ * clipping.
+ * \param outDataHeadroom Pointer to headroom of the SBR output time signal to
+ * prevent clipping.
  *
  * \return  Error code.
  */
-SBR_ERROR sbrDecoder_Apply(HANDLE_SBRDECODER self, INT_PCM *input,
-                           INT_PCM *timeData, const int timeDataSize,
-                           int *numChannels, int *sampleRate,
+SBR_ERROR sbrDecoder_Apply(HANDLE_SBRDECODER self, LONG *input, LONG *timeData,
+                           const int timeDataSize, int *numChannels,
+                           int *sampleRate,
                            const FDK_channelMapDescr *const mapDescr,
                            const int mapIdx, const int coreDecodedOk,
-                           UCHAR *psDecoded);
+                           UCHAR *psDecoded, const INT inDataHeadroom,
+                           INT *outDataHeadroom);
 
 /**
  * \brief       Close SBR decoder instance and free memory.
diff --git a/fdk-aac/libSBRdec/src/HFgen_preFlat.cpp b/fdk-aac/libSBRdec/src/HFgen_preFlat.cpp
index 96adbb9..ad4caba 100644
--- a/fdk-aac/libSBRdec/src/HFgen_preFlat.cpp
+++ b/fdk-aac/libSBRdec/src/HFgen_preFlat.cpp
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -897,30 +897,31 @@ void sbrDecoder_calculateGainVec(FIXP_DBL **sourceBufferReal,
     for (i = startSample; i < stopSample; i++) {
       maxVal |=
           (FIXP_DBL)((LONG)(sourceBufferReal[i][loBand]) ^
-                     ((LONG)sourceBufferReal[i][loBand] >> (SAMPLE_BITS - 1)));
+                     ((LONG)sourceBufferReal[i][loBand] >> (DFRACT_BITS - 1)));
       maxVal |=
           (FIXP_DBL)((LONG)(sourceBufferImag[i][loBand]) ^
-                     ((LONG)sourceBufferImag[i][loBand] >> (SAMPLE_BITS - 1)));
+                     ((LONG)sourceBufferImag[i][loBand] >> (DFRACT_BITS - 1)));
     }
 
     if (maxVal != FL2FX_DBL(0.0f)) {
-      reserve = fixMax(0, CntLeadingZeros(maxVal) - 2);
+      reserve = CntLeadingZeros(maxVal) - 2;
     }
 
     nrg_ov = nrg = (FIXP_DBL)0;
     if (scale_nrg_ov > -31) {
       for (i = startSample; i < overlap; i++) {
-        nrg_ov += (fPow2Div2(sourceBufferReal[i][loBand] << reserve) +
-                   fPow2Div2(sourceBufferImag[i][loBand] << reserve)) >>
-                  sum_scale_ov;
+        nrg_ov +=
+            (fPow2Div2(scaleValue(sourceBufferReal[i][loBand], reserve)) +
+             fPow2Div2(scaleValue(sourceBufferImag[i][loBand], reserve))) >>
+            sum_scale_ov;
       }
     } else {
       scale_nrg_ov = 0;
     }
     if (scale_nrg > -31) {
       for (i = overlap; i < stopSample; i++) {
-        nrg += (fPow2Div2(sourceBufferReal[i][loBand] << reserve) +
-                fPow2Div2(sourceBufferImag[i][loBand] << reserve)) >>
+        nrg += (fPow2Div2(scaleValue(sourceBufferReal[i][loBand], reserve)) +
+                fPow2Div2(scaleValue(sourceBufferImag[i][loBand], reserve))) >>
                sum_scale;
       }
     } else {
diff --git a/fdk-aac/libSBRdec/src/env_calc.cpp b/fdk-aac/libSBRdec/src/env_calc.cpp
index 1242833..0b2f651 100644
--- a/fdk-aac/libSBRdec/src/env_calc.cpp
+++ b/fdk-aac/libSBRdec/src/env_calc.cpp
@@ -151,6 +151,9 @@ amm-info@iis.fraunhofer.de
 
 #include "genericStds.h" /* need FDKpow() for debug outputs */
 
+#define MAX_SFB_NRG_HEADROOM (1)
+#define MAX_VAL_NRG_HEADROOM ((((FIXP_DBL)MAXVAL_DBL) >> MAX_SFB_NRG_HEADROOM))
+
 typedef struct {
   FIXP_DBL nrgRef[MAX_FREQ_COEFFS];
   FIXP_DBL nrgEst[MAX_FREQ_COEFFS];
@@ -699,20 +702,11 @@ static void apply_inter_tes(FIXP_DBL **qmfReal, FIXP_DBL **qmfImag,
       gain_sf[i] += gamma_sf + 1; /* +1 because of fMultDiv2() */
 
       /* set gain to at least 0.2f */
-      FIXP_DBL point_two = FL2FXCONST_DBL(0.8f); /* scaled up by 2 */
-      int point_two_sf = -2;
-
-      FIXP_DBL tmp = gain[i];
-      if (point_two_sf < gain_sf[i]) {
-        point_two >>= gain_sf[i] - point_two_sf;
-      } else {
-        tmp >>= point_two_sf - gain_sf[i];
-      }
-
       /* limit and calculate gain[i]^2 too */
       FIXP_DBL gain_pow2;
       int gain_pow2_sf;
-      if (tmp < point_two) {
+
+      if (fIsLessThan(gain[i], gain_sf[i], FL2FXCONST_DBL(0.2f), 0)) {
         gain[i] = FL2FXCONST_DBL(0.8f);
         gain_sf[i] = -2;
         gain_pow2 = FL2FXCONST_DBL(0.64f);
@@ -739,7 +733,8 @@ static void apply_inter_tes(FIXP_DBL **qmfReal, FIXP_DBL **qmfImag,
             fMin(DFRACT_BITS - 1, new_summand_sf - total_power_high_after_sf);
         total_power_high_after_sf = new_summand_sf;
       } else if (new_summand_sf < total_power_high_after_sf) {
-        subsample_power_high[i] >>= total_power_high_after_sf - new_summand_sf;
+        subsample_power_high[i] >>=
+            fMin(DFRACT_BITS - 1, total_power_high_after_sf - new_summand_sf);
       }
       total_power_high_after += subsample_power_high[i] >> preShift2;
     }
@@ -985,7 +980,8 @@ void calculateSbrEnvelope(
   */
   if (!useLP)
     adj_e = h_sbr_cal_env->filtBufferNoise_e -
-            getScalefactor(h_sbr_cal_env->filtBufferNoise, noSubbands);
+            getScalefactor(h_sbr_cal_env->filtBufferNoise, noSubbands) +
+            (INT)MAX_SFB_NRG_HEADROOM;
 
   /*
     Scan for maximum reference energy to be able
@@ -1005,7 +1001,7 @@ void calculateSbrEnvelope(
        - Smoothing can smear high gains of the previous envelope into the
        current
     */
-    maxSfbNrg_e += 6;
+    maxSfbNrg_e += (6 + MAX_SFB_NRG_HEADROOM);
 
     adj_e = maxSfbNrg_e;
     // final_e should not exist for PVC fixfix framing
@@ -1031,7 +1027,7 @@ void calculateSbrEnvelope(
          - Smoothing can smear high gains of the previous envelope into the
          current
       */
-      maxSfbNrg_e += 6;
+      maxSfbNrg_e += (6 + MAX_SFB_NRG_HEADROOM);
 
       if (borders[i] < hHeaderData->numberTimeSlots)
         /* This envelope affects timeslots that belong to the output frame */
@@ -1477,7 +1473,7 @@ void calculateSbrEnvelope(
 
       for (k = 0; k < noSubbands; k++) {
         int sc = scale_change - pNrgs->nrgGain_e[k] + (sc_change - 1);
-        pNrgs->nrgGain[k] >>= sc;
+        pNrgs->nrgGain[k] >>= fixMin(sc, DFRACT_BITS - 1);
         pNrgs->nrgGain_e[k] += sc;
       }
 
@@ -1485,7 +1481,7 @@ void calculateSbrEnvelope(
         for (k = 0; k < noSubbands; k++) {
           int sc =
               scale_change - h_sbr_cal_env->filtBuffer_e[k] + (sc_change - 1);
-          h_sbr_cal_env->filtBuffer[k] >>= sc;
+          h_sbr_cal_env->filtBuffer[k] >>= fixMin(sc, DFRACT_BITS - 1);
         }
       }
 
