941 research outputs found
Perceptually Motivated Wavelet Packet Transform for Bioacoustic Signal Enhancement
A significant and often unavoidable problem in bioacoustic signal processing is the presence of background noise due to an adverse recording environment. This paper proposes a new bioacoustic signal enhancement technique which can be used on a wide range of species. The technique is based on a perceptually scaled wavelet packet decomposition using a species-specific Greenwood scale function. Spectral estimation techniques, similar to those used for human speech enhancement, are used for estimation of clean signal wavelet coefficients under an additive noise model. The new approach is compared to several other techniques, including basic bandpass filtering as well as classical speech enhancement methods such as spectral subtraction, Wiener filtering, and Ephraim–Malah filtering. Vocalizations recorded from several species are used for evaluation, including the ortolan bunting (Emberiza hortulana), rhesus monkey (Macaca mulatta), and humpback whale (Megaptera novaeanglia), with both additive white Gaussian noise and environment recording noise added across a range of signal-to-noise ratios (SNRs). Results, measured by both SNR and segmental SNR of the enhanced wave forms, indicate that the proposed method outperforms other approaches for a wide range of noise conditions
Group-Sparse Signal Denoising: Non-Convex Regularization, Convex Optimization
Convex optimization with sparsity-promoting convex regularization is a
standard approach for estimating sparse signals in noise. In order to promote
sparsity more strongly than convex regularization, it is also standard practice
to employ non-convex optimization. In this paper, we take a third approach. We
utilize a non-convex regularization term chosen such that the total cost
function (consisting of data consistency and regularization terms) is convex.
Therefore, sparsity is more strongly promoted than in the standard convex
formulation, but without sacrificing the attractive aspects of convex
optimization (unique minimum, robust algorithms, etc.). We use this idea to
improve the recently developed 'overlapping group shrinkage' (OGS) algorithm
for the denoising of group-sparse signals. The algorithm is applied to the
problem of speech enhancement with favorable results in terms of both SNR and
perceptual quality.Comment: 14 pages, 11 figure
Blind MultiChannel Identification and Equalization for Dereverberation and Noise Reduction based on Convolutive Transfer Function
This paper addresses the problems of blind channel identification and
multichannel equalization for speech dereverberation and noise reduction. The
time-domain cross-relation method is not suitable for blind room impulse
response identification, due to the near-common zeros of the long impulse
responses. We extend the cross-relation method to the short-time Fourier
transform (STFT) domain, in which the time-domain impulse responses are
approximately represented by the convolutive transfer functions (CTFs) with
much less coefficients. The CTFs suffer from the common zeros caused by the
oversampled STFT. We propose to identify CTFs based on the STFT with the
oversampled signals and the critical sampled CTFs, which is a good compromise
between the frequency aliasing of the signals and the common zeros problem of
CTFs. In addition, a normalization of the CTFs is proposed to remove the gain
ambiguity across sub-bands. In the STFT domain, the identified CTFs is used for
multichannel equalization, in which the sparsity of speech signals is
exploited. We propose to perform inverse filtering by minimizing the
-norm of the source signal with the relaxed -norm fitting error
between the micophone signals and the convolution of the estimated source
signal and the CTFs used as a constraint. This method is advantageous in that
the noise can be reduced by relaxing the -norm to a tolerance
corresponding to the noise power, and the tolerance can be automatically set.
The experiments confirm the efficiency of the proposed method even under
conditions with high reverberation levels and intense noise.Comment: 13 pages, 5 figures, 5 table
Direct and Residual Subspace Decomposition of Spatial Room Impulse Responses
Psychoacoustic experiments have shown that directional properties of the direct sound, salient reflections, and the late reverberation of an acoustic room response can have a distinct influence on the auditory perception of a given room. Spatial room impulse responses (SRIRs) capture those properties and thus are used for direction-dependent room acoustic analysis and virtual acoustic rendering. This work proposes a subspace method that decomposes SRIRs into a direct part, which comprises the direct sound and the salient reflections, and a residual, to facilitate enhanced analysis and rendering methods by providing individual access to these components. The proposed method is based on the generalized singular value decomposition and interprets the residual as noise that is to be separated from the other components of the reverberation. Large generalized singular values are attributed to the direct part, which is then obtained as a low-rank approximation of the SRIR. By advancing from the end of the SRIR toward the beginning while iteratively updating the residual estimate, the method adapts to spatio-temporal variations of the residual. The method is evaluated using a spatio-spectral error measure and simulated SRIRs of different rooms, microphone arrays, and ratios of direct sound to residual energy. The proposed method creates lower errors than existing approaches in all tested scenarios, including a scenario with two simultaneous reflections. A case study with measured SRIRs shows the applicability of the method under real-world acoustic conditions. A reference implementation is provided
Model-Based Speech Enhancement
Abstract
A method of speech enhancement is developed that reconstructs clean speech from
a set of acoustic features using a harmonic plus noise model of speech. This is a significant
departure from traditional filtering-based methods of speech enhancement.
