diff --git a/.github/workflows/deploy_gh_pages.yml b/.github/workflows/deploy_gh_pages.yml new file mode 100644 index 0000000..96b44ca --- /dev/null +++ b/.github/workflows/deploy_gh_pages.yml @@ -0,0 +1,51 @@ +name: Deploy to GitHub Pages + +on: + push: + branches: + - gh-deploy + +jobs: + build_site: + runs-on: ubuntu-latest + steps: + - name: Checkout + uses: actions/checkout@v3 + + - name: Install Node.js + uses: actions/setup-node@v3 + with: + node-version: 18 + cache: npm + + - name: Install dependencies + run: npm install + + - name: build + env: + BASE_PATH: '/${{ github.event.repository.name }}' + run: | + npm run build + + - name: Upload Artifacts + uses: actions/upload-pages-artifact@v2 + with: + # this should match the `pages` option in your adapter-static options + path: 'build/' + + deploy: + needs: build_site + runs-on: ubuntu-latest + + permissions: + pages: write + id-token: write + + environment: + name: github-pages + url: ${{ steps.deployment.outputs.page_url }} + + steps: + - name: Deploy + id: deployment + uses: actions/deploy-pages@v2 diff --git a/.gitignore b/.gitignore new file mode 100644 index 0000000..6635cf5 --- /dev/null +++ b/.gitignore @@ -0,0 +1,10 @@ +.DS_Store +node_modules +/build +/.svelte-kit +/package +.env +.env.* +!.env.example +vite.config.js.timestamp-* +vite.config.ts.timestamp-* diff --git a/.npmrc b/.npmrc new file mode 100644 index 0000000..b6f27f1 --- /dev/null +++ b/.npmrc @@ -0,0 +1 @@ +engine-strict=true diff --git a/.prettierignore b/.prettierignore new file mode 100644 index 0000000..cc41cea --- /dev/null +++ b/.prettierignore @@ -0,0 +1,4 @@ +# Ignore files for PNPM, NPM and YARN +pnpm-lock.yaml +package-lock.json +yarn.lock diff --git a/.prettierrc b/.prettierrc new file mode 100644 index 0000000..9573023 --- /dev/null +++ b/.prettierrc @@ -0,0 +1,8 @@ +{ + "useTabs": true, + "singleQuote": true, + "trailingComma": "none", + "printWidth": 100, + "plugins": ["prettier-plugin-svelte"], + "overrides": [{ "files": "*.svelte", "options": { "parser": "svelte" } }] +} diff --git a/LICENSE b/LICENSE new file mode 100644 index 0000000..8e5ff78 --- /dev/null +++ b/LICENSE @@ -0,0 +1,21 @@ +MIT License Copyright (c) 2024 flo-bit + +Permission is hereby granted, free of +charge, to any person obtaining a copy of this software and associated +documentation files (the "Software"), to deal in the Software without +restriction, including without limitation the rights to use, copy, modify, merge, +publish, distribute, sublicense, and/or sell copies of the Software, and to +permit persons to whom the Software is furnished to do so, subject to the +following conditions: + +The above copyright notice and this permission notice +(including the next paragraph) shall be included in all copies or substantial +portions of the Software. + +THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF +ANY KIND, EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF +MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO +EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR +OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING +FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN +THE SOFTWARE. diff --git a/README.md b/README.md new file mode 100644 index 0000000..e99b347 --- /dev/null +++ b/README.md @@ -0,0 +1,126 @@ +# sveltekit openai realtime api + +this is a sveltekit port of the [openai-realtime-console](https://github.com/openai/openai-realtime-console) repository. + +it allows you to easily use the openai realtime api in your sveltekit project. + +work in progress, but it should be functional. + +## how to use + +> [!WARNING] +> this is the setup for the client side only version of the realtime api, you will need to use a relay server to use this in production. +> see further down for more information on how to use a relay server. + +1. copy the `src/lib/realtime/` folder from this repository into your sveltekit projects `src/lib/` folder + +2. install the dependency: + +```bash +$ npm i openai/openai-realtime-api-beta +``` + +2. import the `Realtime` component into your svelte file and use it e.g. like so: + +```svelte + + +{#if apiKey} + +{/if} +``` + +see `src/routes/+page.svelte` for a full example. + +## relay server + +for production use, you will need to use a relay server to use the realtime api. + +1. add `OPENAI_API_KEY` to your `.env` file + +2. tell the Realtime component to use the relay server: + +```svelte + +``` + +then you have two options: + +### run the relay server with your sveltekit server + +change your `vite.config.ts` to this: + +```ts +import { realtimeWebSocketServer } from './src/lib/realtime/realtime_server'; +import { sveltekit } from '@sveltejs/kit/vite'; +import { defineConfig } from 'vite'; + +export default defineConfig({ + plugins: [sveltekit(), realtimeWebSocketServer] +}); +``` + +### run the relay server independently + +- copy the `relay-server/` folder (identical to the code in the + [openai-realtime-console](https://github.com/openai/openai-realtime-console) repository) + from this repository into your project + +- install the dependencies: + +```bash +npm i dotenv openai/openai-realtime-api-beta +``` + +- run the relay server: + +```bash +$ node relay-server/index.js +``` + +- add the relay server url to your Realtime component: + +``` + +``` + +## todos + +- [ ] add tests +- [ ] add more documentation +- [ ] show waveforms +- [ ] tool calling +- [ ] update ui + +## license + +MIT diff --git a/eslint.config.js b/eslint.config.js new file mode 100644 index 0000000..a351fa9 --- /dev/null +++ b/eslint.config.js @@ -0,0 +1,33 @@ +import js from '@eslint/js'; +import ts from 'typescript-eslint'; +import svelte from 'eslint-plugin-svelte'; +import prettier from 'eslint-config-prettier'; +import globals from 'globals'; + +/** @type {import('eslint').Linter.FlatConfig[]} */ +export default [ + js.configs.recommended, + ...ts.configs.recommended, + ...svelte.configs['flat/recommended'], + prettier, + ...svelte.configs['flat/prettier'], + { + languageOptions: { + globals: { + ...globals.browser, + ...globals.node + } + } + }, + { + files: ['**/*.svelte'], + languageOptions: { + parserOptions: { + parser: ts.parser + } + } + }, + { + ignores: ['build/', '.svelte-kit/', 'dist/'] + } +]; diff --git a/package-lock.json b/package-lock.json new file mode 100644 index 0000000..7b6c940 --- /dev/null +++ b/package-lock.json @@ -0,0 +1,4085 @@ +{ + "name": "visualization", + "version": "0.0.1", + "lockfileVersion": 3, + "requires": true, + "packages": { + "": { + "name": "visualization", + "version": "0.0.1", + "dependencies": { + "@openai/realtime-api-beta": "github:openai/openai-realtime-api-beta", + "@sveltejs/adapter-static": "^3.0.5", + "dotenv": "^16.4.5" + }, + "devDependencies": { + "@sveltejs/adapter-auto": "^3.0.0", + "@sveltejs/adapter-node": "^5.2.5", + "@sveltejs/kit": "^2.0.0", + 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"@types/ws": "^8.5.12", + "autoprefixer": "^10.4.19", + "eslint": "^9.0.0", + "eslint-config-prettier": "^9.1.0", + "eslint-plugin-svelte": "^2.36.0", + "globals": "^15.0.0", + "postcss": "^8.4.38", + "prettier": "^3.1.1", + "prettier-plugin-svelte": "^3.1.2", + "svelte": "^4.2.7", + "svelte-check": "^3.6.0", + "tailwindcss": "^3.4.3", + "tslib": "^2.4.1", + "typescript": "^5.0.0", + "typescript-eslint": "^8.0.0-alpha.20", + "vite": "^5.0.3" + }, + "type": "module", + "dependencies": { + "@openai/realtime-api-beta": "github:openai/openai-realtime-api-beta", + "@sveltejs/adapter-static": "^3.0.5", + "dotenv": "^16.4.5" + }, + "license": "MIT" +} \ No newline at end of file diff --git a/postcss.config.js b/postcss.config.js new file mode 100644 index 0000000..2e7af2b --- /dev/null +++ b/postcss.config.js @@ -0,0 +1,6 @@ +export default { + plugins: { + tailwindcss: {}, + autoprefixer: {}, + }, +} diff --git a/relay-server/index.js b/relay-server/index.js new file mode 100644 index 0000000..4160921 --- /dev/null +++ b/relay-server/index.js @@ -0,0 +1,18 @@ +import { RealtimeRelay } from './lib/relay.js'; +import dotenv from 'dotenv'; +dotenv.config({ override: true }); + +const OPENAI_API_KEY = process.env.OPENAI_API_KEY; + +if (!OPENAI_API_KEY) { + console.error( + `Environment variable "OPENAI_API_KEY" is required.\n` + + `Please set it in your .env file.` + ); + process.exit(1); +} + +const PORT = parseInt(process.env.PORT) || 8081; + +const relay = new RealtimeRelay(OPENAI_API_KEY); +relay.listen(PORT); diff --git a/relay-server/lib/relay.js b/relay-server/lib/relay.js new file mode 100644 index 0000000..39dc309 --- /dev/null +++ b/relay-server/lib/relay.js @@ -0,0 +1,84 @@ +import { WebSocketServer } from 'ws'; +import { RealtimeClient } from '@openai/realtime-api-beta'; + +export class RealtimeRelay { + constructor(apiKey) { + this.apiKey = apiKey; + this.sockets = new WeakMap(); + this.wss = null; + } + + listen(port) { + this.wss = new WebSocketServer({ port }); + this.wss.on('connection', this.connectionHandler.bind(this)); + this.log(`Listening on ws://localhost:${port}`); + } + + async connectionHandler(ws, req) { + if (!req.url) { + this.log('No URL provided, closing connection.'); + ws.close(); + return; + } + + const url = new URL(req.url, `http://${req.headers.host}`); + const pathname = url.pathname; + + if (pathname !== '/') { + this.log(`Invalid pathname: "${pathname}"`); + ws.close(); + return; + } + + // Instantiate new client + this.log(`Connecting with key "${this.apiKey.slice(0, 3)}..."`); + const client = new RealtimeClient({ apiKey: this.apiKey }); + + // Relay: OpenAI Realtime API Event -> Browser Event + client.realtime.on('server.*', (event) => { + this.log(`Relaying "${event.type}" to Client`); + ws.send(JSON.stringify(event)); + }); + client.realtime.on('close', () => ws.close()); + + // Relay: Browser Event -> OpenAI Realtime API Event + // We need to queue data waiting for the OpenAI connection + const messageQueue = []; + const messageHandler = (data) => { + try { + const event = JSON.parse(data); + this.log(`Relaying "${event.type}" to OpenAI`); + client.realtime.send(event.type, event); + } catch (e) { + console.error(e.message); + this.log(`Error parsing event from client: ${data}`); + } + }; + ws.on('message', (data) => { + if (!client.isConnected()) { + messageQueue.push(data); + } else { + messageHandler(data); + } + }); + ws.on('close', () => client.disconnect()); + + // Connect to OpenAI Realtime API + try { + this.log(`Connecting to OpenAI...