rtp: Support packetization/depacketization of opus

Signed-off-by: Martin Storsjö <martin@martin.st>
This commit is contained in:
Martin Storsjö 2012-10-09 00:51:42 +03:00
parent e04826c34e
commit c136a813d7
3 changed files with 30 additions and 0 deletions

View file

@ -77,6 +77,7 @@ static int is_supported(enum AVCodecID id)
case AV_CODEC_ID_ILBC:
case AV_CODEC_ID_MJPEG:
case AV_CODEC_ID_SPEEX:
case AV_CODEC_ID_OPUS:
return 1;
default:
return 0;
@ -186,6 +187,16 @@ static int rtp_write_header(AVFormatContext *s1)
* 8000, even if the sample rate is 16000. See RFC 3551. */
avpriv_set_pts_info(st, 32, 1, 8000);
break;
case AV_CODEC_ID_OPUS:
if (st->codec->channels > 2) {
av_log(s1, AV_LOG_ERROR, "Multistream opus not supported in RTP\n");
goto fail;
}
/* The opus RTP RFC says that all opus streams should use 48000 Hz
* as clock rate, since all opus sample rates can be expressed in
* this clock rate, and sample rate changes on the fly are supported. */
avpriv_set_pts_info(st, 32, 1, 48000);
break;
case AV_CODEC_ID_ILBC:
if (st->codec->block_align != 38 && st->codec->block_align != 50) {
av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n");
@ -525,6 +536,14 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
case AV_CODEC_ID_MJPEG:
ff_rtp_send_jpeg(s1, pkt->data, size);
break;
case AV_CODEC_ID_OPUS:
if (size > s->max_payload_size) {
av_log(s1, AV_LOG_ERROR,
"Packet size %d too large for max RTP payload size %d\n",
size, s->max_payload_size);
return AVERROR(EINVAL);
}
/* Intentional fallthrough */
default:
/* better than nothing : send the codec raw data */
rtp_send_raw(s1, pkt->data, size);