While cascaded multi-talker ASR (MT-ASR) leverages state-of-the-art foundation models, its performance is often capped by speaker leakage during separation. Prior correction strategies primarily focus on lexical re-labeling for speaker attribution. We propose a complementary pruning-based paradigm that robustly identifies and removes leakage artifacts. Our method utilizes a pre-trained speaker diarization model as a multimodal verifier to prune transcribed segments satisfying a tripartite consensus of temporal containment, lexical cross-validation, and temporal alignment. Results on LibriMix, LibriSpeechMix, and the AMI Meeting corpus show our algorithm consistently reduces cpW ER across diverse overlap conditions. Specifically, on subsets with high speaker leakage, our method achieves relative cpW ER reductions of up to 29%, highlighting its effectiveness in enhancing the reliability of cascaded MT-ASR transcripts in complex acoustic environments.
Figures & tables
Figure 1: Overview of the proposed transcript correction pipeline. The system identifies acoustic leakage using diarization, followed by a tripartite consensus check to prune leaked words.
System
Libri2Mix
LibriSpeechMix
AMI
test-clean
test-both
2spk
SDM
IHM
Joint Mossformer-Diar
↪ Universal-2 (experimental)
3.52 ( ↓ 0.28%)
9.23 ( ↓ 0.86%)
6.10 ( ↑ 18.91%)
34.77 ( ↓ 2.06%)
28.94 ( ↑ 20.78%)
Sepformer
↪ Universal-2 (baseline)
3.24
16.96
3.71
42.10
39.80
↪ Spk-leakage fix (proposed)
3.20 ( ↓ 1.23%)
16.90 ( ↓ 0.35%)
3.53 ( ↓ 4.85%)
39.66 ( ↓ 5.80%)
35.60 ( ↓ 10.55%)
Table 1: Comparison of WER (%) across datasets. ↓ indicates a performance improvement, while ↑ indicates a decline. The improvement/decline percentage for Joint Mossformer-Diar → Universal-2 is relative to the Mossformer → Universal-2 baseline.
Dataset
Baseline
Leakage Fix
Rel. Δ
Sepformer
L2M test-clean ( N=64 )
16.88
14.33
↓ 15.11%
L2M test-both ( N=65 )
31.25
28.19
↓ 9.79%
LSM 2spk ( N=84 )
29.64
21.54
↓ 27.33%
AMI SDM ( N=138 )
71.53
57.54
↓ 19.56%
AMI IHM ( N=175 )
77.75
62.22
↓ 19.97%
Mossformer
L2M test-clean ( N=79 )
18.82
14.74
↓ 21.68%
Table 2: WER (%) comparison for high-leakage samples (transcripts similarity >0.4 ). Gray values ( N ) indicate sample counts; ↓ denotes relative reduction compared to baseline.
Modality
Dataset
Baseline
Text
Acoustic
All
Sepformer
Libri2Mix test-clean
16.88
31.54
18.46
14.33
Libri2Mix test-both
31.25
44.19
38.41
28.19
LibriSpeechMix 2spk
29.64
41.58
23.35
21.54
AMI SDM
71.53
78.17
58.93
57.54
AMI IHM
77.75
82.87
59.84
62.22
Table 3: Ablation study of the Speaker Leakage Fix on high-leakage subsets (transcripts similarity >0.4 ). We compare the baseline against fixes using text-only, acoustic-only, and combined text + acoustic modalities. All values are WER (%).
Multi-talker automatic speech recognition (MT-ASR) remains challenging in the presence of overlapping speech. Hard segmentation introduces irreversible errors, whereas serialized output training (SOT) avoids explicit segmentation but does not condition a pretrained encoder on speaker activity. We propose Soft Posterior Speaker Injection (SPSI). A Soft Posterior Head predicts per-frame speaker posteriors P^ and injects them into Whisper through Multi-layer Feature-wise Linear Modulation (MFLM) and Speaker Memory Prompts (SMP). The benefit of SPSI is largest where overlap is heaviest and under domain transfer. On controlled two-speaker LibriSpeech overlap, SPSI reduces concatenated minimum-permutation word error rate (cpWER) from 61.5% to 60.0% in the high-overlap bin, and from 51.9% to 51.0% on the full set, relative to SOT. By contrast, Speaker CE, SD-CTC, SA-DiCoW, and Pipeline (oracle/est.\ VAD) do not outperform SOT. Freeze-posterior overlap-heavy adaptation reduces held-out LibriCSS cpWER from 42.3% to 36.8% on sessions 8--9, a 5.5-point gain over SOT. The source code is available at https://github.com/HackerHyper/SPSI.
Jian Zhu, Jun Sun, Jiang Yang +6
Zhejiang Lab, Hangzhou, China · Zhejiang International Studies University, Hangzhou, China · University of Science and Technology of China, Hefei, China
Multi-speaker automatic speech recognition (MSASR) aims to jointly predict content transcriptions, speaker identities, and timestamps, thereby addressing the key question of "who spoke what and when" and holds substantial practical value in real-world multi-speaker scenarios. However, MSASR still encounters considerable challenges in the presence of fast turn transitions, overlapping speech, and complex, diverse multi-speaker scenarios. In this work, we propose DiaScriber, an end-to-end multi-speaker diarization and transcription model built on a speech large language model. We first construct diverse data pipelines to cover a wide variety of multi-speaker scenarios and their complexities, including validation and refinement, turn-transition and overlapping-speech simulation, and multimodal annotation. Furthermore, DiaScriber is developed based on the pretrained version of Qwen3.5-Omni through a three-stage training strategy involving continual pretraining, supervised fine-tuning, and reinforcement learning. Experiments show that DiaScriber achieves superior performance over comparison methods across extensive multi-speaker scenario test sets and demonstrates outstanding generalization ability in unseen multi-speaker scenarios.
We introduce target-speaker unlearning ASR (TSU-ASR) task in a fully end-to-end framework for multi-speaker ASR and diarization. Given a multi-speaker utterance and a set of opt-out speakers who do not wish to have their speech transcribed, the task requires an ASR system to transcribe all speakers except the opt-out ones, while still indicating when those speakers are active. As a first step towards tackling this task, we introduce a novel, light-weight Enrollment-Conditioned Gating (ECG) module attachable to a frozen dual-stream speech LLM that enables ASR for new opt-out speakers dynamically during inference, even those who were not seen during initial ECG training phase. Our experiments on both AMI (English) and AliMeeting (Mandarin) datasets show that speech transcription accuracy for corresponding opt-out words or characters falls from 72.3% to 48.2% and from 73.6% to 27.3%, respectively, while retained speakers' transcription error rates maintain more or less the same. Our approach provides a practical solution for modern video conferencing platforms, allowing speakers to dynamically opt-out from automated AI transcriptions without forcefully leaving the meeting sessions, enabling a privacy-preserving interface for potentially millions of online meetings daily.