Commit a41c824c authored by Michael Bradshaw's avatar Michael Bradshaw Committed by Michael Niedermayer

Parse DEFINESOUND tags in swf (fix ticket 1638)

Signed-off-by: 's avatarMichael Bradshaw <mbradshaw@sorensonmedia.com>
Signed-off-by: 's avatarMichael Niedermayer <michaelni@gmx.at>
parent a51540d8
...@@ -44,6 +44,7 @@ ...@@ -44,6 +44,7 @@
#define TAG_FREECHARACTER 3 #define TAG_FREECHARACTER 3
#define TAG_PLACEOBJECT 4 #define TAG_PLACEOBJECT 4
#define TAG_REMOVEOBJECT 5 #define TAG_REMOVEOBJECT 5
#define TAG_DEFINESOUND 14
#define TAG_STREAMHEAD 18 #define TAG_STREAMHEAD 18
#define TAG_STREAMBLOCK 19 #define TAG_STREAMBLOCK 19
#define TAG_JPEG2 21 #define TAG_JPEG2 21
......
...@@ -200,6 +200,44 @@ static int swf_read_packet(AVFormatContext *s, AVPacket *pkt) ...@@ -200,6 +200,44 @@ static int swf_read_packet(AVFormatContext *s, AVPacket *pkt)
ast->codec->sample_rate = 44100 >> (3 - sample_rate_code); ast->codec->sample_rate = 44100 >> (3 - sample_rate_code);
avpriv_set_pts_info(ast, 64, 1, ast->codec->sample_rate); avpriv_set_pts_info(ast, 64, 1, ast->codec->sample_rate);
len -= 4; 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) { } else if (tag == TAG_VIDEOFRAME) {
int ch_id = avio_rl16(pb); int ch_id = avio_rl16(pb);
len -= 2; len -= 2;
......
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