From dc13d0b5ae2ea60861ad0716ce2b7c92be1a38b2 Mon Sep 17 00:00:00 2001
From: Baptiste Coudurier <baptiste.coudurier@gmail.com>
Date: Thu, 8 Mar 2007 22:14:04 +0000
Subject: [PATCH] seems safer to set pts timebase to sample rate, fix some mp3
Originally committed as revision 8300 to svn://svn.ffmpeg.org/ffmpeg/trunk
---
libavformat/swf.c | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/libavformat/swf.c b/libavformat/swf.c
index 49c43431f40..7d889af7d0b 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);
--
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