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ffmpeg.wasm-core
Commit 8c5b5683 authored by Fabrice Bellard's avatar Fabrice Bellard
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fixed audio frame buffering problem (should correct problems on some streams)... 

fixed audio frame buffering problem (should correct problems on some streams) - faster synthesis filter - prototype 'parse_only' support

Originally committed as revision 2173 to svn://svn.ffmpeg.org/ffmpeg/trunk
parent d99ce8d7
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Showing with 183 additions and 82 deletions
libavcodec/mpegaudiodec.c
View file @ 8c5b5683
......@@ -310,7 +310,7 @@ static int decode_init(AVCodecContext * avctx)
static int init=0;
int i, j, k;
if(!init) {
if (!init && !avctx->parse_only) {
/* scale factors table for layer 1/2 */
for(i=0;i<64;i++) {
int shift, mod;
......@@ -737,57 +737,95 @@ static void dct32(int32_t *out, int32_t *tab)
#if FRAC_BITS <= 15
#define OUT_SAMPLE(sum)\
{\
int sum1;\
sum1 = (sum + (1 << (OUT_SHIFT - 1))) >> OUT_SHIFT;\
if (sum1 < -32768)\
sum1 = -32768;\
else if (sum1 > 32767)\
sum1 = 32767;\
*samples = sum1;\
samples += incr;\
static inline int round_sample(int sum)
{
int sum1;
sum1 = (sum + (1 << (OUT_SHIFT - 1))) >> OUT_SHIFT;
if (sum1 < -32768)
sum1 = -32768;
else if (sum1 > 32767)
sum1 = 32767;
return sum1;
}
#define SUM8(off, op) \
{ \
sum op w[0 * 64 + off] * p[0 * 64];\
sum op w[1 * 64 + off] * p[1 * 64];\
sum op w[2 * 64 + off] * p[2 * 64];\
sum op w[3 * 64 + off] * p[3 * 64];\
sum op w[4 * 64 + off] * p[4 * 64];\
sum op w[5 * 64 + off] * p[5 * 64];\
sum op w[6 * 64 + off] * p[6 * 64];\
sum op w[7 * 64 + off] * p[7 * 64];\
}
#if defined(ARCH_POWERPC_405)
/* signed 16x16 -> 32 multiply add accumulate */
#define MACS(rt, ra, rb) \
asm ("maclhw %0, %2, %3" : "=r" (rt) : "0" (rt), "r" (ra), "r" (rb));
/* signed 16x16 -> 32 multiply */
#define MULS(ra, rb) \
({ int __rt; asm ("mullhw %0, %1, %2" : "=r" (__rt) : "r" (ra), "r" (rb)); __rt; })
#else
#define OUT_SAMPLE(sum)\
{\
int sum1;\
sum1 = (int)((sum + (int64_t_C(1) << (OUT_SHIFT - 1))) >> OUT_SHIFT);\
if (sum1 < -32768)\
sum1 = -32768;\
else if (sum1 > 32767)\
sum1 = 32767;\
*samples = sum1;\
samples += incr;\
/* signed 16x16 -> 32 multiply add accumulate */
#define MACS(rt, ra, rb) rt += (ra) * (rb)
/* signed 16x16 -> 32 multiply */
#define MULS(ra, rb) ((ra) * (rb))
#endif
#else
static inline int round_sample(int64_t sum)
{
int sum1;
sum1 = (int)((sum + (int64_t_C(1) << (OUT_SHIFT - 1))) >> OUT_SHIFT);
if (sum1 < -32768)
sum1 = -32768;
else if (sum1 > 32767)
sum1 = 32767;
return sum1;
}
#define SUM8(off, op) \
#define MULS(ra, rb) MUL64(ra, rb)
#endif
#define SUM8(sum, op, w, p) \
{ \
sum op MUL64(w[0 * 64 + off], p[0 * 64]);\