@@ -1576,12 +1572,13 @@ void calculateSbrEnvelope(
           FDK_ASSERT(!iTES_enable); /* not supported */
           if (flags & SBRDEC_ELD_GRID) {
             /* FDKmemset(analysBufferReal[j], 0, 64 * sizeof(FIXP_DBL)); */
-            adjustTimeSlot_EldGrid(&analysBufferReal[j][lowSubband], pNrgs,
-                                   &h_sbr_cal_env->harmIndex, lowSubband,
-                                   noSubbands,
-                                   fMin(scale_change, DFRACT_BITS - 1),
-                                   noNoiseFlag, &h_sbr_cal_env->phaseIndex,
-                                   EXP2SCALE(adj_e) - sbrScaleFactor->lb_scale);
+            adjustTimeSlot_EldGrid(
+                &analysBufferReal[j][lowSubband], pNrgs,
+                &h_sbr_cal_env->harmIndex, lowSubband, noSubbands,
+                fMin(scale_change, DFRACT_BITS - 1), noNoiseFlag,
+                &h_sbr_cal_env->phaseIndex,
+                fMax(EXP2SCALE(adj_e) - sbrScaleFactor->lb_scale,
+                     -(DFRACT_BITS - 1)));
           } else {
             adjustTimeSlotLC(&analysBufferReal[j][lowSubband], pNrgs,
                              &h_sbr_cal_env->harmIndex, lowSubband, noSubbands,
@@ -1830,7 +1827,8 @@ static void equalizeFiltBufferExp(
     diff = (int)(nrgGain_e[band] - filtBuffer_e[band]);
     if (diff > 0) {
       filtBuffer[band] >>=
-          diff; /* Compensate for the scale change by shifting the mantissa. */
+          fMin(diff, DFRACT_BITS - 1); /* Compensate for the scale change by
+                                          shifting the mantissa. */
       filtBuffer_e[band] += diff; /* New gain is bigger, use its exponent */
     } else if (diff < 0) {
       /* The buffered gains seem to be larger, but maybe there
@@ -1850,8 +1848,8 @@ static void equalizeFiltBufferExp(
         filtBuffer_e[band] -= reserve; /* Compensate in the exponent: */
 
         /* For the remaining difference, change the new gain value */
-        diff = fixMin(-(reserve + diff), DFRACT_BITS - 1);
-        nrgGain[band] >>= diff;
+        diff = -(reserve + diff);
+        nrgGain[band] >>= fMin(diff, DFRACT_BITS - 1);
         nrgGain_e[band] += diff;
       }
     }
@@ -2423,6 +2421,9 @@ static void adjustTimeSlot_EldGrid(
   const FIXP_DBL *p_harmonicPhaseX = &harmonicPhaseX[harmIndex][0];
   const INT *p_harmonicPhase = &harmonicPhase[harmIndex][0];
 
+  const FIXP_DBL max_val = MAX_VAL_NRG_HEADROOM >> scale_change;
+  const FIXP_DBL min_val = -max_val;
+
   *(ptrReal - 1) = fAddSaturate(
       *(ptrReal - 1),
       SATURATE_SHIFT(fMultDiv2(p_harmonicPhaseX[lowSubband & 1], pSineLevel[0]),
@@ -2435,7 +2436,8 @@ static void adjustTimeSlot_EldGrid(
     FIXP_DBL sineLevel_curr = *pSineLevel++;
     phaseIndex = (phaseIndex + 1) & (SBR_NF_NO_RANDOM_VAL - 1);
 
-    signalReal = fMultDiv2(*ptrReal, *pGain++) << ((int)scale_change);
+    signalReal = fMax(fMin(fMultDiv2(*ptrReal, *pGain++), max_val), min_val)
+                 << scale_change;
     sbNoise = *pNoiseLevel++;
     if (((INT)sineLevel_curr | noNoiseFlag) == 0) {
       signalReal +=
@@ -2469,7 +2471,8 @@ static void adjustTimeSlot_EldGrid(
     FIXP_DBL sineLevel_curr = *pSineLevel++;
     phaseIndex = (phaseIndex + 1) & (SBR_NF_NO_RANDOM_VAL - 1);
 
-    signalReal = fMultDiv2(*ptrReal, *pGain++) << ((int)scale_change);
+    signalReal = fMax(fMin(fMultDiv2(*ptrReal, *pGain++), max_val), min_val)
+                 << scale_change;
     sbNoise = *pNoiseLevel++;
     if (((INT)sineLevel_curr | noNoiseFlag) == 0) {
       signalReal +=
@@ -2509,6 +2512,8 @@ static void adjustTimeSlotLC(
   FIXP_DBL signalReal, sineLevel, sineLevelNext, sineLevelPrev;
   int tone_count = 0;
   int sineSign = 1;
+  const FIXP_DBL max_val = MAX_VAL_NRG_HEADROOM >> scale_change;
+  const FIXP_DBL min_val = -max_val;
 
 #define C1 ((FIXP_SGL)FL2FXCONST_SGL(2.f * 0.00815f))
 #define C1_CLDFB ((FIXP_SGL)FL2FXCONST_SGL(2.f * 0.16773f))
@@ -2524,7 +2529,8 @@ static void adjustTimeSlotLC(
     of the signal and should be carried out with full accuracy
     (supplying #FRACT_BITS valid bits).
   */
-  signalReal = fMultDiv2(*ptrReal, *pGain++) << ((int)scale_change);
+  signalReal = fMax(fMin(fMultDiv2(*ptrReal, *pGain++), max_val), min_val)
+               << scale_change;
   sineLevel = *pSineLevel++;
   sineLevelNext = (noSubbands > 1) ? pSineLevel[0] : FL2FXCONST_DBL(0.0f);
 
@@ -2552,10 +2558,10 @@ static void adjustTimeSlotLC(
 
       /* save switch and compare operations and reduce to XOR statement */
       if (((harmIndex >> 1) & 0x1) ^ freqInvFlag) {
-        *(ptrReal - 1) += tmp1;
+        *(ptrReal - 1) = fAddSaturate(*(ptrReal - 1), tmp1);
         signalReal -= tmp2;
       } else {
-        *(ptrReal - 1) -= tmp1;
+        *(ptrReal - 1) = fAddSaturate(*(ptrReal - 1), -tmp1);
         signalReal += tmp2;
       }
       *ptrReal++ = signalReal;
@@ -2586,7 +2592,9 @@ static void adjustTimeSlotLC(
 
         /* The next multiplication constitutes the actual envelope adjustment of
          * the signal. */
-        signalReal += fMultDiv2(*ptrReal, *pGain++) << ((int)scale_change);
+        signalReal +=
+            fMax(fMin(fMultDiv2(*ptrReal, *pGain++), max_val), min_val)
+            << scale_change;
 
         pNoiseLevel++;
         *ptrReal++ = signalReal;
@@ -2599,7 +2607,8 @@ static void adjustTimeSlotLC(
         index++;
         /* The next multiplication constitutes the actual envelope adjustment of
          * the signal. */
-        signalReal = fMultDiv2(*ptrReal, *pGain++) << ((int)scale_change);
+        signalReal = fMax(fMin(fMultDiv2(*ptrReal, *pGain++), max_val), min_val)
+                     << scale_change;
 
         if (*pSineLevel++ != FL2FXCONST_DBL(0.0f))
           tone_count++;
@@ -2627,7 +2636,8 @@ static void adjustTimeSlotLC(
     index++;
     /* The next multiplication constitutes the actual envelope adjustment of the
      * signal. */
-    signalReal = fMultDiv2(*ptrReal, *pGain) << ((int)scale_change);
+    signalReal = fMax(fMin(fMultDiv2(*ptrReal, *pGain), max_val), min_val)
+                 << scale_change;
     sineLevelPrev = fMultDiv2(pSineLevel[-1], FL2FX_SGL(0.0163f));
     sineLevel = pSineLevel[0];
 
@@ -2696,6 +2706,9 @@ static void adjustTimeSlotHQ_GainAndNoise(
       /*FL2FXCONST_SGL(1.0f) */ (FIXP_SGL)MAXVAL_SGL - smooth_ratio;
   int index = *ptrPhaseIndex;
   int shift;
+  FIXP_DBL max_val_noise = 0, min_val_noise = 0;
+  const FIXP_DBL max_val = MAX_VAL_NRG_HEADROOM >> scale_change;
+  const FIXP_DBL min_val = -max_val;
 
   *ptrPhaseIndex = (index + noSubbands) & (SBR_NF_NO_RANDOM_VAL - 1);
 
