rtp: Support packetization/depacketization of opus
Signed-off-by: Martin Storsjö <martin@martin.st>
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e04826c34e
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c136a813d7
3 changed files with 30 additions and 0 deletions
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@ -77,6 +77,7 @@ static int is_supported(enum AVCodecID id)
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case AV_CODEC_ID_ILBC:
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case AV_CODEC_ID_MJPEG:
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case AV_CODEC_ID_SPEEX:
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case AV_CODEC_ID_OPUS:
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return 1;
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default:
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return 0;
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@ -186,6 +187,16 @@ static int rtp_write_header(AVFormatContext *s1)
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* 8000, even if the sample rate is 16000. See RFC 3551. */
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avpriv_set_pts_info(st, 32, 1, 8000);
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break;
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case AV_CODEC_ID_OPUS:
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if (st->codec->channels > 2) {
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av_log(s1, AV_LOG_ERROR, "Multistream opus not supported in RTP\n");
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goto fail;
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}
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/* The opus RTP RFC says that all opus streams should use 48000 Hz
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* as clock rate, since all opus sample rates can be expressed in
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* this clock rate, and sample rate changes on the fly are supported. */
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avpriv_set_pts_info(st, 32, 1, 48000);
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break;
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case AV_CODEC_ID_ILBC:
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if (st->codec->block_align != 38 && st->codec->block_align != 50) {
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av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n");
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@ -525,6 +536,14 @@ static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt)
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case AV_CODEC_ID_MJPEG:
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ff_rtp_send_jpeg(s1, pkt->data, size);
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break;
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case AV_CODEC_ID_OPUS:
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if (size > s->max_payload_size) {
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av_log(s1, AV_LOG_ERROR,
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"Packet size %d too large for max RTP payload size %d\n",
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size, s->max_payload_size);
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return AVERROR(EINVAL);
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}
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/* Intentional fallthrough */
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default:
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/* better than nothing : send the codec raw data */
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rtp_send_raw(s1, pkt->data, size);
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