33 accepted papers
Generative diffusion priors have recently achieved state-of-the-art performance in natural image super-resolution, demonstrating a powerful capability to synthesize photorealistic details. However, their direct application to remote sensing image super-resolution (RSISR) reveals significant shortcom
The increasing impact of climate change and extreme weather events has spurred growing interest in deep learning for weather research. However, existing studies often rely on weather data in pixel space, which presents several challenges such as smooth outputs in model outputs, limited applicability…
Minimum Variance Distortionless Response (MVDR) is a classical adaptive beamformer that theoretically ensures the distortionless transmission of signals in the target direction, which makes it popular in real applications. Its noise reduction performance actually depends on the accuracy of the noise…
Acoustic Echo Cancellation (AEC) is an essential speech signal processing technology that removes echoes from microphone inputs to facilitate natural-sounding full-duplex communication. Currently, deep learning-based AEC methods primarily focus on refining model architectures, frequently neglecting…
The performance of traditional beamforming algorithms is influenced by the number of microphones, with performance improving as the number increases. However, in practice, the number of microphones is often limited. In this paper, we propose a novel virtual microphone estimation method that combines…
Vision-Language Models (VLMs) have demonstrated great potential in interpreting remote sensing (RS) images through language-guided semantic. However, the effectiveness of these VLMs critically depends on high-quality image-text training data that captures rich semantic relationships between visual c…
In multi-speaker environments the direction of arrival (DOA) of a target speaker is key for improving speech clarity and extracting target speaker’s voice. However, traditional DOA estimation methods often struggle in the presence of noise, reverberation, and particularly when competing speakers are…
Recent advancements in neural audio codec (NAC) unlock new potential in audio signal processing. Studies have increasingly explored leveraging the latent features of NAC for various speech signal processing tasks. This paper introduces the first approach to speech bandwidth extension (BWE) that util…
Target speaker extraction (TSE) aims to isolate a specific voice from multiple mixed speakers relying on a registerd sample. Since voiceprint features usually vary greatly, current end-to-end neural networks require large model parameters which are computational intensive and impractical for real-ti…
Multi-channel speech enhancement extracts speech using multiple microphones that capture spatial cues. Effectively utilizing directional information is therefore key. Deep learning shows great potential on multi-channel speech enhancement and often takes short-time Fourier Transform (STFT) as inputs…
Target speaker extraction aims to isolate a specific speaker’s voice from a composite of multiple sound sources, guided by an enrollment utterance or called anchor. Current methods predominantly derive speaker embeddings from the anchor and integrate them into the separation network to separate the…
Multi-channel speech enhancement leverages multiple microphones to extract target speech signals amid background noise. Effectively utilizing directional cues is key for robust enhancement. While deep learning shows promise for multi-channel speech processing, most methods operate on short-time Four…
Speech enhancement aims to improve speech quality and intelligibility, especially in noisy environments where background noise degrades speech signals. Currently, deep learning methods achieve great success in speech enhancement, e.g. the representative convolutional recurrent neural network (CRN) a…
According to the mechanism of speech production, speech can be decomposed into excitation and vocal tract which are sparsely represented in cepstral domain. In this study, we propose a neural network for monaural speech enhancement on time-frequency cepstral space that is implemented by inserting a…
This report summarizes our system submission to the ICASSP 2023 Clarity Challenge. The goal of the challenge is to estimate clean binaural speech signals within a 5 ms system delay. Our submitted Inplace Cepstral Speech Enhancement (ICSE) system features the following aspects: First, we developed a…
Most neural network speech enhancement models ignore speech production mathematical models by directly mapping Fourier transform spectrums or waveforms. In this work, we propose a neural source filter network for speech enhancement. Specifically, we use homomorphic signal processing and cepstral ana…
Recently, deep learning is introduced in acoustic echo cancellation (AEC) and achieves remarkable performance. For deep learning-based AEC, the most important problem is generalization ability in diversity scenarios. Different from most methods which process the entire frequency band, we propose inp…
There are many methods for detecting forged audio produced by conversion and synthesis. However, as a simpler method of forgery, splicing has not attracted widespread attention. Based on the characteristic that the tampering operation will cause singularities at high-frequency components, we propose…
In this paper, we study the loss-metric mismatch problem of supervised single-channel speech enhancement system. Most of the existing speech enhancement systems achieve unsatisfying performance since their empirically selected loss functions have semantic gaps with the non-differentiable evaluation…
Bone-conduction (BC) microphones capture speech signals by converting the vibrations of the human skull into electrical signals. BC sensors are insensitive to acoustic noise, but limited in bandwidth. On the other hand, conventional or air-conduction (AC) microphones are capable of capturing full-ba…
Under our previous work on frequency bin-wise independent processing, a dramatic reduction of the computational complexity for recurrent neural networks (RNN) is achieved. So that a massive deployment of RNN in time dimension is realized in this paper, by using the channel-wise long short-term memor…
Speech quality and intelligibility can be severely degraded by back-ground noise in mobile communication. In order to attenuate back-ground noise, speech enhancement systems have been integrated into mobile phones, and a microphone array is typically deployed to improve the enhancement performance.…
This paper deals with the problem of separating target speech signal from reverberant and noisy environment with dual microphones, where the target speech comes from a predefined direction range. First, we apply two differential beamformers with opposite directions to dual-channel inputs. Then, the…
Speaker extraction aims to separate a target speaker from multiple voices which is useful for applications, e.g. teleconference. In many practical cases, it has an opportunity to get a piece voice of the target speaker in advance, which provides useful information for speaker extraction. This paper…
Recently, deep neural networks (DNNs) have achieved incredible performance in speaker verification. However, most of which remains sensitive to environment noise. In this paper, we propose an end-to-end speaker verification framework to enhance the robustness against background noise. The proposed f…
In mobile speech communication, the quality and intelligibility of the received speech can be severely degraded by background noise if the far-end talker is in an adverse acoustic environment. Therefore, speech enhancement algorithms are typically integrated into mobile phones to remove background n…
Speech enhancement aims to separate a target speech from background noise. Recently, speech enhancement has been formulated as a supervised learning problem, in which a learning machine is trained to estimate the target spectrum denoted as mapping-based method or a time-frequency mask denoted as mas…
Direction of arrival (DOA) estimation is an important topic in microphone array processing. Conventional methods work well in relatively clean conditions but suffer from noise and reverberation distortions. Recently, deep learning-based methods show the robustness to noise and reverberation. However…
Supervised speech separation methods train learning machine to cast the noisy speech to the target clean speech. Most of them use mean-square error (MSE) as loss function. However, MSE is not the perfect choice because it doesn't match the human auditory perception. Short-time objective intelligibil…
We propose a speech enhancement algorithm based on single- and multi-microphone processing techniques. The core of the algorithm estimates a time-frequency mask which represents the target speech and use masking-based beamforming to enhance corrupted speech. Specifically, in single-microphone proces…
Pitch is an important characteristic of speech and is useful for many applications. However, pitch determination in noisy conditions is difficult. In this paper, we propose a supervised learning algorithm to estimate pitch using a convolutional neural network (CNN). Specifically, we use a CNN for pi…
The targets of speech separation, whether ideal masks or magnitude spectrograms of interest, have prominent spectro-temporal structures. These characteristics are very worthy to be exploited for speech separation, however, they are usually ignored in previous works. In this paper, we use nonnegative…
Pitch information is an important cue for speech separation. However, pitch estimation in noisy condition is also a task as challenging as speech separation. In this paper, we propose a supervised learning architecture which combines these two problems concisely. The proposed algorithm is based on d…