integrated openai realtime console

This commit is contained in:
RockYang
2024-10-15 19:25:18 +08:00
parent bd852c82b7
commit 48139290ed
18 changed files with 2444 additions and 281 deletions

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import { WavPacker } from './lib/wav_packer.js';
import { AudioAnalysis } from './lib/analysis/audio_analysis.js';
import { WavStreamPlayer } from './lib/wav_stream_player.js';
import { WavRecorder } from './lib/wav_recorder.js';
export { AudioAnalysis, WavPacker, WavStreamPlayer, WavRecorder };

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import {
noteFrequencies,
noteFrequencyLabels,
voiceFrequencies,
voiceFrequencyLabels,
} from './constants.js';
/**
* Output of AudioAnalysis for the frequency domain of the audio
* @typedef {Object} AudioAnalysisOutputType
* @property {Float32Array} values Amplitude of this frequency between {0, 1} inclusive
* @property {number[]} frequencies Raw frequency bucket values
* @property {string[]} labels Labels for the frequency bucket values
*/
/**
* Analyzes audio for visual output
* @class
*/
export class AudioAnalysis {
/**
* Retrieves frequency domain data from an AnalyserNode adjusted to a decibel range
* returns human-readable formatting and labels
* @param {AnalyserNode} analyser
* @param {number} sampleRate
* @param {Float32Array} [fftResult]
* @param {"frequency"|"music"|"voice"} [analysisType]
* @param {number} [minDecibels] default -100
* @param {number} [maxDecibels] default -30
* @returns {AudioAnalysisOutputType}
*/
static getFrequencies(
analyser,
sampleRate,
fftResult,
analysisType = 'frequency',
minDecibels = -100,
maxDecibels = -30,
) {
if (!fftResult) {
fftResult = new Float32Array(analyser.frequencyBinCount);
analyser.getFloatFrequencyData(fftResult);
}
const nyquistFrequency = sampleRate / 2;
const frequencyStep = (1 / fftResult.length) * nyquistFrequency;
let outputValues;
let frequencies;
let labels;
if (analysisType === 'music' || analysisType === 'voice') {
const useFrequencies =
analysisType === 'voice' ? voiceFrequencies : noteFrequencies;
const aggregateOutput = Array(useFrequencies.length).fill(minDecibels);
for (let i = 0; i < fftResult.length; i++) {
const frequency = i * frequencyStep;
const amplitude = fftResult[i];
for (let n = useFrequencies.length - 1; n >= 0; n--) {
if (frequency > useFrequencies[n]) {
aggregateOutput[n] = Math.max(aggregateOutput[n], amplitude);
break;
}
}
}
outputValues = aggregateOutput;
frequencies =
analysisType === 'voice' ? voiceFrequencies : noteFrequencies;
labels =
analysisType === 'voice' ? voiceFrequencyLabels : noteFrequencyLabels;
} else {
outputValues = Array.from(fftResult);
frequencies = outputValues.map((_, i) => frequencyStep * i);
labels = frequencies.map((f) => `${f.toFixed(2)} Hz`);
}
// We normalize to {0, 1}
const normalizedOutput = outputValues.map((v) => {
return Math.max(
0,
Math.min((v - minDecibels) / (maxDecibels - minDecibels), 1),
);
});
const values = new Float32Array(normalizedOutput);
return {
values,
frequencies,
labels,
};
}
/**
* Creates a new AudioAnalysis instance for an HTMLAudioElement
* @param {HTMLAudioElement} audioElement
* @param {AudioBuffer|null} [audioBuffer] If provided, will cache all frequency domain data from the buffer
* @returns {AudioAnalysis}
*/
constructor(audioElement, audioBuffer = null) {
this.fftResults = [];
if (audioBuffer) {
/**
* Modified from
* https://stackoverflow.com/questions/75063715/using-the-web-audio-api-to-analyze-a-song-without-playing
*
* We do this to populate FFT values for the audio if provided an `audioBuffer`
* The reason to do this is that Safari fails when using `createMediaElementSource`
* This has a non-zero RAM cost so we only opt-in to run it on Safari, Chrome is better
*/
const { length, sampleRate } = audioBuffer;
const offlineAudioContext = new OfflineAudioContext({
length,
sampleRate,
});
const source = offlineAudioContext.createBufferSource();
source.buffer = audioBuffer;
const analyser = offlineAudioContext.createAnalyser();
analyser.fftSize = 8192;
analyser.smoothingTimeConstant = 0.1;
source.connect(analyser);
// limit is :: 128 / sampleRate;
// but we just want 60fps - cuts ~1s from 6MB to 1MB of RAM
const renderQuantumInSeconds = 1 / 60;
const durationInSeconds = length / sampleRate;
const analyze = (index) => {
const suspendTime = renderQuantumInSeconds * index;
if (suspendTime < durationInSeconds) {
offlineAudioContext.suspend(suspendTime).then(() => {
const fftResult = new Float32Array(analyser.frequencyBinCount);
analyser.getFloatFrequencyData(fftResult);
this.fftResults.push(fftResult);
analyze(index + 1);
});
}
if (index === 1) {
offlineAudioContext.startRendering();
} else {
offlineAudioContext.resume();
}
};
source.start(0);
analyze(1);
this.audio = audioElement;
this.context = offlineAudioContext;
