Support voice messages on Safari

__mocks__/empty.js

@@ -0,0 +1,2 @@
+// Yes, this is empty.
+module.exports = {};

package.json

@@ -185,7 +185,10 @@
     ],
     "moduleNameMapper": {
       "\\.(gif|png|svg|ttf|woff2)$": "<rootDir>/__mocks__/imageMock.js",
-      "\\$webapp/i18n/languages.json": "<rootDir>/__mocks__/languages.json"
+      "\\$webapp/i18n/languages.json": "<rootDir>/__mocks__/languages.json",
+      "decoderWorker\\.min\\.js": "<rootDir>/__mocks__/empty.js",
+      "decoderWorker\\.min\\.wasm": "<rootDir>/__mocks__/empty.js",
+      "waveWorker\\.min\\.js": "<rootDir>/__mocks__/empty.js"
     },
     "transformIgnorePatterns": [
       "/node_modules/(?!matrix-js-sdk).+$"

src/@types/global.d.ts

@@ -52,6 +52,9 @@ declare global {
             init: () => Promise<void>;
         };
 
+        // Needed for Safari, unknown to TypeScript
+        webkitAudioContext: typeof AudioContext;
+
         mxContentMessages: ContentMessages;
         mxToastStore: ToastStore;
         mxDeviceListener: DeviceListener;

src/rageshake/rageshake.js

@@ -73,7 +73,9 @@ class ConsoleLogger {
 
         // Convert objects and errors to helpful things
         args = args.map((arg) => {
-            if (arg instanceof Error) {
+            if (arg instanceof DOMException) {
+                return arg.message + ` (${arg.name} | ${arg.code}) ` + (arg.stack ? `\n${arg.stack}` : '');
+            } else if (arg instanceof Error) {
                 return arg.message + (arg.stack ? `\n${arg.stack}` : '');
             } else if (typeof (arg) === 'object') {
                 try {

src/voice/Playback.ts

@@ -21,6 +21,7 @@ import {SimpleObservable} from "matrix-widget-api";
 import {IDestroyable} from "../utils/IDestroyable";
 import {PlaybackClock} from "./PlaybackClock";
 import {clamp} from "../utils/numbers";
+import {createAudioContext, decodeOgg} from "./compat";
 
 export enum PlaybackState {
     Decoding = "decoding",
@@ -49,7 +50,7 @@ export class Playback extends EventEmitter implements IDestroyable {
      */
     constructor(private buf: ArrayBuffer, seedWaveform = DEFAULT_WAVEFORM) {
         super();
-        this.context = new AudioContext();
+        this.context = createAudioContext();
         this.resampledWaveform = arrayFastResample(seedWaveform ?? DEFAULT_WAVEFORM, PLAYBACK_WAVEFORM_SAMPLES);
         this.waveformObservable.update(this.resampledWaveform);
         this.clock = new PlaybackClock(this.context);
@@ -91,7 +92,23 @@ export class Playback extends EventEmitter implements IDestroyable {
     }
 
     public async prepare() {
-        this.audioBuf = await this.context.decodeAudioData(this.buf);
+        // Safari compat: promise API not supported on this function
+        this.audioBuf = await new Promise((resolve, reject) => {
+            this.context.decodeAudioData(this.buf, b => resolve(b), async e => {
+                // This error handler is largely for Safari as well, which doesn't support Opus/Ogg
+                // very well.
+                console.error("Error decoding recording: ", e);
+                console.warn("Trying to re-encode to WAV instead...");
+
+                const wav = await decodeOgg(this.buf);
+
+                // noinspection ES6MissingAwait - not needed when using callbacks
+                this.context.decodeAudioData(wav, b => resolve(b), e => {
+                    console.error("Still failed to decode recording: ", e);
+                    reject(e);
+                });
+            });
+        });
 
         // Update the waveform to the real waveform once we have channel data to use. We don't
         // exactly trust the user-provided waveform to be accurate...

