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authorLuca Abeni <lucabe72@email.it>2007-09-14 08:17:06 +0000
committerLuca Abeni <lucabe72@email.it>2007-09-14 08:17:06 +0000
commit171dce486ce85fbb6a54e24394e1d740395437b0 (patch)
tree56c85d4e9ec26692446297bf5b7b2208764093f9 /libavformat/rtp.c
parentf0dd9d4505675daa0f4fda6fcf4274416a23bf24 (diff)
downloadffmpeg-171dce486ce85fbb6a54e24394e1d740395437b0.tar.gz
Support for AAC streaming over RTP. Fragmentation is not implemented yet
Originally committed as revision 10491 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavformat/rtp.c')
-rw-r--r--libavformat/rtp.c6
1 files changed, 6 insertions, 0 deletions
diff --git a/libavformat/rtp.c b/libavformat/rtp.c
index 60256c593c..fb4a66d151 100644
--- a/libavformat/rtp.c
+++ b/libavformat/rtp.c
@@ -28,6 +28,7 @@
#include "rtp_internal.h"
#include "rtp_h264.h"
#include "rtp_mpv.h"
+#include "rtp_aac.h"
//#define DEBUG
@@ -762,6 +763,8 @@ static int rtp_write_header(AVFormatContext *s1)
s->max_payload_size = n * TS_PACKET_SIZE;
s->buf_ptr = s->buf;
break;
+ case CODEC_ID_AAC:
+ s->read_buf_index = 0;
default:
if (st->codec->codec_type == CODEC_TYPE_AUDIO) {
av_set_pts_info(st, 32, 1, st->codec->sample_rate);
@@ -993,6 +996,9 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
case CODEC_ID_MPEG1VIDEO:
ff_rtp_send_mpegvideo(s1, buf1, size);
break;
+ case CODEC_ID_AAC:
+ ff_rtp_send_aac(s1, buf1, size);
+ break;
case CODEC_ID_MPEG2TS:
rtp_send_mpegts_raw(s1, buf1, size);
break;