homebridge-ring 11.0.1 → 11.0.2

package/CHANGELOG.md

@@ -2,6 +2,17 @@
 
 All notable changes to this project will be documented in this file. See [standard-version](https://github.com/conventional-changelog/standard-version) for commit guidelines.
 
+### [11.0.2](https://github.com/dgreif/ring/compare/v11.0.1...v11.0.2) (2022-05-22)
+
+
+### Bug Fixes
+
+* **api:** prevent duplicate events from `onCallEnded` ([14ee6eb](https://github.com/dgreif/ring/commit/14ee6eb90bb6ae568dcfd9e8e176c3001e422f9c)), closes [#941](https://github.com/dgreif/ring/issues/941)
+* handle pcmu audio streams for incoming audio ([06f6ff1](https://github.com/dgreif/ring/commit/06f6ff10ce7c8f2b05b1a12b8f244daa200fa9ab)), closes [#930](https://github.com/dgreif/ring/issues/930)
+* handle pcmu return audio ([7b86182](https://github.com/dgreif/ring/commit/7b8618261a6ccb85b9f760ee1fcc228be00fe5a6)), closes [#930](https://github.com/dgreif/ring/issues/930)
+* **homebride:** use shared engines config to enforce node 16 requirement ([b748cb0](https://github.com/dgreif/ring/commit/b748cb027db0c43ae109e1ea77c6bc83f86e4913))
+* prevent hanging promises from streams that fail to connect ([dd7638e](https://github.com/dgreif/ring/commit/dd7638e32626499c0be501be6b19b5d874fa974f))
+
 ### [11.0.1](https://github.com/dgreif/ring/compare/v11.0.0...v11.0.1) (2022-05-17)
 
 

package/lib/api/streaming/peer-connection.js

@@ -31,6 +31,12 @@ class WeriftPeerConnection {
                         clockRate: 48000,
                         channels: 2,
                     }),
+                    new werift_1.RTCRtpCodecParameters({
+                        mimeType: 'audio/PCMU',
+                        clockRate: 8000,
+                        channels: 1,
+                        payloadType: 0,
+                    }),
                 ],
                 video: [
                     new werift_1.RTCRtpCodecParameters({

package/lib/api/streaming/streaming-connection-base.d.ts

@@ -21,10 +21,11 @@ export declare abstract class StreamingConnectionBase extends Subscribed {
     }): Promise<void>;
     protected abstract sendSessionMessage(method: string, body?: Record<any, any>): void;
     private activated;
-    activate(): Promise<void>;
-    activateCameraSpeaker(): Promise<void>;
+    activate(): void;
+    activateCameraSpeaker(): void;
     protected sendMessage(message: Record<any, any>): void;
     sendAudioPacket(rtp: RtpPacket): void;
+    private hasEnded;
     protected callEnded(): void;
     stop(): void;
 }

