Tuomo Raitio

HMM-Based Speech Synthesis Utilizing Glottal Inverse Filtering

This paper describes an hidden Markov model (HMM)-based speech synthesizer that utilizes glottal inverse filtering for generating natural sounding synthetic speech. In the proposed method, speech is first decomposed into the glottal source signal and the model of the vocal tract filter through glottal inverse filtering, and thus parametrized into excitation and spectral features. The source and filter features are modeled individually in the framework of HMM and generated in the synthesis stage according to the text input. The glottal excitation is synthesized through interpolating and concatenating natural glottal flow pulses, and the excitation signal is further modified according to the spectrum of the desired voice source characteristics. Speech is synthesized by filtering the reconstructed source signal with the vocal tract filter. Experiments show that the proposed system is capable of generating natural sounding speech, and the quality is clearly better compared to two HMM-based speech synthesis systems based on widely used vocoder techniques.

COVAREP — A collaborative voice analysis repository for speech technologies

Speech processing algorithms are often developed demonstrating improvements over the state-of-the-art, but sometimes at the cost of high complexity. This makes algorithm reimplementations based on literature difficult, and thus reliable comparisons between published results and current work are hard to achieve. This paper presents a new collaborative and freely available repository for speech processing algorithms called COVAREP, which aims at fast and easy access to new speech processing algorithms and thus facilitating research in the field. We envisage that COVAREP will allow more reproducible research by strengthening complex implementations through shared contributions and openly available code which can be discussed, commented on and corrected by the community. Presently COVAREP contains contributions from five distinct laboratories and we encourage contributions from across the speech processing research field. In this paper, we provide an overview of the current offerings of COVAREP and also include a demonstration of the algorithms through an emotion classification experiment.

Utilizing glottal source pulse library for generating improved excitation signal for HMM-based speech synthesis

This paper describes a source modeling method for hidden Markov model (HMM) based speech synthesis for improved naturalness. A speech corpus is first decomposed into the glottal source signal and the model of the vocal tract filter using glottal inverse filtering, and parametrized into excitation and spectral features. Additionally, a library of glottal source pulses is extracted from the estimated voice source signal. In the synthesis stage, the excitation signal is generated by selecting appropriate pulses from the library according to the target cost of the excitation features and a concatenation cost between adjacent glottal source pulses. Finally, speech is synthesized by filtering the excitation signal by the vocal tract filter. Experiments show that the naturalness of the synthetic speech is better or equal, and speaker similarity is better, compared to a system using only single glottal source pulse.

Toward a Universal Synthetic Speech Spoofing Detection Using Phase Information

In the field of speaker verification (SV) it is nowadays feasible and relatively easy to create a synthetic voice to deceive a speech driven biometric access system. This paper presents a synthetic speech detector that can be connected at the front-end or at the back-end of a standard SV system, and that will protect it from spoofing attacks coming from state-of-the-art statistical Text to Speech (TTS) systems. The system described is a Gaussian Mixture Model (GMM) based binary classifier that uses natural and copy-synthesized signals obtained from the Wall Street Journal database to train the system models. Three different state-of-the-art vocoders are chosen and modeled using two sets of acoustic parameters: 1) relative phase shift and 2) canonical Mel Frequency Cepstral Coefficients (MFCC) parameters, as baseline. The vocoder dependency of the system and multivocoder modeling features are thoroughly studied. Additional phase-aware vocoders are also tested. Several experiments are carried out, showing that the phase-based parameters perform better and are able to cope with new unknown attacks. The final evaluations, testing synthetic TTS signals obtained from the Blizzard challenge, validate our proposal.

Quasi Closed Phase Glottal Inverse Filtering Analysis With Weighted Linear Prediction

This study presents a new glottal inverse filtering (GIF) technique based on closed phase analysis over multiple fundamental periods. The proposed quasi closed phase (QCP) analysis method utilizes weighted linear prediction (WLP) with a specific attenuated main excitation (AME) weight function that attenuates the contribution of the glottal source in the linear prediction model optimization. This enables the use of the autocorrelation criterion in linear prediction in contrast to the covariance criterion used in conventional closed phase analysis. The QCP method was compared to previously developed methods by using synthetic vowels produced with the conventional source-filter model as well as with a physical modeling approach. The obtained objective measures show that the QCP method improves the GIF performance in terms of errors in typical glottal source parametrizations for both low- and high-pitched vowels. Additionally, QCP was tested in a physiologically oriented vocoder, where the analysis/synthesis quality was evaluated with a subjective listening test indicating improved perceived quality for normal speaking style.

