diff options
| -rw-r--r-- | libavformat/rtpdec.c | 7 | ||||
| -rw-r--r-- | libavformat/rtpenc.c | 19 | ||||
| -rw-r--r-- | libavformat/sdp.c | 4 |
3 files changed, 30 insertions, 0 deletions
diff --git a/libavformat/rtpdec.c b/libavformat/rtpdec.c index 0f4e1d1d86..dac367dd86 100644 --- a/libavformat/rtpdec.c +++ b/libavformat/rtpdec.c @@ -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); diff --git a/libavformat/rtpenc.c b/libavformat/rtpenc.c index 36064ed610..b17c4651b6 100644 --- a/libavformat/rtpenc.c +++ b/libavformat/rtpenc.c @@ -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); diff --git a/libavformat/sdp.c b/libavformat/sdp.c index 91de413dc6..0f7eb2f00d 100644 --- a/libavformat/sdp.c +++ b/libavformat/sdp.c @@ -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; |