package/lib/api/streaming/streaming-connection-base.js

@@ -14,6 +14,7 @@ class StreamingConnectionBase extends subscribed_1.Subscribed {
         this.onCallAnswered = new rxjs_1.ReplaySubject(1);
         this.onCallEnded = new rxjs_1.ReplaySubject(1);
         this.activated = false;
+        this.hasEnded = false;
         if (options.createPeerConnection) {
             // we were passed a custom peer connection factory
             this.pc = options.createPeerConnection();
@@ -50,25 +51,43 @@ class StreamingConnectionBase extends subscribed_1.Subscribed {
             }
         }));
     }
-    async activate() {
+    activate() {
         if (this.activated) {
             return;
         }
         this.activated = true;
-        await (0, rxjs_1.firstValueFrom)(this.pc.onConnectionState.pipe((0, operators_1.filter)((state) => state === 'connected')));
-        (0, util_1.logInfo)('Activating Session');
-        this.sendSessionMessage('activate_session');
+        // Fire and forget this call so that callers don't get hung up waiting for connection (which might not happen)
+        (0, rxjs_1.firstValueFrom)(this.pc.onConnectionState.pipe((0, operators_1.filter)((state) => state === 'connected')))
+            .then(() => {
+            (0, util_1.logInfo)('Activating Session');
+            this.sendSessionMessage('activate_session');
+        })
+            .catch((e) => {
+            (0, util_1.logError)(e);
+        });
     }
-    async activateCameraSpeaker() {
-        await (0, rxjs_1.firstValueFrom)(this.onCallAnswered);
-        this.sendSessionMessage('camera_options', {
-            stealth_mode: false,
+    activateCameraSpeaker() {
+        // Fire and forget this call so that callers don't get hung up waiting for answer (which might not happen)
+        (0, rxjs_1.firstValueFrom)(this.onCallAnswered)
+            .then(() => {
+            this.sendSessionMessage('camera_options', {
+                stealth_mode: false,
+            });
+        })
+            .catch((e) => {
+            (0, util_1.logError)(e);
         });
     }
     sendMessage(message) {
+        if (this.hasEnded) {
+            return;
+        }
         this.ws.send(JSON.stringify(message));
     }
     sendAudioPacket(rtp) {
+        if (this.hasEnded) {
+            return;
+        }
         if (this.pc instanceof peer_connection_1.WeriftPeerConnection) {
             this.pc.returnAudioTrack.writeRtp(rtp);
         }
@@ -77,6 +96,9 @@ class StreamingConnectionBase extends subscribed_1.Subscribed {
         }
     }
     callEnded() {
+        if (this.hasEnded) {
+            return;
+        }
         try {
             this.sendMessage({
                 reason: { code: 0, text: '' },
@@ -87,6 +109,7 @@ class StreamingConnectionBase extends subscribed_1.Subscribed {
         catch (_) {
             // ignore any errors since we are stopping the call
         }
+        this.hasEnded = true;
         this.unsubscribe();
         this.onCallEnded.next();
         this.pc.close();

package/lib/api/streaming/streaming-session.d.ts

@@ -15,18 +15,25 @@ export declare class StreamingSession extends Subscribed {
     private readonly camera;
     private connection;
     readonly onCallEnded: ReplaySubject<void>;
+    private readonly onUsingOpus;
     readonly onVideoRtp: Subject<RtpPacket>;
     readonly onAudioRtp: Subject<RtpPacket>;
     private readonly audioSplitter;
     private readonly videoSplitter;
+    private readonly returnAudioSplitter;
     constructor(camera: RingCamera, connection: RingEdgeConnection | WebrtcConnection);
     private bindToConnection;
     activated: boolean;
-    activate(): Promise<void>;
+    activate(): void;
     cameraSpeakerActivated: boolean;
-    activateCameraSpeaker(): Promise<void>;
+    activateCameraSpeaker(): void;
     reservePort(bufferPorts?: number): Promise<number>;
+    get isUsingOpus(): Promise<boolean>;
     startTranscoding(ffmpegOptions: FfmpegOptions): Promise<void>;
+    transcodeReturnAudio(ffmpegOptions: {
+        input: SpawnInput[];
+    }): Promise<void>;
+    private hasEnded;
     private callEnded;
     stop(): void;
     sendAudioPacket(rtp: RtpPacket): void;

