WebSocket Streaming for TTS API
We are excited to launch WebSocket streaming for our Text-to-Speech API. This feature enables developers to build real-time, low-latency voice experiences by streaming text and receiving synthesized audio over a persistent, bidirectional connection. It is ideal for applications like conversational AI, live chat support, and dynamic content narration where immediate audio feedback is crucial.
Key features include:
- Low-Latency Communication: Stream text and receive audio with minimal delay.
- Bidirectional Streaming: Send text and receive audio on the same connection.
- Efficient: Avoids the overhead of repeated HTTP requests for continuous audio synthesis.
- Real-Time Control: Adjust voice style, speed, and pitch during the session.
Learn more in our WebSocket Streaming documentation.