sum op MUL64(w[1 * 64 + off], p[1 * 64]);\
sum op MUL64(w[2 * 64 + off], p[2 * 64]);\
sum op MUL64(w[3 * 64 + off], p[3 * 64]);\
sum op MUL64(w[4 * 64 + off], p[4 * 64]);\
sum op MUL64(w[5 * 64 + off], p[5 * 64]);\
sum op MUL64(w[6 * 64 + off], p[6 * 64]);\
sum op MUL64(w[7 * 64 + off], p[7 * 64]);\
sum op MULS((w)[0 * 64], p[0 * 64]);\
sum op MULS((w)[1 * 64], p[1 * 64]);\
sum op MULS((w)[2 * 64], p[2 * 64]);\
sum op MULS((w)[3 * 64], p[3 * 64]);\
sum op MULS((w)[4 * 64], p[4 * 64]);\
sum op MULS((w)[5 * 64], p[5 * 64]);\
sum op MULS((w)[6 * 64], p[6 * 64]);\
sum op MULS((w)[7 * 64], p[7 * 64]);\
}
#define SUM8P2(sum1, op1, sum2, op2, w1, w2, p) \
{ \
int tmp;\
tmp = p[0 * 64];\
sum1 op1 MULS((w1)[0 * 64], tmp);\
sum2 op2 MULS((w2)[0 * 64], tmp);\
tmp = p[1 * 64];\
sum1 op1 MULS((w1)[1 * 64], tmp);\
sum2 op2 MULS((w2)[1 * 64], tmp);\
tmp = p[2 * 64];\
sum1 op1 MULS((w1)[2 * 64], tmp);\
sum2 op2 MULS((w2)[2 * 64], tmp);\
tmp = p[3 * 64];\
sum1 op1 MULS((w1)[3 * 64], tmp);\
sum2 op2 MULS((w2)[3 * 64], tmp);\
tmp = p[4 * 64];\
sum1 op1 MULS((w1)[4 * 64], tmp);\
sum2 op2 MULS((w2)[4 * 64], tmp);\
tmp = p[5 * 64];\
sum1 op1 MULS((w1)[5 * 64], tmp);\
sum2 op2 MULS((w2)[5 * 64], tmp);\
tmp = p[6 * 64];\
sum1 op1 MULS((w1)[6 * 64], tmp);\
sum2 op2 MULS((w2)[6 * 64], tmp);\
tmp = p[7 * 64];\
sum1 op1 MULS((w1)[7 * 64], tmp);\
sum2 op2 MULS((w2)[7 * 64], tmp);\
}
#endif
/* 32 sub band synthesis filter. Input: 32 sub band samples, Output:
32 samples. */
......@@ -797,15 +835,16 @@ static void synth_filter(MPADecodeContext *s1,
int32_t sb_samples[SBLIMIT])
{
int32_t tmp[32];
register MPA_INT *synth_buf, *p;
register MPA_INT *w;
register MPA_INT *synth_buf;
const register MPA_INT *w, *w2, *p;
int j, offset, v;
int16_t *samples2;
#if FRAC_BITS <= 15
int sum;
int sum, sum2;
#else
int64_t sum;
int64_t sum, sum2;
#endif
dct32(tmp, sb_samples);
offset = s1->synth_buf_offset[ch];
......@@ -826,32 +865,42 @@ static void synth_filter(MPADecodeContext *s1,
/* copy to avoid wrap */
memcpy(synth_buf + 512, synth_buf, 32 * sizeof(MPA_INT));
samples2 = samples + 31 * incr;
w = window;
for(j=0;j<16;j++) {
sum = 0;
p = synth_buf + 16 + j; /* 0-15 */
SUM8(0, +=);
p = synth_buf + 48 - j; /* 32-47 */
SUM8(32, -=);
OUT_SAMPLE(sum);
w++;
}
p = synth_buf + 32; /* 48 */
w2 = window + 31;
sum = 0;
SUM8(32, -=);
OUT_SAMPLE(sum);
p = synth_buf + 16;
SUM8(sum, +=, w, p);
p = synth_buf + 48;
SUM8(sum, -=, w + 32, p);
*samples = round_sample(sum);
samples += incr;
w++;
for(j=17;j<32;j++) {
/* we calculate two samples at the same time to avoid one memory
access per two sample */
for(j=1;j<16;j++) {
sum = 0;
p = synth_buf + 48 - j; /* 17-31 */
SUM8(0, -=);
p = synth_buf + 16 + j; /* 49-63 */
SUM8(32, -=);
OUT_SAMPLE(sum);
