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Browse files- app.py +25 -15
- index.html +61 -18
app.py
CHANGED
@@ -13,6 +13,7 @@ from fastapi.responses import HTMLResponse
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from fastrtc import (
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AsyncStreamHandler,
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Stream,
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get_twilio_turn_credentials,
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)
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from google import genai
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@@ -62,12 +63,18 @@ class GeminiHandler(AsyncStreamHandler):
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)
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async def start_up(self):
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config = LiveConnectConfig(
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response_modalities=["AUDIO"], # type: ignore
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speech_config=SpeechConfig(
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@@ -78,15 +85,18 @@ class GeminiHandler(AsyncStreamHandler):
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)
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),
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)
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async def stream(self) -> AsyncGenerator[bytes, None]:
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while not self.quit.is_set():
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from fastrtc import (
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AsyncStreamHandler,
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Stream,
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WebRTCError,
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get_twilio_turn_credentials,
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)
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from google import genai
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)
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async def start_up(self):
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if not self.phone_mode:
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await self.wait_for_args()
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api_key, voice_name = self.latest_args[1:]
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else:
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api_key, voice_name = None, "Puck"
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try:
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client = genai.Client(
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api_key=api_key or os.getenv("GEMINI_API_KEY"),
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http_options={"api_version": "v1alpha"},
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)
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except Exception as e:
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raise WebRTCError(str(e))
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config = LiveConnectConfig(
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response_modalities=["AUDIO"], # type: ignore
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speech_config=SpeechConfig(
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)
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),
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)
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try:
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async with client.aio.live.connect(
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model="gemini-2.0-flash-exp", config=config
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) as session:
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async for audio in session.start_stream(
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stream=self.stream(), mime_type="audio/pcm"
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):
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if audio.data:
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array = np.frombuffer(audio.data, dtype=np.int16)
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self.output_queue.put_nowait(array)
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except Exception as e:
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raise WebRTCError(str(e))
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async def stream(self) -> AsyncGenerator[bytes, None]:
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while not self.quit.is_set():
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index.html
CHANGED
@@ -147,11 +147,29 @@
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transform: translateX(-0%) scale(var(--audio-level, 1));
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transition: transform 0.1s ease;
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}
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</style>
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</head>
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<body>
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<div style="text-align: center">
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<h1>Gemini Voice Chat</h1>
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<p>Speak with Gemini using real-time audio streaming</p>
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@@ -229,6 +247,17 @@
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}
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}
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async function setupWebRTC() {
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const config = __RTC_CONFIGURATION__;
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peerConnection = new RTCPeerConnection(config);
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// Create data channel for messages
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dataChannel = peerConnection.createDataChannel('text');
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dataChannel.onmessage =
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// Create and send offer
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const offer = await peerConnection.createOffer();
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});
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const serverResponse = await response.json();
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await peerConnection.setRemoteDescription(serverResponse);
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} catch (err) {
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console.error('Error setting up WebRTC:', err);
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function handleMessage(event) {
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const eventJson = JSON.parse(event.data);
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if (eventJson.type === "send_input") {
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fetch('/input_hook', {
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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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webrtc_id: webrtc_id,
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api_key: apiKeyInput.value,
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voice_name: voiceSelect.value
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})
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});
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}
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}
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@@ -364,6 +406,7 @@
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if (audioContext) {
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audioContext.close();
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}
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}
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startButton.addEventListener('click', () => {
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transform: translateX(-0%) scale(var(--audio-level, 1));
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transition: transform 0.1s ease;
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}
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/* Add styles for toast notifications */
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.toast {
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position: fixed;
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top: 20px;
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left: 50%;
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transform: translateX(-50%);
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background-color: #f44336;
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color: white;
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padding: 16px 24px;
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border-radius: 4px;
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font-size: 14px;
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z-index: 1000;
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display: none;
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box-shadow: 0 2px 5px rgba(0, 0, 0, 0.2);
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}
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</style>
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</head>
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<body>
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<!-- Add toast element after body opening tag -->
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<div id="error-toast" class="toast"></div>
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<div style="text-align: center">
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<h1>Gemini Voice Chat</h1>
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<p>Speak with Gemini using real-time audio streaming</p>
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}
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}
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function showError(message) {
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const toast = document.getElementById('error-toast');
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toast.textContent = message;
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toast.style.display = 'block';
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// Hide toast after 5 seconds
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setTimeout(() => {
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toast.style.display = 'none';
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}, 5000);
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}
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async function setupWebRTC() {
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const config = __RTC_CONFIGURATION__;
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peerConnection = new RTCPeerConnection(config);
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// Create data channel for messages
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dataChannel = peerConnection.createDataChannel('text');
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dataChannel.onmessage = (event) => {
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const eventJson = JSON.parse(event.data);
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if (eventJson.type === "error") {
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showError(eventJson.message);
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} else if (eventJson.type === "send_input") {
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fetch('/input_hook', {
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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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webrtc_id: webrtc_id,
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api_key: apiKeyInput.value,
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voice_name: voiceSelect.value
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})
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});
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}
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};
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// Create and send offer
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const offer = await peerConnection.createOffer();
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});
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const serverResponse = await response.json();
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if (serverResponse.status === 'failed') {
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showError(serverResponse.meta.error === 'concurrency_limit_reached'
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? `Too many connections. Maximum limit is ${serverResponse.meta.limit}`
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: serverResponse.meta.error);
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stop();
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startButton.textContent = 'Start Recording';
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return;
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}
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await peerConnection.setRemoteDescription(serverResponse);
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} catch (err) {
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console.error('Error setting up WebRTC:', err);
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showError('Failed to establish connection. Please try again.');
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stop();
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startButton.textContent = 'Start Recording';
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}
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}
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if (audioContext) {
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audioContext.close();
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}
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updateButtonState();
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}
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startButton.addEventListener('click', () => {
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