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Merge pull request #5989 from matrix-org/travis/voicemessages/safari 

Support voice messages on Safari
pull/21833/head
Travis Ralston 2021-05-11 10:12:31 -06:00 committed by GitHub
parent 03bb3cc951 65f591b69b
commit d1cfde6f12
No known key found for this signature in database
GPG Key ID: 4AEE18F83AFDEB23
7 changed files with 235 additions and 75 deletions
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2
__mocks__/empty.js Normal file
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@ -0,0 +1,2 @@
// Yes, this is empty.
module.exports = {};

5
package.json
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@ -186,7 +186,10 @@
],
"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).+$"

3
src/@types/global.d.ts vendored
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@ -52,6 +52,9 @@ declare global {
init: () => Promise<void>;
};
// Needed for Safari, unknown to TypeScript
webkitAudioContext: typeof AudioContext;
mxContentMessages: ContentMessages;
mxToastStore: ToastStore;
mxDeviceListener: DeviceListener;

4
src/rageshake/rageshake.js
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@ -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 {

21
src/voice/Playback.ts
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@ -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...

193
src/voice/VoiceRecording.ts
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@ -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();
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;
// 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']);
}
break;
// 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");
}
};
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,
// Connect our inputs and outputs
this.recorderSource.connect(this.recorderFFT);
// 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;
};
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;
}
};
} 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;
};
} 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 {
@ -190,6 +221,13 @@ 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;
@ -197,7 +235,16 @@ export class VoiceRecording extends EventEmitter implements IDestroyable {
// 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)

82
src/voice/compat.ts Normal file
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@ -0,0 +1,82 @@
/*
Copyright 2021 The Matrix.org Foundation C.I.C.
Licensed under the Apache License, Version 2.0 (the "License");
you may not use this file except in compliance with the License.
You may obtain a copy of the License at
http://www.apache.org/licenses/LICENSE-2.0
Unless required by applicable law or agreed to in writing, software
distributed under the License is distributed on an "AS IS" BASIS,
WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
See the License for the specific language governing permissions and
limitations under the License.
*/
import {SAMPLE_RATE} from "./VoiceRecording";
// @ts-ignore - we know that this is not a module. We're looking for a path.
import decoderWasmPath from 'opus-recorder/dist/decoderWorker.min.wasm';
import wavEncoderPath from 'opus-recorder/dist/waveWorker.min.js';
import decoderPath from 'opus-recorder/dist/decoderWorker.min.js';
export function createAudioContext(opts?: AudioContextOptions): AudioContext {
if (window.AudioContext) {
return new AudioContext(opts);
} else if (window.webkitAudioContext) {
// While the linter is correct that "a constructor name should not start with
// a lowercase letter", it's also wrong to think that we have control over this.
// eslint-disable-next-line new-cap
return new window.webkitAudioContext(opts);
} else {
throw new Error("Unsupported browser");
}
}
export function decodeOgg(audioBuffer: ArrayBuffer): Promise<ArrayBuffer> {
// Condensed version of decoder example, using a promise:
// https://github.com/chris-rudmin/opus-recorder/blob/master/example/decoder.html
return new Promise((resolve) => { // no reject because the workers don't seem to have a fail path
console.log("Decoder WASM path: " + decoderWasmPath); // so we use the variable (avoid tree shake)
const typedArray = new Uint8Array(audioBuffer);
const decoderWorker = new Worker(decoderPath);
const wavWorker = new Worker(wavEncoderPath);
decoderWorker.postMessage({
command: 'init',
decoderSampleRate: SAMPLE_RATE,
outputBufferSampleRate: SAMPLE_RATE,
});
wavWorker.postMessage({
command: 'init',
wavBitDepth: 24, // standard for 48khz (SAMPLE_RATE)
wavSampleRate: SAMPLE_RATE,
});
decoderWorker.onmessage = (ev) => {
if (ev.data === null) { // null == done
wavWorker.postMessage({command: 'done'});
return;
}
wavWorker.postMessage({
command: 'encode',
buffers: ev.data,
}, ev.data.map(b => b.buffer));
};
wavWorker.onmessage = (ev) => {
if (ev.data.message === 'page') {
// The encoding comes through as a single page
resolve(new Blob([ev.data.page], {type: "audio/wav"}).arrayBuffer());
}
};
decoderWorker.postMessage({
command: 'decode',
pages: typedArray,
}, [typedArray.buffer]);
});
}