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ffmpeg.wasm-core
Commit 81552334 authored by Fabrice Bellard's avatar Fabrice Bellard
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fixed layer1/2 overflow if very loud sound - fixed broken free format decoding... 

fixed layer1/2 overflow if very loud sound - fixed broken free format decoding to pass all mpeg audio standard decoding tests (please avoid patching the parser without having all test streams available - contact me if necessary)

Originally committed as revision 634 to svn://svn.ffmpeg.org/ffmpeg/trunk
parent 31def229
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Showing with 47 additions and 37 deletions
libavcodec/mpegaudiodec.c
View file @ 81552334
......@@ -28,7 +28,9 @@
/* define USE_HIGHPRECISION to have a bit exact (but slower) mpeg
audio decoder */
//#define USE_HIGHPRECISION
#ifdef CONFIG_MPEGAUDIO_HP
#define USE_HIGHPRECISION
#endif
#ifdef USE_HIGHPRECISION
#define FRAC_BITS 23 /* fractional bits for sb_samples and dct */
......@@ -149,9 +151,9 @@ static INT32 scale_factor_mult[15][3];
{ FIXR(1.0 * (v)), FIXR(0.7937005259 * (v)), FIXR(0.6299605249 * (v)) }
static INT32 scale_factor_mult2[3][3] = {
SCALE_GEN(1.0 / 3.0), /* 3 steps */
SCALE_GEN(1.0 / 5.0), /* 5 steps */
SCALE_GEN(1.0 / 9.0), /* 9 steps */
SCALE_GEN(4.0 / 3.0), /* 3 steps */
SCALE_GEN(4.0 / 5.0), /* 5 steps */
SCALE_GEN(4.0 / 9.0), /* 9 steps */
};
/* 2^(n/4) */
......@@ -176,7 +178,8 @@ static inline int l1_unscale(int n, int mant, int scale_factor)
shift >>= 2;
val = MUL64(mant + (-1 << n) + 1, scale_factor_mult[n-1][mod]);
shift += n;
return (int)((val + (1 << (shift - 1))) >> shift);
/* NOTE: at this point, 1 <= shift >= 21 + 15 */
return (int)((val + (1LL << (shift - 1))) >> shift);
}
static inline int l2_unscale_group(int steps, int mant, int scale_factor)
......@@ -186,9 +189,12 @@ static inline int l2_unscale_group(int steps, int mant, int scale_factor)
shift = scale_factor_modshift[scale_factor];
mod = shift & 3;
shift >>= 2;
/* XXX: store the result directly */
val = (2 * (mant - (steps >> 1))) * scale_factor_mult2[steps >> 2][mod];
return (val + (1 << (shift - 1))) >> shift;
val = (mant - (steps >> 1)) * scale_factor_mult2[steps >> 2][mod];
/* NOTE: at this point, 0 <= shift <= 21 */
if (shift > 0)
val = (val + (1 << (shift - 1))) >> shift;
return val;
}
/* compute value^(4/3) * 2^(exponent/4). It normalized to FRAC_BITS */
......@@ -280,7 +286,7 @@ static int int_pow(int i, int *exp_ptr)
eq--;
}
/* now POW_FRAC_ONE <= a < 2 * POW_FRAC_ONE */
#if (POW_FRAC_BITS - 1) > FRAC_BITS
#if POW_FRAC_BITS > FRAC_BITS
a = (a + (1 << (POW_FRAC_BITS - FRAC_BITS - 1))) >> (POW_FRAC_BITS - FRAC_BITS);
/* correct overflow */
if (a >= 2 * (1 << FRAC_BITS)) {
......@@ -303,12 +309,8 @@ static int decode_init(AVCodecContext * avctx)
for(i=0;i<64;i++) {
int shift, mod;
/* 1.0 (i = 3) is normalized to 2 ^ FRAC_BITS */
shift = (i / 3) - 1;
shift = (i / 3);
mod = i % 3;
#if FRAC_BITS <= 15
if (shift > 31)
shift = 31;
#endif
scale_factor_modshift[i] = mod | (shift << 2);
}
......@@ -317,9 +319,9 @@ static int decode_init(AVCodecContext * avctx)
int n, norm;
n = i + 2;
norm = ((INT64_C(1) << n) * FRAC_ONE) / ((1 << n) - 1);
scale_factor_mult[i][0] = MULL(FIXR(1.0), norm);