@@ -2705,6 +2718,8 @@ static void adjustTimeSlotHQ_GainAndNoise(
     shift = fixMin(DFRACT_BITS - 1, -filtBufferNoiseShift);
   } else {
     shift = fixMin(DFRACT_BITS - 1, filtBufferNoiseShift);
+    max_val_noise = MAX_VAL_NRG_HEADROOM >> shift;
+    min_val_noise = -max_val_noise;
   }
 
   if (smooth_ratio > FL2FXCONST_SGL(0.0f)) {
@@ -2720,8 +2735,10 @@ static void adjustTimeSlotHQ_GainAndNoise(
         smoothedNoise = (fMultDiv2(smooth_ratio, filtBufferNoise[k]) >> shift) +
                         fMult(direct_ratio, noiseLevel[k]);
       } else {
-        smoothedNoise = (fMultDiv2(smooth_ratio, filtBufferNoise[k]) << shift) +
-                        fMult(direct_ratio, noiseLevel[k]);
+        smoothedNoise = fMultDiv2(smooth_ratio, filtBufferNoise[k]);
+        smoothedNoise =
+            (fMax(fMin(smoothedNoise, max_val_noise), min_val_noise) << shift) +
+            fMult(direct_ratio, noiseLevel[k]);
       }
 
       /*
@@ -2729,8 +2746,12 @@ static void adjustTimeSlotHQ_GainAndNoise(
         of the signal and should be carried out with full accuracy
         (supplying #DFRACT_BITS valid bits).
       */
-      signalReal = fMultDiv2(*ptrReal, smoothedGain) << ((int)scale_change);
-      signalImag = fMultDiv2(*ptrImag, smoothedGain) << ((int)scale_change);
+      signalReal =
+          fMax(fMin(fMultDiv2(*ptrReal, smoothedGain), max_val), min_val)
+          << scale_change;
+      signalImag =
+          fMax(fMin(fMultDiv2(*ptrImag, smoothedGain), max_val), min_val)
+          << scale_change;
 
       index++;
 
@@ -2752,8 +2773,12 @@ static void adjustTimeSlotHQ_GainAndNoise(
   } else {
     for (k = 0; k < noSubbands; k++) {
       smoothedGain = gain[k];
-      signalReal = fMultDiv2(*ptrReal, smoothedGain) << scale_change;
-      signalImag = fMultDiv2(*ptrImag, smoothedGain) << scale_change;
+      signalReal =
+          fMax(fMin(fMultDiv2(*ptrReal, smoothedGain), max_val), min_val)
+          << scale_change;
+      signalImag =
+          fMax(fMin(fMultDiv2(*ptrImag, smoothedGain), max_val), min_val)
+          << scale_change;
 
       index++;
 
@@ -2859,6 +2884,9 @@ static void adjustTimeSlotHQ(
   int freqInvFlag = (lowSubband & 1);
   FIXP_DBL sineLevel;
   int shift;
+  FIXP_DBL max_val_noise = 0, min_val_noise = 0;
+  const FIXP_DBL max_val = MAX_VAL_NRG_HEADROOM >> scale_change;
+  const FIXP_DBL min_val = -max_val;
 
   *ptrPhaseIndex = (index + noSubbands) & (SBR_NF_NO_RANDOM_VAL - 1);
   *ptrHarmIndex = (harmIndex + 1) & 3;
@@ -2874,10 +2902,13 @@ static void adjustTimeSlotHQ(
 
   filtBufferNoiseShift +=
       1; /* due to later use of fMultDiv2 instead of fMult */
-  if (filtBufferNoiseShift < 0)
+  if (filtBufferNoiseShift < 0) {
     shift = fixMin(DFRACT_BITS - 1, -filtBufferNoiseShift);
-  else
+  } else {
     shift = fixMin(DFRACT_BITS - 1, filtBufferNoiseShift);
+    max_val_noise = MAX_VAL_NRG_HEADROOM >> shift;
+    min_val_noise = -max_val_noise;
+  }
 
   if (smooth_ratio > FL2FXCONST_SGL(0.0f)) {
     for (k = 0; k < noSubbands; k++) {
@@ -2893,8 +2924,10 @@ static void adjustTimeSlotHQ(
         smoothedNoise = (fMultDiv2(smooth_ratio, filtBufferNoise[k]) >> shift) +
                         fMult(direct_ratio, noiseLevel[k]);
       } else {
-        smoothedNoise = (fMultDiv2(smooth_ratio, filtBufferNoise[k]) << shift) +
-                        fMult(direct_ratio, noiseLevel[k]);
+        smoothedNoise = fMultDiv2(smooth_ratio, filtBufferNoise[k]);
+        smoothedNoise =
+            (fMax(fMin(smoothedNoise, max_val_noise), min_val_noise) << shift) +
+            fMult(direct_ratio, noiseLevel[k]);
       }
 
       /*
@@ -2902,8 +2935,12 @@ static void adjustTimeSlotHQ(
         of the signal and should be carried out with full accuracy
         (supplying #DFRACT_BITS valid bits).
       */
-      signalReal = fMultDiv2(*ptrReal, smoothedGain) << ((int)scale_change);
-      signalImag = fMultDiv2(*ptrImag, smoothedGain) << ((int)scale_change);
+      signalReal =
+          fMax(fMin(fMultDiv2(*ptrReal, smoothedGain), max_val), min_val)
+          << scale_change;
+      signalImag =
+          fMax(fMin(fMultDiv2(*ptrImag, smoothedGain), max_val), min_val)
+          << scale_change;
 
       index++;
 
@@ -2956,8 +2993,12 @@ static void adjustTimeSlotHQ(
   } else {
     for (k = 0; k < noSubbands; k++) {
       smoothedGain = gain[k];
-      signalReal = fMultDiv2(*ptrReal, smoothedGain) << scale_change;
-      signalImag = fMultDiv2(*ptrImag, smoothedGain) << scale_change;
+      signalReal =
+          fMax(fMin(fMultDiv2(*ptrReal, smoothedGain), max_val), min_val)
+          << scale_change;
+      signalImag =
+          fMax(fMin(fMultDiv2(*ptrImag, smoothedGain), max_val), min_val)
+          << scale_change;
 
       index++;
 
@@ -3141,6 +3182,11 @@ ResetLimiterBands(
       return SBRDEC_UNSUPPORTED_CONFIG;
     }
 
+    /* Restrict maximum value of limiter band table */
+    if (workLimiterBandTable[tempNoLim] > highSubband) {
+      return SBRDEC_UNSUPPORTED_CONFIG;
+    }
+
     /* Copy limiterbands from working buffer into final destination */
     for (k = 0; k <= nBands; k++) {
       limiterBandTable[k] = workLimiterBandTable[k];
diff --git a/fdk-aac/libSBRdec/src/hbe.cpp b/fdk-aac/libSBRdec/src/hbe.cpp
index 3310dcd..77cb8af 100644
--- a/fdk-aac/libSBRdec/src/hbe.cpp
+++ b/fdk-aac/libSBRdec/src/hbe.cpp
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -957,7 +957,7 @@ QmfTransposerCreate(HANDLE_HBE_TRANSPOSER* hQmfTransposer, const int frameSize,
     hQmfTran->qmfOutBufSize = 2 * (hQmfTran->noCols / 2 + QMF_WIN_LEN - 1);
 
     hQmfTran->inBuf_F =
-        (INT_PCM*)FDKcalloc(QMF_SYNTH_CHANNELS + 20 + 1, sizeof(INT_PCM));
+        (LONG*)FDKcalloc(QMF_SYNTH_CHANNELS + 20 + 1, sizeof(LONG));
     /* buffered time signal needs to be delayed by synthesis_size; max
      * synthesis_size = 20; */
     if (hQmfTran->inBuf_F == NULL) {
@@ -1339,7 +1339,7 @@ static void addHighBandPart(FIXP_DBL g_r_m, FIXP_DBL g_i_m, INT g_e,
     g_r_m = fMultDiv2(tmp_r, factor_m) << shift;
     g_i_m = fMultDiv2(tmp_i, factor_m) << shift;
     g_e = scale_factor_hbe - (g_e + factor_e + gammaCenter_e + add);
-    fMax((INT)0, g_e);
+    g_e = fMax((INT)0, g_e);
     *qmfHBEBufReal_F += g_r_m >> g_e;
     *qmfHBEBufImag_F += g_i_m >> g_e;
   }
diff --git a/fdk-aac/libSBRdec/src/hbe.h b/fdk-aac/libSBRdec/src/hbe.h
index fdffe1e..3556783 100644
--- a/fdk-aac/libSBRdec/src/hbe.h
+++ b/fdk-aac/libSBRdec/src/hbe.h
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -132,6 +132,9 @@ typedef enum {
 } KEEP_STATES_SYNCED_MODE;
 
 struct hbeTransposer {
+  FIXP_DBL anaQmfStates[HBE_QMF_FILTER_STATE_ANA_SIZE];
+  FIXP_QSS synQmfStates[HBE_QMF_FILTER_STATE_SYN_SIZE];
+
   int xOverQmf[MAX_NUM_PATCHES_HBE];
 
   int maxStretch;
@@ -144,7 +147,7 @@ struct hbeTransposer {
   int stopBand;
   int bSbr41;
 
