Searcharxiv⌕ Search

arXiv · 2610.29778

Geometry-Conditioned Target Speaker Extraction based on spatial features incorporating acoustic propagation principles

Abstract

Recently, spatially selective non-linear filter (SSF) was proposed for target speaker extraction, utilizing the direction-of-arrival (DOA) of the target speaker as a spatial cue. To overcome its dependency on a fixed microphone array geometry, geometry-conditioned SSF (GCSSF) was proposed, which maps the target DOA and the microphone positions into abstract high-dimensional embeddings using positional encoding. As an alternative to these high-dimensional embeddings, in this paper we propose utilizing two spatial features incorporating acoustic propagation principles for geometry conditioning: 1) the steering vector of the target speaker, which captures inter-microphone phase differences, and 2) a spatial mask based on GCC-PHAT, which quantifies presence of the target speaker per time-frequency bin. These features are used as input to a geometry-conditioning encoder and integrated via a FiLM layer into the GCSSF system. Experimental results on circular and random arrays demonstrate that conditioning with spatial features improves target extraction performance compared to abstract high-dimensional embeddings, maintaining a clear performance advantage across a wide range of target DOA errors.

Explore related subjects

Keep this discovery

Explore connections, maps & timelines

BibTeXRIS

Jiatong Li, Simon Doclo. 2026-10-21. Geometry-Conditioned Target Speaker Extraction based on spatial features incorporating acoustic propagation principles. https://arxiv.org/abs/2610.29778

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

KEEP EXPLORING

Related papers

A Semi-spontaneous Dutch Speech Dataset for Speech Enhancement and Speech Recognition

We present DRES: a 1.5-hour Dutch realistic elicited (semi-spontaneous) speech dataset from 80 speakers recorded in noisy, public indoor environments. DRES was designed as a test set for the evaluation of state-of-the-art (SotA) automatic speech recognition (ASR) and speech enhancement (SE) models in a real-world scenario: a person speaking in a public indoor space with background talkers and noise. The speech was recorded with a four-channel linear microphone array. In this work we evaluate the speech quality of five well-known single-channel SE algorithms and the recognition performance of eight SotA off-the-shelf ASR models before and after applying SE on the speech of DRES. We found that five out of the eight ASR models have WERs lower than 22\% on DRES, despite the challenging conditions. In contrast to recent work, we did not find a positive effect of modern single-channel SE on ASR performance, emphasizing the importance of evaluating in realistic conditions.

eess.AS↗

Depth through recurrence: Looped transformers for flow-matching TTS

We study how to organize Transformer depth through recurrence in flow-matching text-to-speech, varying the amount, order, and placement of weight reuse. Seven layouts perform 18 block calls per network evaluation under a common training objective and sampler. On Seed-TTS and LibriSpeech-PC, SEQUENCE applies each of nine blocks twice consecutively, retaining competitive intelligibility, speaker similarity, and predicted speech quality at 32 sampling steps with 47.1% fewer parameters than the unshared baseline. Cycling six blocks three times further reduces model size but raises 32-step word error rates relative to cycling nine blocks twice. At matched parameter counts and executed depth, reuse order and sharing position produce different quality trade-offs. These comparisons depend on sampling budget: Prefix and Suffix have similar 32-step word error rates, but Suffix is worse by 3.44 and 5.97 percentage points at four steps on the two datasets, respectively. Only Middle ranks first or second in mean word error rate at 32 and four steps on both datasets. These results show that the organization of recurrent computation affects synthesis quality, and that reuse layouts should be selected for both the sampling budget and the quality dimensions of interest.

eess.AS↗

XingChen-UniFE: A Unified Generative Speech Front-End via Complementary Acoustic-Semantic Modeling

This paper presents XingChen-UniFE, a unified large audio language model (LALM) for general speech restoration (GSR) and target speaker extraction (TSE). The proposed model employs complementary semantic and acoustic representations through a dual-stream encoder, where a self-supervised learning model provides hierarchical semantic features and a learnable acoustic branch captures fine-grained structures from complex spectra. A bidirectional cross-attention module further facilitates information exchange between the two representations. The fused representation conditions the LALM for discrete speech token generation with task and speaker information. Experiments demonstrate strong performance on both GSR and TSE and robust generalization to unseen degradation conditions.

eess.AS↗