Synthesis and perception of breathy, normal, and Lombard speech in the presence of noise

This papers studies the synthesis of speech over a wide vocal effort continuum and its perception in the presence of noise. Three types of speech are recorded and studied along the continuum: breathy, normal, and Lombard speech. Corresponding synthetic voices are created by training and adapting the statistical parametric speech synthesis system GlottHMM. Natural and synthetic speech along the continuum is assessed in listening tests that evaluate the intelligibility, quality, and suitability of speech in three different realistic multichannel noise conditions: silence, moderate street noise, and extreme street noise. The evaluation results show that the synthesized voices with varying vocal effort are rated similarly to their natural counterparts both in terms of intelligibility and suitability.

EXCITATION MODELING FOR HMM-BASED SPEECH SYNTHESIS: BREAKING DOWN THE IMPACT OF PERIODIC AND APERIODIC COMPONENTS

HMM-based speech synthesis generally suffers from typical buzzi-ness due to over-simplified excitation modeling of voiced speech. In order to alleviate this effect, several studies have proposed various new excitation models. No consensus has however been reached on what is the perceptual importance of the accurate modeling of the periodic and aperiodic components of voiced speech, and to what extent they separately contribute in improving naturalness. This paper considers a generalized mixed excitation modeling, common to various existing approaches, in which both periodic and aperiodic components coexist. At least three main factors may alter the quality of synthesis: periodic waveform, noise spectral weighting, and noise time envelope. Based on a large subjective evaluation, the goal of this paper is threefold: i) to evaluate the relative perceptual importance of each factor, ii) to investigate what is the most appropriate method to model the periodic and aperiodic components, and iii) to provide prospective clues for future work in excitation modeling.

A deep generative architecture for postfiltering in statistical parametric speech synthesis

The generated speech of hidden Markov model (HMM)-based statistical parametric speech synthesis still sounds “muffled.” One cause of this degradation in speech quality may be the loss of fine spectral structures. In this paper, we propose to use a deep generative architecture, a deep neural network (DNN) generatively trained, as a postfilter. The network models the conditional probability of the spectrum of natural speech given that of synthetic speech to compensate for such gap between synthetic and natural speech. The proposed probabilistic postfilter is generatively trained by cascading two restricted Boltzmann machines (RBMs) or deep belief networks (DBNs) with one bidirectional associative memory (BAM). We devised two types of DNN postfilters: one operating in the mel-cepstral domain and the other in the higher dimensional spectral domain. We compare these two new data-driven postfilters with other types of postfilters that are currently used in speech synthesis: a fixed mel-cepstral based postfilter, the global variance based parameter generation, and the modulation spectrum-based enhancement. Subjective evaluations using the synthetic voices of a male and female speaker confirmed that the proposed DNN-based postfilter in the spectral domain significantly improved the segmental quality of synthetic speech compared to that with conventional methods.

Comparing glottal-flow-excited statistical parametric speech synthesis methods

This paper studies the performance of glottal flow signal based excitation methods in statistical parametric speech synthesis. The current state of the art in excitationmodeling is reviewed and three excitation methods are selected for experiments. Two of the methods are based on the principal component analysis (PCA) decomposition of estimated glottal flow pulses. While the first one uses only the mean of the pulses, the second method uses 12 principal components in addition to the mean signal for modeling the glottal flow waveform. The third method utilizes a glottal flow pulse library from which pulses are selected according to target and concatenation costs. Subjective listening tests are carried out to determine the quality and similarity of the synthetic speech of one male and one female speaker. The results show that the PCA-based methods are rated best both in quality and similarity, but adding more components does not yield any improvements.

Detection of shouted speech in noise: Human and machine

High vocal effort has characteristic acoustic effects on speech. This study focuses on the utilization of this information by human listeners and a machine-based detection system in the task of detecting shouted speech in the presence of noise. Both female and male speakers read Finnish sentences using normal and shouted voice in controlled conditions, with the sound pressure level recorded. The speech material was artificially corrupted by noise and supplemented with pure noise. The human performance level was statistically evaluated by a listening test, where the subjects labeled noisy samples according to whether shouting was heard or not. A Bayesian detection system was constructed and statistically evaluated. Its performance was compared against that of human listeners, substituting different spectrum analysis methods in the feature extraction stage. Using features capable of taking into account the spectral fine structure (i.e., the fundamental frequency and its harmonics), the machine reached the detection level of humans even in the noisiest conditions. In the listening test, male listeners detected shouted speech significantly better than female listeners, especially with speakers making a smaller vocal effort increase for shouting.