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webrtc: documentation on raw audio packet times

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This commit is contained in:
Koushik Dutta
2023-10-22 22:28:01 -07:00
parent 327acaec76
commit 672f01fd3f
1 changed files with 3 additions and 1 deletions
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4
plugins/webrtc/src/ffmpeg-to-wrtc.ts
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@@ -220,7 +220,7 @@ export async function createTrackForwarder(options: {
const audioRtpTrack: RtpTrack = {
codecCopy: audioCodecCopy,
onRtp: buffer => {
if (false && audioTransceiver.sender.codec.mimeType === "audio/opus") {
if (false && audioTransceiver.sender.codec.mimeType?.toLowerCase() === "audio/opus") {
// this will use 3 20ms frames, 60ms. seems to work up to 6/120ms
if (!opusRepacketizer)
opusRepacketizer = new OpusRepacketizer(3);
@@ -229,6 +229,8 @@ export async function createTrackForwarder(options: {
}
}
else {
// pcm audio can be concatenated.
// hikvision seems to send 40ms duration packets, so 25 packets per second.
audioTransceiver.sender.sendRtp(buffer);
}
},