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| author | Luca Abeni <lucabe72@email.it> | 2007-09-14 08:17:06 +0000 |
|---|---|---|
| committer | Luca Abeni <lucabe72@email.it> | 2007-09-14 08:17:06 +0000 |
| commit | 171dce486ce85fbb6a54e24394e1d740395437b0 (patch) | |
| tree | 56c85d4e9ec26692446297bf5b7b2208764093f9 /libavformat/rtp.c | |
| parent | f0dd9d4505675daa0f4fda6fcf4274416a23bf24 (diff) | |
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.c | 6 |
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; |