WebSocket
The Text-to-Speech WebSockets API is designed to generate audio from partial text input
while ensuring consistency throughout the generated audio. Although highly flexible,
the WebSockets API isn't a one-size-fits-all solution. It's well-suited for scenarios where:
* The input text is being streamed or generated in chunks.
* Word-to-audio alignment information is required.
However, it may not be the best choice when:
* The entire input text is available upfront. Given that the generations are partial,
some buffering is involved, which could potentially result in slightly higher latency compared
to a standard HTTP request.
* You want to quickly experiment or prototype. Working with WebSockets can be harder and more
complex than using a standard HTTP API, which might slow down rapid development and testing.