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https://github.com/FFmpeg/FFmpeg.git
synced 2024-12-23 12:43:46 +02:00
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
This commit is contained in:
Fabrice Bellard
parent
31def22984
commit
8155233413
@ -28,7 +28,9 @@
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/* define USE_HIGHPRECISION to have a bit exact (but slower) mpeg
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audio decoder */
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//#define USE_HIGHPRECISION
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#ifdef CONFIG_MPEGAUDIO_HP
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#define USE_HIGHPRECISION
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#endif
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#ifdef USE_HIGHPRECISION
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#define FRAC_BITS 23 /* fractional bits for sb_samples and dct */
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@ -149,9 +151,9 @@ static INT32 scale_factor_mult[15][3];
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{ FIXR(1.0 * (v)), FIXR(0.7937005259 * (v)), FIXR(0.6299605249 * (v)) }
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static INT32 scale_factor_mult2[3][3] = {
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SCALE_GEN(1.0 / 3.0), /* 3 steps */
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SCALE_GEN(1.0 / 5.0), /* 5 steps */
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SCALE_GEN(1.0 / 9.0), /* 9 steps */
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SCALE_GEN(4.0 / 3.0), /* 3 steps */
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SCALE_GEN(4.0 / 5.0), /* 5 steps */
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SCALE_GEN(4.0 / 9.0), /* 9 steps */
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};
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/* 2^(n/4) */
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@ -176,7 +178,8 @@ static inline int l1_unscale(int n, int mant, int scale_factor)
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shift >>= 2;
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val = MUL64(mant + (-1 << n) + 1, scale_factor_mult[n-1][mod]);
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shift += n;
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return (int)((val + (1 << (shift - 1))) >> shift);
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/* NOTE: at this point, 1 <= shift >= 21 + 15 */
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return (int)((val + (1LL << (shift - 1))) >> shift);
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}
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static inline int l2_unscale_group(int steps, int mant, int scale_factor)
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@ -186,9 +189,12 @@ static inline int l2_unscale_group(int steps, int mant, int scale_factor)
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shift = scale_factor_modshift[scale_factor];
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mod = shift & 3;
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shift >>= 2;
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/* XXX: store the result directly */
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val = (2 * (mant - (steps >> 1))) * scale_factor_mult2[steps >> 2][mod];
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return (val + (1 << (shift - 1))) >> shift;
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val = (mant - (steps >> 1)) * scale_factor_mult2[steps >> 2][mod];
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/* NOTE: at this point, 0 <= shift <= 21 */
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if (shift > 0)
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val = (val + (1 << (shift - 1))) >> shift;
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return val;
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}
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/* compute value^(4/3) * 2^(exponent/4). It normalized to FRAC_BITS */
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@ -280,7 +286,7 @@ static int int_pow(int i, int *exp_ptr)
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eq--;
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}
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/* now POW_FRAC_ONE <= a < 2 * POW_FRAC_ONE */
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#if (POW_FRAC_BITS - 1) > FRAC_BITS
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#if POW_FRAC_BITS > FRAC_BITS
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a = (a + (1 << (POW_FRAC_BITS - FRAC_BITS - 1))) >> (POW_FRAC_BITS - FRAC_BITS);
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/* correct overflow */
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if (a >= 2 * (1 << FRAC_BITS)) {
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@ -303,12 +309,8 @@ static int decode_init(AVCodecContext * avctx)
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for(i=0;i<64;i++) {
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int shift, mod;
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/* 1.0 (i = 3) is normalized to 2 ^ FRAC_BITS */
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shift = (i / 3) - 1;
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shift = (i / 3);
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mod = i % 3;
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#if FRAC_BITS <= 15
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if (shift > 31)
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shift = 31;
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#endif
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scale_factor_modshift[i] = mod | (shift << 2);
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}
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@ -317,9 +319,9 @@ static int decode_init(AVCodecContext * avctx)
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int n, norm;
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n = i + 2;
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norm = ((INT64_C(1) << n) * FRAC_ONE) / ((1 << n) - 1);
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scale_factor_mult[i][0] = MULL(FIXR(1.0), norm);
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scale_factor_mult[i][1] = MULL(FIXR(0.7937005259), norm);
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scale_factor_mult[i][2] = MULL(FIXR(0.6299605249), norm);
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scale_factor_mult[i][0] = MULL(FIXR(1.0 * 2.0), norm);
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scale_factor_mult[i][1] = MULL(FIXR(0.7937005259 * 2.0), norm);
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scale_factor_mult[i][2] = MULL(FIXR(0.6299605249 * 2.0), norm);
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dprintf("%d: norm=%x s=%x %x %x\n",
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i, norm,
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scale_factor_mult[i][0],
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@ -809,6 +811,8 @@ static void synth_filter(MPADecodeContext *s1,
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for(j=0;j<32;j++) {
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v = tmp[j];
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#if FRAC_BITS <= 15
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/* NOTE: can cause a loss in precision if very high amplitude
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sound */
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if (v > 32767)
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v = 32767;
