From 943f3805ea823a99f26a5135f8cd97a852f2e0e3 Mon Sep 17 00:00:00 2001 From: Florian <45694132+flo-bit@users.noreply.github.com> Date: Fri, 4 Oct 2024 20:07:21 +0200 Subject: [PATCH] first commit --- .github/workflows/deploy_gh_pages.yml | 51 + .gitignore | 10 + .npmrc | 1 + .prettierignore | 4 + .prettierrc | 8 + LICENSE | 21 + README.md | 126 + eslint.config.js | 33 + package-lock.json | 4085 +++++++++++++++++ package.json | 44 + postcss.config.js | 6 + relay-server/index.js | 18 + relay-server/lib/relay.js | 84 + src/app.css | 3 + src/app.d.ts | 13 + src/app.html | 18 + src/lib/index.ts | 1 + src/lib/realtime/realtime.svelte | 208 + src/lib/realtime/realtime_server.ts | 73 + src/lib/realtime/wavtools/dist/index.d.ts | 6 + src/lib/realtime/wavtools/dist/index.d.ts.map | 1 + .../dist/lib/analysis/audio_analysis.d.ts | 70 + .../dist/lib/analysis/audio_analysis.d.ts.map | 1 + .../wavtools/dist/lib/analysis/constants.d.ts | 9 + .../dist/lib/analysis/constants.d.ts.map | 1 + .../wavtools/dist/lib/wav_packer.d.ts | 58 + .../wavtools/dist/lib/wav_packer.d.ts.map | 1 + .../wavtools/dist/lib/wav_recorder.d.ts | 167 + .../wavtools/dist/lib/wav_recorder.d.ts.map | 1 + .../wavtools/dist/lib/wav_stream_player.d.ts | 69 + .../dist/lib/wav_stream_player.d.ts.map | 1 + .../dist/lib/worklets/audio_processor.d.ts | 2 + .../lib/worklets/audio_processor.d.ts.map | 1 + .../dist/lib/worklets/stream_processor.d.ts | 3 + .../lib/worklets/stream_processor.d.ts.map | 1 + src/lib/realtime/wavtools/index.js | 6 + .../wavtools/lib/analysis/audio_analysis.js | 203 + .../wavtools/lib/analysis/constants.js | 60 + src/lib/realtime/wavtools/lib/wav_packer.js | 113 + src/lib/realtime/wavtools/lib/wav_recorder.js | 548 +++ .../wavtools/lib/wav_stream_player.js | 160 + .../wavtools/lib/worklets/audio_processor.js | 214 + .../wavtools/lib/worklets/stream_processor.js | 96 + src/lib/ui/Layout.svelte | 15 + src/lib/ui/Switch.svelte | 25 + src/routes/+layout.svelte | 5 + src/routes/+layout.ts | 1 + src/routes/+page.svelte | 124 + svelte.config.js | 21 + tailwind.config.js | 8 + tsconfig.json | 19 + vite.config.ts | 14 + 52 files changed, 6831 insertions(+) create mode 100644 .github/workflows/deploy_gh_pages.yml create mode 100644 .gitignore create mode 100644 .npmrc create mode 100644 .prettierignore create mode 100644 .prettierrc create mode 100644 LICENSE create mode 100644 README.md create mode 100644 eslint.config.js create mode 100644 package-lock.json create mode 100644 package.json create mode 100644 postcss.config.js create mode 100644 relay-server/index.js create mode 100644 relay-server/lib/relay.js create mode 100644 src/app.css create mode 100644 src/app.d.ts create mode 100644 src/app.html create mode 100644 src/lib/index.ts create mode 100644 src/lib/realtime/realtime.svelte create mode 100644 src/lib/realtime/realtime_server.ts create mode 100644 src/lib/realtime/wavtools/dist/index.d.ts create mode 100644 src/lib/realtime/wavtools/dist/index.d.ts.map create mode 100644 src/lib/realtime/wavtools/dist/lib/analysis/audio_analysis.d.ts create mode 100644 src/lib/realtime/wavtools/dist/lib/analysis/audio_analysis.d.ts.map create mode 100644 src/lib/realtime/wavtools/dist/lib/analysis/constants.d.ts create mode 100644 src/lib/realtime/wavtools/dist/lib/analysis/constants.d.ts.map create mode 100644 src/lib/realtime/wavtools/dist/lib/wav_packer.d.ts create mode 100644 src/lib/realtime/wavtools/dist/lib/wav_packer.d.ts.map create mode 100644 src/lib/realtime/wavtools/dist/lib/wav_recorder.d.ts create mode 100644 src/lib/realtime/wavtools/dist/lib/wav_recorder.d.ts.map create mode 100644 src/lib/realtime/wavtools/dist/lib/wav_stream_player.d.ts create mode 100644 src/lib/realtime/wavtools/dist/lib/wav_stream_player.d.ts.map create mode 100644 src/lib/realtime/wavtools/dist/lib/worklets/audio_processor.d.ts create mode 100644 src/lib/realtime/wavtools/dist/lib/worklets/audio_processor.d.ts.map create mode 100644 src/lib/realtime/wavtools/dist/lib/worklets/stream_processor.d.ts create mode 100644 src/lib/realtime/wavtools/dist/lib/worklets/stream_processor.d.ts.map create mode 100644 src/lib/realtime/wavtools/index.js create mode 100644 src/lib/realtime/wavtools/lib/analysis/audio_analysis.js create mode 100644 src/lib/realtime/wavtools/lib/analysis/constants.js create mode 100644 src/lib/realtime/wavtools/lib/wav_packer.js create mode 100644 src/lib/realtime/wavtools/lib/wav_recorder.js create mode 100644 src/lib/realtime/wavtools/lib/wav_stream_player.js create mode 100644 src/lib/realtime/wavtools/lib/worklets/audio_processor.js create mode 100644 src/lib/realtime/wavtools/lib/worklets/stream_processor.js create mode 100644 src/lib/ui/Layout.svelte create mode 100644 src/lib/ui/Switch.svelte create mode 100644 src/routes/+layout.svelte create mode 100644 src/routes/+layout.ts create mode 100644 src/routes/+page.svelte create mode 100644 svelte.config.js create mode 100644 tailwind.config.js create mode 100644 tsconfig.json create mode 100644 vite.config.ts 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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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] +// });