
Real-time Text-to-Speech with WebSockets
Baseten's platform supports real-time text-to-speech (TTS) inference using WebSockets, enabling low-latency, bidirectional communication for streaming audio applications. This builds upon existing capabilities for AI model deployment and serving, specifically enhancing interactive voice applications by leveraging persistent connections for continuous data exchange between clients and the Orpheus TTS model. The platform now also supports NVIDIA Nemotron 3 Diarization models, offering real-time speaker labeling and diarized transcription by integrating with NVIDIA's Streaming Sortformer architecture and Parakeet V2 ASR model. This enables end-to-end diarized transcription pipelines that condition the ASR model on speaker activity features within a single forward pass, improving accuracy and temporal alignment for overlapping speech.
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