update CHANGELOG and create websocker-server instructions
This commit is contained in:
Aleix Conchillo Flaqué
204
examples/websocket-server/index.html
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204
examples/websocket-server/index.html
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<!DOCTYPE html>
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<html lang="en">
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<head>
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<meta charset="UTF-8">
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<meta name="viewport" content="width=device-width, initial-scale=1.0">
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<script src="https://cdn.jsdelivr.net/npm/protobufjs@7.X.X/dist/protobuf.min.js"></script>
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<title>Pipecat WebSocket Client Example</title>
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</head>
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<body>
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<h1>Pipecat WebSocket Client Example</h1>
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<h3><div id="progressText">Loading, wait...</div></h2>
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<button id="startAudioBtn">Start Audio</button>
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<button id="stopAudioBtn">Stop Audio</button>
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<script>
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const SAMPLE_RATE = 16000;
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const NUM_CHANNELS = 1;
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const PLAY_TIME_RESET_THRESHOLD_MS = 1.0;
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// The protobuf type. We will load it later.
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let Frame = null;
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// The websocket connection.
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let ws = null;
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// The audio context
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let audioContext = null;
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// The audio context media stream source
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let source = null;
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// The microphone stream from getUserMedia. SHould be sampled to the
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// proper sample rate.
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let microphoneStream = null;
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// Script processor to get data from microphone.
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let scriptProcessor = null;
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// AudioContext play time.
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let playTime = 0;
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// Last time we received a websocket message.
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let lastMessageTime = 0;
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// Whether we should be playing audio.
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let isPlaying = false;
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let startBtn = document.getElementById('startAudioBtn');
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let stopBtn = document.getElementById('stopAudioBtn');
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const proto = protobuf.load("frames.proto", (err, root) => {
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if (err) {
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throw err;
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}
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Frame = root.lookupType("pipecat.Frame");
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const progressText = document.getElementById("progressText");
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progressText.textContent = "We are ready! Make sure to run the server and then click `Start Audio`.";
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startBtn.disabled = false;
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stopBtn.disabled = true;
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});
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function initWebSocket() {
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ws = new WebSocket('ws://localhost:8765');
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ws.addEventListener('open', () => console.log('WebSocket connection established.'));
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ws.addEventListener('message', handleWebSocketMessage);
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ws.addEventListener('close', (event) => {
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console.log("WebSocket connection closed.", event.code, event.reason);
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stopAudio(false);
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});
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ws.addEventListener('error', (event) => console.error('WebSocket error:', event));
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}
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async function handleWebSocketMessage(event) {
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const arrayBuffer = await event.data.arrayBuffer();
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if (isPlaying) {
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enqueueAudioFromProto(arrayBuffer);
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}
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}
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function enqueueAudioFromProto(arrayBuffer) {
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const parsedFrame = Frame.decode(new Uint8Array(arrayBuffer));
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if (!parsedFrame?.audio) {
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return false;
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}
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// Reset play time if it's been a while we haven't played anything.
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const diffTime = audioContext.currentTime - lastMessageTime;
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if ((playTime == 0) || (diffTime > PLAY_TIME_RESET_THRESHOLD_MS)) {
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playTime = audioContext.currentTime;
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}
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lastMessageTime = audioContext.currentTime;
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// We should be able to use parsedFrame.audio.audio.buffer but for
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// some reason that contains all the bytes from the protobuf message.
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const audioVector = Array.from(parsedFrame.audio.audio);
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const audioArray = new Uint8Array(audioVector);
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audioContext.decodeAudioData(audioArray.buffer, function(buffer) {
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const source = new AudioBufferSourceNode(audioContext);
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source.buffer = buffer;
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source.start(playTime);
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source.connect(audioContext.destination);
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playTime = playTime + buffer.duration;
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});
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}
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function convertFloat32ToS16PCM(float32Array) {
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let int16Array = new Int16Array(float32Array.length);
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for (let i = 0; i < float32Array.length; i++) {
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let clampedValue = Math.max(-1, Math.min(1, float32Array[i]));
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int16Array[i] = clampedValue < 0 ? clampedValue * 32768 : clampedValue * 32767;
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}
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return int16Array;
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}
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function startAudioBtnHandler() {
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if (!navigator.mediaDevices || !navigator.mediaDevices.getUserMedia) {
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alert('getUserMedia is not supported in your browser.');
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return;
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}
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startBtn.disabled = true;
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stopBtn.disabled = false;
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audioContext = new (window.AudioContext || window.webkitAudioContext)({
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latencyHint: "interactive",
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sampleRate: SAMPLE_RATE
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});
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isPlaying = true;
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initWebSocket();
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navigator.mediaDevices.getUserMedia({
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audio: {
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sampleRate: SAMPLE_RATE,
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channelCount: NUM_CHANNELS,
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autoGainControl: true,
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echoCancellation: true,
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noiseSuppression: true,
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}
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}).then((stream) => {
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microphoneStream = stream;
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// 512 is closest thing to 200ms.
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scriptProcessor = audioContext.createScriptProcessor(512, 1, 1);
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source = audioContext.createMediaStreamSource(stream);
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source.connect(scriptProcessor);
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scriptProcessor.connect(audioContext.destination);
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scriptProcessor.onaudioprocess = (event) => {
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if (!ws) {
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return;
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}
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const audioData = event.inputBuffer.getChannelData(0);
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const pcmS16Array = convertFloat32ToS16PCM(audioData);
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const pcmByteArray = new Uint8Array(pcmS16Array.buffer);
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const frame = Frame.create({
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audio: {
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audio: Array.from(pcmByteArray),
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sampleRate: SAMPLE_RATE,
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numChannels: NUM_CHANNELS
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}
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});
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const encodedFrame = new Uint8Array(Frame.encode(frame).finish());
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ws.send(encodedFrame);
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};
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}).catch((error) => console.error('Error accessing microphone:', error));
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}
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function stopAudio(closeWebsocket) {
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isPlaying = false;
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startBtn.disabled = false;
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stopBtn.disabled = true;
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if (ws && closeWebsocket) {
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ws.close();
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ws = null;
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}
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if (scriptProcessor) {
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scriptProcessor.disconnect();
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}
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if (source) {
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source.disconnect();
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}
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}
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function stopAudioBtnHandler() {
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stopAudio(true);
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}
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startBtn.addEventListener('click', startAudioBtnHandler);
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stopBtn.addEventListener('click', stopAudioBtnHandler);
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startBtn.disabled = true;
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stopBtn.disabled = true;
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</script>
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</body>
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</html>
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