Papers for

conference transcription teams

Papers whose findings have a practical use for this group, as judged from the abstract. Open a paper to read what it means in practice.

HINTT compares cascaded and unified speech models for multilingual diarization and transcription

HINTT Submission to the 2nd MLC-SLM Challenge: Comparing Cascaded and Unified Approaches to Diarization and ASR

Abstract: This paper presents the HINTT system submitted to the 2nd Challenge and Workshop on Multilingual Conversational Speech Language Model (MLC-SLM). We address multilingual speaker-attributed ASR, where systems must determine who spoke when and what was spoken. We investigate two modeling strategies for this problem: a cascaded pipeline that combines speaker diarization with speech-LLM-based ASR, and a unified speech LLM that directly generates speaker labels, timestamps, and transcriptions. Our final submission is based on the cascaded pipeline, consisting of a fine-tuned DiariZen diarization model, a fine-tuned Qwen3-ASR model, and LLM-based generative error correction. For comparison, we also fine-tune VibeVoice-ASR as a unified model using the same official training data. All task-specific fine-tuning and model selection are performed using only the official MLC-SLM data, without external data or pseudo-labels. Experimental results demonstrate that the cascaded system remains more reliable under the MLC-SLM Task 1 conditions, while unified speech LLMs offer a promising direction for future speaker-attributed ASR.

Tue 6 OctComputation and Language
The gist
Understanding who said what and when in a conversation across multiple languages is a tricky problem. The paper describes two ways to solve this: one that breaks the task into steps (finding speakers first, then transcribing), and another that tries to do everything at once with a single model. The authors find that, for this challenge, their step-by-step method works more reliably, but the all-in-one approach shows promise for the future.
Open → 2610.08063v1

Self supervised speech model improves multi speaker separation and diarization

SepRQ : Self-Supervised Speech Mixture Representation Learning via Mask-Free, Multi-Scale Source Separation

Abstract: Self-supervised learning (SSL) is standard for speech representation learning, but mainstream models are designed around single-speaker audio, limiting their usefulness in multi-speakers scenarios. We present SepRQ, an open-source SSL framework that replaces masked prediction with a pseudo-source-separation objective over frozen random-projection codebooks. By adopting a novel mask-free, multiresolution approach, SepRQ achieves state-of-the-art performance in Speaker Diarization and Speech Separation on the SUPERB benchmark, surpassing WavLM and other cocktail-party derived SSLs at both Base and Large scales, while requiring only 85.68M inference parameters. SepRQ also demonstrates strong performance across target-speaker tasks requiring enrollment (such as Target-Speaker Automatic Speech Recognition), and on the challenging multi-domain DIHARD 3 diarization dataset. Notably, we report strong separation capabilities on three-speaker mixtures (WSJ0-3Mix), where current SSL literature struggles. While cocktail-party SSLs remain scarce and closed-source, limited to C-HuBERT and the enrollment-based SA-WavLM, we open-source SepRQ to the community.

Sat 3 OctComputation and LanguageMachine Learning
The gist
Understanding multiple people talking at once is hard for many speech recognition systems because they are usually trained on single speakers. The authors created SepRQ, a new system that learns to separate voices without needing masked audio parts for training. SepRQ works well in identifying who spoke when and separating voices, even for three speakers at the same time, which is usually very challenging. It is also open-source, allowing others to use and build on it.
Open → 2610.04690v1

Target speaker speech recognition improves by tuning guidance scales

Asymmetric Classifier-Free Guidance for Target-Speaker ASR

Abstract: Target-speaker automatic speech recognition (TS-ASR) must identify and transcribe a desired speaker under varying overlap and noise conditions. These changes alter the acoustic evidence for the target speaker in the speech mixture, motivating inference-time calibration of speaker conditioning. We introduce asymmetric classifier-free guidance (CFG) for TS-ASR using Whisper: the speaker-conditioned branch predicts the target transcript, while the speaker-unconditioned branch predicts serialized multi-speaker transcripts. CFG adjusts the contribution of speaker conditioning during decoding through a single guidance scale. We select a global guidance scale on target-domain development data and train a lightweight encoder-based predictor to adjust it for each utterance, keeping the recognition model fixed. Under domain shifts, our full system achieves relative word error rate (WER) reductions of up to 21.8% over the condition-only baseline, and 5.6% over standard conditional decoding of the same CFG-trained model. Oracle analysis shows that substantially larger WER reductions are possible through utterance-level scale selection and identifies how beneficial adjustments vary with domain shifts.

Thu 24 SeptComputation and LanguageSound
The gist
Recognizing what a specific person says in noisy or overlapping conversations is hard. The authors show that adjusting how much the recognition system focuses on the target speaker during transcription can help. They combine predictions that include and exclude speaker information and change their influence based on conditions. Their approach reduces errors by picking the right balance for each recording, even when conditions change.
Open → 2609.30476v1

Speaker temporal memory improves multi speaker speech recognition accuracy

STAM-ASR: Speaker-Temporal Anchoring with Memory for Multi-Speaker ASR

Abstract: Natural conversations make both speech recognition and speaker attribution challenging for ASR, as speakers take turns, overlap, and reappear over time. We propose STAM-ASR, Speaker-Temporal Anchoring with Memory, a lightweight framework that extends an already pretrained AudioLLM for multi-speaker ASR. Without relying on an external diarization system, STAM-ASR learns speaker activity and speaker-aware representations directly from intermediate AudioLLM features. Hence providing explicit who and when cues to modulate the AudioLLM's semantic representation without explicit speech separation. STAM-ASR further maintains fixed-size speaker and conversational memories to carry complementary context across turns. We evaluate STAM-ASR on AMI, ICSI, LibriCSS, and NOTSOFAR-1 across close-talk, far-field, overlapping, and cross-domain conditions. Our reported results shows that speaker-temporal conditioning and memory provide complementary benefits, while the gap between reference and predicted speaker activity identifies robust speaker tracking as a key remaining challenge.

Thu 24 SeptSound
The gist
Understanding who is speaking and when they talk is hard in conversations where people speak over each other and come back later. The authors created a method called STAM-ASR that helps speech recognition systems figure out speaker identity and timing without needing separate steps to separate voices. It uses memory to remember who spoke earlier and builds on existing speech AI models. Tests on multiple real-world conversation types show that remembering speaker info and timing helps, but accurately tracking speakers over time remains tough.
Open → 2609.29805v1