Mercurial > libavformat.hg
diff rtpenc.c @ 6476:f06908662125 libavformat
Handle G.722 in RTP, and all the exceptions mandated in RFC 3551
author | mstorsjo |
---|---|
date | Wed, 15 Sep 2010 17:35:39 +0000 |
parents | 37944ce385a0 |
children |
line wrap: on
line diff
--- a/rtpenc.c Tue Sep 14 22:20:46 2010 +0000 +++ b/rtpenc.c Wed Sep 15 17:35:39 2010 +0000 @@ -56,6 +56,7 @@ case CODEC_ID_VORBIS: case CODEC_ID_THEORA: case CODEC_ID_VP8: + case CODEC_ID_ADPCM_G722: return 1; default: return 0; @@ -148,6 +149,11 @@ case CODEC_ID_VP8: av_log(s1, AV_LOG_WARNING, "RTP VP8 payload is still experimental\n"); break; + case CODEC_ID_ADPCM_G722: + /* Due to a historical error, the clock rate for G722 in RTP is + * 8000, even if the sample rate is 16000. See RFC 3551. */ + av_set_pts_info(st, 32, 1, 8000); + break; case CODEC_ID_AMR_NB: case CODEC_ID_AMR_WB: if (!s->max_frames_per_packet) @@ -382,6 +388,12 @@ case CODEC_ID_PCM_S16LE: rtp_send_samples(s1, pkt->data, size, 2 * st->codec->channels); break; + case CODEC_ID_ADPCM_G722: + /* The actual sample size is half a byte per sample, but since the + * stream clock rate is 8000 Hz while the sample rate is 16000 Hz, + * the correct parameter for send_samples is 1 byte per stream clock. */ + rtp_send_samples(s1, pkt->data, size, 1 * st->codec->channels); + break; case CODEC_ID_MP2: case CODEC_ID_MP3: rtp_send_mpegaudio(s1, pkt->data, size);