this.analyser = analyser;
this.sampleRate = sampleRate;
this.audioBuffer = audioBuffer;
} else {
const audioContext = new AudioContext();
const track = audioContext.createMediaElementSource(audioElement);
const analyser = audioContext.createAnalyser();
analyser.fftSize = 8192;
analyser.smoothingTimeConstant = 0.1;
track.connect(analyser);
analyser.connect(audioContext.destination);
this.audio = audioElement;
this.context = audioContext;
this.analyser = analyser;
this.sampleRate = this.context.sampleRate;
this.audioBuffer = null;
}
}
/**
* Gets the current frequency domain data from the playing audio track
* @param {"frequency"|"music"|"voice"} [analysisType]
* @param {number} [minDecibels] default -100
* @param {number} [maxDecibels] default -30
* @returns {AudioAnalysisOutputType}
*/
getFrequencies(
analysisType = 'frequency',
minDecibels = -100,
maxDecibels = -30,
) {
let fftResult = null;
if (this.audioBuffer && this.fftResults.length) {
const pct = this.audio.currentTime / this.audio.duration;
const index = Math.min(
(pct * this.fftResults.length) | 0,
this.fftResults.length - 1,
);
fftResult = this.fftResults[index];
}
return AudioAnalysis.getFrequencies(
this.analyser,
this.sampleRate,
fftResult,
analysisType,
minDecibels,
maxDecibels,
);
}
/**
* Resume the internal AudioContext if it was suspended due to the lack of
* user interaction when the AudioAnalysis was instantiated.
* @returns {Promise<true>}
*/
async resumeIfSuspended() {
if (this.context.state === 'suspended') {
await this.context.resume();
}
return true;
}
}
globalThis.AudioAnalysis = AudioAnalysis;

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/**
* Constants for help with visualization
* Helps map frequency ranges from Fast Fourier Transform
* to human-interpretable ranges, notably music ranges and
* human vocal ranges.
*/
// Eighth octave frequencies
const octave8Frequencies = [
4186.01, 4434.92, 4698.63, 4978.03, 5274.04, 5587.65, 5919.91, 6271.93,
6644.88, 7040.0, 7458.62, 7902.13,
];
// Labels for each of the above frequencies
const octave8FrequencyLabels = [
'C',
'C#',
'D',
'D#',
'E',
'F',
'F#',
'G',
'G#',
'A',
'A#',
'B',
];
/**
* All note frequencies from 1st to 8th octave
* in format "A#8" (A#, 8th octave)
*/
export const noteFrequencies = [];
export const noteFrequencyLabels = [];
for (let i = 1; i <= 8; i++) {
for (let f = 0; f < octave8Frequencies.length; f++) {
const freq = octave8Frequencies[f];
noteFrequencies.push(freq / Math.pow(2, 8 - i));
noteFrequencyLabels.push(octave8FrequencyLabels[f] + i);
}
}
/**
* Subset of the note frequencies between 32 and 2000 Hz
* 6 octave range: C1 to B6
*/
const voiceFrequencyRange = [32.0, 2000.0];
export const voiceFrequencies = noteFrequencies.filter((_, i) => {
return (
noteFrequencies[i] > voiceFrequencyRange[0] &&
noteFrequencies[i] < voiceFrequencyRange[1]
);
});
export const voiceFrequencyLabels = noteFrequencyLabels.filter((_, i) => {
return (
noteFrequencies[i] > voiceFrequencyRange[0] &&
noteFrequencies[i] < voiceFrequencyRange[1]
);
});

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/**
* Raw wav audio file contents
* @typedef {Object} WavPackerAudioType
* @property {Blob} blob
* @property {string} url
* @property {number} channelCount
* @property {number} sampleRate
* @property {number} duration
*/
/**
* Utility class for assembling PCM16 "audio/wav" data
* @class
*/
export class WavPacker {
/**
* Converts Float32Array of amplitude data to ArrayBuffer in Int16Array format
* @param {Float32Array} float32Array
* @returns {ArrayBuffer}
*/
static floatTo16BitPCM(float32Array) {
const buffer = new ArrayBuffer(float32Array.length * 2);
const view = new DataView(buffer);
let offset = 0;
for (let i = 0; i < float32Array.length; i++, offset += 2) {
let s = Math.max(-1, Math.min(1, float32Array[i]));
view.setInt16(offset, s < 0 ? s * 0x8000 : s * 0x7fff, true);
}
return buffer;
}
/**
* Concatenates two ArrayBuffers
* @param {ArrayBuffer} leftBuffer
* @param {ArrayBuffer} rightBuffer
* @returns {ArrayBuffer}
*/
static mergeBuffers(leftBuffer, rightBuffer) {
const tmpArray = new Uint8Array(
leftBuffer.byteLength + rightBuffer.byteLength
);
tmpArray.set(new Uint8Array(leftBuffer), 0);
tmpArray.set(new Uint8Array(rightBuffer), leftBuffer.byteLength);
return tmpArray.buffer;
}
/**
* Packs data into an Int16 format
* @private
* @param {number} size 0 = 1x Int16, 1 = 2x Int16
* @param {number} arg value to pack
* @returns
*/
_packData(size, arg) {
return [
new Uint8Array([arg, arg >> 8]),
new Uint8Array([arg, arg >> 8, arg >> 16, arg >> 24]),
][size];
}
/**
* Packs audio into "audio/wav" Blob
* @param {number} sampleRate
* @param {{bitsPerSample: number, channels: Array<Float32Array>, data: Int16Array}} audio
* @returns {WavPackerAudioType}
*/
pack(sampleRate, audio) {
if (!audio?.bitsPerSample) {
