1,874 research outputs found
Monaural Singing Voice Separation with Skip-Filtering Connections and Recurrent Inference of Time-Frequency Mask
Singing voice separation based on deep learning relies on the usage of
time-frequency masking. In many cases the masking process is not a learnable
function or is not encapsulated into the deep learning optimization.
Consequently, most of the existing methods rely on a post processing step using
the generalized Wiener filtering. This work proposes a method that learns and
optimizes (during training) a source-dependent mask and does not need the
aforementioned post processing step. We introduce a recurrent inference
algorithm, a sparse transformation step to improve the mask generation process,
and a learned denoising filter. Obtained results show an increase of 0.49 dB
for the signal to distortion ratio and 0.30 dB for the signal to interference
ratio, compared to previous state-of-the-art approaches for monaural singing
voice separation
Deep Learning for Environmentally Robust Speech Recognition: An Overview of Recent Developments
Eliminating the negative effect of non-stationary environmental noise is a
long-standing research topic for automatic speech recognition that stills
remains an important challenge. Data-driven supervised approaches, including
ones based on deep neural networks, have recently emerged as potential
alternatives to traditional unsupervised approaches and with sufficient
training, can alleviate the shortcomings of the unsupervised methods in various
real-life acoustic environments. In this light, we review recently developed,
representative deep learning approaches for tackling non-stationary additive
and convolutional degradation of speech with the aim of providing guidelines
for those involved in the development of environmentally robust speech
recognition systems. We separately discuss single- and multi-channel techniques
developed for the front-end and back-end of speech recognition systems, as well
as joint front-end and back-end training frameworks
Deep Denoising for Hearing Aid Applications
Reduction of unwanted environmental noises is an important feature of today's
hearing aids (HA), which is why noise reduction is nowadays included in almost
every commercially available device. The majority of these algorithms, however,
is restricted to the reduction of stationary noises. In this work, we propose a
denoising approach based on a three hidden layer fully connected deep learning
network that aims to predict a Wiener filtering gain with an asymmetric input
context, enabling real-time applications with high constraints on signal delay.
The approach is employing a hearing instrument-grade filter bank and complies
with typical hearing aid demands, such as low latency and on-line processing.
It can further be well integrated with other algorithms in an existing HA
signal processing chain. We can show on a database of real world noise signals
that our algorithm is able to outperform a state of the art baseline approach,
both using objective metrics and subject tests.Comment: submitted to IWAENC 201
Entropy-based feature extraction for electromagnetic discharges classification in high-voltage power generation
This work exploits four entropy measures known as Sample, Permutation, Weighted Permutation, and Dispersion Entropy to extract relevant information from Electromagnetic Interference (EMI) discharge signals that are useful in fault diagnosis of High-Voltage (HV) equipment. Multi-class classification algorithms are used to classify or distinguish between various discharge sources such as Partial Discharges (PD), Exciter, Arcing, micro Sparking and Random Noise. The signals were measured and recorded on different sites followed by EMI expert’s data analysis in order to identify and label the discharge source type contained within the signal. The classification was performed both within each site and across all sites. The system performs well for both cases with extremely high classification accuracy within site. This work demonstrates the ability to extract relevant entropy-based features from EMI discharge sources from time-resolved signals requiring minimal computation making the system ideal for a potential application to online condition monitoring based on EMI
A Recurrent Encoder-Decoder Approach with Skip-filtering Connections for Monaural Singing Voice Separation
The objective of deep learning methods based on encoder-decoder architectures
for music source separation is to approximate either ideal time-frequency masks
or spectral representations of the target music source(s). The spectral
representations are then used to derive time-frequency masks. In this work we
introduce a method to directly learn time-frequency masks from an observed
mixture magnitude spectrum. We employ recurrent neural networks and train them
using prior knowledge only for the magnitude spectrum of the target source. To
assess the performance of the proposed method, we focus on the task of singing
voice separation. The results from an objective evaluation show that our
proposed method provides comparable results to deep learning based methods
which operate over complicated signal representations. Compared to previous
methods that approximate time-frequency masks, our method has increased
performance of signal to distortion ratio by an average of 3.8 dB
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
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