Searcharxiv⌕ Search
Home/Archive/arXiv

arXiv · 2509.02771

Analysis of Speaker Verification Performance Trade-offs with Neural Audio Codec Transmission

Nirmalya Mallick Thakur·Jia Qi Yip·Eng Siong Chng
View original source ↗Read source document ↗Export metadata ↓

Abstract

Neural audio codecs (NACs) have made significant advancements in recent years and are rapidly being adopted in many audio processing pipelines. However, they can introduce audio distortions which degrade speaker verification (SV) performance. This study investigates the impact of both traditional and neural audio codecs at varying bitrates on three state of-the-art SV models evaluated on the VoxCeleb1 dataset. Our findings reveal a consistent degradation in SV performance across all models and codecs as bitrates decrease. Notably, NACs do not fundamentally break SV performance when compared to traditional codecs. They outperform Opus by 6-8% at low-bitrates (< 12 kbps) and remain marginally behind at higher bitrates ($\approx$ 24 kbps), with an EER increase of only 0.4-0.7%. The disparity at higher bitrates is likely due to the primary optimization of NACs for perceptual quality, which can inadvertently discard critical speaker-discriminative features, unlike Opus which was designed to preserve vocal characteristics. Our investigation suggests that NACs are a feasible alternative to traditional codecs, especially under bandwidth limitations. To bridge the gap at higher bitrates, future work should focus on developing speaker-aware NACs or retraining and adapting SV models.

Explore related subjects

cs.SDeess.AS

Keep this discovery

BibTeXRIS

Nirmalya Mallick Thakur, Jia Qi Yip, Eng Siong Chng. 2025-09-02. Analysis of Speaker Verification Performance Trade-offs with Neural Audio Codec Transmission. https://arxiv.org/abs/2509.02771

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↗