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WebSocket Support for TTS API

Our Text-to-Speech (TTS) API supports WebSocket communication, providing a real-time, low-latency streaming experience for applications that require instant speech synthesis. WebSockets allow continuous data exchange, making them ideal for use cases that demand uninterrupted audio generation.

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When to Use WebSockets

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1. Real-Time Streaming

WebSockets are perfect for applications that need real-time speech synthesis, eliminating the delays associated with traditional HTTP requests.

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2. Interactive Applications

For voice assistants, chatbots, and live transcription services, WebSockets ensure smooth, uninterrupted audio playback and response times.

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3. Reduced Latency

A persistent WebSocket connection reduces the need for repeated request-response cycles, significantly improving performance for applications requiring rapid audio generation.

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How It Works

  1. Establish a Connection: The client opens a WebSocket connection to our TTS API.
  2. Send Text Data: The client sends the text payload to be synthesized.
  3. Process in Chunks: The API breaks the text into chunks and processes them individually.
  4. Receive Audio Stream: As each chunk is processed, it is sent back to the client as a base64-encoded audio buffer.
  5. Completion: Once all chunks are processed, a complete message is sent to indicate the end of the stream.

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Implementation Details

The WebSocket TTS API is optimized to handle real-time text-to-speech conversions efficiently. Here are some key implementation aspects:
  • Input Validation: Ensures the provided text and voice ID are valid before processing.
  • Chunk Processing: Long texts are split into smaller chunks (e.g., 240 characters) to optimize processing.
  • Voice Caching: The API fetches and caches voice configurations to reduce redundant database queries.
  • Task Queue System: Tasks are pushed to a Redis-based queue for efficient processing and real-time audio generation.
  • Error Handling: If any chunk fails, an error message is logged and sent to the client.

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Example Request Flow

  1. The client sends a WebSocket message: 
    {
  "text": "Hello, world!",
  "voice_id": "12345",
  "speed": 1.0,
  "sample_rate": 24000
}
  2. The API validates the request and retrieves the voice settings.
  3. The text is split into chunks and processed in the background.
  4. The client receives responses like: 
    {
 "request_id": "047c9091-b770-41d8-b96b-907d1c8406c0",
  "status": "chunk",
  "data": {
    "audio": "<base64_encoded_audio_chunk>"
  }
}
  5. Once all chunks are sent, a final message is sent: 
    {
 "request_id": "047c9091-b770-41d8-b96b-907d1c8406c0",
  "status": "complete",
  "message": "All chunks sent",
  "done": true
}
For implementation details, check our WebSocket API documentation.