diff --git a/src/rageshake/rageshake.js b/src/rageshake/rageshake.js index b886f369df..9512f62e42 100644 --- a/src/rageshake/rageshake.js +++ b/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 { diff --git a/src/voice/VoiceRecording.ts b/src/voice/VoiceRecording.ts index c4a0a78ce5..402bd8beca 100644 --- a/src/voice/VoiceRecording.ts +++ b/src/voice/VoiceRecording.ts @@ -90,78 +90,97 @@ 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 = 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; + // 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"); } - }; + await this.recorderContext.audioWorklet.addModule(mxRecorderWorkletPath); + this.recorderWorklet = new AudioWorkletNode(this.recorderContext, WORKLET_NAME); - 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); + this.recorderSource.connect(this.recorderWorklet); + this.recorderWorklet.connect(this.recorderContext.destination); - // 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; - }; + // 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; + } + }; + + 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 {