throw new Error(`Missing "bitsPerSample"`);
} else if (!audio?.channels) {
throw new Error(`Missing "channels"`);
} else if (!audio?.data) {
throw new Error(`Missing "data"`);
}
const { bitsPerSample, channels, data } = audio;
const output = [
// Header
'RIFF',
this._packData(
1,
4 + (8 + 24) /* chunk 1 length */ + (8 + 8) /* chunk 2 length */
), // Length
'WAVE',
// chunk 1
'fmt ', // Sub-chunk identifier
this._packData(1, 16), // Chunk length
this._packData(0, 1), // Audio format (1 is linear quantization)
this._packData(0, channels.length),
this._packData(1, sampleRate),
this._packData(1, (sampleRate * channels.length * bitsPerSample) / 8), // Byte rate
this._packData(0, (channels.length * bitsPerSample) / 8),
this._packData(0, bitsPerSample),
// chunk 2
'data', // Sub-chunk identifier
this._packData(
1,
(channels[0].length * channels.length * bitsPerSample) / 8
), // Chunk length
data,
];
const blob = new Blob(output, { type: 'audio/mpeg' });
const url = URL.createObjectURL(blob);
return {
blob,
url,
channelCount: channels.length,
sampleRate,
duration: data.byteLength / (channels.length * sampleRate * 2),
};
}
}
globalThis.WavPacker = WavPacker;

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import { AudioProcessorSrc } from './worklets/audio_processor.js';
import { AudioAnalysis } from './analysis/audio_analysis.js';
import { WavPacker } from './wav_packer.js';
/**
* Decodes audio into a wav file
* @typedef {Object} DecodedAudioType
* @property {Blob} blob
* @property {string} url
* @property {Float32Array} values
* @property {AudioBuffer} audioBuffer
*/
/**
* Records live stream of user audio as PCM16 "audio/wav" data
* @class
*/
export class WavRecorder {
/**
* Create a new WavRecorder instance
* @param {{sampleRate?: number, outputToSpeakers?: boolean, debug?: boolean}} [options]
* @returns {WavRecorder}
*/
constructor({
sampleRate = 44100,
outputToSpeakers = false,
debug = false,
} = {}) {
// Script source
this.scriptSrc = AudioProcessorSrc;
// Config
this.sampleRate = sampleRate;
this.outputToSpeakers = outputToSpeakers;
this.debug = !!debug;
this._deviceChangeCallback = null;
this._devices = [];
// State variables
this.stream = null;
this.processor = null;
this.source = null;
this.node = null;
this.recording = false;
// Event handling with AudioWorklet
this._lastEventId = 0;
this.eventReceipts = {};
this.eventTimeout = 5000;
// Process chunks of audio
this._chunkProcessor = () => {};
this._chunkProcessorSize = void 0;
this._chunkProcessorBuffer = {
raw: new ArrayBuffer(0),
mono: new ArrayBuffer(0),
};
}
/**
* Decodes audio data from multiple formats to a Blob, url, Float32Array and AudioBuffer
* @param {Blob|Float32Array|Int16Array|ArrayBuffer|number[]} audioData
* @param {number} sampleRate
* @param {number} fromSampleRate
* @returns {Promise<DecodedAudioType>}
*/
static async decode(audioData, sampleRate = 44100, fromSampleRate = -1) {
const context = new AudioContext({ sampleRate });
let arrayBuffer;
let blob;
if (audioData instanceof Blob) {
if (fromSampleRate !== -1) {
throw new Error(
`Can not specify "fromSampleRate" when reading from Blob`,
);
}
blob = audioData;
arrayBuffer = await blob.arrayBuffer();
} else if (audioData instanceof ArrayBuffer) {
if (fromSampleRate !== -1) {
throw new Error(
`Can not specify "fromSampleRate" when reading from ArrayBuffer`,
);
}
arrayBuffer = audioData;
blob = new Blob([arrayBuffer], { type: 'audio/wav' });
} else {
let float32Array;
let data;
if (audioData instanceof Int16Array) {
data = audioData;
float32Array = new Float32Array(audioData.length);
for (let i = 0; i < audioData.length; i++) {
float32Array[i] = audioData[i] / 0x8000;
}
} else if (audioData instanceof Float32Array) {
float32Array = audioData;
} else if (audioData instanceof Array) {
float32Array = new Float32Array(audioData);
} else {
throw new Error(
`"audioData" must be one of: Blob, Float32Arrray, Int16Array, ArrayBuffer, Array<number>`,
);
}
if (fromSampleRate === -1) {
throw new Error(
`Must specify "fromSampleRate" when reading from Float32Array, In16Array or Array`,
);
} else if (fromSampleRate < 3000) {
throw new Error(`Minimum "fromSampleRate" is 3000 (3kHz)`);
}
if (!data) {
data = WavPacker.floatTo16BitPCM(float32Array);
}
const audio = {
bitsPerSample: 16,
channels: [float32Array],
data,
};
const packer = new WavPacker();
const result = packer.pack(fromSampleRate, audio);
blob = result.blob;
arrayBuffer = await blob.arrayBuffer();
}
const audioBuffer = await context.decodeAudioData(arrayBuffer);
const values = audioBuffer.getChannelData(0);
const url = URL.createObjectURL(blob);
return {
blob,
url,
values,
audioBuffer,
};
}
/**
* Logs data in debug mode
* @param {...any} arguments
* @returns {true}
*/
log() {
if (this.debug) {
this.log(...arguments);
}
return true;
}
/**
* Retrieves the current sampleRate for the recorder
