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always export audiostreams

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Originally committed as revision 7454 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Baptiste Coudurier 2007-01-14 00:54:03 +00:00
parent 69566e4365
commit ec54839ab1
1 changed files with 21 additions and 26 deletions
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47
libavformat/swf.c
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@ -811,35 +811,30 @@ static int swf_read_header(AVFormatContext *s, AVFormatParameters *ap)
get_byte(pb);
v = get_byte(pb);
swf->samples_per_frame = get_le16(pb);
ast = av_new_stream(s, 1);
av_set_pts_info(ast, 24, 1, 1000); /* 24 bit pts in ms */
ast->codec->channels = 1 + (v&1);
ast->codec->codec_type = CODEC_TYPE_AUDIO;
if (v & 0x20)
ast->codec->codec_id = CODEC_ID_MP3;
ast->need_parsing = 1;
switch((v>> 2) & 0x03) {
case 1:
ast->codec->sample_rate = 11025;
break;
case 2:
ast->codec->sample_rate = 22050;
break;
case 3:
ast->codec->sample_rate = 44100;
break;
default:
return AVERROR_IO;
}
if (len > 4)
url_fskip(pb,len-4);
/* if mp3 streaming found, OK */
if ((v & 0x20) != 0) {
ast = av_new_stream(s, 1);
if (!ast)
return -ENOMEM;
av_set_pts_info(ast, 24, 1, 1000); /* 24 bit pts in ms */
ast->codec->channels = 1 + (v&1);
switch((v>> 2) & 0x03) {
case 1:
ast->codec->sample_rate = 11025;
break;
case 2:
ast->codec->sample_rate = 22050;
break;
case 3:
ast->codec->sample_rate = 44100;
break;
default:
av_free(ast);
return AVERROR_IO;
}
ast->codec->codec_type = CODEC_TYPE_AUDIO;
ast->codec->codec_id = CODEC_ID_MP3;
ast->need_parsing = 1;
}
} else {
url_fskip(pb, len);
}