package/lib/api/streaming/streaming-session.js

@@ -1,6 +1,7 @@
 "use strict";
 Object.defineProperty(exports, "__esModule", { value: true });
 exports.StreamingSession = void 0;
+const werift_1 = require("werift");
 const camera_utils_1 = require("@homebridge/camera-utils");
 const rxjs_1 = require("rxjs");
 const ffmpeg_1 = require("../ffmpeg");
@@ -21,48 +22,59 @@ class StreamingSession extends subscribed_1.Subscribed {
         this.camera = camera;
         this.connection = connection;
         this.onCallEnded = new rxjs_1.ReplaySubject(1);
+        this.onUsingOpus = new rxjs_1.ReplaySubject(1);
         this.onVideoRtp = new rxjs_1.Subject();
         this.onAudioRtp = new rxjs_1.Subject();
         this.audioSplitter = new camera_utils_1.RtpSplitter();
         this.videoSplitter = new camera_utils_1.RtpSplitter();
+        this.returnAudioSplitter = new camera_utils_1.RtpSplitter();
         this.activated = false;
         this.cameraSpeakerActivated = false;
+        this.hasEnded = false;
         this.bindToConnection(connection);
     }
     bindToConnection(connection) {
-        this.addSubscriptions(connection.onAudioRtp.subscribe(this.onAudioRtp), connection.onVideoRtp.subscribe(this.onVideoRtp));
+        this.addSubscriptions(connection.onAudioRtp.subscribe(this.onAudioRtp), connection.onVideoRtp.subscribe(this.onVideoRtp), connection.onCallAnswered.subscribe((sdp) => {
+            this.onUsingOpus.next(sdp.toLocaleLowerCase().includes(' opus/'));
+        }));
     }
-    async activate() {
-        if (this.activated) {
+    activate() {
+        if (this.activated || this.hasEnded) {
             return;
         }
         this.activated = true;
-        await this.connection.activate();
+        this.connection.activate();
     }
-    async activateCameraSpeaker() {
-        if (this.cameraSpeakerActivated) {
+    activateCameraSpeaker() {
+        if (this.cameraSpeakerActivated || this.hasEnded) {
             return;
         }
         this.cameraSpeakerActivated = true;
-        await this.connection.activateCameraSpeaker();
+        this.connection.activateCameraSpeaker();
     }
     async reservePort(bufferPorts = 0) {
         const ports = await (0, camera_utils_1.reservePorts)({ count: bufferPorts + 1 });
         return ports[0];
     }
+    get isUsingOpus() {
+        return (0, rxjs_1.firstValueFrom)(this.onUsingOpus);
+    }
     async startTranscoding(ffmpegOptions) {
-        const videoPort = await this.reservePort(1), audioPort = await this.reservePort(1), transcodeVideoStream = ffmpegOptions.video !== false, ffmpegInputArguments = [
+        if (this.hasEnded) {
+            return;
+        }
+        const videoPort = await this.reservePort(1), audioPort = await this.reservePort(1), transcodeVideoStream = ffmpegOptions.video !== false, ringSdp = await (0, rxjs_1.firstValueFrom)(this.connection.onCallAnswered), usingOpus = await this.isUsingOpus, ffmpegInputArguments = [
             '-hide_banner',
             '-protocol_whitelist',
             'pipe,udp,rtp,file,crypto',
-            '-acodec',
-            'libopus',
+            // Ring will answer with either opus or pcmu
+            ...(usingOpus ? ['-acodec', 'libopus'] : []),
             '-f',
             'sdp',
             ...(ffmpegOptions.input || []),
             '-i',
             'pipe:',
-        ], ringSdp = await (0, rxjs_1.firstValueFrom)(this.connection.onCallAnswered), inputSdp = getCleanSdp(ringSdp, transcodeVideoStream)
+        ], inputSdp = getCleanSdp(ringSdp, transcodeVideoStream)
             .replace(/m=audio \d+/, `m=audio ${audioPort}`)
             .replace(/m=video \d+/, `m=video ${videoPort}`), ff = new camera_utils_1.FfmpegProcess({
             ffmpegArgs: ffmpegInputArguments.concat(...(ffmpegOptions.audio || ['-acodec', 'aac']), ...(transcodeVideoStream
@@ -95,19 +107,63 @@ class StreamingSession extends subscribed_1.Subscribed {
         this.onCallEnded.subscribe(() => ff.stop());
         ff.writeStdin(inputSdp);
         // Activate the stream now that ffmpeg is ready to receive
-        await this.activate();
+        this.activate();
+    }
+    async transcodeReturnAudio(ffmpegOptions) {
+        if (this.hasEnded) {
+            return;
+        }
+        const audioOutForwarder = new camera_utils_1.RtpSplitter(({ message }) => {
+            const rtp = werift_1.RtpPacket.deSerialize(message);
+            this.connection.sendAudioPacket(rtp);
+            return null;
+        }), usingOpus = await this.isUsingOpus, ff = new camera_utils_1.FfmpegProcess({
+            ffmpegArgs: [
+                '-hide_banner',
+                '-protocol_whitelist',
+                'pipe,udp,rtp,file,crypto',
+                '-re',
+                '-i',
+                ...ffmpegOptions.input,
+                '-acodec',
+                ...(usingOpus
+                    ? ['libopus', '-ac', 2, '-ar', '48k']
+                    : ['pcm_mulaw', '-ac', 1, '-ar', '8k']),
+                '-flags',
+                '+global_header',
+                '-f',
+                'rtp',
+                `rtp://127.0.0.1:${await audioOutForwarder.portPromise}`,
+            ],
+            ffmpegPath: (0, ffmpeg_1.getFfmpegPath)(),
+            exitCallback: () => this.callEnded(),
+            logLabel: `Return Audio (${this.camera.name})`,
+            logger: {
+                error: util_1.logError,
+                info: util_1.logDebug,
+            },
+        });
+        this.onCallEnded.subscribe(() => ff.stop());
     }
     callEnded() {
+        if (this.hasEnded) {
+            return;
+        }
+        this.hasEnded = true;
         this.unsubscribe();
         this.onCallEnded.next();
         this.connection.stop();
         this.audioSplitter.close();
         this.videoSplitter.close();
+        this.returnAudioSplitter.close();
     }
     stop() {
         this.callEnded();
     }
     sendAudioPacket(rtp) {
+        if (this.hasEnded) {
+            return;
+        }
         this.connection.sendAudioPacket(rtp);
     }
 }

