[FFmpeg-cvslog] Parse DEFINESOUND tags in swf (fix ticket 1638)

[Michael Bradshaw]

ffmpeg | branch: master | Michael Bradshaw <mbradshaw at sorensonmedia.com> | Thu Oct 11 19:48:44 2012 -0600| [a41c824c539cc1fe79b3afd9393a91d22a8065da] | committer: Michael Niedermayer

Parse DEFINESOUND tags in swf (fix ticket 1638)

Signed-off-by: Michael Bradshaw <mbradshaw at sorensonmedia.com>
Signed-off-by: Michael Niedermayer <michaelni at gmx.at>

> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=a41c824c539cc1fe79b3afd9393a91d22a8065da
---

 libavformat/swf.h    |    1 +
 libavformat/swfdec.c |   38 ++++++++++++++++++++++++++++++++++++++
 2 files changed, 39 insertions(+)

diff --git a/libavformat/swf.h b/libavformat/swf.h
index 5f0d486..4e8f877 100644
--- a/libavformat/swf.h
+++ b/libavformat/swf.h
@@ -44,6 +44,7 @@
 #define TAG_FREECHARACTER 3
 #define TAG_PLACEOBJECT   4
 #define TAG_REMOVEOBJECT  5
+#define TAG_DEFINESOUND   14
 #define TAG_STREAMHEAD    18
 #define TAG_STREAMBLOCK   19
 #define TAG_JPEG2         21
diff --git a/libavformat/swfdec.c b/libavformat/swfdec.c
index 3967d82..8a57a34 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;