gemini-live-api-dev

# Gemini Live API Development Skill ## Overview The Live API enables **low-latency, real-time voice and video interactions** with Gemini over WebSockets. It processes continuous streams of audio, video, or text to deliver immediate, human-like spoken responses. Key capabilities: - **Bidirectional audio streaming** — real-time mic-to-speaker conversations - **Video streaming** — send camera/screen frames alongside audio - **Text input/output** — send and receive text within a live session - **Audio transcriptions** — get text transcripts of both input and output audio - **Voice Activity Detection (VAD)** — automatic interruption handling - **Native audio** — affective dialog, proactive audio, thinking - **Function calling** — synchronous and asynchronous tool use - **Google Search grounding** — ground responses in real-time search results - **Session management** — context compression, session resumption, GoAway signals - **Ephemeral tokens** — secure client-side authentication > [!NOTE] > The Live API currently **only supports WebSockets**. For WebRTC support or simplified integration, use a [partner integration](#partner-integrations). ## Models - `gemini-2.5-flash-native-audio-preview-12-2025` — Native audio output, affective dialog, proactive audio, thinking. 128k context window. **This is the recommended model for all Live API use cases.** > [!WARNING] > The following Live API models are **deprecated** and will be shut down. Migrate to `gemini-2.5-flash-native-audio-preview-12-2025`. > - `gemini-live-2.5-flash-preview` — Released June 17, 2025. Shutdown: December 9, 2025. > - `gemini-2.0-flash-live-001` — Released April 9, 2025. Shutdown: December 9, 2025. ## SDKs - **Python**: `google-genai` — `pip install google-genai` - **JavaScript/TypeScript**: `@google/genai` — `npm install @google/genai` > [!WARNING] > Legacy SDKs `google-generativeai` (Python) and `@google/generative-ai` (JS) are deprecated. Use the new SDKs above. ## Partner Integrations To streamline real-time audio/video app development, use a third-party integration supporting the Gemini Live API over **WebRTC** or **WebSockets**: - [LiveKit](https://docs.livekit.io/agents/models/realtime/plugins/gemini/) — Use the Gemini Live API with LiveKit Agents. - [Pipecat by Daily](https://docs.pipecat.ai/guides/features/gemini-live) — Create a real-time AI chatbot using Gemini Live and Pipecat. - [Fishjam by Software Mansion](https://docs.fishjam.io/tutorials/gemini-live-integration) — Create live video and audio streaming applications with Fishjam. - [Vision Agents by Stream](https://visionagents.ai/integrations/gemini) — Build real-time voice and video AI applications with Vision Agents. - [Voximplant](https://voximplant.com/products/gemini-client) — Connect inbound and outbound calls to Live API with Voximplant. - [Firebase AI SDK](https://firebase.google.com/docs/ai-logic/live-api?api=dev) — Get started with the Gemini Live API using Firebase AI Logic. ## Audio Formats - **Input**: Raw PCM, little-endian, 16-bit, mono. 16kHz native (will resample others). MIME type: `audio/pcm;rate=16000` - **Output**: Raw PCM, little-endian, 16-bit, mono. 24kHz sample rate. > [!IMPORTANT] > Use `send_realtime_input` / `sendRealtimeInput` for all real-time user input (audio, video, **and text**). Use `send_client_content` / `sendClientContent` **only** for incremental conversation history updates (appending prior turns to context), not for sending new user messages. > [!WARNING] > Do **not** use `media` in `sendRealtimeInput`. Use the specific keys: `audio` for audio data, `video` for images/video frames, and `text` for text input. --- ## Quick Start ### Authentication #### Python ```python from google import genai client = genai.Client(api_key="YOUR_API_KEY") ``` #### JavaScript ```js import { GoogleGenAI } from '@google/genai'; const ai = new GoogleGenAI({ apiKey: 'YOUR_API_KEY' }); ``` ### Connecting to the Live API #### Python ```python from google.genai import types config = types.LiveConnectConfig( response_modalities=[types.Modality.AUDIO], system_instruction=types.Content( parts=[types.Part(text="You are a helpful assistant.")] ) ) async with client.aio.live.connect(model="gemini-2.5-flash-native-audio-preview-12-2025", config=config) as session: pass # Session is now active ``` #### JavaScript ```js const session = await ai.live.connect({ model: 'gemini-2.5-flash-native-audio-preview-12-2025', config: { responseModalities: ['audio'], systemInstruction: { parts: [{ text: 'You are a helpful assistant.' }] } }, callbacks: { onopen: () => console.log('Connected'), onmessage: (response) => console.log('Message:', response), onerror: (error) => console.error('Error:', error), onclose: () => console.log('Closed') } }); ``` ### Sending Text #### Python ```python await session.send_realtime_input(text="Hello, how are you?") ``` #### JavaScript ```js session.sendRealtimeInput({ text: 'Hello, how are you?' }); ``` ### Sending Audio #### Python ```python await session.send_realtime_input( audio=types.Blob(data=chunk, mime_type="audio/pcm;rate=16000") ) ``` #### JavaScript ```js session.sendRealtimeInput({ audio: { data: chunk.toString('base64'), mimeType: 'audio/pcm;rate=16000' } }); ``` ### Sending Video #### Python ```python # frame: raw JPEG-encoded bytes await session.send_realtime_input( video=types.Blob(data=frame, mime_type="image/jpeg") ) ``` #### JavaScript ```js session.sendRealtimeInput({ video: { data: frame.toString('base64'), mimeType: 'image/jpeg' } }); ``` ### Receiving Audio and Text #### Python ```python async for response in session.receive(): content = response.server_content if content: # Audio if content.model_turn: for part in content.model_turn.parts: if part.inline_data: audio_data = part.inline_data.data # Transcription if content.input_transcription: print(f"User: {content.input_transcription.text}") if content.output_transcription: print(f"Gemini: {content.output_transcription.text}") # Interruption if content.interrupted is True: pass # Stop playback, clear audio queue ``` #### JavaScript ```js // Inside the onmessage callback const content = response.serverContent; if (content?.modelTurn?.parts) { for (const part of content.modelTurn.parts) { if (part.inlineData) { const audioData = part.inlineData.data; // Base64 encoded } } } if (content?.inputTranscription) console.log('User:', content.inputTranscription.text); if (content?.outputTranscription) console.log('Gemini:', content.outputTranscription.text); if (content?.interrupted) { /* Stop playback, clear audio queue */ } ``` --- ## Limitations - **Response modality** — Only `TEXT` **or** `AUDIO` per session, not both - **Audio-only session** — 15 min without compression - **Audio+video session** — 2 min without compression - **Connection lifetime** — ~10 min (use session resumption) - **Context window** — 128k tokens (native audio) / 32k tokens (standard) - **Code execution** — Not supported - **URL context** — Not supported ## Best Practices 1. **Use headphones** when testing mic audio to prevent echo/self-interruption 2. **Enable context window compression** for sessions longer than 15 minutes 3. **Implement session resumption** to handle connection resets gracefully 4. **Use ephemeral tokens** for client-side deployments — never expose API keys in browsers 5. **Use `send_realtime_input`** for all real-time user input (audio, video, text). Reserve `send_client_content` only for injecting conversation history 6. **Send `audioStreamEnd`** when the mic is paused to flush cached audio 7. **Clear audio playback queues** on interruption signals ## How to use the Gemini API For detailed API documentation, fetch from the official docs index: **llms.txt URL**: `https://ai.google.dev/gemini-api/docs/llms.txt` This index contains links to all documentation pages in `.md.txt` format. Use web fetch tools to: 1. Fetch `llms.txt` to discover available documentation pages 2. Fetch specific pages (e.g., `https://ai.google.dev/gemini-api/docs/live-session.md.txt`) ### Key Documentation Pages > [!IMPORTANT] > Those are not all the documentation pages. Use the `llms.txt` index to discover available documentation pages - [Live API Overview](https://ai.google.dev/gemini-api/docs/live.md.txt) — getting started, raw WebSocket usage - [Live API Capabilities Guide](https://ai.google.dev/gemini-api/docs/live-guide.md.txt) — voice config, transcription config, native audio (affective dialog, proactive audio, thinking), VAD configuration, media resolution - [Live API Tool Use](https://ai.google.dev/gemini-api/docs/live-tools.md.txt) — function calling (sync and async), Google Search grounding - [Session Management](https://ai.google.dev/gemini-api/docs/live-session.md.txt) — context window compression, session resumption, GoAway signals - [Ephemeral Tokens](https://ai.google.dev/gemini-api/docs/ephemeral-tokens.md.txt) — secure client-side authentication for browser/mobile - [WebSockets API Reference](https://ai.google.dev/api/live.md.txt) — raw WebSocket protocol details ## Supported Languages The Live API supports 70 languages including: English, Spanish, French, German, Italian, Portuguese, Chinese, Japanese, Korean, Hindi, Arabic, Russian, and many more. Native audio models automatically detect and switch languages.