Mercurial > libavformat.hg
changeset 1636:783969dab069 libavformat
always export audiostreams
| author | bcoudurier |
|---|---|
| date | Sun, 14 Jan 2007 00:54:03 +0000 |
| parents | 59d2c40950ca |
| children | 50bbf9658436 |
| files | swf.c |
| diffstat | 1 files changed, 21 insertions(+), 26 deletions(-) [+] |
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line diff
--- a/swf.c Sun Jan 14 00:46:37 2007 +0000 +++ b/swf.c Sun Jan 14 00:54:03 2007 +0000 @@ -811,35 +811,30 @@ 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); }