CI / ruff · pytest (pull_request) Successful in 2m27s
Microphone button in the chat composer and in free-text steps. The browser
records 16 kHz mono WAV (no server-side ffmpeg), /me/{token}/stt transcribes,
the text lands in the field for the patient to check and edit — never sent
automatically, so safety and scrub() apply on send. Hidden when the server has
voice input off or the browser lacks a microphone API. nginx: 4 MB body limit.
Refs #117
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
120 lines
4.3 KiB
TypeScript
120 lines
4.3 KiB
TypeScript
/*
|
|
* Voice recording for the patient's messages (#117, slide 11): microphone → 16 kHz mono 16-bit WAV in the browser,
|
|
* so the server needs no ffmpeg and Chrome, Android and iOS Safari produce the same format (ADR-014).
|
|
* The audio lives only in memory until it is sent for transcription; nothing is stored on the device.
|
|
*/
|
|
|
|
export const TARGET_RATE = 16_000;
|
|
export const MAX_SECONDS = 60;
|
|
|
|
type WindowWithWebkit = Window & { webkitAudioContext?: typeof AudioContext };
|
|
|
|
export function voiceSupported(): boolean {
|
|
const w = window as WindowWithWebkit;
|
|
return typeof navigator.mediaDevices?.getUserMedia === "function" && Boolean(window.AudioContext || w.webkitAudioContext);
|
|
}
|
|
|
|
export class Recorder {
|
|
private ctx: AudioContext | null = null;
|
|
private stream: MediaStream | null = null;
|
|
private node: ScriptProcessorNode | null = null;
|
|
private chunks: Float32Array[] = [];
|
|
private rate = 48_000;
|
|
|
|
/** `onLevel` gets the loudness 0..1 of each audio block (for the waveform). */
|
|
async start(onLevel: (level: number) => void): Promise<void> {
|
|
this.stream = await navigator.mediaDevices.getUserMedia({
|
|
audio: { channelCount: 1, echoCancellation: true, noiseSuppression: true, autoGainControl: true },
|
|
});
|
|
const Ctx = window.AudioContext || (window as WindowWithWebkit).webkitAudioContext!;
|
|
this.ctx = new Ctx();
|
|
await this.ctx.resume();
|
|
this.rate = this.ctx.sampleRate;
|
|
const source = this.ctx.createMediaStreamSource(this.stream);
|
|
// ScriptProcessorNode is deprecated but works everywhere (incl. older iOS Safari) without a worklet file.
|
|
this.node = this.ctx.createScriptProcessor(4096, 1, 1);
|
|
this.node.onaudioprocess = (e) => {
|
|
const data = new Float32Array(e.inputBuffer.getChannelData(0));
|
|
this.chunks.push(data);
|
|
let sum = 0;
|
|
for (let i = 0; i < data.length; i++) sum += data[i] * data[i];
|
|
onLevel(Math.min(1, Math.sqrt(sum / data.length) * 4));
|
|
};
|
|
source.connect(this.node);
|
|
this.node.connect(this.ctx.destination);
|
|
}
|
|
|
|
/** Stops the microphone and returns the WAV, or null if nothing was recorded. */
|
|
stop(): Blob | null {
|
|
const chunks = this.chunks;
|
|
const rate = this.rate;
|
|
this.release();
|
|
if (!chunks.length) return null;
|
|
return encodeWav(downsample(concat(chunks), rate, TARGET_RATE), TARGET_RATE);
|
|
}
|
|
|
|
cancel(): void {
|
|
this.release();
|
|
}
|
|
|
|
private release(): void {
|
|
this.node?.disconnect();
|
|
if (this.node) this.node.onaudioprocess = null;
|
|
this.stream?.getTracks().forEach((t) => t.stop());
|
|
void this.ctx?.close();
|
|
this.node = null;
|
|
this.stream = null;
|
|
this.ctx = null;
|
|
this.chunks = [];
|
|
}
|
|
}
|
|
|
|
function concat(chunks: Float32Array[]): Float32Array {
|
|
const out = new Float32Array(chunks.reduce((n, c) => n + c.length, 0));
|
|
let pos = 0;
|
|
for (const c of chunks) {
|
|
out.set(c, pos);
|
|
pos += c.length;
|
|
}
|
|
return out;
|
|
}
|
|
|
|
/** Averaging downsampler: good enough for speech, avoids aliasing better than dropping samples. */
|
|
function downsample(input: Float32Array, from: number, to: number): Float32Array {
|
|
if (from === to) return input;
|
|
const ratio = from / to;
|
|
const out = new Float32Array(Math.floor(input.length / ratio));
|
|
for (let i = 0; i < out.length; i++) {
|
|
const start = Math.floor(i * ratio);
|
|
const end = Math.min(input.length, Math.floor((i + 1) * ratio));
|
|
let sum = 0;
|
|
for (let j = start; j < end; j++) sum += input[j];
|
|
out[i] = sum / Math.max(1, end - start);
|
|
}
|
|
return out;
|
|
}
|
|
|
|
function encodeWav(samples: Float32Array, rate: number): Blob {
|
|
const buf = new ArrayBuffer(44 + samples.length * 2);
|
|
const v = new DataView(buf);
|
|
const str = (o: number, s: string) => [...s].forEach((ch, i) => v.setUint8(o + i, ch.charCodeAt(0)));
|
|
str(0, "RIFF");
|
|
v.setUint32(4, 36 + samples.length * 2, true);
|
|
str(8, "WAVE");
|
|
str(12, "fmt ");
|
|
v.setUint32(16, 16, true);
|
|
v.setUint16(20, 1, true); // PCM
|
|
v.setUint16(22, 1, true); // mono
|
|
v.setUint32(24, rate, true);
|
|
v.setUint32(28, rate * 2, true); // byte rate
|
|
v.setUint16(32, 2, true); // block align
|
|
v.setUint16(34, 16, true);
|
|
str(36, "data");
|
|
v.setUint32(40, samples.length * 2, true);
|
|
for (let i = 0; i < samples.length; i++) {
|
|
const s = Math.max(-1, Math.min(1, samples[i]));
|
|
v.setInt16(44 + i * 2, s < 0 ? s * 0x8000 : s * 0x7fff, true);
|
|
}
|
|
return new Blob([buf], { type: "audio/wav" });
|
|
}
|