SearcharxivSearch

arXiv · 2603.25947

UPV_RIR_DB: A Structured Room Impulse Response Database with Hierarchical Metadata and Acoustic Indicators

Abstract

This paper presents UPV_RIR_DB, a structured database of measured room impulse responses (RIRs) designed to provide acoustic data with explicit spatial metadata and traceable acquisition parameters. The dataset currently contains 166 multichannel RIR files measured in three rooms of the Universitat Polit\`ecnica de Val\`encia (UPV). Each multichannel RIR file contains impulse responses for multiple source-receiver pairs, with each pair covering a 25 cm2 area - the typical size of a personal sound zone. Considering the number of sources and receiver channels associated with each microphone modality, the database contains a total of 18,976 single impulse responses. A hierarchical organization is adopted in which directory structure and metadata jointly describe the measurement context. Each room includes a metadata file containing acquisition parameters, hardware description, spatial coordinates of zones and microphones, and acoustic indicators such as reverberation time. A central index links each RIR file with its experimental context, ensuring traceability and enabling reproducible analysis. The resulting database provides a consistent framework for storing, inspecting, and reusing real RIR measurements while preserving compatibility with both MATLAB- and JSON-based workflows. The UPV_RIR_DB dataset is publicly available through the open repository Zenodo.

Explore related subjects

Keep this discovery

Explore connections, maps & timelines

BibTeXRIS

Jesús García-Gamborino, Laura Fuster, Daniel de la Prida, Luis A. Azpicueta-Ruiz, Gema Piñero. 2026-03-26. UPV_RIR_DB: A Structured Room Impulse Response Database with Hierarchical Metadata and Acoustic Indicators. https://arxiv.org/abs/2603.25947

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

KEEP EXPLORING

Related papers

Diarization Error Decomposition Under Pause Annotation Ambiguity

Speaker diarization evaluation is sensitive to ambiguity in pause annotation, which can inflate diarization error rate (DER) or obscure genuine model errors. We show that morphological closing, which has been used for pause-tolerant diarization evaluation, discards segment-level distinctions. Instead, we propose an exact, overlap-aware decomposition of standard DER into a pause-attributable component, consisting of errors compatible with pause filling, and a residual core component that can serve as a proxy for intrinsic diarization errors. The decomposition leaves DER unchanged, while the pause-attributable and core components vary monotonically with the pause threshold and eventually saturate. Experiments spanning synthetic transformations, annotation mismatch, cross-domain evaluation, and tight-boundary diarization show that the decomposition reveals error sources not apparent from standard DER.

eess.AS

Less can be More: What Aspects of Speech Drive End-of-Turn Detection

In conversational AI, detecting when a speaker has finished talking is crucial for natural turn taking. While recent work incorporates semantics, the relative contribution of different modalities remains unclear. We present a controlled ablation of acoustic, prosodic, and semantic signals for streaming end of turn detection using a lightweight trimodal classifier. Under identical training conditions, the acoustic prosodic combination achieves the best balance of accuracy and latency, achieving utterance F1 of 0.93 with 7.8% false alarms at 400ms median latency. Adding text increases premature detections without improving performance. Feature space analysis confirms that prosodic features have the strongest class separability, while text representations overlap substantially. These findings suggest that turn-taking is primarily conveyed through intonation and silence patterns rather than semantic completeness, enabling faster and more reliable systems without expensive text inference.

eess.AS

Downstream-Task-Aware Unified Source Separation

Task-aware unified source separation (TUSS) enables a single model to handle diverse separation tasks by conditioning on input prompts. However, conventional TUSS does not account for downstream task requirements, such as whether the enhanced speech will be used for human listening or automatic speech recognition (ASR). In this paper, we propose a prompt extension framework for TUSS that incorporates downstream task information into the input prompts and switches the loss function according to the given prompt during training, enabling outputs with different signal characteristics at inference time. Specifically, we introduce an ASR-dedicated prompt paired with a regularized loss function that reduces speech artifacts to improve ASR robustness, while the standard prompt is paired with the conventional SNR loss function. Experiments on the LibriSpeech and JNAS corpora demonstrate that the proposed joint-training scheme enables a single model to improve ASR performance over noisy input across a wide range of SNR conditions by selecting the ASR-dedicated prompt, while maintaining general speech enhancement quality when the standard prompt is used.

eess.AS