* @returns {number}
*/
getSampleRate() {
return this.sampleRate;
}
/**
* Retrieves the current status of the recording
* @returns {"ended"|"paused"|"recording"}
*/
getStatus() {
if (!this.processor) {
return 'ended';
} else if (!this.recording) {
return 'paused';
} else {
return 'recording';
}
}
/**
* Sends an event to the AudioWorklet
* @private
* @param {string} name
* @param {{[key: string]: any}} data
* @param {AudioWorkletNode} [_processor]
* @returns {Promise<{[key: string]: any}>}
*/
async _event(name, data = {}, _processor = null) {
_processor = _processor || this.processor;
if (!_processor) {
throw new Error('Can not send events without recording first');
}
const message = {
event: name,
id: this._lastEventId++,
data,
};
_processor.port.postMessage(message);
const t0 = new Date().valueOf();
while (!this.eventReceipts[message.id]) {
if (new Date().valueOf() - t0 > this.eventTimeout) {
throw new Error(`Timeout waiting for "${name}" event`);
}
await new Promise((res) => setTimeout(() => res(true), 1));
}
const payload = this.eventReceipts[message.id];
delete this.eventReceipts[message.id];
return payload;
}
/**
* Sets device change callback, remove if callback provided is `null`
* @param {(Array<MediaDeviceInfo & {default: boolean}>): void|null} callback
* @returns {true}
*/
listenForDeviceChange(callback) {
if (callback === null && this._deviceChangeCallback) {
navigator.mediaDevices.removeEventListener(
'devicechange',
this._deviceChangeCallback,
);
this._deviceChangeCallback = null;
} else if (callback !== null) {
// Basically a debounce; we only want this called once when devices change
// And we only want the most recent callback() to be executed
// if a few are operating at the same time
let lastId = 0;
let lastDevices = [];
const serializeDevices = (devices) =>
devices
.map((d) => d.deviceId)
.sort()
.join(',');
const cb = async () => {
let id = ++lastId;
const devices = await this.listDevices();
if (id === lastId) {
if (serializeDevices(lastDevices) !== serializeDevices(devices)) {
lastDevices = devices;
callback(devices.slice());
}
}
};
navigator.mediaDevices.addEventListener('devicechange', cb);
cb();
this._deviceChangeCallback = cb;
}
return true;
}
/**
* Manually request permission to use the microphone
* @returns {Promise<true>}
*/
async requestPermission() {
const permissionStatus = await navigator.permissions.query({
name: 'microphone',
});
if (permissionStatus.state === 'denied') {
window.alert('You must grant microphone access to use this feature.');
} else if (permissionStatus.state === 'prompt') {
try {
const stream = await navigator.mediaDevices.getUserMedia({
audio: true,
});
const tracks = stream.getTracks();
tracks.forEach((track) => track.stop());
} catch (e) {
window.alert('You must grant microphone access to use this feature.');
}
}
return true;
}
/**
* List all eligible devices for recording, will request permission to use microphone
* @returns {Promise<Array<MediaDeviceInfo & {default: boolean}>>}
*/
async listDevices() {
if (
!navigator.mediaDevices ||
!('enumerateDevices' in navigator.mediaDevices)
) {
throw new Error('Could not request user devices');
}
await this.requestPermission();
const devices = await navigator.mediaDevices.enumerateDevices();
const audioDevices = devices.filter(
(device) => device.kind === 'audioinput',
);
const defaultDeviceIndex = audioDevices.findIndex(
(device) => device.deviceId === 'default',
);
const deviceList = [];
if (defaultDeviceIndex !== -1) {
let defaultDevice = audioDevices.splice(defaultDeviceIndex, 1)[0];
let existingIndex = audioDevices.findIndex(
(device) => device.groupId === defaultDevice.groupId,
);
if (existingIndex !== -1) {
defaultDevice = audioDevices.splice(existingIndex, 1)[0];
}
defaultDevice.default = true;
deviceList.push(defaultDevice);
}
return deviceList.concat(audioDevices);
}
/**
* Begins a recording session and requests microphone permissions if not already granted
* Microphone recording indicator will appear on browser tab but status will be "paused"
* @param {string} [deviceId] if no device provided, default device will be used
* @returns {Promise<true>}
*/
async begin(deviceId) {
if (this.processor) {
throw new Error(
`Already connected: please call .end() to start a new session`,
);
}
if (
!navigator.mediaDevices ||
!('getUserMedia' in navigator.mediaDevices)
) {
throw new Error('Could not request user media');
}
try {
const config = { audio: true };
if (deviceId) {
config.audio = { deviceId: { exact: deviceId } };
}
this.stream = await navigator.mediaDevices.getUserMedia(config);
} catch (err) {
throw new Error('Could not start media stream');
}
const context = new AudioContext({ sampleRate: this.sampleRate });
const source = context.createMediaStreamSource(this.stream);
// Load and execute the module script.
