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Add tts-to-webrtc example
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‎tts-to-webrtc/README.md‎

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# WebRTC Text-to-Speech Example
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This is an example app that combines WebRTC with OpenAI's Text-to-Speech API to stream audio in real-time.
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## Prerequisites
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- Go 1.20 or later
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- An OpenAI API key
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- Web browser with WebRTC support (Chrome, Firefox, Safari, etc.)
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## Installation
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1. Clone the repository:
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```bash
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git clone <https://github.com/pion/example-webrtc-applications>
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cd tts-to-webrtc
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```
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2. Install module dependencies:
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[Resampler](https://github.com/dh1tw/gosamplerate) and [opus encoder](https://github.com/hraban/opus) packages are using cgo modules and need to setup. Follow the instructions below to install the required packages.
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Linux:
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using apt (Ubuntu), yum (Centos)...etc.
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```bash
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$ sudo apt install libsamplerate0 pkg-config libopus-dev libopusfile-dev
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```
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MacOS
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using Homebrew:
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```bash
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$ brew install libsamplerate pkg-config opus opusfile
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```
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3. Install Go dependencies:
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```bash
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export GO111MODULE=on
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go install github.com/pion/example-webrtc-applications/v4/tts-to-webrtc@latest
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```
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## Configuration
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Set your OpenAI API key as an environment variable:
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```bash
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export OPENAI_API_KEY=your_api_key_here
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```
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## Running the Application
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1. Start the server:
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```bash
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go run main.go
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```
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2. Open your web browser and navigate to:
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```
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http://localhost:8080
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```
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## Usage
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1. Click the "Connect" button to establish a WebRTC connection
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2. Wait for the connection status to show "connected"
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3. Type some text in the textarea
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4. Click "Convert to Speech" to hear the text being spoken
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## Technical Details
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- The application uses OpenAI's TTS API to convert text to speech
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- Audio is streamed using WebRTC with Opus codec
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- Sample rate conversion is handled automatically (24kHz to 48kHz)
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- The server implements a simple audio buffer to handle streaming
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## License
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This project is licensed under the MIT License - see the LICENSE file for details.

‎tts-to-webrtc/index.html‎

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<!DOCTYPE html>
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<html>
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<head>
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<title>WebRTC TTS Demo</title>
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<meta name="viewport" content="width=device-width, initial-scale=1">
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<style>
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body {
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font-family: Arial, sans-serif;
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max-width: 800px;
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margin: 0 auto;
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padding: 20px;
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}
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.form-group {
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margin-bottom: 15px;
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}
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textarea {
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width: 100%;
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padding: 8px;
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}
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button {
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padding: 10px 20px;
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background-color: #4CAF50;
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color: white;
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border: none;
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border-radius: 4px;
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cursor: pointer;
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}
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button:hover {
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background-color: #45a049;
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}
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button.connected {
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background-color: #dc3545;
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}
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button.connected:hover {
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background-color: #c82333;
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}
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</style>
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</head>
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<body>
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<h1>Text to Speech with WebRTC</h1>
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<div id="connectionStatus" style="margin-bottom: 20px; padding: 10px; background-color: #f0f0f0; border-radius: 4px;">
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<div>Connection State: <span id="connectionState">new</span></div>
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<div>ICE Connection: <span id="iceConnectionState">new</span></div>
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<div>Signaling State: <span id="signalingState">new</span></div>
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</div>
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<div class="form-group">
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<button id="connectButton" onclick="toggleConnection()">Connect</button>
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</div>
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<div class="form-group">
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<textarea id="textInput" rows="4" placeholder="Enter text to convert to speech"></textarea>
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</div>
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<div class="form-group">
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<button id="convertButton" onclick="submitText()" disabled>Convert to Speech</button>
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</div>
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<script>
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let pc;
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let isConnected = false;
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async function toggleConnection() {
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const connectButton = document.getElementById('connectButton');
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if (!isConnected) {
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connectButton.disabled = true;
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await initWebRTC();
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} else {
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if (pc) {
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await pc.close();
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pc = null;
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}
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document.getElementById('convertButton').disabled = true;
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document.getElementById('connectionState').textContent = 'new';
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document.getElementById('iceConnectionState').textContent = 'new';
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document.getElementById('signalingState').textContent = 'new';
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connectButton.textContent = 'Connect';
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connectButton.classList.remove('connected');
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isConnected = false;
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}
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}
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async function initWebRTC() {
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pc = new RTCPeerConnection({
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iceServers: [{
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urls: 'stun:stun.l.google.com:19302'
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}]
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});
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// Add connection state monitoring
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pc.onconnectionstatechange = () => {
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const state = pc.connectionState;
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const connectButton = document.getElementById('connectButton');
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document.getElementById('connectionState').textContent = state;
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if (state === 'connected') {
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document.getElementById('convertButton').disabled = false;
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connectButton.disabled = false;
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connectButton.textContent = 'Disconnect';
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connectButton.classList.add('connected');
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isConnected = true;
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} else {
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document.getElementById('convertButton').disabled = true;
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}
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};
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pc.oniceconnectionstatechange = () => {
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document.getElementById('iceConnectionState').textContent = pc.iceConnectionState;
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};
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pc.onsignalingstatechange = () => {
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document.getElementById('signalingState').textContent = pc.signalingState;
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};
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pc.ontrack = function(event) {
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const audio = new Audio();
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audio.srcObject = event.streams[0];
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audio.play();
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};
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// Create promise to wait for ICE gathering
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const iceCandidatesComplete = new Promise((resolve) => {
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pc.onicegatheringstatechange = () => {
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if (pc.iceGatheringState === 'complete') {
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resolve();
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}
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};
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});
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pc.addTransceiver('audio', { direction: 'recvonly' });
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const offer = await pc.createOffer({
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offerToReceiveAudio: true
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});
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await pc.setLocalDescription(offer);
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// Wait for ICE gathering to complete
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await iceCandidatesComplete;
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const response = await fetch('/webrtc', {
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method: 'POST',
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headers: {
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'Content-Type': 'application/json'
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},
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body: JSON.stringify({
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sdp: JSON.stringify(pc.localDescription)
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})
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});
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const answer = await response.json();
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await pc.setRemoteDescription(new RTCSessionDescription(answer));
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}
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async function submitText() {
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const text = document.getElementById('textInput').value;
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if (!text) return;
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const response = await fetch('/tts', {
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method: 'POST',
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headers: {
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'Content-Type': 'application/json'
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},
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body: JSON.stringify({
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text: text
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})
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});
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const audioBlob = await response.blob();
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const audio = new Audio(URL.createObjectURL(audioBlob));
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audio.play();
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}
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// Remove the automatic initWebRTC() call at the end
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</script>
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</body>
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</html>

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