SearcharxivSearch

arXiv · 2209.04077

Prediction method of Soundscape Impressions using Environmental Sounds and Aerial Photographs

Abstract

We investigate an method for quantifying city characteristics based on impressions of a sound environment. The quantification of the city characteristics will be beneficial to government policy planning, tourism projects, etc. In this study, we try to predict two soundscape impressions, meaning pleasantness and eventfulness, using sound data collected by the cloud-sensing method. The collected sounds comprise meta information of recording location using Global Positioning System. Furthermore, the soundscape impressions and sound-source features are separately assigned to the cloud-sensing sounds by assessments defined using Swedish Soundscape-Quality Protocol, assessing the quality of the acoustic environment. The prediction models are built using deep neural networks with multi-layer perceptron for the input of 10-second sound and the aerial photographs of its location. An acoustic feature comprises equivalent noise level and outputs of octave-band filters every second, and statistics of them in 10~s. An image feature is extracted from an aerial photograph using ResNet-50 and autoencoder architecture. We perform comparison experiments to demonstrate the benefit of each feature. As a result of the comparison, aerial photographs and sound-source features are efficient to predict impression information. Additionally, even if the sound-source features are predicted using acoustic and image features, the features also show fine results to predict the soundscape impression close to the result of oracle sound-source features.

Explore related subjects

Keep this discovery

BibTeXRIS

Yusuke Ono, Sunao Hara, Masanobu Abe. 2022-09-09. Prediction method of Soundscape Impressions using Environmental Sounds and Aerial Photographs. https://arxiv.org/abs/2209.04077

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