Encode as RTP stream

[mkhahani]

What would be the equivalent code for the following command? We plan to use beamcoder in a WebRTC project if is possible.

ffmpeg \
  -re \
  -v info \
  -i video.mp4 \
  -map 0:v:0 \
  -pix_fmt yuv420p -c:v libvpx -b:v 1000k -deadline realtime -cpu-used 4 \
  -f tee \
  "[select=v:f=rtp:ssrc=2222:payload_type=101]rtp://192.168.1.100:10090"

[Bahlinc-Dev]

First of all you are only encoding video, but not sending audio because you inserted a "tee" which makes multiple muxers, but then only are using the video stream part. The payload type also does not look right, should be 103 I think?

Anyways, to send video via rtp you would use this:

`//video stream to rtp
const vmux = await beamcoder.muxer({
format_name: 'rtp'
});

vmux.newStream(demux.streams[0]); // get stream from demuxer or make a new one

await vmux.openIO({
url: 'rtp://192.168.1.100:10090' + '?rtcpport=' // where to send the stream remove rtcp part if not needed
});

await vmux.writeHeader({
payload_type: 103, // IMPORTANT. libav sets a default of -1 for the payload type, webrtc does not like that
ssrc: 1111, // stream identifier, can be any positive integer number (must be expected by your webrtc implementation)
});

let vDiff;
while (true) {
const packet = await demux.read();

// video packets
if (packet.stream_index === 0) {
  if (!vDiff) vDiff = packet.pts; // if first packet is > 0 pts, subtract that out so we always forward a stream starting at zero (players like that)
  packet.pts = packet.pts - vDiff;
  packet.dts = packet.pts; // unless you have b frames, dts can equal pts.  You probably don't have b frames.
  await vmux.writeFrame({
    packet: packet
  });
}

}`

If you plan on passing audio also, opus requires a few steps depending on the source. You probably need a filter instead of just passing from decoder to encoder. Especially if changing sample rate from source.

(edit: not sure why the formatting here is messed up?)