diff options
author | Michael Bradshaw <mbradshaw@sorensonmedia.com> | 2012-10-12 05:48:44 +0400 |
---|---|---|
committer | Michael Niedermayer <michaelni@gmx.at> | 2012-10-12 19:37:34 +0400 |
commit | a41c824c539cc1fe79b3afd9393a91d22a8065da (patch) | |
tree | 75781289b6793195c777cae791fc47e8d8bd476d /libavformat/swfdec.c | |
parent | a51540d81167547f1c7c2b8f545c246711611071 (diff) |
Parse DEFINESOUND tags in swf (fix ticket 1638)
Signed-off-by: Michael Bradshaw <mbradshaw@sorensonmedia.com>
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavformat/swfdec.c')
-rw-r--r-- | libavformat/swfdec.c | 38 |
1 files changed, 38 insertions, 0 deletions
diff --git a/libavformat/swfdec.c b/libavformat/swfdec.c index 3967d82ef3..8a57a34d04 100644 --- a/libavformat/swfdec.c +++ b/libavformat/swfdec.c @@ -200,6 +200,44 @@ static int swf_read_packet(AVFormatContext *s, AVPacket *pkt) ast->codec->sample_rate = 44100 >> (3 - sample_rate_code); avpriv_set_pts_info(ast, 64, 1, ast->codec->sample_rate); len -= 4; + } else if (tag == TAG_DEFINESOUND) { + /* audio stream */ + int sample_rate_code; + int ch_id = avio_rl16(pb); + + for (i=0; i<s->nb_streams; i++) { + st = s->streams[i]; + if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO && st->id == ch_id) + goto skip; + } + + // FIXME: 8-bit uncompressed PCM audio will be interpreted as 16-bit + // FIXME: The entire audio stream is stored in a single chunk/tag. Normally, + // these are smaller audio streams in DEFINESOUND tags, but it's technically + // possible they could be huge. Break it up into multiple packets if it's big. + v = avio_r8(pb); + ast = avformat_new_stream(s, NULL); + if (!ast) + return -1; + ast->id = ch_id; + ast->codec->channels = 1 + (v&1); + ast->codec->codec_type = AVMEDIA_TYPE_AUDIO; + ast->codec->codec_id = ff_codec_get_id(swf_audio_codec_tags, (v>>4) & 15); + ast->need_parsing = AVSTREAM_PARSE_FULL; + sample_rate_code= (v>>2) & 3; + ast->codec->sample_rate = 44100 >> (3 - sample_rate_code); + avpriv_set_pts_info(ast, 64, 1, ast->codec->sample_rate); + ast->duration = avio_rl32(pb); // number of samples + if (((v>>4) & 15) == 2) { // MP3 sound data record + ast->skip_samples = avio_rl16(pb); + len -= 2; + } + len -= 7; + if ((res = av_get_packet(pb, pkt, len)) < 0) + return res; + pkt->pos = pos; + pkt->stream_index = ast->index; + return pkt->size; } else if (tag == TAG_VIDEOFRAME) { int ch_id = avio_rl16(pb); len -= 2; |