SearcharxivSearch

arXiv · 2506.14864

pycnet-audio: A Python package to support bioacoustics data processing

Abstract

Passive acoustic monitoring is an emerging approach in wildlife research that leverages recent improvements in purpose-made automated recording units (ARUs). The general approach is to deploy ARUs in the field to record on a programmed schedule for extended periods (weeks or months), after which the audio data are retrieved. These data must then be processed, typically either by measuring or analyzing characteristics of the audio itself (e.g. calculating acoustic indices), or by searching for some signal of interest within the recordings, e.g. vocalizations or other sounds produced by some target species, anthropogenic or environmental noise, etc. In the latter case, some method is required to locate the signal(s) of interest within the audio. While very small datasets can simply be searched manually, even modest projects can produce audio datasets on the order of 105 hours of recordings, making manual review impractical and necessitating some form of automated detection. pycnet-audio (Ruff 2024) is intended to provide a practical processing workflow for acoustic data, built around the PNW-Cnet model, which was initially developed by the U.S. Forest Service to support population monitoring of northern spotted owls (Strix occidentalis caurina) and other forest owls (Lesmeister and Jenkins 2022; Ruff et al. 2020). PNW-Cnet has been expanded to detect vocalizations of ca. 80 forest wildlife species and numerous forms of anthropogenic and environmental noise (Ruff et al. 2021, 2023).

Explore related subjects

Keep this discovery

BibTeXRIS

Zachary J. Ruff, Damon B. Lesmeister. 2025-06-17. pycnet-audio: A Python package to support bioacoustics data processing. https://arxiv.org/abs/2506.14864

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