-  INT_PCM *inBuf_F;
+  LONG *inBuf_F;
   FIXP_DBL **qmfInBufReal_F;
   FIXP_DBL **qmfInBufImag_F;
 
@@ -156,9 +159,6 @@ struct hbeTransposer {
   FIXP_DBL const *synthesisQmfPreModCos_F;
   FIXP_DBL const *synthesisQmfPreModSin_F;
 
-  FIXP_QAS anaQmfStates[HBE_QMF_FILTER_STATE_ANA_SIZE];
-  FIXP_QSS synQmfStates[HBE_QMF_FILTER_STATE_SYN_SIZE];
-
   FIXP_DBL **qmfHBEBufReal_F;
   FIXP_DBL **qmfHBEBufImag_F;
 
diff --git a/fdk-aac/libSBRdec/src/lpp_tran.cpp b/fdk-aac/libSBRdec/src/lpp_tran.cpp
index 6acb626..93e1158 100644
--- a/fdk-aac/libSBRdec/src/lpp_tran.cpp
+++ b/fdk-aac/libSBRdec/src/lpp_tran.cpp
@@ -1014,8 +1014,8 @@ void lppTransposerHBE(
                               pSettings->nCols) +
                    lowBandShift);
 
-    dynamicScale = fixMax(
-        0, dynamicScale - 1); /* one additional bit headroom to prevent -1.0 */
+    dynamicScale =
+        dynamicScale - 1; /* one additional bit headroom to prevent -1.0 */
 
     /*
     Scale temporal QMF buffer.
@@ -1194,6 +1194,9 @@ void lppTransposerHBE(
     } else { /* bw <= 0 */
 
       int descale = fixMin(DFRACT_BITS - 1, (LPC_SCALE_FACTOR + dynamicScale));
+      dynamicScale +=
+          1; /* prevent negativ scale factor due to 'one additional bit
+                headroom' */
 
       for (i = startSample; i < stopSample; i++) {
         FIXP_DBL accu1, accu2;
@@ -1210,9 +1213,9 @@ void lppTransposerHBE(
                 dynamicScale;
 
         qmfBufferReal[i][loBand] =
-            (lowBandReal[LPC_ORDER + i] >> descale) + (accu1 << 1);
+            (lowBandReal[LPC_ORDER + i] >> descale) + (accu1 << (1 + 1));
         qmfBufferImag[i][loBand] =
-            (lowBandImag[LPC_ORDER + i] >> descale) + (accu2 << 1);
+            (lowBandImag[LPC_ORDER + i] >> descale) + (accu2 << (1 + 1));
       }
     } /* bw <= 0 */
 
diff --git a/fdk-aac/libSBRdec/src/pvc_dec.cpp b/fdk-aac/libSBRdec/src/pvc_dec.cpp
index b477122..e1e3c2c 100644
--- a/fdk-aac/libSBRdec/src/pvc_dec.cpp
+++ b/fdk-aac/libSBRdec/src/pvc_dec.cpp
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -534,7 +534,8 @@ void pvcDecodeTimeSlot(PVC_STATIC_DATA *pPvcStaticData,
     for (ksg = ksg_start; ksg < PVC_NBLOW; ksg++) {
       for (band = sg_borders[ksg]; band < sg_borders[ksg + 1]; band++) {
         /* The division by 8 == (RATE*lbw) is required algorithmically */
-        E[ksg] += (fPow2Div2(qmfR[band]) + fPow2Div2(qmfI[band])) >> 2;
+        E[ksg] +=
+            ((fPow2Div2(qmfR[band]) >> 1) + (fPow2Div2(qmfI[band]) >> 1)) >> 3;
       }
     }
   }
@@ -542,7 +543,7 @@ void pvcDecodeTimeSlot(PVC_STATIC_DATA *pPvcStaticData,
     if (E[ksg] > (FIXP_DBL)0) {
       /* 10/log2(10) = 0.752574989159953 * 2^2 */
       int exp_log;
-      FIXP_DBL nrg = CalcLog2(E[ksg], 2 * qmfExponent, &exp_log);
+      FIXP_DBL nrg = CalcLog2(E[ksg], 2 * qmfExponent + 2, &exp_log);
       nrg = fMult(nrg, FL2FXCONST_SGL(LOG10FAC));
       nrg = scaleValue(nrg, exp_log - PVC_ESG_EXP + 2);
       pEsg[ksg] = fMax(nrg, FL2FXCONST_DBL(-10.0 / (1 << PVC_ESG_EXP)));
@@ -603,22 +604,22 @@ void pvcDecodeTimeSlot(PVC_STATIC_DATA *pPvcStaticData,
       E_high_exp[ksg] = 0;
 
       /* residual part */
-      accu = ((LONG)(SCHAR)*pTab2++) << (DFRACT_BITS - 8 - PVC_ESG_EXP +
+      accu = ((LONG)(SCHAR)*pTab2++) << (DFRACT_BITS - 8 - PVC_ESG_EXP - 2 +
                                          pPvcDynamicData->pScalingCoef[3]);
 
       /* linear combination of lower grouped energies part */
       for (kb = 0; kb < PVC_NBLOW; kb++) {
         predCoeff = (FIXP_SGL)(
             (SHORT)(SCHAR)pTab1[kb * pPvcDynamicData->nbHigh + ksg] << 8);
-        predCoeff_exp = pPvcDynamicData->pScalingCoef[kb] +
-                        1; /* +1 to compensate for Div2 */
-        accu += fMultDiv2(E[kb], predCoeff) << predCoeff_exp;
+        predCoeff_exp = -(pPvcDynamicData->pScalingCoef[kb] + 1 -
+                          2); /* +1 to compensate for Div2; -2 for accu */
+        accu += fMultDiv2(E[kb], predCoeff) >> predCoeff_exp;
       }
       /* convert back to linear domain */
       accu = fMult(accu, FL2FXCONST_SGL(LOG10FAC_INV));
-      accu = f2Pow(
-          accu, PVC_ESG_EXP - 1,
-          &predCoeff_exp); /* -1 compensates for exponent of LOG10FAC_INV */
+      accu = f2Pow(accu, PVC_ESG_EXP - 1 + 2,
+                   &predCoeff_exp); /* -1 compensates for exponent of
+                                       LOG10FAC_INV; +2 for accu */
       predictedEsgSlot[ksg] = accu;
       E_high_exp[ksg] = predCoeff_exp;
       if (predCoeff_exp > E_high_exp_max) {
@@ -628,8 +629,8 @@ void pvcDecodeTimeSlot(PVC_STATIC_DATA *pPvcStaticData,
 