A major challenge with this approach is to estimate accurately the acoustic features
(voicing, fundamental frequency, spectral envelope and phase) from noisy speech.
This is achieved using maximum a-posteriori (MAP) estimation methods that operate
on the noisy speech. In each case a prior model of the relationship between the
noisy speech features and the estimated acoustic feature is required. These models
are approximated using speaker-independent GMMs of the clean speech features
that are adapted to speaker-dependent models using MAP adaptation and for noise
using the Unscented Transform.
Objective results are presented to optimise the proposed system and a set of subjective
tests compare the approach with traditional enhancement methods. Threeway
listening tests examining signal quality, background noise intrusiveness and
overall quality show the proposed system to be highly robust to noise, performing
significantly better than conventional methods of enhancement in terms of background
noise intrusiveness. However, the proposed method is shown to reduce signal
quality, with overall quality measured to be roughly equivalent to that of the Wiener
filter
Noise-Robust Voice Conversion
A persistent challenge in speech processing is the presence of noise that reduces the quality of speech signals. Whether natural speech is used as input or speech is the desirable output to be synthesized, noise degrades the performance of these systems and causes output speech to be unnatural. Speech enhancement deals with such a problem, typically seeking to improve the input speech or post-processes the (re)synthesized speech. An intriguing complement to post-processing speech signals is voice conversion, in which speech by one person (source speaker) is made to sound as if spoken by a different person (target speaker). Traditionally, the majority of speech enhancement and voice conversion methods rely on parametric modeling of speech. A promising complement to parametric models is an inventory-based approach, which is the focus of this work. In inventory-based speech systems, one records an inventory of clean speech signals as a reference. Noisy speech (in the case of enhancement) or target speech (in the case of conversion) can then be replaced by the best-matching clean speech in the inventory, which is found via a correlation search method. Such an approach has the potential to alleviate intelligibility and unnaturalness issues often encountered by parametric modeling speech processing systems. This work investigates and compares inventory-based speech enhancement methods with conventional ones. In addition, the inventory search method is applied to estimate source speaker characteristics for voice conversion in noisy environments. Two noisy-environment voice conversion systems were constructed for a comparative study: a direct voice conversion system and an inventory-based voice conversion system, both with limited noise filtering at the front end. Results from this work suggest that the inventory method offers encouraging improvements over the direct conversion method
Photorealistic retrieval of occluded facial information using a performance-driven face model
Facial occlusions can cause both human observers and computer algorithms
to fail in a variety of important tasks such as facial action analysis and
expression classification. This is because the missing information is not
reconstructed accurately enough for the purpose of the task in hand. Most
current computer methods that are used to tackle this problem implement
complex three-dimensional polygonal face models that are generally timeconsuming
to produce and unsuitable for photorealistic reconstruction of
missing facial features and behaviour.
In this thesis, an image-based approach is adopted to solve the occlusion
problem. A dynamic computer model of the face is used to retrieve the
occluded facial information from the driver faces. The model consists of a
set of orthogonal basis actions obtained by application of principal
component analysis (PCA) on image changes and motion fields extracted
from a sequence of natural facial motion (Cowe 2003). Examples of
occlusion affected facial behaviour can then be projected onto the model to
compute coefficients of the basis actions and thus produce photorealistic
performance-driven animations.
Visual inspection shows that the PCA face model recovers aspects of
expressions in those areas occluded in the driver sequence, but the expression is generally muted. To further investigate this finding, a database
of test sequences affected by a considerable set of artificial and natural
occlusions is created. A number of suitable metrics is developed to measure
the accuracy of the reconstructions. Regions of the face that are most
important for performance-driven mimicry and that seem to carry the best
information about global facial configurations are revealed using Bubbles,
thus in effect identifying facial areas that are most sensitive to occlusions.
Recovery of occluded facial information is enhanced by applying an
appropriate scaling factor to the respective coefficients of the basis actions
obtained by PCA. This method improves the reconstruction of the facial
actions emanating from the occluded areas of the face. However, due to the
fact that PCA produces bases that encode composite, correlated actions,
such an enhancement also tends to affect actions in non-occluded areas of
the face. To avoid this, more localised controls for facial actions are
produced using independent component analysis (ICA). Simple projection
of the data onto an ICA model is not viable due to the non-orthogonality of
the extracted bases. Thus occlusion-affected mimicry is first generated using
the PCA model and then enhanced by accordingly manipulating the
independent components that are subsequently extracted from the mimicry.
This combination of methods yields significant improvements and results in
photorealistic reconstructions of occluded facial actions
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