342 research outputs found
Score-Informed Source Separation for Musical Audio Recordings [An overview]
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The impact of exploiting spectro-temporal context in computational speech segregation
The experimental data from the study:
https://asa.scitation.org/doi/10.1121/1.5020273
Group 1 contains results, masks and audio from the models of the 16 GMM component segregation system
Group 2 contains results, masks and audio from the models of the 64 GMM component segregation system
There are three folders:
Audio:
The CLUE sentences that were used for the listener study
IBM = Ideal Binary Mask, UP = UnProcessed, EBM = Estimated Binary Mask.
The IBM and UP are stored in one of the configuration folders (Front-end), that is:
Audio\Group1\Front-end\icra_01_10sec_matched\UP
Audio\Group1\Front-end\icra_01_10sec_matched\IBM
Audio\Group1\Front-end\icra_01_10sec_matched\EBM
Results:
The computed metrics for group 1 & 2 as well as Word Recognition Scores (WRSs) from the listener study
BinaryMasks:
a priori SNR masks, IBMs and EBMs from group 1 and 2.
Developed with Matlab R2016a
Structured Sparsity Models for Multiparty Speech Recovery from Reverberant Recordings
We tackle the multi-party speech recovery problem through modeling the
acoustic of the reverberant chambers. Our approach exploits structured sparsity
models to perform room modeling and speech recovery. We propose a scheme for
characterizing the room acoustic from the unknown competing speech sources
relying on localization of the early images of the speakers by sparse
approximation of the spatial spectra of the virtual sources in a free-space
model. The images are then clustered exploiting the low-rank structure of the
spectro-temporal components belonging to each source. This enables us to
identify the early support of the room impulse response function and its unique
map to the room geometry. To further tackle the ambiguity of the reflection
ratios, we propose a novel formulation of the reverberation model and estimate
the absorption coefficients through a convex optimization exploiting joint
sparsity model formulated upon spatio-spectral sparsity of concurrent speech
representation. The acoustic parameters are then incorporated for separating
individual speech signals through either structured sparse recovery or inverse
filtering the acoustic channels. The experiments conducted on real data
recordings demonstrate the effectiveness of the proposed approach for
multi-party speech recovery and recognition.Comment: 31 page
Influence of binary mask estimation errors on robust speaker identification
Missing-data strategies have been developed to improve the noise-robustness of automatic speech recognition systems in adverse acoustic conditions. This is achieved by classifying time-frequency (T-F) units into reliable and unreliable components, as indicated by a so-called binary mask. Different approaches have been proposed to handle unreliable feature components, each with distinct advantages. The direct masking (DM) approach attenuates unreliable T-F units in the spectral domain, which allows the extraction of conventionally used mel-frequency cepstral coefficients (MFCCs). Instead of attenuating unreliable components in the feature extraction front-end, full marginalization (FM) discards unreliable feature components in the classification back-end. Finally, bounded marginalization (BM) can be used to combine the evidence from both reliable and unreliable feature components during classification. Since each of these approaches utilizes the knowledge about reliable and unreliable feature components in a different way, they will respond differently to estimation errors in the binary mask. The goal of this study was to identify the most effective strategy to exploit knowledge about reliable and unreliable feature components in the context of automatic speaker identification (SID). A systematic evaluation under ideal and non-ideal conditions demonstrated that the robustness to errors in the binary mask varied substantially across the different missing-data strategies. Moreover, full and bounded marginalization showed complementary performances in stationary and non-stationary background noises and were subsequently combined using a simple score fusion. This approach consistently outperformed individual SID systems in all considered experimental conditions
HiFi-GAN: High-Fidelity Denoising and Dereverberation Based on Speech Deep Features in Adversarial Networks
Real-world audio recordings are often degraded by factors such as noise,
reverberation, and equalization distortion. This paper introduces HiFi-GAN, a
deep learning method to transform recorded speech to sound as though it had
been recorded in a studio. We use an end-to-end feed-forward WaveNet
architecture, trained with multi-scale adversarial discriminators in both the
time domain and the time-frequency domain. It relies on the deep feature
matching losses of the discriminators to improve the perceptual quality of
enhanced speech. The proposed model generalizes well to new speakers, new
speech content, and new environments. It significantly outperforms
state-of-the-art baseline methods in both objective and subjective experiments.Comment: Accepted by INTERSPEECH 202
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