try {
await context.audioWorklet.addModule(this.scriptSrc);
} catch (e) {
console.error(e);
throw new Error(`Could not add audioWorklet module: ${this.scriptSrc}`);
}
const processor = new AudioWorkletNode(context, 'audio_processor');
processor.port.onmessage = (e) => {
const { event, id, data } = e.data;
if (event === 'receipt') {
this.eventReceipts[id] = data;
} else if (event === 'chunk') {
if (this._chunkProcessorSize) {
const buffer = this._chunkProcessorBuffer;
this._chunkProcessorBuffer = {
raw: WavPacker.mergeBuffers(buffer.raw, data.raw),
mono: WavPacker.mergeBuffers(buffer.mono, data.mono),
};
if (
this._chunkProcessorBuffer.mono.byteLength >=
this._chunkProcessorSize
) {
this._chunkProcessor(this._chunkProcessorBuffer);
this._chunkProcessorBuffer = {
raw: new ArrayBuffer(0),
mono: new ArrayBuffer(0),
};
}
} else {
this._chunkProcessor(data);
}
}
};
const node = source.connect(processor);
const analyser = context.createAnalyser();
analyser.fftSize = 8192;
analyser.smoothingTimeConstant = 0.1;
node.connect(analyser);
if (this.outputToSpeakers) {
// eslint-disable-next-line no-console
console.warn(
'Warning: Output to speakers may affect sound quality,\n' +
'especially due to system audio feedback preventative measures.\n' +
'use only for debugging',
);
analyser.connect(context.destination);
}
this.source = source;
this.node = node;
this.analyser = analyser;
this.processor = processor;
return true;
}
/**
* Gets the current frequency domain data from the recording track
* @param {"frequency"|"music"|"voice"} [analysisType]
* @param {number} [minDecibels] default -100
* @param {number} [maxDecibels] default -30
* @returns {import('./analysis/audio_analysis.js').AudioAnalysisOutputType}
*/
getFrequencies(
analysisType = 'frequency',
minDecibels = -100,
maxDecibels = -30,
) {
if (!this.processor) {
throw new Error('Session ended: please call .begin() first');
}
return AudioAnalysis.getFrequencies(
this.analyser,
this.sampleRate,
null,
analysisType,
minDecibels,
maxDecibels,
);
}
/**
* Pauses the recording
* Keeps microphone stream open but halts storage of audio
* @returns {Promise<true>}
*/
async pause() {
if (!this.processor) {
throw new Error('Session ended: please call .begin() first');
} else if (!this.recording) {
throw new Error('Already paused: please call .record() first');
}
if (this._chunkProcessorBuffer.raw.byteLength) {
this._chunkProcessor(this._chunkProcessorBuffer);
}
this.log('Pausing ...');
await this._event('stop');
this.recording = false;
return true;
}
/**
* Start recording stream and storing to memory from the connected audio source
* @param {(data: { mono: Int16Array; raw: Int16Array }) => any} [chunkProcessor]
* @param {number} [chunkSize] chunkProcessor will not be triggered until this size threshold met in mono audio
* @returns {Promise<true>}
*/
async record(chunkProcessor = () => {}, chunkSize = 8192) {
if (!this.processor) {
throw new Error('Session ended: please call .begin() first');
} else if (this.recording) {
throw new Error('Already recording: please call .pause() first');
} else if (typeof chunkProcessor !== 'function') {
throw new Error(`chunkProcessor must be a function`);
}
this._chunkProcessor = chunkProcessor;
this._chunkProcessorSize = chunkSize;
this._chunkProcessorBuffer = {
raw: new ArrayBuffer(0),
mono: new ArrayBuffer(0),
};
this.log('Recording ...');
await this._event('start');
this.recording = true;
return true;
}
/**
* Clears the audio buffer, empties stored recording
* @returns {Promise<true>}
*/
async clear() {
if (!this.processor) {
throw new Error('Session ended: please call .begin() first');
}
await this._event('clear');
return true;
}
/**
* Reads the current audio stream data
* @returns {Promise<{meanValues: Float32Array, channels: Array<Float32Array>}>}
*/
async read() {
if (!this.processor) {
throw new Error('Session ended: please call .begin() first');
}
this.log('Reading ...');
const result = await this._event('read');
return result;
}
/**
* Saves the current audio stream to a file
* @param {boolean} [force] Force saving while still recording
* @returns {Promise<import('./wav_packer.js').WavPackerAudioType>}
*/
async save(force = false) {
if (!this.processor) {
throw new Error('Session ended: please call .begin() first');
}
if (!force && this.recording) {
throw new Error(
'Currently recording: please call .pause() first, or call .save(true) to force',
);
}
this.log('Exporting ...');
