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rtp: Support packetization/depacketization of opus

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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
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7
libavformat/rtpdec.c
View File

@ -55,6 +55,12 @@ static RTPDynamicProtocolHandler speex_dynamic_handler = {
.codec_id = AV_CODEC_ID_SPEEX,
};
static RTPDynamicProtocolHandler opus_dynamic_handler = {
.enc_name = "opus",
.codec_type = AVMEDIA_TYPE_AUDIO,
.codec_id = AV_CODEC_ID_OPUS,
};
/* statistics functions */
static RTPDynamicProtocolHandler *RTPFirstDynamicPayloadHandler= NULL;
@ -85,6 +91,7 @@ void av_register_rtp_dynamic_payload_handlers(void)
ff_register_dynamic_payload_handler(&ff_qcelp_dynamic_handler);
ff_register_dynamic_payload_handler(&realmedia_mp3_dynamic_handler);
ff_register_dynamic_payload_handler(&speex_dynamic_handler);
ff_register_dynamic_payload_handler(&opus_dynamic_handler);
ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfv_handler);
ff_register_dynamic_payload_handler(&ff_ms_rtp_asf_pfa_handler);

19
libavformat/rtpenc.c
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);

4
libavformat/sdp.c
View File

@ -576,6 +576,10 @@ static char *sdp_write_media_attributes(char *buff, int size, AVCodecContext *c,
av_strlcatf(buff, size, "a=rtpmap:%d speex/%d\r\n",
payload_type, c->sample_rate);
break;
case AV_CODEC_ID_OPUS:
av_strlcatf(buff, size, "a=rtpmap:%d opus/48000\r\n",
payload_type);
break;
default:
/* Nothing special to do here... */
break;