src/voice/VoiceRecording.ts

@@ -19,16 +19,17 @@ import encoderPath from 'opus-recorder/dist/encoderWorker.min.js';
 import {MatrixClient} from "matrix-js-sdk/src/client";
 import CallMediaHandler from "../CallMediaHandler";
 import {SimpleObservable} from "matrix-widget-api";
-import {clamp} from "../utils/numbers";
+import {clamp, percentageOf, percentageWithin} from "../utils/numbers";
 import EventEmitter from "events";
 import {IDestroyable} from "../utils/IDestroyable";
 import {Singleflight} from "../utils/Singleflight";
 import {PayloadEvent, WORKLET_NAME} from "./consts";
 import {UPDATE_EVENT} from "../stores/AsyncStore";
 import {Playback} from "./Playback";
+import {createAudioContext} from "./compat";
 
 const CHANNELS = 1; // stereo isn't important
-const SAMPLE_RATE = 48000; // 48khz is what WebRTC uses. 12khz is where we lose quality.
+export const SAMPLE_RATE = 48000; // 48khz is what WebRTC uses. 12khz is where we lose quality.
 const BITRATE = 24000; // 24kbps is pretty high quality for our use case in opus.
 const TARGET_MAX_LENGTH = 120; // 2 minutes in seconds. Somewhat arbitrary, though longer == larger files.
 const TARGET_WARN_TIME_LEFT = 10; // 10 seconds, also somewhat arbitrary.
@@ -55,6 +56,7 @@ export class VoiceRecording extends EventEmitter implements IDestroyable {
     private recorderStream: MediaStream;
     private recorderFFT: AnalyserNode;
     private recorderWorklet: AudioWorkletNode;
+    private recorderProcessor: ScriptProcessorNode;
     private buffer = new Uint8Array(0); // use this.audioBuffer to access
     private mxc: string;
     private recording = false;
@@ -90,78 +92,107 @@ export class VoiceRecording extends EventEmitter implements IDestroyable {
     }
 