     /* rescale output vector according to largest exponent */
     for (ksg = 0; ksg < pPvcDynamicData->nbHigh; ksg++) {
-      int scale = E_high_exp[ksg] - E_high_exp_max;
-      predictedEsgSlot[ksg] = scaleValue(predictedEsgSlot[ksg], scale);
+      int scale = fMin(E_high_exp_max - E_high_exp[ksg], DFRACT_BITS - 1);
+      predictedEsgSlot[ksg] = predictedEsgSlot[ksg] >> scale;
     }
     *predictedEsg_exp = E_high_exp_max;
   }
diff --git a/fdk-aac/libSBRdec/src/sbr_crc.cpp b/fdk-aac/libSBRdec/src/sbr_crc.cpp
deleted file mode 100644
index ba0fd05..0000000
--- a/fdk-aac/libSBRdec/src/sbr_crc.cpp
+++ /dev/null
@@ -1,192 +0,0 @@
-/* -----------------------------------------------------------------------------
-Software License for The Fraunhofer FDK AAC Codec Library for Android
-
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
-Forschung e.V. All rights reserved.
-
- 1.    INTRODUCTION
-The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software
-that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding
-scheme for digital audio. This FDK AAC Codec software is intended to be used on
-a wide variety of Android devices.
-
-AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient
-general perceptual audio codecs. AAC-ELD is considered the best-performing
-full-bandwidth communications codec by independent studies and is widely
-deployed. AAC has been standardized by ISO and IEC as part of the MPEG
-specifications.
-
-Patent licenses for necessary patent claims for the FDK AAC Codec (including
-those of Fraunhofer) may be obtained through Via Licensing
-(www.vialicensing.com) or through the respective patent owners individually for
-the purpose of encoding or decoding bit streams in products that are compliant
-with the ISO/IEC MPEG audio standards. Please note that most manufacturers of
-Android devices already license these patent claims through Via Licensing or
-directly from the patent owners, and therefore FDK AAC Codec software may
-already be covered under those patent licenses when it is used for those
-licensed purposes only.
-
-Commercially-licensed AAC software libraries, including floating-point versions
-with enhanced sound quality, are also available from Fraunhofer. Users are
-encouraged to check the Fraunhofer website for additional applications
-information and documentation.
-
-2.    COPYRIGHT LICENSE
-
-Redistribution and use in source and binary forms, with or without modification,
-are permitted without payment of copyright license fees provided that you
-satisfy the following conditions:
-
-You must retain the complete text of this software license in redistributions of
-the FDK AAC Codec or your modifications thereto in source code form.
-
-You must retain the complete text of this software license in the documentation
-and/or other materials provided with redistributions of the FDK AAC Codec or
-your modifications thereto in binary form. You must make available free of
-charge copies of the complete source code of the FDK AAC Codec and your
-modifications thereto to recipients of copies in binary form.
-
-The name of Fraunhofer may not be used to endorse or promote products derived
-from this library without prior written permission.
-
-You may not charge copyright license fees for anyone to use, copy or distribute
-the FDK AAC Codec software or your modifications thereto.
-
-Your modified versions of the FDK AAC Codec must carry prominent notices stating
-that you changed the software and the date of any change. For modified versions
-of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android"
-must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK
-AAC Codec Library for Android."
-
-3.    NO PATENT LICENSE
-
-NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without
-limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE.
-Fraunhofer provides no warranty of patent non-infringement with respect to this
-software.
-
-You may use this FDK AAC Codec software or modifications thereto only for
-purposes that are authorized by appropriate patent licenses.
-
-4.    DISCLAIMER
-
-This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright
-holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES,
-including but not limited to the implied warranties of merchantability and
-fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
-CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary,
-or consequential damages, including but not limited to procurement of substitute
-goods or services; loss of use, data, or profits, or business interruption,
-however caused and on any theory of liability, whether in contract, strict
-liability, or tort (including negligence), arising in any way out of the use of
-this software, even if advised of the possibility of such damage.
-
-5.    CONTACT INFORMATION
-
-Fraunhofer Institute for Integrated Circuits IIS
-Attention: Audio and Multimedia Departments - FDK AAC LL
-Am Wolfsmantel 33
-91058 Erlangen, Germany
-
-www.iis.fraunhofer.de/amm
-amm-info@iis.fraunhofer.de
------------------------------------------------------------------------------ */
-
-/**************************** SBR decoder library ******************************
-
-   Author(s):
-
-   Description:
-
-*******************************************************************************/
-
-/*!
-  \file
-  \brief  CRC check coutines
-*/
-
-#include "sbr_crc.h"
-
-#include "FDK_bitstream.h"
-#include "transcendent.h"
-
-#define MAXCRCSTEP 16
-#define MAXCRCSTEP_LD 4
-
-/*!
-  \brief     crc calculation
-*/
-static ULONG calcCRC(HANDLE_CRC hCrcBuf, ULONG bValue, int nBits) {
-  int i;
-  ULONG bMask = (1UL << (nBits - 1));
-
-  for (i = 0; i < nBits; i++, bMask >>= 1) {
-    USHORT flag = (hCrcBuf->crcState & hCrcBuf->crcMask) ? 1 : 0;
-    USHORT flag1 = (bMask & bValue) ? 1 : 0;
-
-    flag ^= flag1;
-    hCrcBuf->crcState <<= 1;
-    if (flag) hCrcBuf->crcState ^= hCrcBuf->crcPoly;
-  }
-
-  return (hCrcBuf->crcState);
-}
-
-/*!
-  \brief     crc
-*/
-static int getCrc(HANDLE_FDK_BITSTREAM hBs, ULONG NrBits) {
-  int i;
-  CRC_BUFFER CrcBuf;
-
-  CrcBuf.crcState = SBR_CRC_START;
-  CrcBuf.crcPoly = SBR_CRC_POLY;
-  CrcBuf.crcMask = SBR_CRC_MASK;
-
-  int CrcStep = NrBits >> MAXCRCSTEP_LD;
-
-  int CrcNrBitsRest = (NrBits - CrcStep * MAXCRCSTEP);
-  ULONG bValue;
-
-  for (i = 0; i < CrcStep; i++) {
-    bValue = FDKreadBits(hBs, MAXCRCSTEP);
-    calcCRC(&CrcBuf, bValue, MAXCRCSTEP);
-  }
-
-  bValue = FDKreadBits(hBs, CrcNrBitsRest);
-  calcCRC(&CrcBuf, bValue, CrcNrBitsRest);
-
-  return (CrcBuf.crcState & SBR_CRC_RANGE);
-}
-
-/*!
-  \brief   crc interface
-  \return  1: CRC OK, 0: CRC check failure
-*/
-int SbrCrcCheck(HANDLE_FDK_BITSTREAM hBs, /*!< handle to bit-buffer  */
-                LONG NrBits)              /*!< max. CRC length       */
-{
-  int crcResult = 1;
-  ULONG NrCrcBits;
-  ULONG crcCheckResult;
-  LONG NrBitsAvailable;
-  ULONG crcCheckSum;
-
-  crcCheckSum = FDKreadBits(hBs, 10);
-
-  NrBitsAvailable = FDKgetValidBits(hBs);
-  if (NrBitsAvailable <= 0) {
-    return 0;
-  }
-
-  NrCrcBits = fixMin((INT)NrBits, (INT)NrBitsAvailable);
-
-  crcCheckResult = getCrc(hBs, NrCrcBits);
-  FDKpushBack(hBs, (NrBitsAvailable - FDKgetValidBits(hBs)));
-
-  if (crcCheckResult != crcCheckSum) {
-    crcResult = 0;
-  }
-
-  return (crcResult);
-}
diff --git a/fdk-aac/libSBRdec/src/sbr_crc.h b/fdk-aac/libSBRdec/src/sbr_crc.h
deleted file mode 100644
index 9633717..0000000
--- a/fdk-aac/libSBRdec/src/sbr_crc.h
+++ /dev/null
@@ -1,138 +0,0 @@
-/* -----------------------------------------------------------------------------
-Software License for The Fraunhofer FDK AAC Codec Library for Android
-
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
-Forschung e.V. All rights reserved.
-
- 1.    INTRODUCTION
-The Fraunhofer FDK AAC Codec Library for Android ("FDK AAC Codec") is software
-that implements the MPEG Advanced Audio Coding ("AAC") encoding and decoding
-scheme for digital audio. This FDK AAC Codec software is intended to be used on
-a wide variety of Android devices.
-
-AAC's HE-AAC and HE-AAC v2 versions are regarded as today's most efficient
-general perceptual audio codecs. AAC-ELD is considered the best-performing
-full-bandwidth communications codec by independent studies and is widely
-deployed. AAC has been standardized by ISO and IEC as part of the MPEG
-specifications.
-
-Patent licenses for necessary patent claims for the FDK AAC Codec (including
-those of Fraunhofer) may be obtained through Via Licensing
-(www.vialicensing.com) or through the respective patent owners individually for
-the purpose of encoding or decoding bit streams in products that are compliant
-with the ISO/IEC MPEG audio standards. Please note that most manufacturers of
-Android devices already license these patent claims through Via Licensing or
-directly from the patent owners, and therefore FDK AAC Codec software may
-already be covered under those patent licenses when it is used for those
-licensed purposes only.
-
-Commercially-licensed AAC software libraries, including floating-point versions
-with enhanced sound quality, are also available from Fraunhofer. Users are
-encouraged to check the Fraunhofer website for additional applications
-information and documentation.
-
-2.    COPYRIGHT LICENSE
-
-Redistribution and use in source and binary forms, with or without modification,
-are permitted without payment of copyright license fees provided that you
-satisfy the following conditions:
-
-You must retain the complete text of this software license in redistributions of
-the FDK AAC Codec or your modifications thereto in source code form.
-
-You must retain the complete text of this software license in the documentation
-and/or other materials provided with redistributions of the FDK AAC Codec or
-your modifications thereto in binary form. You must make available free of
-charge copies of the complete source code of the FDK AAC Codec and your
-modifications thereto to recipients of copies in binary form.
-
-The name of Fraunhofer may not be used to endorse or promote products derived
-from this library without prior written permission.
-
-You may not charge copyright license fees for anyone to use, copy or distribute
-the FDK AAC Codec software or your modifications thereto.
-
-Your modified versions of the FDK AAC Codec must carry prominent notices stating
-that you changed the software and the date of any change. For modified versions
-of the FDK AAC Codec, the term "Fraunhofer FDK AAC Codec Library for Android"
-must be replaced by the term "Third-Party Modified Version of the Fraunhofer FDK
-AAC Codec Library for Android."
-
-3.    NO PATENT LICENSE
-
-NO EXPRESS OR IMPLIED LICENSES TO ANY PATENT CLAIMS, including without
-limitation the patents of Fraunhofer, ARE GRANTED BY THIS SOFTWARE LICENSE.
-Fraunhofer provides no warranty of patent non-infringement with respect to this
-software.
-
-You may use this FDK AAC Codec software or modifications thereto only for
-purposes that are authorized by appropriate patent licenses.
-
-4.    DISCLAIMER
-
-This FDK AAC Codec software is provided by Fraunhofer on behalf of the copyright
-holders and contributors "AS IS" and WITHOUT ANY EXPRESS OR IMPLIED WARRANTIES,
-including but not limited to the implied warranties of merchantability and
-fitness for a particular purpose. IN NO EVENT SHALL THE COPYRIGHT HOLDER OR
-CONTRIBUTORS BE LIABLE for any direct, indirect, incidental, special, exemplary,
-or consequential damages, including but not limited to procurement of substitute
-goods or services; loss of use, data, or profits, or business interruption,
-however caused and on any theory of liability, whether in contract, strict
-liability, or tort (including negligence), arising in any way out of the use of
-this software, even if advised of the possibility of such damage.
-
-5.    CONTACT INFORMATION
-
-Fraunhofer Institute for Integrated Circuits IIS
-Attention: Audio and Multimedia Departments - FDK AAC LL
-Am Wolfsmantel 33
-91058 Erlangen, Germany
-
-www.iis.fraunhofer.de/amm
-amm-info@iis.fraunhofer.de
------------------------------------------------------------------------------ */
-
-/**************************** SBR decoder library ******************************
-
-   Author(s):
-
-   Description:
-
-*******************************************************************************/
-
-/*!
-  \file
-  \brief  CRC checking routines
-*/
-#ifndef SBR_CRC_H
-#define SBR_CRC_H
-
-#include "sbrdecoder.h"
-
-#include "FDK_bitstream.h"
-
-/* some useful crc polynoms:
-
-crc5: x^5+x^4+x^2+x^1+1
-crc6: x^6+x^5+x^3+x^2+x+1
-crc7: x^7+x^6+x^2+1
-crc8: x^8+x^2+x+x+1
-*/
-
-/* default SBR CRC */ /* G(x) = x^10 + x^9 + x^5 + x^4 + x + 1 */
-#define SBR_CRC_POLY 0x0233
-#define SBR_CRC_MASK 0x0200
-#define SBR_CRC_START 0x0000
-#define SBR_CRC_RANGE 0x03FF
-
-typedef struct {
-  USHORT crcState;
-  USHORT crcMask;
-  USHORT crcPoly;
-} CRC_BUFFER;
-
-typedef CRC_BUFFER *HANDLE_CRC;
-
-int SbrCrcCheck(HANDLE_FDK_BITSTREAM hBitBuf, LONG NrCrcBits);
-
-#endif
diff --git a/fdk-aac/libSBRdec/src/sbr_dec.cpp b/fdk-aac/libSBRdec/src/sbr_dec.cpp
index 30611e7..b1fb0da 100644
--- a/fdk-aac/libSBRdec/src/sbr_dec.cpp
+++ b/fdk-aac/libSBRdec/src/sbr_dec.cpp
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -259,17 +259,18 @@ static void copyHarmonicSpectrum(int *xOverQmf, FIXP_DBL **qmfReal,
 