const exportData = await this._event('export');
const packer = new WavPacker();
const result = packer.pack(this.sampleRate, exportData.audio);
return result;
}
/**
* Ends the current recording session and saves the result
* @returns {Promise<import('./wav_packer.js').WavPackerAudioType>}
*/
async end() {
if (!this.processor) {
throw new Error('Session ended: please call .begin() first');
}
const _processor = this.processor;
this.log('Stopping ...');
await this._event('stop');
this.recording = false;
const tracks = this.stream.getTracks();
tracks.forEach((track) => track.stop());
this.log('Exporting ...');
const exportData = await this._event('export', {}, _processor);
this.processor.disconnect();
this.source.disconnect();
this.node.disconnect();
this.analyser.disconnect();
this.stream = null;
this.processor = null;
this.source = null;
this.node = null;
const packer = new WavPacker();
const result = packer.pack(this.sampleRate, exportData.audio);
return result;
}
/**
* Performs a full cleanup of WavRecorder instance
* Stops actively listening via microphone and removes existing listeners
* @returns {Promise<true>}
*/
async quit() {
this.listenForDeviceChange(null);
if (this.processor) {
await this.end();
}
return true;
}
}
globalThis.WavRecorder = WavRecorder;

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import { StreamProcessorSrc } from './worklets/stream_processor.js';
import { AudioAnalysis } from './analysis/audio_analysis.js';
/**
* Plays audio streams received in raw PCM16 chunks from the browser
* @class
*/
export class WavStreamPlayer {
/**
* Creates a new WavStreamPlayer instance
* @param {{sampleRate?: number}} options
* @returns {WavStreamPlayer}
*/
constructor({ sampleRate = 44100 } = {}) {
this.scriptSrc = StreamProcessorSrc;
this.sampleRate = sampleRate;
this.context = null;
this.stream = null;
this.analyser = null;
this.trackSampleOffsets = {};
this.interruptedTrackIds = {};
}
/**
* Connects the audio context and enables output to speakers
* @returns {Promise<true>}
*/
async connect() {
this.context = new AudioContext({ sampleRate: this.sampleRate });
if (this.context.state === 'suspended') {
await this.context.resume();
}
try {
await this.context.audioWorklet.addModule(this.scriptSrc);
} catch (e) {
console.error(e);
throw new Error(`Could not add audioWorklet module: ${this.scriptSrc}`);
}
const analyser = this.context.createAnalyser();
analyser.fftSize = 8192;
analyser.smoothingTimeConstant = 0.1;
this.analyser = analyser;
return true;
}
/**
* Gets the current frequency domain data from the playing track
* @param {"frequency"|"music"|"voice"} [analysisType]
* @param {number} [minDecibels] default -100
* @param {number} [maxDecibels] default -30
* @returns {import('./analysis/audio_analysis.js').AudioAnalysisOutputType}
*/
getFrequencies(
analysisType = 'frequency',
minDecibels = -100,
maxDecibels = -30
) {
if (!this.analyser) {
throw new Error('Not connected, please call .connect() first');
}
return AudioAnalysis.getFrequencies(
this.analyser,
this.sampleRate,
null,
analysisType,
minDecibels,
maxDecibels
);
}
/**
* Starts audio streaming
* @private
* @returns {Promise<true>}
*/
_start() {
const streamNode = new AudioWorkletNode(this.context, 'stream_processor');
streamNode.connect(this.context.destination);
streamNode.port.onmessage = (e) => {
const { event } = e.data;
if (event === 'stop') {
streamNode.disconnect();
this.stream = null;
} else if (event === 'offset') {
const { requestId, trackId, offset } = e.data;
const currentTime = offset / this.sampleRate;
this.trackSampleOffsets[requestId] = { trackId, offset, currentTime };
}
};
this.analyser.disconnect();
streamNode.connect(this.analyser);
this.stream = streamNode;
return true;
}
/**
* Adds 16BitPCM data to the currently playing audio stream
* You can add chunks beyond the current play point and they will be queued for play
* @param {ArrayBuffer|Int16Array} arrayBuffer
* @param {string} [trackId]
* @returns {Int16Array}
*/
add16BitPCM(arrayBuffer, trackId = 'default') {
if (typeof trackId !== 'string') {
throw new Error(`trackId must be a string`);
} else if (this.interruptedTrackIds[trackId]) {
return;
}
if (!this.stream) {
this._start();
}
let buffer;
if (arrayBuffer instanceof Int16Array) {
buffer = arrayBuffer;
} else if (arrayBuffer instanceof ArrayBuffer) {
buffer = new Int16Array(arrayBuffer);
} else {
throw new Error(`argument must be Int16Array or ArrayBuffer`);
}
