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author | Fraunhofer IIS FDK <audio-fdk@iis.fraunhofer.de> | 2019-10-29 13:06:00 +0100 |
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committer | Jean-Michel Trivi <jmtrivi@google.com> | 2019-12-27 08:59:49 -0800 |
commit | 68cdcfc6b82f88ed1f616b7e3a0133b0072f4bf4 (patch) | |
tree | b90b8f8ae2324242453d7d69ccd95ced9b5d8958 | |
parent | 2ce3e76d0fd843e1cede9c747392b716039db596 (diff) | |
download | fdk-aac-68cdcfc6b82f88ed1f616b7e3a0133b0072f4bf4.tar.gz fdk-aac-68cdcfc6b82f88ed1f616b7e3a0133b0072f4bf4.tar.bz2 fdk-aac-68cdcfc6b82f88ed1f616b7e3a0133b0072f4bf4.zip |
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.cpp | 97 |
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++; |