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dont set the sampling rate just because 1 mp3 packet header says so (fixes playback speed on some old mencoder generated avis which where then dumped to mp3)

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Originally committed as revision 6837 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Michael Niedermayer 2006-10-30 02:19:55 +00:00
parent c0d8052b50
commit 498c544ad2
4 changed files with 12 additions and 12 deletions
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2
libavcodec/mpegaudio.h
View File

@ -72,7 +72,7 @@ typedef int32_t MPA_INT;
#endif
int l2_select_table(int bitrate, int nb_channels, int freq, int lsf);
int mpa_decode_header(AVCodecContext *avctx, uint32_t head);
int mpa_decode_header(AVCodecContext *avctx, uint32_t head, int *sample_rate);
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,

11
libavcodec/mpegaudiodec.c
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@ -1190,7 +1190,7 @@ static int decode_header(MPADecodeContext *s, uint32_t header)
/* useful helper to get mpeg audio stream infos. Return -1 if error in
header, otherwise the coded frame size in bytes */
int mpa_decode_header(AVCodecContext *avctx, uint32_t head)
int mpa_decode_header(AVCodecContext *avctx, uint32_t head, int *sample_rate)
{
MPADecodeContext s1, *s = &s1;
@ -1217,7 +1217,7 @@ int mpa_decode_header(AVCodecContext *avctx, uint32_t head)
break;
}
avctx->sample_rate = s->sample_rate;
*sample_rate = s->sample_rate;
avctx->channels = s->nb_channels;
avctx->bit_rate = s->bit_rate;
avctx->sub_id = s->layer;
@ -2547,7 +2547,6 @@ retry:
return -1;
}
/* update codec info */
avctx->sample_rate = s->sample_rate;
avctx->channels = s->nb_channels;
avctx->bit_rate = s->bit_rate;
avctx->sub_id = s->layer;
@ -2574,9 +2573,11 @@ retry:
}
out_size = mp_decode_frame(s, out_samples, buf, buf_size);
if(out_size>=0)
if(out_size>=0){
*data_size = out_size;
else
avctx->sample_rate = s->sample_rate;
//FIXME maybe move the other codec info stuff from above here too
}else
av_log(avctx, AV_LOG_DEBUG, "Error while decoding MPEG audio frame.\n"); //FIXME return -1 / but also return the number of bytes consumed
s->frame_size = 0;
return buf_size;

7
libavcodec/parser.c
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@ -666,11 +666,10 @@ static int mpegaudio_parse(AVCodecParserContext *s1,
}
if ((s->inbuf_ptr - s->inbuf) >= MPA_HEADER_SIZE) {
got_header:
sr= avctx->sample_rate;
header = (s->inbuf[0] << 24) | (s->inbuf[1] << 16) |
(s->inbuf[2] << 8) | s->inbuf[3];
ret = mpa_decode_header(avctx, header);
ret = mpa_decode_header(avctx, header, &sr);
if (ret < 0) {
s->header_count= -2;
/* no sync found : move by one byte (inefficient, but simple!) */
@ -694,8 +693,8 @@ static int mpegaudio_parse(AVCodecParserContext *s1,
}
#endif
}
if(s->header_count <= 0)
avctx->sample_rate= sr; //FIXME ugly
if(s->header_count > 1)
avctx->sample_rate= sr;
}
} else
#if 0

4
libavformat/mp3.c
View File

@ -247,7 +247,7 @@ static void id3_create_tag(AVFormatContext *s, uint8_t *buf)
static int mp3_read_probe(AVProbeData *p)
{
int max_frames, first_frames;
int fsize, frames;
int fsize, frames, sample_rate;
uint32_t header;
uint8_t *buf, *buf2, *end;
AVCodecContext avctx;
@ -267,7 +267,7 @@ static int mp3_read_probe(AVProbeData *p)
for(frames = 0; buf2 < end; frames++) {
header = (buf2[0] << 24) | (buf2[1] << 16) | (buf2[2] << 8) | buf2[3];
fsize = mpa_decode_header(&avctx, header);
fsize = mpa_decode_header(&avctx, header, &sample_rate);
if(fsize < 0)
break;
buf2 += fsize;