`); + await client.connect(); + } catch (e) { + this.log(`Error connecting to OpenAI: ${e.message}`); + ws.close(); + return; + } + this.log(`Connected to OpenAI successfully!`); + while (messageQueue.length) { + messageHandler(messageQueue.shift()); + } + } + + log(...args) { + console.log(`[RealtimeRelay]`, ...args); + } +} diff --git a/src/app.css b/src/app.css new file mode 100644 index 0000000..b5c61c9 --- /dev/null +++ b/src/app.css @@ -0,0 +1,3 @@ +@tailwind base; +@tailwind components; +@tailwind utilities; diff --git a/src/app.d.ts b/src/app.d.ts new file mode 100644 index 0000000..743f07b --- /dev/null +++ b/src/app.d.ts @@ -0,0 +1,13 @@ +// See https://kit.svelte.dev/docs/types#app +// for information about these interfaces +declare global { + namespace App { + // interface Error {} + // interface Locals {} + // interface PageData {} + // interface PageState {} + // interface Platform {} + } +} + +export {}; diff --git a/src/app.html b/src/app.html new file mode 100644 index 0000000..8ce1ebe --- /dev/null +++ b/src/app.html @@ -0,0 +1,18 @@ + + + + + + + + + svelte realtime api + %sveltekit.head% + + +
%sveltekit.body%
+ + diff --git a/src/lib/index.ts b/src/lib/index.ts new file mode 100644 index 0000000..856f2b6 --- /dev/null +++ b/src/lib/index.ts @@ -0,0 +1 @@ +// place files you want to import through the `$lib` alias in this folder. diff --git a/src/lib/realtime/realtime.svelte b/src/lib/realtime/realtime.svelte new file mode 100644 index 0000000..e328b61 --- /dev/null +++ b/src/lib/realtime/realtime.svelte @@ -0,0 +1,208 @@ + diff --git a/src/lib/realtime/realtime_server.ts b/src/lib/realtime/realtime_server.ts new file mode 100644 index 0000000..860920e --- /dev/null +++ b/src/lib/realtime/realtime_server.ts @@ -0,0 +1,73 @@ +import { WebSocketServer } from 'ws'; +import { RealtimeClient } from '@openai/realtime-api-beta'; +import { loadEnv } from 'vite'; + +export const realtimeWebSocketServer = { + name: 'realtimeWebSocketServer', + configureServer(server) { + console.log('Configuring WebSocket Server...'); + if (!server.httpServer) return; + + const wss = new WebSocketServer({ noServer: true }); + + server.httpServer.on('upgrade', (request, socket, head) => { + // if starts with /api/chat, then upgrade the connection + if (request.url?.startsWith('/api/realtime')) { + wss.handleUpgrade(request, socket, head, (ws) => { + wss.emit('connection', ws, request); + }); + } + }); + + wss.on('connection', async (ws) => { + const env = loadEnv(process.env.NODE_ENV, process.cwd(), ''); + + const apiKey = env.OPENAI_API_KEY ?? 'your-fallback-api-key'; + console.log(`Connecting with key "${apiKey.slice(0, 10)}..."`); + + const messageQueue = []; + + const client = new RealtimeClient({ apiKey }); + + const messageHandler = (data) => { + try { + const event = JSON.parse(data); + console.log(`Relaying "${event.type}" to OpenAI`); + client.realtime.send(event.type, event); + } catch (e) { + console.error(e.message); + console.log(`Error parsing event from client: ${data}`); + } + }; + + client.realtime.on('server.*', (event) => { + console.log(`Relaying "${event.type}" to Client`); + ws.send(JSON.stringify(event)); + }); + + client.realtime.on('close', () => ws.close()); + + ws.on('message', (data) => { + if (!client.isConnected()) { + messageQueue.push(data); + } else { + messageHandler(data); + } + }); + + ws.on('close', () => client.disconnect()); + + try { + console.log('Connecting to OpenAI...'); + await client.connect(); + console.log('Connected to OpenAI successfully!'); + } catch (e) { + console.log(`Error connecting to OpenAI: ${e.message}`); + ws.close(); + } + while (messageQueue.length) { + messageHandler(messageQueue.shift()); + } + }); + } +}; diff --git a/src/lib/realtime/wavtools/dist/index.d.ts b/src/lib/realtime/wavtools/dist/index.d.ts new file mode 100644 index 0000000..9529532 --- /dev/null +++ b/src/lib/realtime/wavtools/dist/index.d.ts @@ -0,0 +1,6 @@ +import { AudioAnalysis } from './lib/analysis/audio_analysis.js'; +import { WavPacker } from './lib/wav_packer.js'; +import { WavStreamPlayer } from './lib/wav_stream_player.js'; +import { WavRecorder } from './lib/wav_recorder.js'; +export { AudioAnalysis, WavPacker, WavStreamPlayer, WavRecorder }; +//# sourceMappingURL=index.d.ts.map \ No newline at end of file diff --git a/src/lib/realtime/wavtools/dist/index.d.ts.map b/src/lib/realtime/wavtools/dist/index.d.ts.map new file mode 100644 index 0000000..a80c055 --- /dev/null +++ b/src/lib/realtime/wavtools/dist/index.d.ts.map @@ -0,0 +1 @@ +{"version":3,"file":"index.d.ts","sourceRoot":"","sources":["../index.js"],"names":[],"mappings":"8BAC8B,kCAAkC;0BADtC,qBAAqB;gCAEf,4BAA4B;4BAChC,uBAAuB"} \ No newline at end of file diff --git a/src/lib/realtime/wavtools/dist/lib/analysis/audio_analysis.d.ts b/src/lib/realtime/wavtools/dist/lib/analysis/audio_analysis.d.ts new file mode 100644 index 0000000..fc50758 --- /dev/null +++ b/src/lib/realtime/wavtools/dist/lib/analysis/audio_analysis.d.ts @@ -0,0 +1,70 @@ +/** + * 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: AnalyserNode, sampleRate: number, fftResult?: Float32Array, analysisType?: "frequency" | "music" | "voice", minDecibels?: number, maxDecibels?: number): AudioAnalysisOutputType; + /** + * 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: HTMLAudioElement, audioBuffer?: AudioBuffer | null); + fftResults: any[]; + audio: HTMLAudioElement; + context: any; + analyser: any; + sampleRate: any; + audioBuffer: any; + /** + * 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" | "music" | "voice", minDecibels?: number, maxDecibels?: number): AudioAnalysisOutputType; + /** + * Resume the internal AudioContext if it was suspended due to the lack of + * user interaction when the AudioAnalysis was instantiated. + * @returns {Promise} + */ + resumeIfSuspended(): Promise; +} +/** + * Output of AudioAnalysis for the frequency domain of the audio + */ +export type AudioAnalysisOutputType = { + /** + * Amplitude of this frequency between {0, 1} inclusive + */ + values: Float32Array; + /** + * Raw frequency bucket values + */ + frequencies: number[]; + /** + * Labels for the frequency bucket values + */ + labels: string[]; +}; +//# sourceMappingURL=audio_analysis.d.ts.map \ No newline at end of file diff --git a/src/lib/realtime/wavtools/dist/lib/analysis/audio_analysis.d.ts.map b/src/lib/realtime/wavtools/dist/lib/analysis/audio_analysis.d.ts.map new file mode 100644 index 0000000..abb292b --- /dev/null +++ b/src/lib/realtime/wavtools/dist/lib/analysis/audio_analysis.d.ts.map @@ -0,0 +1 @@ +{"version":3,"file":"audio_analysis.d.ts","sourceRoot":"","sources":["../../../lib/analysis/audio_analysis.js"],"names":[],"mappings":"AAOA;;;;;;GAMG;AAEH;;;GAGG;AACH;IACE;;;;;;;;;;OAUG;IACH,gCARW,YAAY,cACZ,MAAM,cACN,YAAY,iBACZ,WAAW,GAAC,OAAO,GAAC,OAAO,gBAC3B,MAAM,gBACN,MAAM,GACJ,uBAAuB,CAwDnC;IAED;;;;;OAKG;IACH,0BAJW,gBAAgB,gBAChB,WAAW,GAAC,IAAI,EAkE1B;IA9DC,kBAAoB;IA2ClB,wBAAyB;IACzB,aAAkC;IAClC,cAAwB;IACxB,gBAA4B;IAC5B,iBAA8B;IAiBlC;;;;;;OAMG;IACH,8BALW,WAAW,GAAC,OAAO,GAAC,OAAO,gBAC3B,MAAM,gBACN,MAAM,GACJ,uBAAuB,CAwBnC;IAED;;;;OAIG;IACH,qBAFa,OAAO,CAAC,IAAI,CAAC,CAOzB;CACF;;;;;;;;YA9La,YAAY;;;;iBACZ,MAAM,EAAE;;;;YACR,MAAM,EAAE"} \ No newline at end of file diff --git a/src/lib/realtime/wavtools/dist/lib/analysis/constants.d.ts b/src/lib/realtime/wavtools/dist/lib/analysis/constants.d.ts new file mode 100644 index 0000000..868ba15 --- /dev/null +++ b/src/lib/realtime/wavtools/dist/lib/analysis/constants.d.ts @@ -0,0 +1,9 @@ +/** + * All note frequencies from 1st to 8th octave + * in format "A#8" (A#, 8th octave) + */ +export const noteFrequencies: any[]; +export const noteFrequencyLabels: any[]; +export const voiceFrequencies: any[]; +export const voiceFrequencyLabels: any[]; +//# sourceMappingURL=constants.d.ts.map \ No newline at end of file diff --git a/src/lib/realtime/wavtools/dist/lib/analysis/constants.d.ts.map b/src/lib/realtime/wavtools/dist/lib/analysis/constants.d.ts.map new file mode 100644 index 0000000..0f5d851 --- /dev/null +++ b/src/lib/realtime/wavtools/dist/lib/analysis/constants.d.ts.map @@ -0,0 +1 @@ +{"version":3,"file":"constants.d.ts","sourceRoot":"","sources":["../../../lib/analysis/constants.js"],"names":[],"mappings":"AA6BA;;;GAGG;AACH,oCAAkC;AAClC,wCAAsC;AActC,qCAKG;AACH,yCAKG"} \ No newline at end of file diff --git a/src/lib/realtime/wavtools/dist/lib/wav_packer.d.ts b/src/lib/realtime/wavtools/dist/lib/wav_packer.d.ts new file mode 100644 index 0000000..4fe1187 --- /dev/null +++ b/src/lib/realtime/wavtools/dist/lib/wav_packer.d.ts @@ -0,0 +1,58 @@ +/** + * 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: Float32Array): ArrayBuffer; + /** + * Concatenates two ArrayBuffers + * @param {ArrayBuffer} leftBuffer + * @param {ArrayBuffer} rightBuffer + * @returns {ArrayBuffer} + */ + static mergeBuffers(leftBuffer: ArrayBuffer, rightBuffer: ArrayBuffer): ArrayBuffer; + /** + * Packs data into an Int16 format + * @private + * @param {number} size 0 = 1x Int16, 1 = 2x Int16 + * @param {number} arg value to pack + * @returns + */ + private _packData; + /** + * Packs audio into "audio/wav" Blob + * @param {number} sampleRate + * @param {{bitsPerSample: number, channels: Array, data: Int16Array}} audio + * @returns {WavPackerAudioType} + */ + pack(sampleRate: number, audio: { + bitsPerSample: number; + channels: Array; + data: Int16Array; + }): WavPackerAudioType; +} +/** + * Raw wav audio file contents + */ +export type WavPackerAudioType = { + blob: Blob; + url: string; + channelCount: number; + sampleRate: number; + duration: number; +}; +//# sourceMappingURL=wav_packer.d.ts.map \ No newline at end of file diff --git a/src/lib/realtime/wavtools/dist/lib/wav_packer.d.ts.map b/src/lib/realtime/wavtools/dist/lib/wav_packer.d.ts.map new file mode 100644 index 0000000..96477a9 --- /dev/null +++ b/src/lib/realtime/wavtools/dist/lib/wav_packer.d.ts.map @@ -0,0 +1 @@ +{"version":3,"file":"wav_packer.d.ts","sourceRoot":"","sources":["../../lib/wav_packer.js"],"names":[],"mappings":"AAAA;;;;;;;;GAQG;AAEH;;;GAGG;AACH;IACE;;;;OAIG;IACH,qCAHW,YAAY,GACV,WAAW,CAWvB;IAED;;;;;OAKG;IACH,gCAJW,WAAW,eACX,WAAW,GACT,WAAW,CASvB;IAED;;;;;;OAMG;IACH,kBAKC;IAED;;;;;OAKG;IACH,iBAJW,MAAM,SACN;QAAC,aAAa,EAAE,MAAM,CAAC;QAAC,QAAQ,EAAE,KAAK,CAAC,YAAY,CAAC,CAAC;QAAC,IAAI,EAAE,UAAU,CAAA;KAAC,GACtE,kBAAkB,CA6C9B;CACF;;;;;UA3Ga,IAAI;SACJ,MAAM;kBACN,MAAM;gBACN,MAAM;cACN,MAAM"} \ No newline at end of file diff --git a/src/lib/realtime/wavtools/dist/lib/wav_recorder.d.ts b/src/lib/realtime/wavtools/dist/lib/wav_recorder.d.ts new file mode 100644 index 0000000..03cd269 --- /dev/null +++ b/src/lib/realtime/wavtools/dist/lib/wav_recorder.d.ts @@ -0,0 +1,167 @@ +/** + * 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 { + /** + * 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} + */ + static decode(audioData: Blob | Float32Array | Int16Array | ArrayBuffer | number[], sampleRate?: number, fromSampleRate?: number): Promise; + /** + * Create a new WavRecorder instance + * @param {{sampleRate?: number, outputToSpeakers?: boolean, debug?: boolean}} [options] + * @returns {WavRecorder} + */ + constructor({ sampleRate, outputToSpeakers, debug, }?: { + sampleRate?: number; + outputToSpeakers?: boolean; + debug?: boolean; + }); + scriptSrc: any; + sampleRate: number; + outputToSpeakers: boolean; + debug: boolean; + _deviceChangeCallback: () => Promise; + _devices: any[]; + stream: any; + processor: any; + source: any; + node: any; + recording: boolean; + _lastEventId: number; + eventReceipts: {}; + eventTimeout: number; + _chunkProcessor: () => void; + _chunkProcessorBuffer: { + raw: ArrayBuffer; + mono: ArrayBuffer; + }; + /** + * Logs data in debug mode + * @param {...any} arguments + * @returns {true} + */ + log(...args: any[]): true; + /** + * Retrieves the current sampleRate for the recorder + * @returns {number} + */ + getSampleRate(): number; + /** + * Retrieves the current status of the recording + * @returns {"ended"|"paused"|"recording"} + */ + getStatus(): "ended" | "paused" | "recording"; + /** + * Sends an event to the AudioWorklet + * @private + * @param {string} name + * @param {{[key: string]: any}} data + * @param {AudioWorkletNode} [_processor] + * @returns {Promise<{[key: string]: any}>} + */ + private _event; + /** + * Sets device change callback, remove if callback provided is `null` + * @param {(Array): void|null} callback + * @returns {true} + */ + listenForDeviceChange(callback: any): true; + /** + * Manually request permission to use the microphone + * @returns {Promise} + */ + requestPermission(): Promise; + /** + * List all eligible devices for recording, will request permission to use microphone + * @returns {Promise>} + */ + listDevices(): Promise>; + /** + * 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} + */ + begin(deviceId?: string): Promise; + analyser: any; + /** + * 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" | "music" | "voice", minDecibels?: number, maxDecibels?: number): import("./analysis/audio_analysis.js").AudioAnalysisOutputType; + /** + * Pauses the recording + * Keeps microphone stream open but halts storage of audio + * @returns {Promise} + */ + pause(): Promise; + /** + * 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} + */ + record(chunkProcessor?: (data: { + mono: Int16Array; + raw: Int16Array; + }) => any, chunkSize?: number): Promise; + _chunkProcessorSize: number; + /** + * Clears the audio buffer, empties stored recording + * @returns {Promise} + */ + clear(): Promise; + /** + * Reads the current audio stream data + * @returns {Promise<{meanValues: Float32Array, channels: Array}>} + */ + read(): Promise<{ + meanValues: Float32Array; + channels: Array; + }>; + /** + * Saves the current audio stream to a file + * @param {boolean} [force] Force saving while still recording + * @returns {Promise} + */ + save(force?: boolean): Promise; + /** + * Ends the current recording session and saves the result + * @returns {Promise} + */ + end(): Promise; + /** + * Performs a full cleanup of WavRecorder instance + * Stops actively listening via microphone and removes existing listeners + * @returns {Promise} + */ + quit(): Promise; +} +/** + * Decodes audio into a wav file + */ +export type DecodedAudioType = { + blob: Blob; + url: string; + values: Float32Array; + audioBuffer: AudioBuffer; +}; +//# sourceMappingURL=wav_recorder.d.ts.map \ No newline at end of file diff --git a/src/lib/realtime/wavtools/dist/lib/wav_recorder.d.ts.map b/src/lib/realtime/wavtools/dist/lib/wav_recorder.d.ts.map new file mode 100644 index 0000000..7954106 --- /dev/null +++ b/src/lib/realtime/wavtools/dist/lib/wav_recorder.d.ts.map @@ -0,0 +1 @@ +{"version":3,"file":"wav_recorder.d.ts","sourceRoot":"","sources":["../../lib/wav_recorder.js"],"names":[],"mappings":"AAIA;;;;;;;GAOG;AAEH;;;GAGG;AACH;IAsCE;;;;;;OAMG;IACH,yBALW,IAAI,GAAC,YAAY,GAAC,UAAU,GAAC,WAAW,GAAC,MAAM,EAAE,eACjD,MAAM,mBACN,MAAM,GACJ,OAAO,CAAC,gBAAgB,CAAC,CAqErC;IA/GD;;;;OAIG;IACH,uDAHW;QAAC,UAAU,CAAC,EAAE,MAAM,CAAC;QAAC,gBAAgB,CAAC,EAAE,OAAO,CAAC;QAAC,KAAK,CAAC,EAAE,OAAO,CAAA;KAAC,EAiC5E;IAxBC,eAAkC;IAElC,mBAA4B;IAC5B,0BAAwC;IACxC,eAAoB;IACpB,2CAAiC;IACjC,gBAAkB;IAElB,YAAkB;IAClB,eAAqB;IACrB,YAAkB;IAClB,UAAgB;IAChB,mBAAsB;IAEtB,qBAAqB;IACrB,kBAAuB;IACvB,qBAAwB;IAExB,4BAA+B;IAE/B;;;MAGC;IA+EH;;;;OAIG;IACH,qBAFa,IAAI,CAOhB;IAED;;;OAGG;IACH,iBAFa,MAAM,CAIlB;IAED;;;OAGG;IACH,aAFa,OAAO,GAAC,QAAQ,GAAC,WAAW,CAUxC;IAED;;;;;;;OAOG;IACH,eAqBC;IAED;;;;OAIG;IACH,sCAFa,IAAI,CAmChB;IAED;;;OAGG;IACH,qBAFa,OAAO,CAAC,IAAI,CAAC,CAoBzB;IAED;;;OAGG;IACH,eAFa,OAAO,CAAC,KAAK,CAAC,eAAe,GAAG;QAAC,OAAO,EAAE,OAAO,CAAA;KAAC,CAAC,CAAC,CA8BhE;IAED;;;;;OAKG;IACH,iBAHW,MAAM,GACJ,OAAO,CAAC,IAAI,CAAC,CAkFzB;IAHC,cAAwB;IAK1B;;;;;;OAMG;IACH,8BALW,WAAW,GAAC,OAAO,GAAC,OAAO,gBAC3B,MAAM,gBACN,MAAM,GACJ,OAAO,8BAA8B,EAAE,uBAAuB,CAkB1E;IAED;;;;OAIG;IACH,SAFa,OAAO,CAAC,IAAI,CAAC,CAezB;IAED;;;;;OAKG;IACH,wBAJW,CAAC,IAAI,EAAE;QAAE,IAAI,EAAE,UAAU,CAAC;QAAC,GAAG,EAAE,UAAU,CAAA;KAAE,KAAK,GAAG,cACpD,MAAM,GACJ,OAAO,CAAC,IAAI,CAAC,CAoBzB;IATC,4BAAoC;IAWtC;;;OAGG;IACH,SAFa,OAAO,CAAC,IAAI,CAAC,CAQzB;IAED;;;OAGG;IACH,QAFa,OAAO,CAAC;QAAC,UAAU,EAAE,YAAY,CAAC;QAAC,QAAQ,EAAE,KAAK,CAAC,YAAY,CAAC,CAAA;KAAC,CAAC,CAS9E;IAED;;;;OAIG;IACH,aAHW,OAAO,GACL,OAAO,CAAC,OAAO,iBAAiB,EAAE,kBAAkB,CAAC,CAgBjE;IAED;;;OAGG;IACH,OAFa,OAAO,CAAC,OAAO,iBAAiB,EAAE,kBAAkB,CAAC,CA8BjE;IAED;;;;OAIG;IACH,QAFa,OAAO,CAAC,IAAI,CAAC,CAQzB;CACF;;;;;UA1hBa,IAAI;SACJ,MAAM;YACN,YAAY;iBACZ,WAAW"} \ No newline at end of file diff --git a/src/lib/realtime/wavtools/dist/lib/wav_stream_player.d.ts b/src/lib/realtime/wavtools/dist/lib/wav_stream_player.d.ts new file mode 100644 index 0000000..91a2263 --- /dev/null +++ b/src/lib/realtime/wavtools/dist/lib/wav_stream_player.d.ts @@ -0,0 +1,69 @@ +/** + * 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 }?: { + sampleRate?: number; + }); + scriptSrc: any; + sampleRate: number; + context: any; + stream: any; + analyser: any; + trackSampleOffsets: {}; + interruptedTrackIds: {}; + /** + * Connects the audio context and enables output to speakers + * @returns {Promise} + */ + connect(): Promise; + /** + * 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" | "music" | "voice", minDecibels?: number, maxDecibels?: number): import("./analysis/audio_analysis.js").AudioAnalysisOutputType; + /** + * Starts audio streaming + * @private + * @returns {Promise} + */ + private _start; + /** + * 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: ArrayBuffer | Int16Array, trackId?: string): Int16Array; + /** + * Gets the offset (sample count) of the currently playing stream + * @param {boolean} [interrupt] + * @returns {{trackId: string|null, offset: number, currentTime: number}} + */ + getTrackSampleOffset(interrupt?: boolean): { + trackId: string | null; + offset: number; + currentTime: number; + }; + /** + * Strips the current stream and returns the sample offset of the audio + * @param {boolean} [interrupt] + * @returns {{trackId: string|null, offset: number, currentTime: number}} + */ + interrupt(): { + trackId: string | null; + offset: number; + currentTime: number; + }; +} +//# sourceMappingURL=wav_stream_player.d.ts.map \ No newline at end of file diff --git a/src/lib/realtime/wavtools/dist/lib/wav_stream_player.d.ts.map b/src/lib/realtime/wavtools/dist/lib/wav_stream_player.d.ts.map new file mode 100644 index 0000000..500126c --- /dev/null +++ b/src/lib/realtime/wavtools/dist/lib/wav_stream_player.d.ts.map @@ -0,0 +1 @@ +{"version":3,"file":"wav_stream_player.d.ts","sourceRoot":"","sources":["../../lib/wav_stream_player.js"],"names":[],"mappings":"AAGA;;;GAGG;AACH;IACE;;;;OAIG;IACH,6BAHW;QAAC,UAAU,CAAC,EAAE,MAAM,CAAA;KAAC,EAW/B;IAPC,eAAmC;IACnC,mBAA4B;IAC5B,aAAmB;IACnB,YAAkB;IAClB,cAAoB;IACpB,uBAA4B;IAC5B,wBAA6B;IAG/B;;;OAGG;IACH,WAFa,OAAO,CAAC,IAAI,CAAC,CAkBzB;IAED;;;;;;OAMG;IACH,8BALW,WAAW,GAAC,OAAO,GAAC,OAAO,gBAC3B,MAAM,gBACN,MAAM,GACJ,OAAO,8BAA8B,EAAE,uBAAuB,CAkB1E;IAED;;;;OAIG;IACH,eAkBC;IAED;;;;;;OAMG;IACH,yBAJW,WAAW,GAAC,UAAU,YACtB,MAAM,GACJ,UAAU,CAqBtB;IAED;;;;OAIG;IACH,iCAHW,OAAO,GACL;QAAC,OAAO,EAAE,MAAM,GAAC,IAAI,CAAC;QAAC,MAAM,EAAE,MAAM,CAAC;QAAC,WAAW,EAAE,MAAM,CAAA;KAAC,CAqBvE;IAED;;;;OAIG;IACH,aAFa;QAAC,OAAO,EAAE,MAAM,GAAC,IAAI,CAAC;QAAC,MAAM,EAAE,MAAM,CAAC;QAAC,WAAW,EAAE,MAAM,CAAA;KAAC,CAIvE;CACF"} \ No newline at end of file diff --git a/src/lib/realtime/wavtools/dist/lib/worklets/audio_processor.d.ts b/src/lib/realtime/wavtools/dist/lib/worklets/audio_processor.d.ts new file mode 100644 index 0000000..8b7c8ac --- /dev/null +++ b/src/lib/realtime/wavtools/dist/lib/worklets/audio_processor.d.ts @@ -0,0 +1,2 @@ +export const AudioProcessorSrc: any; +//# sourceMappingURL=audio_processor.d.ts.map \ No newline at end of file diff --git a/src/lib/realtime/wavtools/dist/lib/worklets/audio_processor.d.ts.map b/src/lib/realtime/wavtools/dist/lib/worklets/audio_processor.d.ts.map new file mode 100644 index 0000000..d651100 --- /dev/null +++ b/src/lib/realtime/wavtools/dist/lib/worklets/audio_processor.d.ts.map @@ -0,0 +1 @@ +{"version":3,"file":"audio_processor.d.ts","sourceRoot":"","sources":["../../../lib/worklets/audio_processor.js"],"names":[],"mappings":"AAqNA,oCAAqC"} \ No newline at end of file diff --git a/src/lib/realtime/wavtools/dist/lib/worklets/stream_processor.d.ts b/src/lib/realtime/wavtools/dist/lib/worklets/stream_processor.d.ts new file mode 100644 index 0000000..627da71 --- /dev/null +++ b/src/lib/realtime/wavtools/dist/lib/worklets/stream_processor.d.ts @@ -0,0 +1,3 @@ +export const StreamProcessorWorklet: "\nclass StreamProcessor extends AudioWorkletProcessor {\n constructor() {\n super();\n this.hasStarted = false;\n this.hasInterrupted = false;\n this.outputBuffers = [];\n this.bufferLength = 128;\n this.write = { buffer: new Float32Array(this.bufferLength), trackId: null };\n this.writeOffset = 0;\n this.trackSampleOffsets = {};\n this.port.onmessage = (event) => {\n if (event.data) {\n const payload = event.data;\n if (payload.event === 'write') {\n const int16Array = payload.buffer;\n const float32Array = new Float32Array(int16Array.length);\n for (let i = 0; i < int16Array.length; i++) {\n float32Array[i] = int16Array[i] / 0x8000; // Convert Int16 to Float32\n }\n this.writeData(float32Array, payload.trackId);\n } else if (\n payload.event === 'offset' ||\n payload.event === 'interrupt'\n ) {\n const requestId = payload.requestId;\n const trackId = this.write.trackId;\n const offset = this.trackSampleOffsets[trackId] || 0;\n this.port.postMessage({\n event: 'offset',\n requestId,\n trackId,\n offset,\n });\n if (payload.event === 'interrupt') {\n this.hasInterrupted = true;\n }\n } else {\n throw new Error(`Unhandled event \"${payload.event}\"`);\n }\n }\n };\n }\n\n writeData(float32Array, trackId = null) {\n let { buffer } = this.write;\n let offset = this.writeOffset;\n for (let i = 0; i < float32Array.length; i++) {\n buffer[offset++] = float32Array[i];\n if (offset >= buffer.length) {\n this.outputBuffers.push(this.write);\n this.write = { buffer: new Float32Array(this.bufferLength), trackId };\n buffer = this.write.buffer;\n offset = 0;\n }\n }\n this.writeOffset = offset;\n return true;\n }\n\n process(inputs, outputs, parameters) {\n const output = outputs[0];\n const outputChannelData = output[0];\n const outputBuffers = this.outputBuffers;\n if (this.hasInterrupted) {\n this.port.postMessage({ event: 'stop' });\n return false;\n } else if (outputBuffers.length) {\n this.hasStarted = true;\n const { buffer, trackId } = outputBuffers.shift();\n for (let i = 0; i < outputChannelData.length; i++) {\n outputChannelData[i] = buffer[i] || 0;\n }\n if (trackId) {\n this.trackSampleOffsets[trackId] =\n this.trackSampleOffsets[trackId] || 0;\n this.trackSampleOffsets[trackId] += buffer.length;\n }\n return true;\n } else if (this.hasStarted) {\n this.port.postMessage({ event: 'stop' });\n return false;\n } else {\n return true;\n }\n }\n}\n\nregisterProcessor('stream_processor', StreamProcessor);\n"; +export const StreamProcessorSrc: any; +//# sourceMappingURL=stream_processor.d.ts.map \ No newline at end of file diff --git a/src/lib/realtime/wavtools/dist/lib/worklets/stream_processor.d.ts.map b/src/lib/realtime/wavtools/dist/lib/worklets/stream_processor.d.ts.map new file mode 100644 index 0000000..c372e0b --- /dev/null +++ b/src/lib/realtime/wavtools/dist/lib/worklets/stream_processor.d.ts.map @@ -0,0 +1 @@ +{"version":3,"file":"stream_processor.d.ts","sourceRoot":"","sources":["../../../lib/worklets/stream_processor.js"],"names":[],"mappings":"AAAA,q4FAyFE;AAMF,qCAAsC"} \ No newline at end of file diff --git a/src/lib/realtime/wavtools/index.js b/src/lib/realtime/wavtools/index.js new file mode 100644 index 0000000..7123894 --- /dev/null +++ b/src/lib/realtime/wavtools/index.js @@ -0,0 +1,6 @@ +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 }; diff --git a/src/lib/realtime/wavtools/lib/analysis/audio_analysis.js b/src/lib/realtime/wavtools/lib/analysis/audio_analysis.js new file mode 100644 index 0000000..4af34d5 --- /dev/null +++ b/src/lib/realtime/wavtools/lib/analysis/audio_analysis.js @@ -0,0 +1,203 @@ +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} + */ + async resumeIfSuspended() { + if (this.context.state === 'suspended') { + await this.context.resume(); + } + return true; + } +} + +globalThis.AudioAnalysis = AudioAnalysis; diff --git a/src/lib/realtime/wavtools/lib/analysis/constants.js b/src/lib/realtime/wavtools/lib/analysis/constants.js new file mode 100644 index 0000000..f14da38 --- /dev/null +++ b/src/lib/realtime/wavtools/lib/analysis/constants.js @@ -0,0 +1,60 @@ +/** + * 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] + ); +}); diff --git a/src/lib/realtime/wavtools/lib/wav_packer.js b/src/lib/realtime/wavtools/lib/wav_packer.js new file mode 100644 index 0000000..7146b7f --- /dev/null +++ b/src/lib/realtime/wavtools/lib/wav_packer.js @@ -0,0 +1,113 @@ +/** + * 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, 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; diff --git a/src/lib/realtime/wavtools/lib/wav_recorder.js b/src/lib/realtime/wavtools/lib/wav_recorder.js new file mode 100644 index 0000000..a4f1d04 --- /dev/null +++ b/src/lib/realtime/wavtools/lib/wav_recorder.js @@ -0,0 +1,548 @@ +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} + */ + 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`, + ); + } + 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): 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} + */ + 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>} + */ + 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} + */ + 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} + */ + 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} + */ + 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} + */ + 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}>} + */ + 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} + */ + 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} + */ + 