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webrtc: add some debug code

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This commit is contained in:
Koushik Dutta
2022-05-17 11:45:26 -07:00
parent 5ee18780b9
commit 5bae16977f
2 changed files with 9 additions and 5 deletions
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1
plugins/webrtc/src/main.ts
View File

@@ -167,6 +167,7 @@ export class WebRTCPlugin extends AutoenableMixinProvider implements DeviceCreat
if (interfaces.includes(ScryptedInterface.RTCSignalingChannel)) {
const ret = [
ScryptedInterface.RTCSignalingChannel,
// ScryptedInterface.Settings,
];
if (type === ScryptedDeviceType.Speaker) {
ret.push(ScryptedInterface.Intercom);

13
plugins/webrtc/src/wrtc-to-rtsp.ts
View File

@@ -110,13 +110,16 @@ export async function createRTCPeerConnectionSource(options: {
let audioTrack: string;
let videoTrack: string;
let audioTransceiver: RTCRtpTransceiver;
const useRtspJitterBuffer = false;
const doSetup = async (setup: RTCAVSignalingSetup) => {
let gotAudio = false;
let gotVideo = false;
audioTransceiver = pc.addTransceiver("audio", setup.audio as any);
audioTransceiver.onTrack.subscribe((track) => {
if (useUdp) {
if (useUdp || !useRtspJitterBuffer) {
track.onReceiveRtp.subscribe(rtp => {
if (!gotAudio) {
gotAudio = true;
@@ -147,13 +150,13 @@ export async function createRTCPeerConnectionSource(options: {
}
}
}
jitter.pipe(new RtspOutput())
jitter.pipe(new RtspOutput());
}
});
const videoTransceiver = pc.addTransceiver("video", setup.video as any);
videoTransceiver.onTrack.subscribe((track) => {
if (useUdp) {
if (useUdp || !useRtspJitterBuffer) {
track.onReceiveRtp.subscribe(rtp => {
if (!gotVideo) {
gotVideo = true;
@@ -184,7 +187,7 @@ export async function createRTCPeerConnectionSource(options: {
}
}
}
jitter.pipe(new RtspOutput())
jitter.pipe(new RtspOutput());
}
track.onReceiveRtp.once(() => {
@@ -397,7 +400,7 @@ export async function createRTCPeerConnectionSource(options: {
}
else {
const url = `rtsp://127.0.0.1:${port}`;
const mediaStreamUrl : MediaStreamUrl = {
const mediaStreamUrl: MediaStreamUrl = {
url,
mediaStreamOptions,
};