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| author | Martin Storsjö <martin@martin.st> | 2010-09-15 17:35:39 +0000 |
|---|---|---|
| committer | Martin Storsjö <martin@martin.st> | 2010-09-15 17:35:39 +0000 |
| commit | 0048a2a8d347c9a81a781f4126023018f1b29527 (patch) | |
| tree | eb9e71c1e3edaa6eface988067c0fac44bd5656b /libavformat/rtpdec.c | |
| parent | 82eac2f3216534c065c5023e5599720bd17bed26 (diff) | |
| download | ffmpeg-0048a2a8d347c9a81a781f4126023018f1b29527.tar.gz | |
Handle G.722 in RTP, and all the exceptions mandated in RFC 3551
Originally committed as revision 25125 to svn://svn.ffmpeg.org/ffmpeg/trunk
Diffstat (limited to 'libavformat/rtpdec.c')
| -rw-r--r-- | libavformat/rtpdec.c | 7 |
1 files changed, 7 insertions, 0 deletions
diff --git a/libavformat/rtpdec.c b/libavformat/rtpdec.c index debc14c90b..942b8d71c8 100644 --- a/libavformat/rtpdec.c +++ b/libavformat/rtpdec.c @@ -365,6 +365,13 @@ RTPDemuxContext *rtp_parse_open(AVFormatContext *s1, AVStream *st, URLContext *r case CODEC_ID_H264: st->need_parsing = AVSTREAM_PARSE_FULL; break; + case CODEC_ID_ADPCM_G722: + av_set_pts_info(st, 32, 1, st->codec->sample_rate); + /* According to RFC 3551, the stream clock rate is 8000 + * even if the sample rate is 16000. */ + if (st->codec->sample_rate == 8000) + st->codec->sample_rate = 16000; + break; default: if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { av_set_pts_info(st, 32, 1, st->codec->sample_rate); |