scale_factor_mult[i][1] = MULL(FIXR(0.7937005259), norm);
scale_factor_mult[i][2] = MULL(FIXR(0.6299605249), norm);
scale_factor_mult[i][0] = MULL(FIXR(1.0 * 2.0), norm);
scale_factor_mult[i][1] = MULL(FIXR(0.7937005259 * 2.0), norm);
scale_factor_mult[i][2] = MULL(FIXR(0.6299605249 * 2.0), norm);
dprintf("%d: norm=%x s=%x %x %x\n",
i, norm,
scale_factor_mult[i][0],
......@@ -809,6 +811,8 @@ static void synth_filter(MPADecodeContext *s1,
for(j=0;j<32;j++) {
v = tmp[j];
#if FRAC_BITS <= 15
/* NOTE: can cause a loss in precision if very high amplitude
sound */
if (v > 32767)
v = 32767;
else if (v < -32768)
......@@ -1069,11 +1073,10 @@ static int decode_header(MPADecodeContext *s, UINT32 header)
/* extract frequency */
sample_rate_index = (header >> 10) & 3;
sample_rate = mpa_freq_tab[sample_rate_index] >> (s->lsf + mpeg25);
if (sample_rate == 0)
return 1;
sample_rate_index += 3 * (s->lsf + mpeg25);
s->sample_rate_index = sample_rate_index;
s->error_protection = ((header >> 16) & 1) ^ 1;
s->sample_rate = sample_rate;
bitrate_index = (header >> 12) & 0xf;
padding = (header >> 9) & 1;
......@@ -1131,7 +1134,6 @@ static int decode_header(MPADecodeContext *s, UINT32 header)
break;
}
}
s->sample_rate = sample_rate;
#if defined(DEBUG)
printf("layer%d, %d Hz, %d kbits/s, ",
......@@ -1962,10 +1964,19 @@ void sample_dump(int fnum, INT32 *tab, int n)
{
static FILE *files[16], *f;
char buf[512];
int i;
INT32 v;
f = files[fnum];
if (!f) {
sprintf(buf, "/tmp/out%d.pcm", fnum);
sprintf(buf, "/tmp/out%d.%s.pcm",
fnum,
#ifdef USE_HIGHPRECISION
"hp"
#else
"lp"
#endif
);
f = fopen(buf, "w");
if (!f)
return;
......@@ -1973,18 +1984,20 @@ void sample_dump(int fnum, INT32 *tab, int n)
}
if (fnum == 0) {
int i;
static int pos = 0;
printf("pos=%d\n", pos);
for(i=0;i<n;i++) {
printf(" %f", (double)tab[i] / 32768.0);
printf(" %0.4f", (double)tab[i] / FRAC_ONE);
if ((i % 18) == 17)
printf("\n");
}
pos += n;
}
fwrite(tab, 1, n * sizeof(INT32), f);
for(i=0;i<n;i++) {
/* normalize to 23 frac bits */
v = tab[i] << (23 - FRAC_BITS);
fwrite(&v, 1, sizeof(INT32), f);
}
}
#endif
......@@ -2273,11 +2286,11 @@ static int mp_decode_layer3(MPADecodeContext *s)
sample_dump(0, g->sb_hybrid, 576);
#endif
compute_antialias(s, g);
#ifdef DEBUG
#if defined(DEBUG)
sample_dump(1, g->sb_hybrid, 576);
#endif
compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
#ifdef DEBUG
#if defined(DEBUG)
sample_dump(2, &s->sb_samples[ch][18 * gr][0], 576);
#endif
}
......@@ -2389,17 +2402,14 @@ static int decode_frame(AVCodecContext * avctx,
s->free_format_frame_size = 0;
} else {
if (decode_header(s, header) == 1) {
/* free format: compute frame size */
/* free format: prepare to compute frame size */
s->frame_size = -1;
memcpy(s->inbuf, s->inbuf + 1, s->inbuf_ptr - s->inbuf - 1);
s->inbuf_ptr--;
} else {
/* update codec info */
avctx->sample_rate = s->sample_rate;
avctx->channels = s->nb_channels;
avctx->bit_rate = s->bit_rate;
avctx->frame_size = s->frame_size;
}
/* update codec info */
avctx->sample_rate = s->sample_rate;
avctx->channels = s->nb_channels;
avctx->bit_rate = s->bit_rate;
avctx->frame_size = s->frame_size;
}
}
} else if (s->frame_size == -1) {
......
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