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authorFraunhofer IIS FDK <audio-fdk@iis.fraunhofer.de>2019-10-29 13:06:00 +0100
committerJean-Michel Trivi <jmtrivi@google.com>2019-12-27 08:59:49 -0800
commit68cdcfc6b82f88ed1f616b7e3a0133b0072f4bf4 (patch)
treeb90b8f8ae2324242453d7d69ccd95ced9b5d8958
parent2ce3e76d0fd843e1cede9c747392b716039db596 (diff)
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Avoid signed integer overflows in multiple adjustTimeSlot*() functions.
Bug: 145669628 Test: atest DecoderTestXheAac ; atest DecoderTestAacDrc Change-Id: Id0f3ee9d3413e534505c0c7eea379ce42880ca79
-rw-r--r--libSBRdec/src/env_calc.cpp97
1 files changed, 72 insertions, 25 deletions
diff --git a/libSBRdec/src/env_calc.cpp b/libSBRdec/src/env_calc.cpp
index 23ad509..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++;