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Handle G.722 in RTP, and all the exceptions mandated in RFC 3551

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Originally committed as revision 25125 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Martin Storsjö
2010-09-15 17:35:39 +00:00
parent 82eac2f321
commit 0048a2a8d3
4 changed files with 26 additions and 1 deletions
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12
libavformat/rtpenc.c
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@ -56,6 +56,7 @@ static int is_supported(enum CodecID id)
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 @@ static int rtp_write_header(AVFormatContext *s1)
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 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
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);