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webrtc: update werift with fixed track removal. add support for replaceTrack.

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This commit is contained in:
Koushik Dutta
2022-10-16 20:26:02 -07:00
parent c13604bed9
commit aa09a4960a
5 changed files with 39 additions and 19 deletions
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2
external/werift vendored

Submodule external/werift updated: 26cd73cedd...530a2bc747

4
plugins/webrtc/package-lock.json generated
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@@ -1,12 +1,12 @@
{
"name": "@scrypted/webrtc",
"version": "0.0.77",
"version": "0.0.78",
"lockfileVersion": 2,
"requires": true,
"packages": {
"": {
"name": "@scrypted/webrtc",
"version": "0.0.77",
"version": "0.0.78",
"dependencies": {
"@koush/werift": "file:../../external/werift/packages/webrtc",
"@scrypted/common": "file:../../common",

2
plugins/webrtc/package.json
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@@ -1,6 +1,6 @@
{
"name": "@scrypted/webrtc",
"version": "0.0.77",
"version": "0.0.78",
"scripts": {
"scrypted-setup-project": "scrypted-setup-project",
"prescrypted-setup-project": "scrypted-package-json",

40
plugins/webrtc/src/ffmpeg-to-wrtc.ts
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@@ -1,7 +1,7 @@
import { MediaStreamTrack, RTCPeerConnection, RTCRtpTransceiver, RtpPacket } from "@koush/werift";
import { Deferred } from "@scrypted/common/src/deferred";
import sdk, { BufferConverter, BufferConvertorOptions, FFmpegInput, FFmpegTranscodeStream, Intercom, MediaObject, MediaStreamDestination, RequestMediaStream, RTCAVSignalingSetup, RTCConnectionManagement, RTCMediaObjectTrack, RTCSignalingOptions, RTCSignalingSession, ScryptedDevice, ScryptedDeviceBase, ScryptedInterface, ScryptedMimeTypes } from "@scrypted/sdk";
import sdk, { BufferConverter, BufferConvertorOptions, FFmpegInput, FFmpegTranscodeStream, Intercom, MediaObject, MediaStreamDestination, RequestMediaStream, RTCAVSignalingSetup, RTCConnectionManagement, RTCMediaObjectTrack, RTCMediaObjectTrackOptions, RTCSignalingOptions, RTCSignalingSession, ScryptedDevice, ScryptedDeviceBase, ScryptedInterface, ScryptedMimeTypes } from "@scrypted/sdk";
import type { WebRTCPlugin } from "./main";
import { ScryptedSessionControl } from "./session-control";
import { requiredAudioCodecs, requiredVideoCodec } from "./webrtc-required-codecs";
@@ -230,16 +230,23 @@ class WebRTCTrack implements RTCMediaObjectTrack {
control: ScryptedSessionControl;
removed = new Deferred<void>();
constructor(public connectionManagement: WebRTCConnectionManagement, public video: RTCRtpTransceiver, public audio: RTCRtpTransceiver, public intercom: Intercom) {
constructor(public connectionManagement: WebRTCConnectionManagement, public video: RTCRtpTransceiver, public audio: RTCRtpTransceiver, intercom: Intercom) {
this.control = new ScryptedSessionControl(async () => { }, intercom, audio);
}
async replace(mediaObject: MediaObject): Promise<void> {
throw new Error("Method not implemented.");
const { atrack, vtrack, createTrackForwarder, intercom } = await this.connectionManagement.createTracks(mediaObject);
this.cleanup(true);
this.video.sender.replaceTrack(vtrack);
this.audio.sender.replaceTrack(atrack);
const { atrack, vtrack, createTrackForwarder, intercom } = await this.connectionManagement.createTracks(mediaObject);
this.removed = new Deferred();
this.control = new ScryptedSessionControl(async () => { }, intercom, this.audio);
const f = await createTrackForwarder(this.video, this.audio)
waitClosed(this.connectionManagement.pc).finally(() => f.kill());
this.removed.promise.finally(() => f.kill());
await this.video.sender.replaceTrack(vtrack);
await this.audio.sender.replaceTrack(atrack);
}
cleanup(cleanupTrackOnly: boolean) {
@@ -265,6 +272,10 @@ class WebRTCTrack implements RTCMediaObjectTrack {
setPlayback(options: { audio: boolean; video: boolean; }): Promise<void> {
return this.control.setPlayback(options);
}
async setOptions(options: RTCMediaObjectTrackOptions): Promise<void> {
console.log('updating destination');
}
}
export class WebRTCConnectionManagement implements RTCConnectionManagement {
@@ -303,7 +314,7 @@ export class WebRTCConnectionManagement implements RTCConnectionManagement {
this.weriftSignalingSession = new WeriftSignalingSession(console, this.pc);
}
async createTracks(mediaObject: MediaObject) {
async createTracks(mediaObject: MediaObject, options?: RTCMediaObjectTrackOptions) {
let requestMediaStream: RequestMediaStream;
try {
@@ -313,6 +324,17 @@ export class WebRTCConnectionManagement implements RTCConnectionManagement {
requestMediaStream = async () => mediaObject;
}
if (options?.destination) {
const { destination } = options;
const wrapped = requestMediaStream;
requestMediaStream = async requestOptions => {
return wrapped({
...requestOptions,
destination,
});
}
}
let intercom: Intercom;
try {
const device = await sdk.mediaManager.convertMediaObject<ScryptedDevice & Intercom>(mediaObject, ScryptedMimeTypes.ScryptedDevice);
@@ -381,11 +403,11 @@ export class WebRTCConnectionManagement implements RTCConnectionManagement {
}
}
async addTrack(mediaObject: MediaObject, options?: {
async addTrack(mediaObject: MediaObject, options?: RTCMediaObjectTrackOptions & {
videoMid?: string,
audioMid?: string,
}) {
const { atrack, vtrack, createTrackForwarder, intercom } = await this.createTracks(mediaObject);
const { atrack, vtrack, createTrackForwarder, intercom } = await this.createTracks(mediaObject, options);
const videoTransceiver = this.pc.addTransceiver(vtrack, {
direction: 'sendonly',

10
plugins/webrtc/src/main.ts
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@@ -23,11 +23,6 @@ defaultPeerConfig.headerExtensions = {
audio: [],
};
const supportedTypes = [
ScryptedDeviceType.Camera,
ScryptedDeviceType.Doorbell,
];
mediaManager.addConverter({
fromMimeType: ScryptedMimeTypes.ScryptedDevice,
toMimeType: ScryptedMimeTypes.RequestMediaStream,
@@ -252,7 +247,10 @@ export class WebRTCPlugin extends AutoenableMixinProvider implements DeviceCreat
return ret;
}
else if (supportedTypes.includes(type) && interfaces.includes(ScryptedInterface.VideoCamera)) {
else if ([
ScryptedDeviceType.Camera,
ScryptedDeviceType.Doorbell,
].includes(type) && interfaces.includes(ScryptedInterface.VideoCamera)) {
return [
ScryptedInterface.RTCSignalingChannel,
// ScryptedInterface.Settings,