 void sbr_dec(
     HANDLE_SBR_DEC hSbrDec,             /*!< handle to Decoder channel */
-    INT_PCM *timeIn,                    /*!< pointer to input time signal */
-    INT_PCM *timeOut,                   /*!< pointer to output time signal */
+    LONG *timeIn,                       /*!< pointer to input time signal */
+    LONG *timeOut,                      /*!< pointer to output time signal */
     HANDLE_SBR_DEC hSbrDecRight,        /*!< handle to Decoder channel right */
-    INT_PCM *timeOutRight,              /*!< pointer to output time signal */
+    LONG *timeOutRight,                 /*!< pointer to output time signal */
     const int strideOut,                /*!< Time data traversal strideOut */
     HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */
     HANDLE_SBR_FRAME_DATA hFrameData,   /*!< Control data of current frame */
     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 int codecFrameSize) {
+    HANDLE_PS_DEC h_ps_d, const UINT flags, const int codecFrameSize,
+    const INT sbrInDataHeadroom) {
   int i, slot, reserve;
   int saveLbScale;
   int lastSlotOffs;
@@ -278,7 +279,7 @@ void sbr_dec(
   /* temporary pointer / variable for QMF;
      required as we want to use temporary buffer
      creating one frame delay for HBE in LP mode */
-  INT_PCM *pTimeInQmf = timeIn;
+  LONG *pTimeInQmf = timeIn;
 
   /* Number of QMF timeslots in the overlap buffer: */
   int ov_len = hSbrDec->LppTrans.pSettings->overlap;
@@ -341,8 +342,8 @@ void sbr_dec(
   } else {
     C_AALLOC_SCRATCH_START(qmfTemp, FIXP_DBL, 2 * (64));
     qmfAnalysisFiltering(&hSbrDec->qmfDomainInCh->fb, pReal, pImag,
-                         &hSbrDec->qmfDomainInCh->scaling, pTimeInQmf, 0, 1,
-                         qmfTemp);
+                         &hSbrDec->qmfDomainInCh->scaling, pTimeInQmf,
+                         0 + sbrInDataHeadroom, 1, qmfTemp);
 
     C_AALLOC_SCRATCH_END(qmfTemp, FIXP_DBL, 2 * (64));
   }
@@ -658,7 +659,7 @@ void sbr_dec(
 
   if (!(flags & SBRDEC_PS_DECODED)) {
     if (!(flags & SBRDEC_SKIP_QMF_SYN)) {
-      int outScalefactor = 0;
+      int outScalefactor = -(8);
 
       if (h_ps_d != NULL) {
         h_ps_d->procFrameBased = 1; /* we here do frame based processing */
@@ -743,6 +744,7 @@ void sbr_dec(
       */
       FDK_ASSERT(hSbrDec->qmfDomainInCh->pGlobalConf->nBandsSynthesis <=
                  QMF_MAX_SYNTHESIS_BANDS);
+      qmfChangeOutScalefactor(synQmfRight, -(8));
       FDKmemcpy(synQmfRight->FilterStates, synQmf->FilterStates,
                 9 * hSbrDec->qmfDomainInCh->pGlobalConf->nBandsSynthesis *
                     sizeof(FIXP_QSS));
@@ -814,7 +816,8 @@ void sbr_dec(
                   : scaleFactorLowBand_no_ov,
               scaleFactorHighBand, synQmf->lsb, synQmf->usb);
 
-          outScalefactorL = outScalefactorR = 1; /* psDiffScale! (MPEG-PS) */
+          outScalefactorL = outScalefactorR =
+              1 + sbrInDataHeadroom; /* psDiffScale! (MPEG-PS) */
         }
 
         sbrDecoder_drcApplySlot(/* right channel */
@@ -831,6 +834,9 @@ void sbr_dec(
         outScalefactorL += maxShift;
 
         if (!(flags & SBRDEC_SKIP_QMF_SYN)) {
+          qmfChangeOutScalefactor(synQmf, -(8));
+          qmfChangeOutScalefactor(synQmfRight, -(8));
+
           qmfSynthesisFilteringSlot(
               synQmfRight, rQmfReal, /* QMF real buffer */
               rQmfImag,              /* QMF imag buffer */
diff --git a/fdk-aac/libSBRdec/src/sbr_dec.h b/fdk-aac/libSBRdec/src/sbr_dec.h
index 156da03..eb9c394 100644
--- a/fdk-aac/libSBRdec/src/sbr_dec.h
+++ b/fdk-aac/libSBRdec/src/sbr_dec.h
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -176,17 +176,18 @@ typedef SBR_CHANNEL *HANDLE_SBR_CHANNEL;
 
 void sbr_dec(
     HANDLE_SBR_DEC hSbrDec,             /*!< handle to Decoder channel */
-    INT_PCM *timeIn,                    /*!< pointer to input time signal */
-    INT_PCM *timeOut,                   /*!< pointer to output time signal */
+    LONG *timeIn,                       /*!< pointer to input time signal */
+    LONG *timeOut,                      /*!< pointer to output time signal */
     HANDLE_SBR_DEC hSbrDecRight,        /*!< handle to Decoder channel right */
-    INT_PCM *timeOutRight,              /*!< pointer to output time signal */
+    LONG *timeOutRight,                 /*!< pointer to output time signal */
     INT strideOut,                      /*!< Time data traversal strideOut */
     HANDLE_SBR_HEADER_DATA hHeaderData, /*!< Static control data */
     HANDLE_SBR_FRAME_DATA hFrameData,   /*!< Control data of current frame */
     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 int codecFrameSize);
+    HANDLE_PS_DEC h_ps_d, const UINT flags, const int codecFrameSize,
+    const INT sbrInDataHeadroom);
 
