[SCM] libav/experimental: seems safer to set pts timebase to sample rate, fix some mp3
siretart at users.alioth.debian.org
siretart at users.alioth.debian.org
Sun Jun 30 15:57:30 UTC 2013
The following commit has been merged in the experimental branch:
commit dc13d0b5ae2ea60861ad0716ce2b7c92be1a38b2
Author: Baptiste Coudurier <baptiste.coudurier at gmail.com>
Date: Thu Mar 8 22:14:04 2007 +0000
seems safer to set pts timebase to sample rate, fix some mp3
Originally committed as revision 8300 to svn://svn.ffmpeg.org/ffmpeg/trunk
diff --git a/libavformat/swf.c b/libavformat/swf.c
index 49c4343..7d889af 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);
--
Libav/FFmpeg packaging
More information about the pkg-multimedia-commits
mailing list