Caption latency ruins the viewer experience when captions lag far behind your speech. Here's what causes latency and how to get it as close to real-time as possible.
Start Free TrialCaption latency is the delay between when you speak and when the caption text appears on screen. For live streaming, caption latency determines whether captions feel like a live experience or a delayed echo. At under 500ms, captions feel near-instantaneous. At 2-5 seconds (common with some caption methods), they feel disconnected and frustrating. At 10+ seconds (YouTube's auto-captions), they're essentially useless for following live conversation.
StreamTranslate is designed for sub-500ms end-to-end caption latency. This section explains what contributes to latency and how to minimize it on your end.
Your microphone audio needs to be captured by OBS and sent to StreamTranslate's servers. With a wired ethernet connection, this typically adds 20-50ms. The audio buffer size in your operating system's audio settings can affect this — smaller buffers mean lower latency.
Deepgram Nova-2 processes audio in real-time streaming mode, generating partial transcripts as you speak rather than waiting for complete sentences. This is the primary reason StreamTranslate achieves sub-500ms latency — partial results are shown progressively, not buffered until a complete sentence is detected.
If you're using translation, the transcribed text is sent to a translation API and returned. This adds approximately 50-150ms depending on server load and language pair. Total latency with translation still typically stays under 600-700ms for supported language pairs.
OBS Browser Source has a small processing overhead for rendering the caption overlay. This is typically 20-50ms and is not significantly reducible.
Wi-Fi introduces variable latency (jitter) that can cause noticeable caption delays or dropped audio packets. Wired ethernet provides consistent, low-latency connectivity that keeps caption latency predictable and minimal.
In your operating system's audio settings, set your audio interface or microphone sample rate to 48kHz with the smallest buffer size that doesn't cause audio glitches (typically 256 samples or 512 samples). Smaller buffers mean audio reaches StreamTranslate faster.
Deepgram Nova-2 processes speech in real time and shows progressive partial results. The system can start displaying a word before you finish the sentence. Speaking clearly with natural pacing (not artificially slow, but not rushing) helps the STT engine generate accurate partial results quickly.
OBS Browser Source rendering competes with other browser and application processes. A system with fewer active processes has more resources for rendering the caption overlay quickly.
StreamTranslate routes your audio to the closest available server region. If you're experiencing higher-than-expected latency, check your account settings for server region options. Physical distance between you and the processing server adds real-time latency.
Local solutions like LocalVocal (Whisper on GPU) buffer audio into segments of several seconds before processing — this is fundamental to how batch-mode Whisper works and results in 2-5 second latency. YouTube's auto-captions buffer audio for 5-10 seconds. Google Speech API-based plugins buffer for 1-3 seconds and add browser overhead.
StreamTranslate uses Deepgram Nova-2 in streaming mode — audio is processed in a continuous stream with partial results returned progressively. This is what enables sub-500ms latency. It's not a configuration trick; it's an architectural difference in how the STT engine processes audio.
For live conversation, interaction, and game commentary — yes, lower is better. But for some use cases (educational streams where the viewer benefits from slight text delay, or ASMR where the deliberate pace makes higher latency less noticeable), the difference between 400ms and 800ms is imperceptible to most viewers. Sub-500ms is the target; anything under 1 second is generally acceptable for live stream captions.
Common causes: Wi-Fi instead of wired ethernet, large audio buffer size in system settings, high system CPU/memory usage, or physical distance from StreamTranslate's nearest server. Try wired ethernet first — it's the most impactful fix.
StreamTranslate achieves sub-500ms end-to-end caption latency under optimal conditions. With translation enabled, latency is typically 500-700ms. Both are significantly faster than alternative captioning solutions.
Whisper in LocalVocal processes audio in batch segments (several seconds of audio at once) rather than in a continuous stream. This fundamental architecture difference means 2-5 second latency vs StreamTranslate's under-500ms streaming approach.
Translation adds approximately 50-150ms to caption latency. Total latency with translation enabled is typically 500-700ms — still significantly faster than YouTube's auto-captions or Whisper-based solutions.
Yes. Wi-Fi introduces variable latency (jitter) that causes audio packet delays. Wired ethernet provides consistent low latency. Switching from Wi-Fi to ethernet often reduces observable caption latency by 100-200ms.
Set your audio interface buffer to 256 samples at 48kHz if your system can handle it without audio glitches. If you get crackles or pops, increase to 512 samples. The lowest buffer size that's stable gives you minimum audio delivery latency to StreamTranslate.