4,130 research outputs found
Weakly Supervised Audio Source Separation via Spectrum Energy Preserved Wasserstein Learning
Separating audio mixtures into individual instrument tracks has been a long
standing challenging task. We introduce a novel weakly supervised audio source
separation approach based on deep adversarial learning. Specifically, our loss
function adopts the Wasserstein distance which directly measures the
distribution distance between the separated sources and the real sources for
each individual source. Moreover, a global regularization term is added to
fulfill the spectrum energy preservation property regardless separation. Unlike
state-of-the-art weakly supervised models which often involve deliberately
devised constraints or careful model selection, our approach need little prior
model specification on the data, and can be straightforwardly learned in an
end-to-end fashion. We show that the proposed method performs competitively on
public benchmark against state-of-the-art weakly supervised methods
Deep speech inpainting of time-frequency masks
Transient loud intrusions, often occurring in noisy environments, can
completely overpower speech signal and lead to an inevitable loss of
information. While existing algorithms for noise suppression can yield
impressive results, their efficacy remains limited for very low signal-to-noise
ratios or when parts of the signal are missing. To address these limitations,
here we propose an end-to-end framework for speech inpainting, the
context-based retrieval of missing or severely distorted parts of
time-frequency representation of speech. The framework is based on a
convolutional U-Net trained via deep feature losses, obtained using speechVGG,
a deep speech feature extractor pre-trained on an auxiliary word classification
task. Our evaluation results demonstrate that the proposed framework can
recover large portions of missing or distorted time-frequency representation of
speech, up to 400 ms and 3.2 kHz in bandwidth. In particular, our approach
provided a substantial increase in STOI & PESQ objective metrics of the
initially corrupted speech samples. Notably, using deep feature losses to train
the framework led to the best results, as compared to conventional approaches.Comment: Accepted to InterSpeech202
The Conversation: Deep Audio-Visual Speech Enhancement
Our goal is to isolate individual speakers from multi-talker simultaneous
speech in videos. Existing works in this area have focussed on trying to
separate utterances from known speakers in controlled environments. In this
paper, we propose a deep audio-visual speech enhancement network that is able
to separate a speaker's voice given lip regions in the corresponding video, by
predicting both the magnitude and the phase of the target signal. The method is
applicable to speakers unheard and unseen during training, and for
unconstrained environments. We demonstrate strong quantitative and qualitative
results, isolating extremely challenging real-world examples.Comment: To appear in Interspeech 2018. We provide supplementary material with
interactive demonstrations on
http://www.robots.ox.ac.uk/~vgg/demo/theconversatio
Block-Online Multi-Channel Speech Enhancement Using DNN-Supported Relative Transfer Function Estimates
This work addresses the problem of block-online processing for multi-channel
speech enhancement. Such processing is vital in scenarios with moving speakers
and/or when very short utterances are processed, e.g., in voice assistant
scenarios. We consider several variants of a system that performs beamforming
supported by DNN-based voice activity detection (VAD) followed by
post-filtering. The speaker is targeted through estimating relative transfer
functions between microphones. Each block of the input signals is processed
independently in order to make the method applicable in highly dynamic
environments. Owing to the short length of the processed block, the statistics
required by the beamformer are estimated less precisely. The influence of this
inaccuracy is studied and compared to the processing regime when recordings are
treated as one block (batch processing). The experimental evaluation of the
proposed method is performed on large datasets of CHiME-4 and on another
dataset featuring moving target speaker. The experiments are evaluated in terms
of objective and perceptual criteria (such as signal-to-interference ratio
(SIR) or perceptual evaluation of speech quality (PESQ), respectively).
Moreover, word error rate (WER) achieved by a baseline automatic speech
recognition system is evaluated, for which the enhancement method serves as a
front-end solution. The results indicate that the proposed method is robust
with respect to short length of the processed block. Significant improvements
in terms of the criteria and WER are observed even for the block length of 250
ms.Comment: 10 pages, 8 figures, 4 tables. Modified version of the article
accepted for publication in IET Signal Processing journal. Original results
unchanged, additional experiments presented, refined discussion and
conclusion
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