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else if (v < -32768)
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@ -1069,11 +1073,10 @@ static int decode_header(MPADecodeContext *s, UINT32 header)
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/* extract frequency */
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sample_rate_index = (header >> 10) & 3;
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sample_rate = mpa_freq_tab[sample_rate_index] >> (s->lsf + mpeg25);
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if (sample_rate == 0)
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return 1;
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sample_rate_index += 3 * (s->lsf + mpeg25);
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s->sample_rate_index = sample_rate_index;
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s->error_protection = ((header >> 16) & 1) ^ 1;
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s->sample_rate = sample_rate;
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bitrate_index = (header >> 12) & 0xf;
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padding = (header >> 9) & 1;
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@ -1131,7 +1134,6 @@ static int decode_header(MPADecodeContext *s, UINT32 header)
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break;
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}
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}
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s->sample_rate = sample_rate;
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#if defined(DEBUG)
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printf("layer%d, %d Hz, %d kbits/s, ",
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@ -1962,10 +1964,19 @@ void sample_dump(int fnum, INT32 *tab, int n)
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{
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static FILE *files[16], *f;
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char buf[512];
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int i;
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INT32 v;
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f = files[fnum];
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if (!f) {
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sprintf(buf, "/tmp/out%d.pcm", fnum);
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sprintf(buf, "/tmp/out%d.%s.pcm",
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fnum,
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#ifdef USE_HIGHPRECISION
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"hp"
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#else
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"lp"
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#endif
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);
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f = fopen(buf, "w");
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if (!f)
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return;
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@ -1973,18 +1984,20 @@ void sample_dump(int fnum, INT32 *tab, int n)
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}
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if (fnum == 0) {
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int i;
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static int pos = 0;
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printf("pos=%d\n", pos);
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for(i=0;i<n;i++) {
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printf(" %f", (double)tab[i] / 32768.0);
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printf(" %0.4f", (double)tab[i] / FRAC_ONE);
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if ((i % 18) == 17)
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printf("\n");
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}
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pos += n;
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}
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fwrite(tab, 1, n * sizeof(INT32), f);
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for(i=0;i<n;i++) {
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/* normalize to 23 frac bits */
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v = tab[i] << (23 - FRAC_BITS);
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fwrite(&v, 1, sizeof(INT32), f);
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}
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}
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#endif
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@ -2273,11 +2286,11 @@ static int mp_decode_layer3(MPADecodeContext *s)
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sample_dump(0, g->sb_hybrid, 576);
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#endif
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compute_antialias(s, g);
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#ifdef DEBUG
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#if defined(DEBUG)
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sample_dump(1, g->sb_hybrid, 576);
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#endif
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compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
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#ifdef DEBUG
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#if defined(DEBUG)
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sample_dump(2, &s->sb_samples[ch][18 * gr][0], 576);
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#endif
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}
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@ -2389,17 +2402,14 @@ static int decode_frame(AVCodecContext * avctx,
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s->free_format_frame_size = 0;
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} else {
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if (decode_header(s, header) == 1) {
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/* free format: compute frame size */
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/* free format: prepare to compute frame size */
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s->frame_size = -1;
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memcpy(s->inbuf, s->inbuf + 1, s->inbuf_ptr - s->inbuf - 1);
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s->inbuf_ptr--;
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} else {
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/* update codec info */
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avctx->sample_rate = s->sample_rate;
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avctx->channels = s->nb_channels;
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avctx->bit_rate = s->bit_rate;
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avctx->frame_size = s->frame_size;
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}
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/* update codec info */
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avctx->sample_rate = s->sample_rate;
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avctx->channels = s->nb_channels;
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avctx->bit_rate = s->bit_rate;
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avctx->frame_size = s->frame_size;
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}
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}
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} else if (s->frame_size == -1) {
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