Why TTS Latency Grows With Response Length and the Architecture That Stops It
Blog post from Deepgram
Low-latency text-to-speech evaluation should distinguish time to first playable audio from total synthesis and playback completion time, since headline TTFB figures from short, lightly loaded demos often obscure latency growth on longer responses and under concurrency. Autoregressive TTS architectures generally scale sequentially with output length, while non-autoregressive approaches can generate faster but may delay initial output or involve trade-offs in prosody and expressiveness; streaming runtimes reduce perceived delay by overlapping generation and playback but do not eliminate total generation work or prevent stuttering when synthesis falls behind real time. The discussion emphasizes that production voice agents are especially vulnerable to delays during long disclosures, read-backs, regulated scripts, and high-demand periods, where lengthy requests also increase queueing on shared hardware. It recommends testing vendors with realistic prompt lengths, simulated LLM token streaming, and target concurrency while logging text-ready, first decodable audio, first playback, and last playback timestamps, reporting p50 through p99 latency and real-time factor rather than relying on a single median TTFB.
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