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Avoid signed integer overflows in multiple adjustTimeSlot*() functions. 

Bug: 145669628
Test: atest DecoderTestXheAac ; atest DecoderTestAacDrc
Change-Id: Id0f3ee9d3413e534505c0c7eea379ce42880ca79
This commit is contained in:
Fraunhofer IIS FDK 2019-10-29 13:06:00 +01:00 committed by Jean-Michel Trivi
parent 2ce3e76d0f
commit 68cdcfc6b8
1 changed files with 72 additions and 25 deletions
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93
libSBRdec/src/env_calc.cpp
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@ -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)) {
@ -2723,7 +2744,9 @@ static void adjustTimeSlotHQ_GainAndNoise(
smoothedNoise = (fMultDiv2(smooth_ratio, filtBufferNoise[k]) >> shift) +
fMult(direct_ratio, noiseLevel[k]);
} else {
smoothedNoise = (fMultDiv2(smooth_ratio, filtBufferNoise[k]) << shift) +
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++) {
@ -2896,7 +2933,9 @@ static void adjustTimeSlotHQ(
smoothedNoise = (fMultDiv2(smooth_ratio, filtBufferNoise[k]) >> shift) +
fMult(direct_ratio, noiseLevel[k]);
} else {
smoothedNoise = (fMultDiv2(smooth_ratio, filtBufferNoise[k]) << shift) +
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++;