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| author | Baptiste Coudurier <baptiste.coudurier@gmail.com> | 2007-03-08 22:14:04 +0000 |
|---|---|---|
| committer | Baptiste Coudurier <baptiste.coudurier@gmail.com> | 2007-03-08 22:14:04 +0000 |
| commit | dc13d0b5ae2ea60861ad0716ce2b7c92be1a38b2 (patch) | |
| tree | 4b004b6f35c5aeb295741a6718991141c9a9e0d2 /libavformat/swf.c | |
| parent | b912ef3fc924a2cd2ee29d004b6d2bb9c37862ba (diff) | |
seems safer to set pts timebase to sample rate, fix some mp3
Originally committed as revision 8300 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavformat/swf.c')
| -rw-r--r-- | libavformat/swf.c | 2 |
1 files changed, 1 insertions, 1 deletions
diff --git a/libavformat/swf.c b/libavformat/swf.c index 49c43431f4..7d889af7d0 100644 --- a/libavformat/swf.c +++ b/libavformat/swf.c @@ -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); |