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.
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.
If you are using a voice_id corresponding to a voice clone, you should explicitly set the model parameter to "lightning-large" in the Smallest client or payload.
If you are using a voice_id corresponding to a voice clone, you should explicitly set the model parameter to "lightning-large" in the Smallest client or payload.
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.The TextToAudioStream processor streams raw audio data without WAV headers for better streaming efficiency. These raw audio chunks can be:
.wav or .mp3) for later use.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.