[SCM] libav/experimental: Change mpeg audio parser so it only sets frame_size, channels and bit_rate after it has a few valid frames. Fixes issue762
siretart at users.alioth.debian.org
siretart at users.alioth.debian.org
Sun Jun 30 16:33:57 UTC 2013
The following commit has been merged in the experimental branch:
commit c96bd21227e594856f8fd0610fd213b002056383
Author: Michael Niedermayer <michaelni at gmx.at>
Date: Sun Dec 21 23:50:16 2008 +0000
Change mpeg audio parser so it only sets frame_size, channels and bit_rate
after it has a few valid frames.
Fixes issue762
Originally committed as revision 16259 to svn://svn.ffmpeg.org/ffmpeg/trunk
diff --git a/libavcodec/mpegaudio.h b/libavcodec/mpegaudio.h
index 21de2da..33af18d 100644
--- a/libavcodec/mpegaudio.h
+++ b/libavcodec/mpegaudio.h
@@ -128,7 +128,7 @@ typedef struct HuffTable {
} HuffTable;
int ff_mpa_l2_select_table(int bitrate, int nb_channels, int freq, int lsf);
-int ff_mpa_decode_header(AVCodecContext *avctx, uint32_t head, int *sample_rate);
+int ff_mpa_decode_header(AVCodecContext *avctx, uint32_t head, int *sample_rate, int *channels, int *frame_size, int *bitrate);
void ff_mpa_synth_init(MPA_INT *window);
void ff_mpa_synth_filter(MPA_INT *synth_buf_ptr, int *synth_buf_offset,
MPA_INT *window, int *dither_state,
diff --git a/libavcodec/mpegaudio_parser.c b/libavcodec/mpegaudio_parser.c
index e7cb743..e4b17b0 100644
--- a/libavcodec/mpegaudio_parser.c
+++ b/libavcodec/mpegaudio_parser.c
@@ -44,7 +44,7 @@ typedef struct MpegAudioParseContext {
/* useful helper to get mpeg audio stream infos. Return -1 if error in
header, otherwise the coded frame size in bytes */
-int ff_mpa_decode_header(AVCodecContext *avctx, uint32_t head, int *sample_rate)
+int ff_mpa_decode_header(AVCodecContext *avctx, uint32_t head, int *sample_rate, int *channels, int *frame_size, int *bit_rate)
{
MPADecodeContext s1, *s = &s1;
s1.avctx = avctx;
@@ -58,23 +58,23 @@ int ff_mpa_decode_header(AVCodecContext *avctx, uint32_t head, int *sample_rate)
switch(s->layer) {
case 1:
- avctx->frame_size = 384;
+ *frame_size = 384;
break;
case 2:
- avctx->frame_size = 1152;
+ *frame_size = 1152;
break;
default:
case 3:
if (s->lsf)
- avctx->frame_size = 576;
+ *frame_size = 576;
else
- avctx->frame_size = 1152;
+ *frame_size = 1152;
break;
}
*sample_rate = s->sample_rate;
- avctx->channels = s->nb_channels;
- avctx->bit_rate = s->bit_rate;
+ *channels = s->nb_channels;
+ *bit_rate = s->bit_rate;
avctx->sub_id = s->layer;
return s->frame_size;
}
@@ -92,7 +92,7 @@ static int mpegaudio_parse(AVCodecParserContext *s1,
const uint8_t *buf, int buf_size)
{
MpegAudioParseContext *s = s1->priv_data;
- int len, ret, sr;
+ int len, ret, sr, channels, bit_rate, frame_size;
uint32_t header;
const uint8_t *buf_ptr;
@@ -123,7 +123,7 @@ static int mpegaudio_parse(AVCodecParserContext *s1,
got_header:
header = AV_RB32(s->inbuf);
- ret = ff_mpa_decode_header(avctx, header, &sr);
+ ret = ff_mpa_decode_header(avctx, header, &sr, &channels, &frame_size, &bit_rate);
if (ret < 0) {
s->header_count= -2;
/* no sync found : move by one byte (inefficient, but simple!) */
@@ -146,8 +146,12 @@ static int mpegaudio_parse(AVCodecParserContext *s1,
s->frame_size = -1;
}
#endif
- if(s->header_count > 1)
+ if(s->header_count > 1){
avctx->sample_rate= sr;
+ avctx->channels = channels;
+ avctx->frame_size = frame_size;
+ avctx->bit_rate = bit_rate;
+ }
}
}
} else
diff --git a/libavformat/mp3.c b/libavformat/mp3.c
index 407d4f0..0f3a5fe 100644
--- a/libavformat/mp3.c
+++ b/libavformat/mp3.c
@@ -383,7 +383,7 @@ static int mp3_read_probe(AVProbeData *p)
for(frames = 0; buf2 < end; frames++) {
header = AV_RB32(buf2);
- fsize = ff_mpa_decode_header(&avctx, header, &sample_rate);
+ fsize = ff_mpa_decode_header(&avctx, header, &sample_rate, &sample_rate, &sample_rate, &sample_rate);
if(fsize < 0)
break;
buf2 += fsize;
--
Libav/FFmpeg packaging
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