597 research outputs found
Glottal-synchronous speech processing
Glottal-synchronous speech processing is a field of speech science where the pseudoperiodicity
of voiced speech is exploited. Traditionally, speech processing involves segmenting
and processing short speech frames of predefined length; this may fail to exploit the inherent
periodic structure of voiced speech which glottal-synchronous speech frames have
the potential to harness. Glottal-synchronous frames are often derived from the glottal
closure instants (GCIs) and glottal opening instants (GOIs).
The SIGMA algorithm was developed for the detection of GCIs and GOIs from
the Electroglottograph signal with a measured accuracy of up to 99.59%. For GCI and
GOI detection from speech signals, the YAGA algorithm provides a measured accuracy
of up to 99.84%. Multichannel speech-based approaches are shown to be more robust to
reverberation than single-channel algorithms.
The GCIs are applied to real-world applications including speech dereverberation,
where SNR is improved by up to 5 dB, and to prosodic manipulation where the importance
of voicing detection in glottal-synchronous algorithms is demonstrated by subjective
testing. The GCIs are further exploited in a new area of data-driven speech modelling,
providing new insights into speech production and a set of tools to aid deployment into
real-world applications. The technique is shown to be applicable in areas of speech coding,
identification and artificial bandwidth extension of telephone speec
Towards High-Quality and Efficient Speech Bandwidth Extension with Parallel Amplitude and Phase Prediction
Speech bandwidth extension (BWE) refers to widening the frequency bandwidth
range of speech signals, enhancing the speech quality towards brighter and
fuller. This paper proposes a generative adversarial network (GAN) based BWE
model with parallel prediction of Amplitude and Phase spectra, named AP-BWE,
which achieves both high-quality and efficient wideband speech waveform
generation. The proposed AP-BWE generator is entirely based on convolutional
neural networks (CNNs). It features a dual-stream architecture with mutual
interaction, where the amplitude stream and the phase stream communicate with
each other and respectively extend the high-frequency components from the input
narrowband amplitude and phase spectra. To improve the naturalness of the
extended speech signals, we employ a multi-period discriminator at the waveform
level and design a pair of multi-resolution amplitude and phase discriminators
at the spectral level, respectively. Experimental results demonstrate that our
proposed AP-BWE achieves state-of-the-art performance in terms of speech
quality for BWE tasks targeting sampling rates of both 16 kHz and 48 kHz. In
terms of generation efficiency, due to the all-convolutional architecture and
all-frame-level operations, the proposed AP-BWE can generate 48 kHz waveform
samples 292.3 times faster than real-time on a single RTX 4090 GPU and 18.1
times faster than real-time on a single CPU. Notably, to our knowledge, AP-BWE
is the first to achieve the direct extension of the high-frequency phase
spectrum, which is beneficial for improving the effectiveness of existing BWE
methods.Comment: Submitted to IEEE/ACM Transactions on Audio, Speech, and Language
Processin
Robust speech recognition under band-limited channels and other channel distortions
Tesis doctoral inédita. Universidad Autónoma de Madrid, Escuela Politécnica Superior, junio de 200
Computer Models for Musical Instrument Identification
PhDA particular aspect in the perception of sound is concerned with what is commonly
termed as texture or timbre. From a perceptual perspective, timbre is what allows us
to distinguish sounds that have similar pitch and loudness. Indeed most people are
able to discern a piano tone from a violin tone or able to distinguish different voices
or singers.
This thesis deals with timbre modelling. Specifically, the formant theory of timbre
is the main theme throughout. This theory states that acoustic musical instrument
sounds can be characterised by their formant structures. Following this principle, the
central point of our approach is to propose a computer implementation for building
musical instrument identification and classification systems.
Although the main thrust of this thesis is to propose a coherent and unified
approach to the musical instrument identification problem, it is oriented towards the
development of algorithms that can be used in Music Information Retrieval (MIR)
frameworks. Drawing on research in speech processing, a complete supervised system
taking into account both physical and perceptual aspects of timbre is described.
The approach is composed of three distinct processing layers. Parametric models
that allow us to represent signals through mid-level physical and perceptual representations
are considered. Next, the use of the Line Spectrum Frequencies as spectral
envelope and formant descriptors is emphasised. Finally, the use of generative and
discriminative techniques for building instrument and database models is investigated.
