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seems safer to set pts timebase to sample rate, fix some mp3

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Originally committed as revision 8300 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Baptiste Coudurier 2007-03-08 22:14:04 +00:00
parent b912ef3fc9
commit dc13d0b5ae
1 changed files with 1 additions and 1 deletions
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2
libavformat/swf.c
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@ -679,7 +679,6 @@ static int swf_read_header(AVFormatContext *s, AVFormatParameters *ap)
v = get_byte(pb);
swf->samples_per_frame = get_le16(pb);
ast = av_new_stream(s, -1); /* -1 to avoid clash with video stream ch_id */
av_set_pts_info(ast, 64, 256, swf->frame_rate); /* XXX same as video stream */
swf->audio_stream_index = ast->index;
ast->codec->channels = 1 + (v&1);
ast->codec->codec_type = CODEC_TYPE_AUDIO;
@ -689,6 +688,7 @@ static int swf_read_header(AVFormatContext *s, AVFormatParameters *ap)
if (!sample_rate_code)
return AVERROR_IO;
ast->codec->sample_rate = 11025 << (sample_rate_code-1);
av_set_pts_info(ast, 64, 1, ast->codec->sample_rate);
if (len > 4)
url_fskip(pb,len-4);