1,788 research outputs found
Voice SourceWaveform Analysis and Synthesis Using Principal Component Analysis and Gaussian Mixture Modelling
The paper presents a voice source waveform modeling techniques based on principal component analysis (PCA) and Gaussian mixture modeling (GMM). The voice source is obtained by inverse-filtering speech with the estimated vocal tract filter. This decomposition is useful in speech analysis, synthesis, recognition and coding. Here, a data-driven approach is presented for signal decomposition and classification based on the principal components of the voice source. The principal components are analyzed and the 'prototype' voice source signals corresponding to the Gaussian mixture means are examined. We show how an unknown signal can be decomposed into its components and/or prototypes and resynthesized. We show how the techniques are suited for both low bitrate or high quality analysis/synthesis schemes
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
Voice source characterization for prosodic and spectral manipulation
The objective of this dissertation is to study and develop techniques to decompose the speech signal into its two main
components: voice source and vocal tract. Our main efforts are on the glottal pulse analysis and characterization. We want to
explore the utility of this model in different areas of speech processing: speech synthesis, voice conversion or emotion detection
among others. Thus, we will study different techniques for prosodic and spectral manipulation. One of our requirements is that
the methods should be robust enough to work with the large databases typical of speech synthesis. We use a speech production
model in which the glottal flow produced by the vibrating vocal folds goes through the vocal (and nasal) tract cavities and its
radiated by the lips. Removing the effect of the vocal tract from the speech signal to obtain the glottal pulse is known as inverse
filtering. We use a parametric model fo the glottal pulse directly in the source-filter decomposition phase.
In order to validate the accuracy of the parametrization algorithm, we designed a synthetic corpus using LF glottal parameters
reported in the literature, complemented with our own results from the vowel database. The results show that our method gives
satisfactory results in a wide range of glottal configurations and at different levels of SNR. Our method using the whitened
residual compared favorably to this reference, achieving high quality ratings (Good-Excellent). Our full parametrized system
scored lower than the other two ranking in third place, but still higher than the acceptance threshold (Fair-Good).
Next we proposed two methods for prosody modification, one for each of the residual representations explained above. The first
method used our full parametrization system and frame interpolation to perform the desired changes in pitch and duration. The
second method used resampling on the residual waveform and a frame selection technique to generate a new sequence of
frames to be synthesized. The results showed that both methods are rated similarly (Fair-Good) and that more work is needed in
order to achieve quality levels similar to the reference methods.
As part of this dissertation, we have studied the application of our models in three different areas: voice conversion, voice quality
analysis and emotion recognition. We have included our speech production model in a reference voice conversion system, to
evaluate the impact of our parametrization in this task. The results showed that the evaluators preferred our method over the
original one, rating it with a higher score in the MOS scale. To study the voice quality, we recorded a small database consisting of
isolated, sustained Spanish vowels in four different phonations (modal, rough, creaky and falsetto) and were later also used in
our study of voice quality. Comparing the results with those reported in the literature, we found them to generally agree with
previous findings. Some differences existed, but they could be attributed to the difficulties in comparing voice qualities produced
by different speakers. At the same time we conducted experiments in the field of voice quality identification, with very good
results. We have also evaluated the performance of an automatic emotion classifier based on GMM using glottal measures. For
each emotion, we have trained an specific model using different features, comparing our parametrization to a baseline system
using spectral and prosodic characteristics. The results of the test were very satisfactory, showing a relative error reduction of
more than 20% with respect to the baseline system. The accuracy of the different emotions detection was also high, improving
the results of previously reported works using the same database. Overall, we can conclude that the glottal source parameters
extracted using our algorithm have a positive impact in the field of automatic emotion classification
Analysis and correction of the helium speech effect by autoregressive signal processing
SIGLELD:D48902/84 / BLDSC - British Library Document Supply CentreGBUnited Kingdo
Models and analysis of vocal emissions for biomedical applications