package/lib/homebridge/camera-source.js

@@ -26,8 +26,8 @@ function getSessionConfig(srtpOptions) {
     };
 }
 class StreamingSessionWrapper {
-    constructor(liveCall, prepareStreamRequest, ringCamera, start) {
-        this.liveCall = liveCall;
+    constructor(streamingSession, prepareStreamRequest, ringCamera, start) {
+        this.streamingSession = streamingSession;
         this.prepareStreamRequest = prepareStreamRequest;
         this.ringCamera = ringCamera;
         this.start = start;
@@ -63,15 +63,15 @@ class StreamingSessionWrapper {
             onReturnPacketReceived.next(null);
             return null;
         });
-        liveCall.addSubscriptions((0, rxjs_1.merge)((0, rxjs_1.of)(true).pipe((0, operators_1.delay)(15000)), onReturnPacketReceived)
+        streamingSession.addSubscriptions((0, rxjs_1.merge)((0, rxjs_1.of)(true).pipe((0, operators_1.delay)(15000)), onReturnPacketReceived)
             .pipe((0, operators_1.debounceTime)(5000))
             .subscribe(() => {
             (0, util_1.logInfo)(`Live stream for ${this.ringCamera.name} appears to be inactive. (${getDurationSeconds(start)}s)`);
-            liveCall.stop();
+            streamingSession.stop();
         }));
         // Periodically send a blank RTCP packet to the HomeKit video port
         // Without this, HomeKit assumes the stream is dead after 30 second and sends a stop request
-        liveCall.addSubscriptions((0, rxjs_1.interval)(500).subscribe(() => {
+        streamingSession.addSubscriptions((0, rxjs_1.interval)(500).subscribe(() => {
             const senderInfo = new werift_1.RtcpSenderInfo({
                 ntpTimestamp: BigInt(0),
                 packetCount: 0,
@@ -95,7 +95,7 @@ class StreamingSessionWrapper {
         // use to encrypt Ring video to HomeKit
         videoSrtpSession = new werift_1.SrtpSession(getSessionConfig(this.videoSrtp));
         // Set up packet forwarding for video stream
-        this.liveCall.addSubscriptions(this.liveCall.onVideoRtp.subscribe(({ header, payload }) => {
+        this.streamingSession.addSubscriptions(this.streamingSession.onVideoRtp.subscribe(({ header, payload }) => {
             header.ssrc = this.videoSsrc;
             header.payloadType = request.video.pt;
             const encryptedPacket = videoSrtpSession.encrypt(payload, header);
@@ -112,9 +112,9 @@ class StreamingSessionWrapper {
         }));
         const shouldTranscodeAudio = await this.libfdkAacInstalledPromise;
         if (!shouldTranscodeAudio) {
-            return this.liveCall.activate();
+            return this.streamingSession.activate();
         }
-        const transcodingPromise = this.liveCall.startTranscoding({
+        const transcodingPromise = this.streamingSession.startTranscoding({
             input: ['-vn'],
             audio: [
                 '-map',
@@ -167,20 +167,20 @@ class StreamingSessionWrapper {
         }, audioSrtpSession = new werift_1.SrtpSession(getSessionConfig(remoteAudioSrtpOptions)), returnAudioTranscodedSplitter = new camera_utils_1.RtpSplitter(({ message }) => {
             if (!cameraSpeakerActive) {
                 cameraSpeakerActive = true;
