SearcharxivSearch

arXiv · 2606.21457

DisSpeech: Low-Resource Controllable Mandarin Stuttered Speech Synthesis for ASR Augmentation

Abstract

Stuttered speech recognition remains challenging, with disfluencies such as repetitions, prolongations, and blocks disrupting speech continuity and acoustic patterns. This problem is further aggravated in Mandarin scenarios by the limited availability of stuttered speech data, which makes it difficult to train robust ASR models for diverse disfluency patterns. To address this problem, this paper proposes DisSpeech, a discrete speech token-based framework for low-resource controllable Mandarin stuttered speech synthesis and ASR data augmentation. The proposed framework introduces explicit stuttering event labels to control different disfluency patterns. Text and stuttering event labels are mapped into semantic speech tokens by a non-autoregressive masked generative Transformer, followed by prosody-aware acoustic reconstruction with explicit pitch and energy modeling. With fine-tuning using less than 50 hours of Mandarin stuttered speech, DisSpeech can generate controllable stuttered speech with competitive speech quality. Experimental results show that the proposed method outperforms previous stuttered speech synthesis methods in both speech quality and event controllability. Furthermore, the synthesized stuttered speech effectively improves multiple ASR models, with Qwen3-ASR-0.6B achieving a state-of-the-art CER of 4.19% on the evaluated Mandarin stuttered speech recognition task, while causing only slight degradation on fluent speech.

Explore related subjects

Keep this discovery

BibTeXRIS

Yao Lu. 2026-06-19. DisSpeech: Low-Resource Controllable Mandarin Stuttered Speech Synthesis for ASR Augmentation. https://arxiv.org/abs/2606.21457

Cite the original work for its findings. Save a collection to share your selection of sources.

KEEP EXPLORING

Related papers

Xiaomi-CocktailASR-1 Technical Report

Recently, large language model (LLM) based ASR models have achieved significant progress, yet they generally lack support for multi-speaker scenarios, where the cocktail party problem remains a critical bottleneck for further advancing ASR. Existing TS-ASR methods, including end-to-end architectures with speaker embeddings and latest LLM-based explorations suffer from degraded single-speaker performance and the inability to reject when the target speaker is absent. In this paper, we propose Xiaomi-CocktailASR-1, an LLM-based end-to-end TS-ASR architecture. By utilizing reference speech as voiceprint prompts, it directly transcribes the target speaker's speech without requiring speech separation. Xiaomi-CocktailASR-1 maintains competitive performance in single-speaker scenarios, comparable to mainstream ASR models. It also features a negative sample rejection capability, outputting empty text when the target speaker is absent from the mixed speech. Additionally, Xiaomi-CocktailASR-1 supports a Chain-of-Thought (CoT) reasoning mode to provide explicit reasoning steps. Extensive experiments on various synthetic and real-world multispeaker benchmarks demonstrate that Xiaomi-CocktailASR-1 achieves state-of-the-art performance, effectively addressing the cocktail party problem through a unified architecture that balances multispeaker and single-speaker recognition accuracy, along with rejection capability.

cs.SD

EConv-TasNet: Efficient Conv-TasNet for Effective Speech Separation

Conv-TasNet has served as a strong baseline for time-domain speech separation, and many studies have extended it with advanced architectures such as dual-path networks, U-Nets, and attention mechanisms. However, these methods often introduce high computational cost and complexity, limiting their deployment in resource-constrained scenarios. To address this issue, we propose eConv-TasNet, an efficient variant of Conv-TasNet that improves both effectiveness and efficiency without relying on resource-intensive modules. The proposed model consists of a group-wise early-splitting (GES) module and a multi-group feature aggregation (MGFA) module. GES generates discriminative speaker embeddings at intermediate stages, while MGFA progressively aggregates these group-level representations for refined mask estimation. Experimental results show that eConv-TasNet reduces model size by 22.4%, accelerates inference by 18.9%, and improves SI-SNRi by 14.0%-28.0% across three public benchmarks. Moreover, it achieves competitive performance compared with state-of-the-art methods while requiring significantly fewer parameters and lower inference cost. These results demonstrate a favorable efficiency-effectiveness trade-off for edge deployment.

cs.SD