Speech / Serving
verifiedStreaming ASR
A transcript that arrives after you stop talking is a different product from one that appears as you speak. Streaming recognition emits words while the audio is still arriving, which means committing to a word before hearing what follows it. Every bit of future context that makes the guess better also makes the caption later, and past roughly 300 ms a listener stops reading and starts waiting.
The constraint is architectural rather than an optimisation: an encoder that attends over the whole utterance cannot run until the utterance ends. Streaming models bound how far ahead each frame may see — chunked attention, a fixed right context, or none at all — and pay for it in accuracy. RNN-T is the common choice because its output depends on previous outputs without depending on future inputs, which is the one dependency streaming can afford.
With right context R, frame i attends over frames j ≤ i + R, so the output at i waits for R further frames of audio: a latency of R·H/sr seconds at hop H. Chunked attention amortises that — a chunk of C frames shares one boundary, giving mean latency about (C/2 + R)·H/sr — which is why chunking beats per-frame lookahead at equal accuracy.
16 queries against 16 keys; a brighter cell means more of that query's attention went to that key. Each query sees up to 2 positions ahead of itself; the rest is masked.
Encoder frames attending to encoder frames, with everything past a fixed horizon masked. Drag the lookahead up to watch each frame gain context it has to wait for — every column of it is delay the listener sits through before the word appears.
Reviewed by opendroid · 2026-08-18
- arXiv:1811.06621 — Streaming End-to-end Speech Recognition For Mobile Devices
- arXiv:2010.10759 — Emformer: Efficient Memory Transformer Based Acoustic Model For Low Latency Streaming Speech Recognition