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相关论文: Low-Latency Deep Clustering For Speech Separation

200 篇论文

The performance of automatic speech recognition (ASR) systems severely degrades when multi-talker speech overlap occurs. In meeting environments, speech separation is typically performed to improve the robustness of ASR systems. Recently,…

音频与语音处理 · 电气工程与系统科学 2023-01-18 Hassan Taherian , DeLiang Wang

In this paper two different approaches to enhance the performance of the most challenging component of a Speaker Diarization system are presented, i.e. the speaker clustering part. A processing step is proposed enhancing the input features…

音频与语音处理 · 电气工程与系统科学 2019-09-04 Dimitrios Dimitriadis

This paper describes an effective unsupervised speaker indexing approach. We suggest a two stage algorithm to speed-up the state-of-the-art algorithm based on the Bayesian Information Criterion (BIC). In the first stage of the merging…

声音 · 计算机科学 2010-09-27 Konstantin Biatov

Multi-speaker automatic speech recognition (ASR) aims to transcribe conversational speech involving multiple speakers, requiring the model to capture not only what was said, but also who said it and sometimes when it was spoken. Recent…

音频与语音处理 · 电气工程与系统科学 2026-04-27 Li Li , Ming Cheng , Weixin Zhu , Yannan Wang , Juan Liu , Ming Li

Speaker Diarization (SD) is a crucial component of modern end-to-end ASR pipelines. Traditional SD systems, which are typically audio-based and operate independently of ASR, often introduce speaker errors, particularly during speaker…

音频与语音处理 · 电气工程与系统科学 2025-01-16 Anurag Kumar , Rohit Paturi , Amber Afshan , Sundararajan Srinivasan

In this work, we propose a classifier for distinguishing device-directed queries from background speech in the context of interactions with voice assistants. Applications include rejection of false wake-ups or unintended interactions as…

计算与语言 · 计算机科学 2018-08-09 Sri Harish Mallidi , Roland Maas , Kyle Goehner , Ariya Rastrow , Spyros Matsoukas , Björn Hoffmeister

In this paper, we introduce a causal low-latency low-complexity approach for binaural multichannel blind speaker separation in noisy reverberant conditions. The model, referred to as Group Communication Binaural Filter and Sum Network…

音频与语音处理 · 电气工程与系统科学 2023-12-11 Nils L. Westhausen , Bernd T. Meyer

This study investigates phase reconstruction for deep learning based monaural talker-independent speaker separation in the short-time Fourier transform (STFT) domain. The key observation is that, for a mixture of two sources, with their…

声音 · 计算机科学 2018-11-26 Zhong-Qiu Wang , Ke Tan , DeLiang Wang

Significant performance degradation of automatic speech recognition (ASR) systems is observed when the audio signal contains cross-talk. One of the recently proposed approaches to solve the problem of multi-speaker ASR is the deep…

声音 · 计算机科学 2019-09-26 Tobias Menne , Ilya Sklyar , Ralf Schlüter , Hermann Ney

Effective attention modules have played a crucial role in the success of Transformer-based large language models (LLMs), but the quadratic time and memory complexities of these attention modules also pose a challenge when processing long…

分布式、并行与集群计算 · 计算机科学 2024-06-07 Ao Sun , Weilin Zhao , Xu Han , Cheng Yang , Zhiyuan Liu , Chuan Shi , Maosong Sun

State-of-the-art language models (LMs) represented by long-short term memory recurrent neural networks (LSTM-RNNs) and Transformers are becoming increasingly complex and expensive for practical applications. Low-bit neural network…

计算与语言 · 计算机科学 2021-12-22 Junhao Xu , Jianwei Yu , Shoukang Hu , Xunying Liu , Helen Meng

Nowadays, there is a strong need to deploy the target speaker separation (TSS) model on mobile devices with a limitation of the model size and computational complexity. To better perform TSS for mobile voice communication, we first make a…

声音 · 计算机科学 2021-06-08 Yuanyuan Bao , Yanze Xu , Na Xu , Wenjing Yang , Hongfeng Li , Shicong Li , Yongtao Jia , Fei Xiang , Jincheng He , Ming Li

Scaling test-time computation--generating and analyzing multiple or sequential outputs for a single input--has become a promising strategy for improving the reliability and quality of large language models (LLMs), as evidenced by advances…

计算与语言 · 计算机科学 2025-06-03 Sungjae Lee , Hoyoung Kim , Jeongyeon Hwang , Eunhyeok Park , Jungseul Ok

Many speech enhancement (SE) methods rely on continuous representations. Recently, discrete audio tokens have been explored to enable autoregressive generation for SE. However, it remains unclear whether discretization itself consistently…

声音 · 计算机科学 2026-03-24 Jingyi Li , Luca Della Libera , Mirco Ravanelli , Cem Subakan

Complex-valued processing has brought deep learning-based speech enhancement and signal extraction to a new level. Typically, the process is based on a time-frequency (TF) mask which is applied to a noisy spectrogram, while complex masks…

音频与语音处理 · 电气工程与系统科学 2022-02-02 Hendrik Schröter , Alberto N. Escalante-B. , Tobias Rosenkranz , Andreas Maier

In this paper, we present an improved feedforward sequential memory networks (FSMN) architecture, namely Deep-FSMN (DFSMN), by introducing skip connections between memory blocks in adjacent layers. These skip connections enable the…

神经与进化计算 · 计算机科学 2018-03-15 Shiliang Zhang , Ming Lei , Zhijie Yan , Lirong Dai

Audio source separation is usually achieved by estimating the short-time Fourier transform (STFT) magnitude of each source, and then applying a spectrogram inversion algorithm to retrieve time-domain signals. In particular, the multiple…

声音 · 计算机科学 2020-04-22 Paul Magron , Tuomas Virtanen

A robust multichannel speaker diarization and separation system is proposed by exploiting the spatio-temporal activity of the speakers. The system is realized in a hybrid architecture that combines the array signal processing units and the…

音频与语音处理 · 电气工程与系统科学 2024-01-31 Yicheng Hsu , Ssuhan Chen , Mingsian R. Bai

Speech separation has been very successful with deep learning techniques. Substantial effort has been reported based on approaches over spectrogram, which is well known as the standard time-and-frequency cross-domain representation for…

声音 · 计算机科学 2019-04-17 Gene-Ping Yang , Chao-I Tuan , Hung-Yi Lee , Lin-shan Lee

LPCNet is an efficient vocoder that combines linear prediction and deep neural network modules to keep the computational complexity low. In this work, we present two techniques to further reduce it's complexity, aiming for a low-cost LPCNet…