sum2 = 0;
p = synth_buf + 16 + j;
SUM8P2(sum, +=, sum2, -=, w, w2, p);
p = synth_buf + 48 - j;
SUM8P2(sum, -=, sum2, -=, w + 32, w2 + 32, p);
*samples = round_sample(sum);
samples += incr;
*samples2 = round_sample(sum2);
samples2 -= incr;
w++;
w2--;
}
p = synth_buf + 32;
sum = 0;
SUM8(sum, -=, w + 32, p);
*samples = round_sample(sum);
offset = (offset - 32) & 511;
s1->synth_buf_offset[ch] = offset;
}
......@@ -1157,6 +1206,47 @@ static int decode_header(MPADecodeContext *s, uint32_t header)
return 0;
}
/* useful helper to get mpeg audio stream infos. Return -1 if error in
header */
int mp_decode_header(int *sample_rate_ptr,
int *nb_channels_ptr,
int *coded_frame_size_ptr,
int *decoded_frame_size_ptr,
uint32_t head)
{
MPADecodeContext s1, *s = &s1;
int decoded_frame_size;
if (check_header(head) != 0)
return -1;
if (decode_header(s, head) != 0) {
return -1;
}
switch(s->layer) {
case 1:
decoded_frame_size = 384;
break;
case 2:
decoded_frame_size = 1152;
break;
default:
case 3:
if (s->lsf)
decoded_frame_size = 576;
else
decoded_frame_size = 1152;
break;
}
*sample_rate_ptr = s->sample_rate;
*nb_channels_ptr = s->nb_channels;
*coded_frame_size_ptr = s->frame_size;
*decoded_frame_size_ptr = decoded_frame_size * 2 * s->nb_channels;
return 0;
}
/* return the number of decoded frames */
static int mp_decode_layer1(MPADecodeContext *s)
{
......@@ -2391,7 +2481,20 @@ static int decode_frame(AVCodecContext * avctx,
avctx->sample_rate = s->sample_rate;
avctx->channels = s->nb_channels;
avctx->bit_rate = s->bit_rate;
avctx->frame_size = s->frame_size;
switch(s->layer) {
case 1:
avctx->frame_size = 384;
break;
case 2:
avctx->frame_size = 1152;
break;
case 3:
if (s->lsf)
avctx->frame_size = 576;
else
avctx->frame_size = 1152;
break;
}
}
}
} else if (s->frame_size == -1) {
......@@ -2457,15 +2560,22 @@ static int decode_frame(AVCodecContext * avctx,
buf_ptr += len;
s->inbuf_ptr += len;
buf_size -= len;
} else {
out_size = mp_decode_frame(s, out_samples);
}
next_data:
if (s->frame_size > 0 &&
(s->inbuf_ptr - s->inbuf) >= s->frame_size) {
if (avctx->parse_only) {
/* simply return the frame data */
*(uint8_t **)data = s->inbuf;
out_size = s->inbuf_ptr - s->inbuf;
} else {
out_size = mp_decode_frame(s, out_samples);
}
s->inbuf_ptr = s->inbuf;
s->frame_size = 0;
*data_size = out_size;
break;
}
next_data:
;
}
return buf_ptr - buf;
}
......@@ -2480,6 +2590,7 @@ AVCodec mp2_decoder =
NULL,
NULL,
decode_frame,
CODEC_CAP_PARSE_ONLY,
};
AVCodec mp3_decoder =
......@@ -2492,15 +2603,5 @@ AVCodec mp3_decoder =
NULL,
NULL,
decode_frame,
CODEC_CAP_PARSE_ONLY,
};
#undef C1
#undef C2
#undef C3
#undef C4
#undef C5
#undef C6
#undef C7
#undef C8
#undef FRAC_BITS
#undef HEADER_SIZE
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