I need to process streaming text input for text-to-speech (TTS) and receive the response as a streaming audio buffer with minimal latency. Is there any documentation available on how to achieve this? I tried accessing the documentation, but the GitHub link appears to be broken:
I need to process streaming text input for text-to-speech (TTS) and receive the response as a streaming audio buffer with minimal latency. Is there any documentation available on how to achieve this? I tried accessing the documentation, but the GitHub link appears to be broken:
GitHub Link