     private async makeRecorder() {
-        this.recorderStream = await navigator.mediaDevices.getUserMedia({
-            audio: {
-                channelCount: CHANNELS,
-                noiseSuppression: true, // browsers ignore constraints they can't honour
-                deviceId: CallMediaHandler.getAudioInput(),
-            },
-        });
-        this.recorderContext = new AudioContext({
-            // latencyHint: "interactive", // we don't want a latency hint (this causes data smoothing)
-        });
-        this.recorderSource = this.recorderContext.createMediaStreamSource(this.recorderStream);
-        this.recorderFFT = this.recorderContext.createAnalyser();
-
-        // Bring the FFT time domain down a bit. The default is 2048, and this must be a power
-        // of two. We use 64 points because we happen to know down the line we need less than
-        // that, but 32 would be too few. Large numbers are not helpful here and do not add
-        // precision: they introduce higher precision outputs of the FFT (frequency data), but
-        // it makes the time domain less than helpful.
-        this.recorderFFT.fftSize = 64;
-
-        // Set up our worklet. We use this for timing information and waveform analysis: the
-        // web audio API prefers this be done async to avoid holding the main thread with math.
-        const mxRecorderWorkletPath = document.body.dataset.vectorRecorderWorkletScript;
-        if (!mxRecorderWorkletPath) {
-            throw new Error("Unable to create recorder: no worklet script registered");
-        }
-        await this.recorderContext.audioWorklet.addModule(mxRecorderWorkletPath);
-        this.recorderWorklet = new AudioWorkletNode(this.recorderContext, WORKLET_NAME);
-
-        // Connect our inputs and outputs
-        this.recorderSource.connect(this.recorderFFT);
-        this.recorderSource.connect(this.recorderWorklet);
-        this.recorderWorklet.connect(this.recorderContext.destination);
-
-        // Dev note: we can't use `addEventListener` for some reason. It just doesn't work.
-        this.recorderWorklet.port.onmessage = (ev) => {
-            switch (ev.data['ev']) {
-                case PayloadEvent.Timekeep:
-                    this.processAudioUpdate(ev.data['timeSeconds']);
-                    break;
-                case PayloadEvent.AmplitudeMark:
-                    // Sanity check to make sure we're adding about one sample per second
-                    if (ev.data['forSecond'] === this.amplitudes.length) {
-                        this.amplitudes.push(ev.data['amplitude']);
+        try {
+            this.recorderStream = await navigator.mediaDevices.getUserMedia({
+                audio: {
+                    channelCount: CHANNELS,
+                    noiseSuppression: true, // browsers ignore constraints they can't honour
+                    deviceId: CallMediaHandler.getAudioInput(),
+                },
+            });
+            this.recorderContext = createAudioContext({
+                // latencyHint: "interactive", // we don't want a latency hint (this causes data smoothing)
+            });
+            this.recorderSource = this.recorderContext.createMediaStreamSource(this.recorderStream);
+            this.recorderFFT = this.recorderContext.createAnalyser();
+
+            // Bring the FFT time domain down a bit. The default is 2048, and this must be a power
+            // of two. We use 64 points because we happen to know down the line we need less than
+            // that, but 32 would be too few. Large numbers are not helpful here and do not add
+            // precision: they introduce higher precision outputs of the FFT (frequency data), but
+            // it makes the time domain less than helpful.
+            this.recorderFFT.fftSize = 64;
+
+            // Set up our worklet. We use this for timing information and waveform analysis: the
+            // web audio API prefers this be done async to avoid holding the main thread with math.
+            const mxRecorderWorkletPath = document.body.dataset.vectorRecorderWorkletScript;
+            if (!mxRecorderWorkletPath) {
+                // noinspection ExceptionCaughtLocallyJS
+                throw new Error("Unable to create recorder: no worklet script registered");
+            }
+
+            // Connect our inputs and outputs
+            this.recorderSource.connect(this.recorderFFT);
+
+            if (this.recorderContext.audioWorklet) {
+                await this.recorderContext.audioWorklet.addModule(mxRecorderWorkletPath);
+                this.recorderWorklet = new AudioWorkletNode(this.recorderContext, WORKLET_NAME);
+                this.recorderSource.connect(this.recorderWorklet);
+                this.recorderWorklet.connect(this.recorderContext.destination);
+
+                // Dev note: we can't use `addEventListener` for some reason. It just doesn't work.
+                this.recorderWorklet.port.onmessage = (ev) => {
+                    switch (ev.data['ev']) {
+                        case PayloadEvent.Timekeep:
+                            this.processAudioUpdate(ev.data['timeSeconds']);
+                            break;
+                        case PayloadEvent.AmplitudeMark:
+                            // Sanity check to make sure we're adding about one sample per second
+                            if (ev.data['forSecond'] === this.amplitudes.length) {
+                                this.amplitudes.push(ev.data['amplitude']);
+                            }
+                            break;
                     }
-                    break;
+                };
+            } else {
+                // Safari fallback: use a processor node instead, buffered to 1024 bytes of data
+                // like the worklet is.
+                this.recorderProcessor = this.recorderContext.createScriptProcessor(1024, CHANNELS, CHANNELS);
+                this.recorderSource.connect(this.recorderProcessor);
+                this.recorderProcessor.connect(this.recorderContext.destination);
+                this.recorderProcessor.addEventListener("audioprocess", this.onAudioProcess);
             }
-        };
-
-        this.recorder = new Recorder({
-            encoderPath, // magic from webpack
-            encoderSampleRate: SAMPLE_RATE,
-            encoderApplication: 2048, // voice (default is "audio")
-            streamPages: true, // this speeds up the encoding process by using CPU over time
-            encoderFrameSize: 20, // ms, arbitrary frame size we send to the encoder
-            numberOfChannels: CHANNELS,
-            sourceNode: this.recorderSource,
-            encoderBitRate: BITRATE,
-
-            // We use low values for the following to ease CPU usage - the resulting waveform
-            // is indistinguishable for a voice message. Note that the underlying library will
-            // pick defaults which prefer the highest possible quality, CPU be damned.
-            encoderComplexity: 3, // 0-10, 10 is slow and high quality.
-            resampleQuality: 3, // 0-10, 10 is slow and high quality
-        });
-        this.recorder.ondataavailable = (a: ArrayBuffer) => {
-            const buf = new Uint8Array(a);
-            const newBuf = new Uint8Array(this.buffer.length + buf.length);
-            newBuf.set(this.buffer, 0);
-            newBuf.set(buf, this.buffer.length);
-            this.buffer = newBuf;
-        };
+
+            this.recorder = new Recorder({
+                encoderPath, // magic from webpack
+                encoderSampleRate: SAMPLE_RATE,
+                encoderApplication: 2048, // voice (default is "audio")
+                streamPages: true, // this speeds up the encoding process by using CPU over time
+                encoderFrameSize: 20, // ms, arbitrary frame size we send to the encoder
+                numberOfChannels: CHANNELS,
+                sourceNode: this.recorderSource,
+                encoderBitRate: BITRATE,
+
+                // We use low values for the following to ease CPU usage - the resulting waveform
+                // is indistinguishable for a voice message. Note that the underlying library will
+                // pick defaults which prefer the highest possible quality, CPU be damned.
+                encoderComplexity: 3, // 0-10, 10 is slow and high quality.
+                resampleQuality: 3, // 0-10, 10 is slow and high quality
+            });
+            this.recorder.ondataavailable = (a: ArrayBuffer) => {
+                const buf = new Uint8Array(a);
+                const newBuf = new Uint8Array(this.buffer.length + buf.length);
+                newBuf.set(this.buffer, 0);
+                newBuf.set(buf, this.buffer.length);
+                this.buffer = newBuf;
+            };
+        } catch (e) {
+            console.error("Error starting recording: ", e);
+            if (e instanceof DOMException) { // Unhelpful DOMExceptions are common - parse them sanely
+                console.error(`${e.name} (${e.code}): ${e.message}`);
+            }
+
+            // Clean up as best as possible
+            if (this.recorderStream) this.recorderStream.getTracks().forEach(t => t.stop());
+            if (this.recorderSource) this.recorderSource.disconnect();
+            if (this.recorder) this.recorder.close();
+            if (this.recorderContext) {
+                // noinspection ES6MissingAwait - not important that we wait
+                this.recorderContext.close();
+            }
+
+            throw e; // rethrow so upstream can handle it
+        }
     }
 