 SBR_ERROR
 createSbrDec(SBR_CHANNEL *hSbrChannel, HANDLE_SBR_HEADER_DATA hHeaderData,
diff --git a/fdk-aac/libSBRdec/src/sbr_ram.cpp b/fdk-aac/libSBRdec/src/sbr_ram.cpp
index 8b35fd2..a759d71 100644
--- a/fdk-aac/libSBRdec/src/sbr_ram.cpp
+++ b/fdk-aac/libSBRdec/src/sbr_ram.cpp
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -109,9 +109,6 @@ amm-info@iis.fraunhofer.de
 
 #include "sbr_ram.h"
 
-#define WORKBUFFER1_TAG 2
-#define WORKBUFFER2_TAG 3
-
 /*!
   \name StaticSbrData
 
diff --git a/fdk-aac/libSBRdec/src/sbr_ram.h b/fdk-aac/libSBRdec/src/sbr_ram.h
index e00f8b5..452f835 100644
--- a/fdk-aac/libSBRdec/src/sbr_ram.h
+++ b/fdk-aac/libSBRdec/src/sbr_ram.h
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -170,6 +170,9 @@ struct SBR_DECODER_INSTANCE {
                        flushed consecutively. */
 
   UINT flags;
+
+  INT sbrInDataHeadroom; /* Headroom of the SBR input time signal to prevent
+                            clipping */
 };
 
 H_ALLOC_MEM(Ram_SbrDecElement, SBR_DECODER_ELEMENT)
diff --git a/fdk-aac/libSBRdec/src/sbrdec_freq_sca.cpp b/fdk-aac/libSBRdec/src/sbrdec_freq_sca.cpp
index 165f94b..e187656 100644
--- a/fdk-aac/libSBRdec/src/sbrdec_freq_sca.cpp
+++ b/fdk-aac/libSBRdec/src/sbrdec_freq_sca.cpp
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2018 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -230,6 +230,8 @@ static UCHAR getStopBand(
       }
     }
 
+    stopMin = fMin(stopMin, 64);
+
     /*
       Choose a stop band between k1 and 64 depending on stopFreq (0..13),
       based on a logarithmic scale.
@@ -523,7 +525,8 @@ static FIXP_SGL calcFactorPerBand(int k_start, int k_stop, int num_bands) {
       step = FL2FXCONST_DBL(0.0f);
     }
   }
-  return FX_DBL2FX_SGL(bandfactor << 1);
+  return (bandfactor >= FL2FXCONST_DBL(0.5)) ? (FIXP_SGL)MAXVAL_SGL
+                                             : FX_DBL2FX_SGL(bandfactor << 1);
 }
 
 /*!
diff --git a/fdk-aac/libSBRdec/src/sbrdecoder.cpp b/fdk-aac/libSBRdec/src/sbrdecoder.cpp
index c827ced..b101a4a 100644
--- a/fdk-aac/libSBRdec/src/sbrdecoder.cpp
+++ b/fdk-aac/libSBRdec/src/sbrdecoder.cpp
@@ -1,7 +1,7 @@
 /* -----------------------------------------------------------------------------
 Software License for The Fraunhofer FDK AAC Codec Library for Android
 
-© Copyright  1995 - 2019 Fraunhofer-Gesellschaft zur Förderung der angewandten
+© Copyright  1995 - 2020 Fraunhofer-Gesellschaft zur Förderung der angewandten
 Forschung e.V. All rights reserved.
 
  1.    INTRODUCTION
@@ -143,24 +143,22 @@ amm-info@iis.fraunhofer.de
 #include "env_extr.h"
 #include "sbr_dec.h"
 #include "env_dec.h"
-#include "sbr_crc.h"
+#include "FDK_crc.h"
 #include "sbr_ram.h"
 #include "sbr_rom.h"
 #include "lpp_tran.h"
 #include "transcendent.h"
 
-#include "FDK_crc.h"
-
 #include "sbrdec_drc.h"
 
 #include "psbitdec.h"
 
 /* Decoder library info */
 #define SBRDECODER_LIB_VL0 3
-#define SBRDECODER_LIB_VL1 0
+#define SBRDECODER_LIB_VL1 1
 #define SBRDECODER_LIB_VL2 0
 #define SBRDECODER_LIB_TITLE "SBR Decoder"
-#ifdef __ANDROID__
+#ifdef SUPPRESS_BUILD_DATE_INFO
 #define SBRDECODER_LIB_BUILD_DATE ""
 #define SBRDECODER_LIB_BUILD_TIME ""
 #else
@@ -619,10 +617,6 @@ SBR_ERROR sbrDecoder_InitElement(
       self->numSbrChannels -= self->pSbrElement[elementIndex]->nChannels;
     }
 
-    /* Save element ID for sanity checks and to have a fallback for concealment.
-     */
-    self->pSbrElement[elementIndex]->elementID = elementID;
-
     /* Determine amount of channels for this element */
     switch (elementID) {
       case ID_NONE:
@@ -655,12 +649,16 @@ SBR_ERROR sbrDecoder_InitElement(
     }
 
     /* Sanity check to avoid memory leaks */
-    if (elChannels < self->pSbrElement[elementIndex]->nChannels) {
+    if (elChannels < self->pSbrElement[elementIndex]->nChannels ||
+        (self->numSbrChannels + elChannels) > (8) + (1)) {
       self->numSbrChannels += self->pSbrElement[elementIndex]->nChannels;
       sbrError = SBRDEC_PARSE_ERROR;
       goto bail;
     }
 
+    /* Save element ID for sanity checks and to have a fallback for concealment.
+     */
+    self->pSbrElement[elementIndex]->elementID = elementID;
     self->pSbrElement[elementIndex]->nChannels = elChannels;
 
     for (ch = 0; ch < elChannels; ch++) {
@@ -1134,18 +1132,22 @@ SBR_ERROR sbrDecoder_Parse(HANDLE_SBRDECODER self, HANDLE_FDK_BITSTREAM hBs,
   SBR_HEADER_STATUS headerStatus = HEADER_NOT_PRESENT;
 
   INT startPos = FDKgetValidBits(hBs);
-  INT CRCLen = 0;
+  FDK_CRCINFO crcInfo;
+  INT crcReg = 0;
+  USHORT sbrCrc = 0;
+  UINT crcPoly;
+  UINT crcStartValue = 0;
+  UINT crcLen;
+
   HANDLE_FDK_BITSTREAM hBsOriginal = hBs;
   FDK_BITSTREAM bsBwd;
 
-  FDK_CRCINFO crcInfo;
-  INT crcReg = 0;
-  USHORT drmSbrCrc = 0;
   const int fGlobalIndependencyFlag = acFlags & AC_INDEP;
   const int bs_pvc = acElFlags[elementIndex] & AC_EL_USAC_PVC;
   const int bs_interTes = acElFlags[elementIndex] & AC_EL_USAC_ITES;
   int stereo;
   int fDoDecodeSbrData = 1;
+  int alignBits = 0;
 
   int lastSlot, lastHdrSlot = 0, thisHdrSlot = 0;
 