This book of Proceedings collects the papers presented at the 3rd International Workshop on Models and Analysis of Vocal Emissions for Biomedical Applications, MAVEBA 2003, held 10-12 December 2003, Firenze, Italy. The workshop is organised every two years, and aims to stimulate contacts between specialists active in research and industrial developments, in the area of voice analysis for biomedical applications. The scope of the Workshop includes all aspects of voice modelling and analysis, ranging from fundamental research to all kinds of biomedical applications and related established and advanced technologies
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
The DiTME Project: interdisciplinary research in music technology
This paper profiles the emergence of a significant body of research in audio engineering within the Faculties of Engineering and Applied Arts at Dublin Institute of Technology. Over a period of five years the group has had significant success in completing a Strand 3 research project entitled Digital Tools for Music Education (DiTME)
A review of state-of-the-art speech modelling methods for the parameterisation of expressive synthetic speech
This document will review a sample of available voice modelling and transformation techniques, in view of an application in expressive unit-selection based speech synthesis in the framework of the PAVOQUE project. The underlying idea is to introduce some parametric modification capabilities at the level of the synthesis system, in order to compensate for the sparsity and rigidity, in terms of available emotional speaking styles, of the databases used to define speech synthesis voices. For this work, emotion-related parametric modifications will be restricted to the domains of voice quality and prosody, as suggested by several reviews addressing the vocal correlates of emotions (Schröder, 2001; Schröder, 2004; Roehling et al., 2006). The present report will start with a review of some techniques related to voice quality modelling and modification. First, it will explore the techniques related to glottal flow modelling. Then, it will review the domain of cross-speaker voice transformations, in view of a transposition to the domain of cross-emotion voice transformations. This topic will be exposed from the perspective of the parametric spectral modelling of speech and then from the perspective of available spectral transformation techniques. Then, the domain of prosodic parameterisation and modification will be reviewed
HMM-based speech synthesis using an acoustic glottal source model
Parametric speech synthesis has received increased attention in recent years following
the development of statistical HMM-based speech synthesis. However, the speech
produced using this method still does not sound as natural as human speech and there
is limited parametric flexibility to replicate voice quality aspects, such as breathiness.
The hypothesis of this thesis is that speech naturalness and voice quality can be
more accurately replicated by a HMM-based speech synthesiser using an acoustic glottal
source model, the Liljencrants-Fant (LF) model, to represent the source component
of speech instead of the traditional impulse train.
Two different analysis-synthesis methods were developed during this thesis, in order
to integrate the LF-model into a baseline HMM-based speech synthesiser, which is
based on the popular HTS system and uses the STRAIGHT vocoder. The first method,
which is called Glottal Post-Filtering (GPF), consists of passing a chosen LF-model
signal through a glottal post-filter to obtain the source signal and then generating
speech, by passing this source signal through the spectral envelope filter. The system
which uses the GPF method (HTS-GPF system) is similar to the baseline system,
but it uses a different source signal instead of the impulse train used by STRAIGHT.
The second method, called Glottal Spectral Separation (GSS), generates speech by
passing the LF-model signal through the vocal tract filter. The major advantage of the
synthesiser which incorporates the GSS method, named HTS-LF, is that the acoustic
properties of the LF-model parameters are automatically learnt by the HMMs.
In this thesis, an initial perceptual experiment was conducted to compare the LFmodel
to the impulse train. The results showed that the LF-model was significantly
better, both in terms of speech naturalness and replication of two basic voice qualities
(breathy and tense). In a second perceptual evaluation, the HTS-LF system was better
than the baseline system, although the difference between the two had been expected to
be more significant. A third experiment was conducted to evaluate the HTS-GPF system
and an improved HTS-LF system, in terms of speech naturalness, voice similarity
and intelligibility. The results showed that the HTS-GPF system performed similarly
to the baseline. However, the HTS-LF system was significantly outperformed by the
baseline. Finally, acoustic measurements were performed on the synthetic speech to
investigate the speech distortion in the HTS-LF system. The results indicated that a
problem in replicating the rapid variations of the vocal tract filter parameters at transitions
between voiced and unvoiced sounds is the most significant cause of speech
distortion. This problem encourages future work to further improve the system
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