Searcharxiv⌕ Search
Home/Archive/arXiv

arXiv · 2303.08239

Facilitating deep acoustic phenotyping: A basic coding scheme of infant vocalisations preluding computational analysis, machine learning and clinical reasoning

Tomas Kulvicius·Sigrun Lang·Claudius AA Widmann·Nina Hansmann·Daniel Holzinger·Luise Poustka·Dajie Zhang·Peter B Marschik
View original source ↗Read source document ↗Export metadata ↓

Abstract

Theoretical background: early verbal development is not yet fully understood, especially in its formative phase. Research question: can a reliable, easy-to-use coding scheme for the classification of early infant vocalizations be defined that is applicable as a basis for further analysis of language development? Methods: in a longitudinal study of 45 neurotypical infants, we analyzed vocalizations of the first 4 months of life. Audio segments were assigned to 5 classes: (1) Voiced and (2) Voiceless vocalizations; (3) Defined signal; (4) Non-target; (5) Nonassignable. Results: Two female coders with different experience achieved high agreement without intensive training. Discussion and Conclusion: The reliable scheme can be used in research and clinical settings for efficient coding of infant vocalizations, as a basis for detailed manual and machine analyses.

Explore related subjects

cs.SDeess.AS

Keep this discovery

Explore connections, maps & timelines

BibTeXRIS

Tomas Kulvicius, Sigrun Lang, Claudius AA Widmann, Nina Hansmann, Daniel Holzinger, Luise Poustka, Dajie Zhang, Peter B Marschik. 2023-03-14. Facilitating deep acoustic phenotyping: A basic coding scheme of infant vocalisations preluding computational analysis, machine learning and clinical reasoning. https://doi.org/10.1026/0942-5403%2Fa000418

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

KEEP EXPLORING

Related papers

arXiv2026

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↗
arXiv2026

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↗