Latency determines how "live" your live stream actually is — and it affects captions just as much as video. StreamTranslate keeps caption latency under 400ms so your subtitles feel perfectly synchronized.
Try Low-Latency Captions FreeLatency in live streaming refers to the delay between something happening in real life and viewers seeing it on their screens. This delay accumulates across multiple stages of the streaming pipeline: video/audio capture and encoding on your PC, transmission via RTMP to the streaming platform's ingest servers, transcoding and packaging into viewer-friendly formats, CDN distribution to edge servers globally, and final buffering in the viewer's player.
Total stream latency on Twitch in standard mode is typically 10-30 seconds. YouTube Live can be 5-30 seconds depending on latency mode selected. Low-latency modes reduce this to 2-7 seconds with some tradeoffs in quality and reliability. Ultra-low-latency modes using WebRTC can get below 1 second but are limited to specific platforms and setups.
For viewer experience, moderate latency (5-30 seconds) is often acceptable for entertainment content since viewers aren't interacting with you in real time in most scenarios. Interactive gaming streams where viewer participation matters benefit from lower latency settings, but this comes at the cost of increased buffering risk.
Caption latency is distinct from stream latency. Captions need to be generated and displayed on the stream in near-real-time relative to your speech, so that when the delayed stream reaches viewers, the captions are already visible and synchronized with your words.
The caption latency pipeline includes: audio capture (near zero), transmission to ASR servers via WebSocket (10-50ms), Deepgram Nova-2 transcription processing (100-200ms), optional NMT translation (50-100ms additional), WebSocket return to browser source (10-50ms), and DOM rendering (near zero). StreamTranslate's total caption latency is consistently under 400ms end-to-end.
This matters because viewers watching your stream — even with 30 seconds of platform latency — see captions that are synchronized with your speech. The captions appear on the stream video at the right moment because they were generated within 400ms of you speaking. By the time the delayed video reaches viewers, the captions are already baked into the right frame of content.
StreamTranslate delivers captions in under 400ms from speech to overlay — fast enough that captions appear synchronized with your words for all viewers.
Deepgram Nova-2 processes audio in real-time chunks rather than waiting for complete sentences, keeping ASR latency at 100-200ms even for long sentences.
StreamTranslate uses persistent WebSocket connections throughout the pipeline, eliminating connection overhead that would add latency to each caption update.
Latency is the delay between an event occurring and viewers seeing it, accumulated across encoding, ingest, transcoding, CDN distribution, and player buffering.
StreamTranslate delivers captions in under 400ms end-to-end — fast enough that captions feel synchronized with your speech for all viewers.
Caption latency comes from audio capture, transmission to ASR servers, transcription, optional translation, and rendering. StreamTranslate optimizes every step.
Caption latency and stream latency are independent. Captions bake into your stream in real time, so viewers see synchronized captions regardless of their platform's stream delay.
By using Deepgram Nova-2 streaming ASR that processes audio chunks in real time and optimized WebSocket pipelines that eliminate connection overhead between steps.