Synthesize streaming Text to Speech

The TextToAudioStream class provides real-time text-to-speech (TTS) conversion by streaming text directly into audio output. This feature is particularly useful in applications that require instant feedback, such as voice assistants, live captioning systems, or interactive chatbots, where text is continuously generated and needs to be converted into speech on-the-fly.

This example demonstrates how to stream text from a large language model (LLM) and process it into speech, utilizing the TextToAudioStream class with both synchronous and asynchronous TTS engines.

Example Overview

In this example, text is generated using an LLM (Groq in this case, you can use any LLM), and the generated text is then passed to a TTS system (Smallest API) for real-time audio synthesis. The audio is saved as a .wav file. This entire process happens asynchronously to ensure smooth performance, especially when dealing with large or continuous streams of text.

Code Walkthrough

Stream through a WebSocket

Saving to a file

Parameters

  • tts_instance: The instance of the TTS engine (either Smallest or AsyncSmallest) used to generate speech from the text.
  • queue_timeout: The wait time (in seconds) for new text to be received before attempting to generate speech. Default is 5.0 seconds.
  • max_retries: The maximum number of retries for failed synthesis attempts. Default is 3.

Output Format

The TextToAudioStream processor streams raw audio data without WAV headers for better streaming efficiency. These raw audio chunks can be:

  • Played directly through an audio device for real-time feedback.
  • Saved to a file (e.g., .wav or .mp3) for later use.
  • Streamed over a network to a client device or service.
  • Further processed for additional applications, such as speech analytics or audio effects.

This approach allows you to handle continuous streams of text and convert them into real-time speech, making it ideal for interactive applications where immediate audio feedback is crucial.