Our system is evaluated under realistic recording conditions using databases of isolated
notes and melodic phrases
Statistical models for natural sounds
It is important to understand the rich structure of natural sounds in order to solve important
tasks, like automatic speech recognition, and to understand auditory processing
in the brain. This thesis takes a step in this direction by characterising the statistics of
simple natural sounds. We focus on the statistics because perception often appears to
depend on them, rather than on the raw waveform. For example the perception of auditory
textures, like running water, wind, fire and rain, depends on summary-statistics,
like the rate of falling rain droplets, rather than on the exact details of the physical
source.
In order to analyse the statistics of sounds accurately it is necessary to improve a
number of traditional signal processing methods, including those for amplitude demodulation,
time-frequency analysis, and sub-band demodulation. These estimation tasks
are ill-posed and therefore it is natural to treat them as Bayesian inference problems.
The new probabilistic versions of these methods have several advantages. For example,
they perform more accurately on natural signals and are more robust to noise,
they can also fill-in missing sections of data, and provide error-bars. Furthermore,
free-parameters can be learned from the signal. Using these new algorithms we demonstrate
that the energy, sparsity, modulation depth and modulation time-scale in each
sub-band of a signal are critical statistics, together with the dependencies between the
sub-band modulators. In order to validate this claim, a model containing co-modulated
coloured noise carriers is shown to be capable of generating a range of realistic sounding
auditory textures.
Finally, we explored the connection between the statistics of natural sounds and perception.
We demonstrate that inference in the model for auditory textures qualitatively
replicates the primitive grouping rules that listeners use to understand simple acoustic
scenes. This suggests that the auditory system is optimised for the statistics of natural
sounds
Recommended from our members
Time-Frequency Analysis as Probabilistic Inference
This is the final published version. It was originally published by IEEE at http://ieeexplore.ieee.org/xpl/articleDetails.jsp?arnumber=6918491.This paper proposes a new view of time-frequency analysis framed in terms of probabilistic inference. Natural signals are assumed to be formed by the superposition of distinct time-frequency components, with the analytic goal being to infer these components by application of Bayes' rule. The framework serves to unify various existing models for natural time-series; it relates to both the Wiener and Kalman filters, and with suitable assumptions yields inferential interpretations of the short-time Fourier transform, spectrogram, filter bank, and wavelet representations. Value is gained by placing time-frequency analysis on the same probabilistic basis as is often employed in applications such as denoising, source separation, or recognition. Uncertainty in the time-frequency representation can be propagated correctly to application-specific stages, improving the handing of noise and missing data. Probabilistic learning allows modules to be co-adapted; thus, the time-frequency representation can be adapted to both the demands of the application and the time-varying statistics of the signal at hand. Similarly, the application module can be adapted to fine properties of the signal propagated by the initial time-frequency processing. We demonstrate these benefits by combining probabilistic time-frequency representations with non-negative matrix factorization, finding benefits in audio denoising and inpainting tasks, albeit with higher computational cost than incurred by the standard approach.Funding was provided by EPSRC (grant numbers EP/G050821/1 and
EP/L000776/1) and Google (R.E.T.) and by the Gatsby Charitable Foundation
(M.S.)
Recent Advances in Signal Processing
The signal processing task is a very critical issue in the majority of new technological inventions and challenges in a variety of applications in both science and engineering fields. Classical signal processing techniques have largely worked with mathematical models that are linear, local, stationary, and Gaussian. They have always favored closed-form tractability over real-world accuracy. These constraints were imposed by the lack of powerful computing tools. During the last few decades, signal processing theories, developments, and applications have matured rapidly and now include tools from many areas of mathematics, computer science, physics, and engineering. This book is targeted primarily toward both students and researchers who want to be exposed to a wide variety of signal processing techniques and algorithms. It includes 27 chapters that can be categorized into five different areas depending on the application at hand. These five categories are ordered to address image processing, speech processing, communication systems, time-series analysis, and educational packages respectively. The book has the advantage of providing a collection of applications that are completely independent and self-contained; thus, the interested reader can choose any chapter and skip to another without losing continuity
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