@@ -1277,27 +1279,23 @@ SBR_ERROR sbrDecoder_Parse(HANDLE_SBRDECODER self, HANDLE_FDK_BITSTREAM hBs,
   if (fDoDecodeSbrData) {
     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:
         case AOT_DRM_SURROUND:
-          drmSbrCrc = (USHORT)FDKreadBits(hBs, 8);
-          /* Setup CRC decoder */
-          FDKcrcInit(&crcInfo, 0x001d, 0xFFFF, 8);
-          /* Start CRC region */
-          crcReg = FDKcrcStartReg(&crcInfo, hBs, 0);
+          crcPoly = 0x001d;
+          crcLen = 8;
+          crcStartValue = 0x000000ff;
           break;
         default:
-          CRCLen = bsPayLen - 10; /* change: 0 => i */
-          if (CRCLen < 0) {
-            fDoDecodeSbrData = 0;
-          } else {
-            fDoDecodeSbrData = SbrCrcCheck(hBs, CRCLen);
-          }
+          crcPoly = 0x0633;
+          crcLen = 10;
+          crcStartValue = 0x00000000;
           break;
       }
+      sbrCrc = (USHORT)FDKreadBits(hBs, crcLen);
+      /* Setup CRC decoder */
+      FDKcrcInit(&crcInfo, crcPoly, crcStartValue, crcLen);
+      /* Start CRC region */
+      crcReg = FDKcrcStartReg(&crcInfo, hBs, 0);
     }
   } /* if (fDoDecodeSbrData) */
 
@@ -1450,35 +1448,6 @@ SBR_ERROR sbrDecoder_Parse(HANDLE_SBRDECODER self, HANDLE_FDK_BITSTREAM hBs,
         valBits = (INT)FDKgetValidBits(hBs);
       }
 
-      if (crcFlag) {
-        switch (self->coreCodec) {
-          case AOT_ER_AAC_ELD: {
-            /* late crc check for eld */
-            INT payloadbits =
-                (INT)startPos - (INT)FDKgetValidBits(hBs) - startPos;
-            INT crcLen = payloadbits - 10;
-            FDKpushBack(hBs, payloadbits);
-            fDoDecodeSbrData = SbrCrcCheck(hBs, crcLen);
-            FDKpushFor(hBs, crcLen);
-          } break;
-          case AOT_DRM_AAC:
-          case AOT_DRM_SURROUND:
-            /* End CRC region */
-            FDKcrcEndReg(&crcInfo, hBs, crcReg);
-            /* Check CRC */
-            if ((FDKcrcGetCRC(&crcInfo) ^ 0xFF) != drmSbrCrc) {
-              fDoDecodeSbrData = 0;
-              if (headerStatus != HEADER_NOT_PRESENT) {
-                headerStatus = HEADER_ERROR;
-                hSbrHeader->syncState = SBR_NOT_INITIALIZED;
-              }
-            }
-            break;
-          default:
-            break;
-        }
-      }
-
       /* sanity check of remaining bits */
       if (valBits < 0) {
         fDoDecodeSbrData = 0;
@@ -1489,7 +1458,7 @@ SBR_ERROR sbrDecoder_Parse(HANDLE_SBRDECODER self, HANDLE_FDK_BITSTREAM hBs,
           case AOT_AAC_LC: {
             /* This sanity check is only meaningful with General Audio
              * bitstreams */
-            int alignBits = valBits & 0x7;
+            alignBits = valBits & 0x7;
 
             if (valBits > alignBits) {
               fDoDecodeSbrData = 0;
@@ -1508,6 +1477,20 @@ SBR_ERROR sbrDecoder_Parse(HANDLE_SBRDECODER self, HANDLE_FDK_BITSTREAM hBs,
     errorStatus = SBRDEC_PARSE_ERROR;
   }
 
+  if (crcFlag && (hSbrHeader->syncState >= SBR_HEADER) && fDoDecodeSbrData) {
+    FDKpushFor(hBs, alignBits);
+    FDKcrcEndReg(&crcInfo, hBs, crcReg); /* End CRC region */
+    FDKpushBack(hBs, alignBits);
+    /* Check CRC */
+    if ((FDKcrcGetCRC(&crcInfo) ^ crcStartValue) != sbrCrc) {
+      fDoDecodeSbrData = 0;
+      if (headerStatus != HEADER_NOT_PRESENT) {
+        headerStatus = HEADER_ERROR;
+        hSbrHeader->syncState = SBR_NOT_INITIALIZED;
+      }
+    }
+  }
+
   if (!fDoDecodeSbrData) {
     /* Set error flag for this slot to trigger concealment */
     setFrameErrorFlag(self->pSbrElement[elementIndex], FRAME_ERROR);
@@ -1587,10 +1570,10 @@ bail:
  * \return SBRDEC_OK if successfull, else error code
  */
 static SBR_ERROR sbrDecoder_DecodeElement(
-    HANDLE_SBRDECODER self, QDOM_PCM *input, INT_PCM *timeData,
-    const int timeDataSize, const FDK_channelMapDescr *const mapDescr,
-    const int mapIdx, int channelIndex, const int elementIndex,
-    const int numInChannels, int *numOutChannels, const int psPossible) {
+    HANDLE_SBRDECODER self, LONG *input, LONG *timeData, const int timeDataSize,
+    const FDK_channelMapDescr *const mapDescr, const int mapIdx,
+    int channelIndex, const int elementIndex, const int numInChannels,
+    int *numOutChannels, const int psPossible) {
   SBR_DECODER_ELEMENT *hSbrElement = self->pSbrElement[elementIndex];
   HANDLE_SBR_CHANNEL *pSbrChannel =
       self->pSbrElement[elementIndex]->pSbrChannel;
@@ -1760,7 +1743,7 @@ static SBR_ERROR sbrDecoder_DecodeElement(
             timeData + offset1, strideOut, hSbrHeader, hFrameDataLeft,
             &pSbrChannel[0]->prevFrameData,
             (hSbrHeader->syncState == SBR_ACTIVE), h_ps_d, self->flags,
-            codecFrameSize);
+            codecFrameSize, self->sbrInDataHeadroom);
 
     if (stereo) {
       /* Process right channel */
@@ -1768,7 +1751,7 @@ static SBR_ERROR sbrDecoder_DecodeElement(
               timeData + offset1, NULL, NULL, strideOut, hSbrHeader,
               hFrameDataRight, &pSbrChannel[1]->prevFrameData,
               (hSbrHeader->syncState == SBR_ACTIVE), NULL, self->flags,
-              codecFrameSize);
+              codecFrameSize, self->sbrInDataHeadroom);
     }
 
     C_ALLOC_SCRATCH_END(pPsScratch, struct PS_DEC_COEFFICIENTS, 1)
@@ -1788,14 +1771,14 @@ static SBR_ERROR sbrDecoder_DecodeElement(
       int copyFrameSize =
           codecFrameSize * self->pQmfDomain->QmfDomainOut->fb.no_channels;
       copyFrameSize /= self->pQmfDomain->QmfDomainIn->fb.no_channels;
-      INT_PCM *ptr;
+      LONG *ptr;
       INT i;
       FDK_ASSERT(strideOut == 2);
 
       ptr = timeData;
       for (i = copyFrameSize >> 1; i--;) {
-        INT_PCM tmp; /* This temporal variable is required because some
-                        compilers can't do *ptr++ = *ptr++ correctly. */
+        LONG tmp; /* This temporal variable is required because some compilers
+                     can't do *ptr++ = *ptr++ correctly. */
         tmp = *ptr++;
         *ptr++ = tmp;
         tmp = *ptr++;
@@ -1808,12 +1791,13 @@ static SBR_ERROR sbrDecoder_DecodeElement(
   return errorStatus;
 }
 
-SBR_ERROR sbrDecoder_Apply(HANDLE_SBRDECODER self, INT_PCM *input,
-                           INT_PCM *timeData, const int timeDataSize,
-                           int *numChannels, int *sampleRate,
+SBR_ERROR sbrDecoder_Apply(HANDLE_SBRDECODER self, LONG *input, LONG *timeData,
+                           const int timeDataSize, int *numChannels,
+                           int *sampleRate,
                            const FDK_channelMapDescr *const mapDescr,
                            const int mapIdx, const int coreDecodedOk,
-                           UCHAR *psDecoded) {
+                           UCHAR *psDecoded, const INT inDataHeadroom,
+                           INT *outDataHeadroom) {
   SBR_ERROR errorStatus = SBRDEC_OK;
 
   int psPossible;
@@ -1850,6 +1834,9 @@ SBR_ERROR sbrDecoder_Apply(HANDLE_SBRDECODER self, INT_PCM *input,
     psPossible = 0;
   }
 
+  self->sbrInDataHeadroom = inDataHeadroom;
+  *outDataHeadroom = (INT)(8);
+
   /* Make sure that even if no SBR data was found/parsed *psDecoded is returned
    * 1 if psPossible was 0. */
   if (psPossible == 0) {