Snap for 6104206 from 47b23c88cfe81bbcc9ff0c8de1608283de18a475 to rvc-release

Change-Id: I37ae8a78b271948e025a2e6c071453480dcc9a52

3 files changed, 97 insertions, 41 deletions

diff --git a/libSBRdec/src/env_calc.cpp b/libSBRdec/src/env_calc.cpp
index 41c9e88..81f03f3 100644
--- a/libSBRdec/src/env_calc.cpp
+++ b/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];
@@ -986,7 +989,8 @@ 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
@@ -1006,7 +1010,7 @@ 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
@@ -1032,7 +1036,7 @@ 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 */
@@ -2426,6 +2430,9 @@ 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]),
@@ -2438,7 +2445,8 @@ 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 +=
@@ -2472,7 +2480,8 @@ 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 +=
@@ -2512,6 +2521,8 @@ 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))
@@ -2527,7 +2538,8 @@ 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);
 
@@ -2555,10 +2567,10 @@ 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;
@@ -2589,7 +2601,9 @@ 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;
@@ -2602,7 +2616,8 @@ 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++;
@@ -2630,7 +2645,8 @@ 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];
 
@@ -2699,6 +2715,9 @@ 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);
 
@@ -2708,6 +2727,8 @@ 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)) {
@@ -2723,8 +2744,10 @@ 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]);
       }
 
       /*
@@ -2732,8 +2755,12 @@ 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++;
 
@@ -2755,8 +2782,12 @@ 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++;
 
@@ -2862,6 +2893,9 @@ 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;
@@ -2877,10 +2911,13 @@ 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++) {
@@ -2896,8 +2933,10 @@ 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]);
       }
 
       /*
@@ -2905,8 +2944,12 @@ 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++;
 
@@ -2959,8 +3002,12 @@ 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++;
 
@@ -3144,6 +3191,11 @@ 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/libSBRdec/src/lpp_tran.cpp b/libSBRdec/src/lpp_tran.cpp
index 6acb626..93e1158 100644
--- a/libSBRdec/src/lpp_tran.cpp
+++ b/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/libSBRdec/src/pvc_dec.cpp b/libSBRdec/src/pvc_dec.cpp
index b477122..e1e3c2c 100644
--- a/libSBRdec/src/pvc_dec.cpp
+++ b/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;
   }