This work presents Whisper-Flash, a two-layer drafter built on a property of speech recognition: the words still to be written have already been spoken, which reads Whisper's encoded audio and accepted decoder states and proposes eight tokens in a single forward pass.
Abstract
Whisper is a widely used encoder-decoder model for speech recognition. Its encoder reads an utterance in one parallel pass, but its decoder writes the transcript one token at a time, which dominates inference time. Speculative decoding shortens such loops without changing their output: a small drafter guesses several upcoming tokens, and the original model verifies them all in one forward pass. We present Whisper-Flash, a two-layer drafter built on a property of speech recognition: the words still to be written have already been spoken. It reads Whisper's encoded audio and accepted decoder states and proposes eight tokens in a single forward pass. On the complete LibriSpeech test sets, Whisper-Flash processes $3.16\times/2.85\times$ as much audio per second as greedy decoding with identical outputs, and it remains faster at batch sizes up to 96 and under temperature sampling. Ablations show that direct access to the audio matters most.
Requiring no training or auxiliary computation, stride-k subsampling exploits Whisper's preprocessing redundancy, indicating that its audio-token interface carries more capacity than downstream tasks require.
This work presents TontaubeV1, a model that preserves natural prosody while enabling streaming from a single consumer GPU, and is designed primarily for English and German, with additional multilingual support.
Real-time conversational agents require speaker diarization that streams and runs on a CPU. Most systems apply an utterance-level speaker encoder to short, heavily overlapping chunks, which wastes computation and leaves the model optimized for the wrong task. We instead turn a state-of-the-art speaker recognition archi...
A deployable system for streaming Thai ASR that lets a user steer its vocabulary at decode time, without retraining is presented, which lifts keyword recall from 16.6% to 20.7% at no accuracy cost and negligible overhead.
Warit Sirichotedumrong, Tanawin Samutsin, Shahira Wani et al.· 0 citations
Full-duplex voice agents make many small, closed decisions, which current systems answer by slow autoregressive decoding. We propose DuplexJev, which feeds ASR-encoder hidden states through a small connector into a frozen LLM and reads each question as a single-token distribution over its options. Nothing is decoded, a...
Jie-Hua Jin, Zi-Yin Ma, Min Yin et al.· 1 citation
A mechanistic analysis of paralinguistic information in four open source models using the Expresso dataset with controlled speaking styles identifies a gap between what models encode and what they use, highlighting a key limitation in current audio language models.