What are the common latency issues in real-time TTS streaming?
Asked on Oct 09, 2025
Answer
Latency issues in real-time Text-to-Speech (TTS) streaming often arise from network delays, processing time, and the efficiency of the TTS engine itself. These factors can affect the speed at which text is converted to audio and delivered to the end-user, impacting the overall user experience.
Example Concept: In real-time TTS streaming, latency can be minimized by optimizing the network bandwidth, using efficient TTS algorithms that process text quickly, and implementing buffering strategies to pre-load audio segments. Additionally, selecting a TTS engine with low processing overhead and ensuring a stable internet connection can significantly reduce delays.
Additional Comment:
- Network latency can be reduced by using Content Delivery Networks (CDNs) to cache and deliver audio closer to the user.
- Consider using a TTS service with a high-performance API that supports streaming to handle large volumes of text efficiently.
- Implementing client-side buffering can help smooth out any potential delays in audio playback.
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