     private get audioBuffer(): Uint8Array {
@@ -190,14 +221,30 @@ export class VoiceRecording extends EventEmitter implements IDestroyable {
         return this.mxc;
     }
 
+    private onAudioProcess = (ev: AudioProcessingEvent) => {
+        this.processAudioUpdate(ev.playbackTime);
+
+        // We skip the functionality of the worklet regarding waveform calculations: we
+        // should get that information pretty quick during the playback info.
+    };
+
     private processAudioUpdate = (timeSeconds: number) => {
         if (!this.recording) return;
 
         // The time domain is the input to the FFT, which means we use an array of the same
         // size. The time domain is also known as the audio waveform. We're ignoring the
         // output of the FFT here (frequency data) because we're not interested in it.
         const data = new Float32Array(this.recorderFFT.fftSize);
-        this.recorderFFT.getFloatTimeDomainData(data);
+        if (!this.recorderFFT.getFloatTimeDomainData) {
+            // Safari compat
+            const data2 = new Uint8Array(this.recorderFFT.fftSize);
+            this.recorderFFT.getByteTimeDomainData(data2);
+            for (let i = 0; i < data2.length; i++) {
+                data[i] = percentageWithin(percentageOf(data2[i], 0, 256), -1, 1);
+            }
+        } else {
+            this.recorderFFT.getFloatTimeDomainData(data);
+        }
 
         // We can't just `Array.from()` the array because we're dealing with 32bit floats
         // and the built-in function won't consider that when converting between numbers.
@@ -268,7 +315,11 @@ export class VoiceRecording extends EventEmitter implements IDestroyable {
             // Disconnect the source early to start shutting down resources
             await this.recorder.stop(); // stop first to flush the last frame
             this.recorderSource.disconnect();
-            this.recorderWorklet.disconnect();
+            if (this.recorderWorklet) this.recorderWorklet.disconnect();
+            if (this.recorderProcessor) {
+                this.recorderProcessor.disconnect();
+                this.recorderProcessor.removeEventListener("audioprocess", this.onAudioProcess);
+            }
 
             // close the context after the recorder so the recorder doesn't try to
             // connect anything to the context (this would generate a warning)