this.stream.port.postMessage({ event: 'write', buffer, trackId });
return buffer;
}
/**
* Gets the offset (sample count) of the currently playing stream
* @param {boolean} [interrupt]
* @returns {{trackId: string|null, offset: number, currentTime: number}}
*/
async getTrackSampleOffset(interrupt = false) {
if (!this.stream) {
return null;
}
const requestId = crypto.randomUUID();
this.stream.port.postMessage({
event: interrupt ? 'interrupt' : 'offset',
requestId,
});
let trackSampleOffset;
while (!trackSampleOffset) {
trackSampleOffset = this.trackSampleOffsets[requestId];
await new Promise((r) => setTimeout(() => r(), 1));
}
const { trackId } = trackSampleOffset;
if (interrupt && trackId) {
this.interruptedTrackIds[trackId] = true;
}
return trackSampleOffset;
}
/**
* Strips the current stream and returns the sample offset of the audio
* @param {boolean} [interrupt]
* @returns {{trackId: string|null, offset: number, currentTime: number}}
*/
async interrupt() {
return this.getTrackSampleOffset(true);
}
}
globalThis.WavStreamPlayer = WavStreamPlayer;

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const AudioProcessorWorklet = `
class AudioProcessor extends AudioWorkletProcessor {
constructor() {
super();
this.port.onmessage = this.receive.bind(this);
this.initialize();
}
initialize() {
this.foundAudio = false;
this.recording = false;
this.chunks = [];
}
/**
* Concatenates sampled chunks into channels
* Format is chunk[Left[], Right[]]
*/
readChannelData(chunks, channel = -1, maxChannels = 9) {
let channelLimit;
if (channel !== -1) {
if (chunks[0] && chunks[0].length - 1 < channel) {
throw new Error(
\`Channel \${channel} out of range: max \${chunks[0].length}\`
);
}
channelLimit = channel + 1;
} else {
channel = 0;
channelLimit = Math.min(chunks[0] ? chunks[0].length : 1, maxChannels);
}
const channels = [];
for (let n = channel; n < channelLimit; n++) {
const length = chunks.reduce((sum, chunk) => {
return sum + chunk[n].length;
}, 0);
const buffers = chunks.map((chunk) => chunk[n]);
const result = new Float32Array(length);
let offset = 0;
for (let i = 0; i < buffers.length; i++) {
result.set(buffers[i], offset);
offset += buffers[i].length;
}
channels[n] = result;
}
return channels;
}
/**
* Combines parallel audio data into correct format,
* channels[Left[], Right[]] to float32Array[LRLRLRLR...]
*/
formatAudioData(channels) {
if (channels.length === 1) {
// Simple case is only one channel
const float32Array = channels[0].slice();
const meanValues = channels[0].slice();
return { float32Array, meanValues };
} else {
const float32Array = new Float32Array(
channels[0].length * channels.length
);
const meanValues = new Float32Array(channels[0].length);
for (let i = 0; i < channels[0].length; i++) {
const offset = i * channels.length;
let meanValue = 0;
for (let n = 0; n < channels.length; n++) {
float32Array[offset + n] = channels[n][i];
meanValue += channels[n][i];
}
meanValues[i] = meanValue / channels.length;
}
return { float32Array, meanValues };
}
}
/**
* Converts 32-bit float data to 16-bit integers
*/
floatTo16BitPCM(float32Array) {
const buffer = new ArrayBuffer(float32Array.length * 2);
const view = new DataView(buffer);
let offset = 0;
for (let i = 0; i < float32Array.length; i++, offset += 2) {
let s = Math.max(-1, Math.min(1, float32Array[i]));
view.setInt16(offset, s < 0 ? s * 0x8000 : s * 0x7fff, true);
}
return buffer;
}
/**
* Retrieves the most recent amplitude values from the audio stream
* @param {number} channel
*/
getValues(channel = -1) {
const channels = this.readChannelData(this.chunks, channel);
const { meanValues } = this.formatAudioData(channels);
return { meanValues, channels };
}
/**
* Exports chunks as an audio/wav file
*/
export() {
const channels = this.readChannelData(this.chunks);
const { float32Array, meanValues } = this.formatAudioData(channels);
const audioData = this.floatTo16BitPCM(float32Array);
return {
meanValues: meanValues,
audio: {
bitsPerSample: 16,
channels: channels,
data: audioData,
},
};
}
receive(e) {
const { event, id } = e.data;
let receiptData = {};
switch (event) {
case 'start':
this.recording = true;
break;
case 'stop':
this.recording = false;
break;
case 'clear':
this.initialize();
break;
case 'export':
receiptData = this.export();
break;
case 'read':
receiptData = this.getValues();
break;
default:
break;
}
// Always send back receipt
this.port.postMessage({ event: 'receipt', id, data: receiptData });
}
sendChunk(chunk) {