-                this.liveCall.activateCameraSpeaker().catch(util_1.logError);
+                this.streamingSession.activateCameraSpeaker();
             }
             // decrypt the message
             try {
                 const rtp = werift_1.RtpPacket.deSerialize(message);
                 rtp.payload = audioSrtpSession.decrypt(rtp.payload);
                 // send to Ring - werift will handle encryption and other header params
-                this.liveCall.sendAudioPacket(rtp);
+                this.streamingSession.sendAudioPacket(rtp);
             }
             catch (_) {
                 // deSerialize will sometimes fail, but the errors can be ignored
             }
             return null;
-        }), returnAudioTranscoder = new camera_utils_1.ReturnAudioTranscoder({
+        }), usingOpus = await this.streamingSession.isUsingOpus, returnAudioTranscoder = new camera_utils_1.ReturnAudioTranscoder({
             prepareStreamRequest: this.prepareStreamRequest,
             incomingAudioOptions: {
                 ssrc: this.audioSsrc,
@@ -188,13 +188,11 @@ class StreamingSessionWrapper {
             },
             outputArgs: [
                 '-acodec',
-                'libopus',
+                ...(usingOpus
+                    ? ['libopus', '-ac', 2, '-ar', '48k']
+                    : ['pcm_mulaw', '-ac', 1, '-ar', '8k']),
                 '-flags',
                 '+global_header',
-                '-ac',
-                2,
-                '-ar',
-                '48k',
                 '-f',
                 'rtp',
                 `rtp://127.0.0.1:${await returnAudioTranscodedSplitter.portPromise}`,
@@ -207,7 +205,7 @@ class StreamingSessionWrapper {
             logLabel: `Return Audio (${this.ringCamera.name})`,
             returnAudioSplitter: this.audioSplitter,
         });
-        this.liveCall.onCallEnded.pipe((0, operators_1.take)(1)).subscribe(() => {
+        this.streamingSession.onCallEnded.pipe((0, operators_1.take)(1)).subscribe(() => {
             returnAudioTranscoder.stop();
             returnAudioTranscodedSplitter.close();
         });
@@ -217,7 +215,7 @@ class StreamingSessionWrapper {
     stop() {
         this.audioSplitter.close();
         this.videoSplitter.close();
-        this.liveCall.stop();
+        this.streamingSession.stop();
     }
 }
 class CameraSource {

package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "homebridge-ring",
-  "version": "11.0.1",
+  "version": "11.0.2",
   "description": "Homebridge plugin for Ring doorbells, cameras, security alarm system and smart lighting",
   "main": "lib/homebridge/index.js",
   "bin": {
@@ -42,7 +42,7 @@
     "@eneris/push-receiver": "^3.0.1",
     "@homebridge/camera-utils": "^2.1.0",
     "@homebridge/plugin-ui-utils": "^0.0.19",
-    "werift": "^0.15.0",
+    "werift": "^0.15.1",
     "@types/socket.io-client": "1.4.36",
     "colors": "^1.4.0",
     "debug": "^4.3.4",
@@ -81,8 +81,8 @@
     "typescript": "4.6.4"
   },
   "engines": {
-    "node": ">=8.3.0",
-    "homebridge": ">=0.4.0"
+    "node": "^16",
+    "homebridge": ">=1.4.0"
   },
   "bugs": {
     "url": "https://github.com/dgreif/ring/issues"