mirror of https://git.ffmpeg.org/ffmpeg.git
avcodec/aacdec_template: Fix undefined integer overflow in apply_tns()
Fixes: runtime error: signed integer overflow: -2147483648 - 1202286525 cannot be represented in type 'int'
Fixes: 2071/clusterfuzz-testcase-minimized-6036414271586304
Found-by: continuous fuzzing process https://github.com/google/oss-fuzz/tree/master/projects/ffmpeg
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
(cherry picked from commit 0ef8f03133
)
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
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@ -35,6 +35,7 @@
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#define AAC_RENAME(x) x ## _fixed
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#define AAC_RENAME_32(x) x ## _fixed_32
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typedef int INTFLOAT;
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typedef unsigned UINTFLOAT; ///< Equivalent to INTFLOAT, Used as temporal cast to avoid undefined sign overflow operations.
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typedef int64_t INT64FLOAT;
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typedef int16_t SHORTFLOAT;
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typedef SoftFloat AAC_FLOAT;
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@ -83,6 +84,7 @@ typedef int AAC_SIGNE;
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#define AAC_RENAME(x) x
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#define AAC_RENAME_32(x) x
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typedef float INTFLOAT;
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typedef float UINTFLOAT;
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typedef float INT64FLOAT;
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typedef float SHORTFLOAT;
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typedef float AAC_FLOAT;
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@ -2389,7 +2389,7 @@ static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt,
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* @param decode 1 if tool is used normally, 0 if tool is used in LTP.
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* @param coef spectral coefficients
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*/
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static void apply_tns(INTFLOAT coef[1024], TemporalNoiseShaping *tns,
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static void apply_tns(INTFLOAT coef_param[1024], TemporalNoiseShaping *tns,
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IndividualChannelStream *ics, int decode)
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{
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const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb);
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@ -2397,6 +2397,7 @@ static void apply_tns(INTFLOAT coef[1024], TemporalNoiseShaping *tns,
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int bottom, top, order, start, end, size, inc;
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INTFLOAT lpc[TNS_MAX_ORDER];
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INTFLOAT tmp[TNS_MAX_ORDER+1];
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UINTFLOAT *coef = coef_param;
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for (w = 0; w < ics->num_windows; w++) {
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bottom = ics->num_swb;
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@ -2426,7 +2427,7 @@ static void apply_tns(INTFLOAT coef[1024], TemporalNoiseShaping *tns,
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// ar filter
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for (m = 0; m < size; m++, start += inc)
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for (i = 1; i <= FFMIN(m, order); i++)
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coef[start] -= AAC_MUL26(coef[start - i * inc], lpc[i - 1]);
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coef[start] -= AAC_MUL26((INTFLOAT)coef[start - i * inc], lpc[i - 1]);
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} else {
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// ma filter
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for (m = 0; m < size; m++, start += inc) {
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