const channels = this.readChannelData([chunk]);
const { float32Array, meanValues } = this.formatAudioData(channels);
const rawAudioData = this.floatTo16BitPCM(float32Array);
const monoAudioData = this.floatTo16BitPCM(meanValues);
this.port.postMessage({
event: 'chunk',
data: {
mono: monoAudioData,
raw: rawAudioData,
},
});
}
process(inputList, outputList, parameters) {
// Copy input to output (e.g. speakers)
// Note that this creates choppy sounds with Mac products
const sourceLimit = Math.min(inputList.length, outputList.length);
for (let inputNum = 0; inputNum < sourceLimit; inputNum++) {
const input = inputList[inputNum];
const output = outputList[inputNum];
const channelCount = Math.min(input.length, output.length);
for (let channelNum = 0; channelNum < channelCount; channelNum++) {
input[channelNum].forEach((sample, i) => {
output[channelNum][i] = sample;
});
}
}
const inputs = inputList[0];
// There's latency at the beginning of a stream before recording starts
// Make sure we actually receive audio data before we start storing chunks
let sliceIndex = 0;
if (!this.foundAudio) {
for (const channel of inputs) {
sliceIndex = 0; // reset for each channel
if (this.foundAudio) {
break;
}
if (channel) {
for (const value of channel) {
if (value !== 0) {
// find only one non-zero entry in any channel
this.foundAudio = true;
break;
} else {
sliceIndex++;
}
}
}
}
}
if (inputs && inputs[0] && this.foundAudio && this.recording) {
// We need to copy the TypedArray, because the \`process\`
// internals will reuse the same buffer to hold each input
const chunk = inputs.map((input) => input.slice(sliceIndex));
this.chunks.push(chunk);
this.sendChunk(chunk);
}
return true;
}
}
registerProcessor('audio_processor', AudioProcessor);
`;
const script = new Blob([AudioProcessorWorklet], {
type: 'application/javascript',
});
const src = URL.createObjectURL(script);
export const AudioProcessorSrc = src;

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export const StreamProcessorWorklet = `
class StreamProcessor extends AudioWorkletProcessor {
constructor() {
super();
this.hasStarted = false;
this.hasInterrupted = false;
this.outputBuffers = [];
this.bufferLength = 128;
this.write = { buffer: new Float32Array(this.bufferLength), trackId: null };
this.writeOffset = 0;
this.trackSampleOffsets = {};
this.port.onmessage = (event) => {
if (event.data) {
const payload = event.data;
if (payload.event === 'write') {
const int16Array = payload.buffer;
const float32Array = new Float32Array(int16Array.length);
for (let i = 0; i < int16Array.length; i++) {
float32Array[i] = int16Array[i] / 0x8000; // Convert Int16 to Float32
}
this.writeData(float32Array, payload.trackId);
} else if (
payload.event === 'offset' ||
payload.event === 'interrupt'
) {
const requestId = payload.requestId;
const trackId = this.write.trackId;
const offset = this.trackSampleOffsets[trackId] || 0;
this.port.postMessage({
event: 'offset',
requestId,
trackId,
offset,
});
if (payload.event === 'interrupt') {
this.hasInterrupted = true;
}
} else {
throw new Error(\`Unhandled event "\${payload.event}"\`);
}
}
};
}
writeData(float32Array, trackId = null) {
let { buffer } = this.write;
let offset = this.writeOffset;
for (let i = 0; i < float32Array.length; i++) {
buffer[offset++] = float32Array[i];
if (offset >= buffer.length) {
this.outputBuffers.push(this.write);
this.write = { buffer: new Float32Array(this.bufferLength), trackId };
buffer = this.write.buffer;
offset = 0;
}
}
this.writeOffset = offset;
return true;
}
process(inputs, outputs, parameters) {
const output = outputs[0];
const outputChannelData = output[0];
const outputBuffers = this.outputBuffers;
if (this.hasInterrupted) {
this.port.postMessage({ event: 'stop' });
return false;
} else if (outputBuffers.length) {
this.hasStarted = true;
const { buffer, trackId } = outputBuffers.shift();
for (let i = 0; i < outputChannelData.length; i++) {
outputChannelData[i] = buffer[i] || 0;
}
if (trackId) {
this.trackSampleOffsets[trackId] =
this.trackSampleOffsets[trackId] || 0;
this.trackSampleOffsets[trackId] += buffer.length;
}
return true;
} else if (this.hasStarted) {
this.port.postMessage({ event: 'stop' });
return false;
} else {
return true;
}
}
}
registerProcessor('stream_processor', StreamProcessor);
`;
const script = new Blob([StreamProcessorWorklet], {
type: 'application/javascript',
});
const src = URL.createObjectURL(script);
export const StreamProcessorSrc = src;