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} + */ + async quit() { + this.listenForDeviceChange(null); + if (this.processor) { + await this.end(); + } + return true; + } +} + +globalThis.WavRecorder = WavRecorder; diff --git a/src/lib/realtime/wavtools/lib/wav_stream_player.js b/src/lib/realtime/wavtools/lib/wav_stream_player.js new file mode 100644 index 0000000..500eff6 --- /dev/null +++ b/src/lib/realtime/wavtools/lib/wav_stream_player.js @@ -0,0 +1,160 @@ +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} + */ + 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} + */ + _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; diff --git a/src/lib/realtime/wavtools/lib/worklets/audio_processor.js b/src/lib/realtime/wavtools/lib/worklets/audio_processor.js new file mode 100644 index 0000000..61dd7ec --- /dev/null +++ b/src/lib/realtime/wavtools/lib/worklets/audio_processor.js @@ -0,0 +1,214 @@ +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; diff --git a/src/lib/realtime/wavtools/lib/worklets/stream_processor.js b/src/lib/realtime/wavtools/lib/worklets/stream_processor.js new file mode 100644 index 0000000..d3c794a --- /dev/null +++ b/src/lib/realtime/wavtools/lib/worklets/stream_processor.js @@ -0,0 +1,96 @@ +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; diff --git a/src/lib/ui/Layout.svelte b/src/lib/ui/Layout.svelte new file mode 100644 index 0000000..62b40cf --- /dev/null +++ b/src/lib/ui/Layout.svelte @@ -0,0 +1,15 @@ + + +
+
+ +
+
+ +
+
+ +
+
diff --git a/src/lib/ui/Switch.svelte b/src/lib/ui/Switch.svelte new file mode 100644 index 0000000..cf86ee2 --- /dev/null +++ b/src/lib/ui/Switch.svelte @@ -0,0 +1,25 @@ + + + + {#each options as option, index} + + {/each} + diff --git a/src/routes/+layout.svelte b/src/routes/+layout.svelte new file mode 100644 index 0000000..2e511e0 --- /dev/null +++ b/src/routes/+layout.svelte @@ -0,0 +1,5 @@ + + + diff --git a/src/routes/+layout.ts b/src/routes/+layout.ts new file mode 100644 index 0000000..189f71e --- /dev/null +++ b/src/routes/+layout.ts @@ -0,0 +1 @@ +export const prerender = true; diff --git a/src/routes/+page.svelte b/src/routes/+page.svelte new file mode 100644 index 0000000..880eafc --- /dev/null +++ b/src/routes/+page.svelte @@ -0,0 +1,124 @@ + + + +
+
+
svelte openai realtime api
+ + +
+
+ +
+ {#key items} + {#each items as item, index} + {#if item.formatted && index > 0} +
+ {item.role}: + {item.role === 'user' + ? item.formatted.transcript || '(empty)' + : item.formatted.transcript || '(truncated)'} +
+ {/if} + {/each} + {/key} +
+ +
+
+
+
turn detection
+ +
+ + {#if turnDetection === 'none' && isConnected} + + {/if} + + +
+
+ + +{#if apiKey} + +{/if} diff --git a/svelte.config.js b/svelte.config.js new file mode 100644 index 0000000..14a5c3d --- /dev/null +++ b/svelte.config.js @@ -0,0 +1,21 @@ +import adapter from '@sveltejs/adapter-static'; +import { vitePreprocess } from '@sveltejs/vite-plugin-svelte'; + +/** @type {import('@sveltejs/kit').Config} */ +const config = { + // Consult https://kit.svelte.dev/docs/integrations#preprocessors + // for more information about preprocessors + preprocess: vitePreprocess(), + + kit: { + // adapter-auto only supports some environments, see https://kit.svelte.dev/docs/adapter-auto for a list. + // If your environment is not supported, or you settled on a specific environment, switch out the adapter. + // See https://kit.svelte.dev/docs/adapters for more information about adapters. + adapter: adapter(), + paths: { + base: '/svelte-realtime-api' + } + } +}; + +export default config; diff --git a/tailwind.config.js b/tailwind.config.js new file mode 100644 index 0000000..bfe0bc2 --- /dev/null +++ b/tailwind.config.js @@ -0,0 +1,8 @@ +/** @type {import('tailwindcss').Config} */ +export default { + content: ['./src/**/*.{html,js,svelte,ts}'], + theme: { + extend: {} + }, + plugins: [] +}; diff --git a/tsconfig.json b/tsconfig.json new file mode 100644 index 0000000..fc93cbd --- /dev/null +++ b/tsconfig.json @@ -0,0 +1,19 @@ +{ + "extends": "./.svelte-kit/tsconfig.json", + "compilerOptions": { + "allowJs": true, + "checkJs": true, + "esModuleInterop": true, + "forceConsistentCasingInFileNames": true, + "resolveJsonModule": true, + "skipLibCheck": true, + "sourceMap": true, + "strict": true, + "moduleResolution": "bundler" + } + // Path aliases are handled by https://kit.svelte.dev/docs/configuration#alias + // except $lib which is handled by https://kit.svelte.dev/docs/configuration#files + // + // If you want to overwrite includes/excludes, make sure to copy over the relevant includes/excludes + // from the referenced tsconfig.json - TypeScript does not merge them in +} diff --git a/vite.config.ts b/vite.config.ts new file mode 100644 index 0000000..ccd3604 --- /dev/null +++ b/vite.config.ts @@ -0,0 +1,14 @@ +import { sveltekit } from '@sveltejs/kit/vite'; +import { defineConfig } from 'vite'; + +export default defineConfig({ + plugins: [sveltekit()] +}); + +// import { realtimeWebSocketServer } from './src/lib/realtime/realtime_server'; +// import { sveltekit } from '@sveltejs/kit/vite'; +// import { defineConfig } from 'vite'; + +// export default defineConfig({ +// plugins: [sveltekit(), realtimeWebSocketServer] +// });