We propose an objective measurement method for pitch extractors' responses to frequency-modulated signals.It enables us to evaluate different pitch extractors with unified criteria.The method uses extended time-stretched pulses combined by binary orthogonal sequences.It provides simultaneous measurement results consisting of the linear and the non-linear timeinvariant responses and random and time-varying responses.We tested representative pitch extractors using fundamental frequencies spanning 80 Hz to 800 Hz with 1/48 octave steps and produced more than 2000 modulation frequency response plots.We found that making scientific visualization by animating these plots enables us to understand different pitch extractors' behavior at once.Such efficient and effortless inspection is impossible by inspecting all individual plots.The proposed measurement method with visualization leads to further improvement of the performance of one of the extractors mentioned above.In other words, our procedure turns the specific pitch extractor into the best reliable measuring equipment that is crucial for scientific research.We open-sourced MATLAB codes of the proposed objective measurement method and visualization procedure.

We introduce a new member of TSP (Time Stretched Pulse) for acoustic and speech measurement infrastructure, based on a simple all-pass filter and systematic randomization. This new infrastructure fundamentally upgrades our previous measurement procedure, which enables simultaneous measurement of multiple attributes, including non-linear ones without requiring extra filtering nor post-processing. Our new proposal establishes a theoretically solid, flexible, and extensible foundation in acoustic measurement. Moreover, it is general enough to provide versatile research tools for other fields, such as biological signal analysis. We illustrate using acoustic measurements and data augmentation as representative examples among various prospective applications. We open-sourced MATLAB implementation. It consists of an interactive and real-time acoustic tool, MATLAB functions, and supporting materials.

Auditory feedback plays an essential role in the regulation of the fundamental frequency of voiced sounds. The fundamental frequency also responds to auditory stimulation other than the speaker's voice. We propose to use this response of the fundamental frequency of sustained vowels to frequency-modulated test signals for investigating involuntary control of voice pitch. This involuntary response is difficult to identify and isolate by the conventional paradigm, which uses step-shaped pitch perturbation. We recently developed a versatile measurement method using a mixture of orthogonal sequences made from a set of extended time-stretched pulses (TSP). In this article, we extended our approach and designed a set of test signals using the mixture to modulate the fundamental frequency of artificial signals. For testing the response, the experimenter presents the modulated signal aurally while the subject is voicing sustained vowels. We developed a tool for conducting this test quickly and interactively. We make the tool available as an open-source and also provide executable GUI-based applications. Preliminary tests revealed that the proposed method consistently provides compensatory responses with about 100 ms latency, representing involuntary control. Finally, we discuss future applications of the proposed method for objective and non-invasive auditory response measurements.

We introduce a simple and linear SNR (strictly speaking, periodic to random power ratio) estimator (0dB to 80dB without additional calibration/linearization) for providing reliable descriptions of aperiodicity in speech corpus. The main idea of this method is to estimate the background random noise level without directly extracting the background noise. The proposed method is applicable to a wide variety of time windowing functions with very low sidelobe levels. The estimate combines the frequency derivative and the time-frequency derivative of the mapping from filter center frequency to the output instantaneous frequency. This procedure can replace the periodicity detection and aperiodicity estimation subsystems of recently introduced open source vocoder, YANG vocoder. Source code of MATLAB implementation of this method will also be open sourced.

We formulated and implemented a procedure to generate aliasing-free excitation source signals. It uses a new antialiasing filter in the continuous time domain followed by an IIR digital filter for response equalization. We introduced a cosine-series-based general design procedure for the new antialiasing function. We applied this new procedure to implement the antialiased Fujisaki-Ljungqvist model. We also applied it to revise our previous implementation of the antialiased Fant-Liljencrants model. A combination of these signals and a lattice implementation of the time varying vocal tract model provides a reliable and flexible basis to test fo extractors and source aperiodicity analysis methods. MATLAB implementations of these antialiased excitation source models are available as part of our open source tools for speech science.

A number of studies, with either voiced or unvoiced speech, have demonstrated that a speaker's geometric mean formant frequency (MFF) has a large effect on the perception of the speaker's size, as would be expected.One study with unvoiced speech showed that lifting the slope of the speech spectrum by 6 dB/octave also led to a reduction in the perceived size of the speaker.This paper reports an analogous experiment to determine whether lifting the slope of the speech spectrum by 6 dB/octave affects the perception of speaker size with voiced speech (words).The results showed that voiced speech with high-frequency enhancement was perceived to arise from smaller speakers.On average, the point of subjective equality in MFF discrimination was reduced by about 5%.However, there were large individual differences; some listeners were effectively insensitive to spectral enhancement of 6 dB/octave; others showed a consistent effect of the same enhancement.The results suggest that models of speaker size perception will need to include a listener specific parameter for the effect of spectral slope.

A group delay-based excitation source analysis and design method is introduced for extension of TANDEM-STRAIGHT, a speech analysis, modification and synthesis system. This introduction makes all components of the system be based on interference-free representations. They are power spectrum, instantaneous frequency and group delay representations. This unification has potential to solve the major weak point of VOCODER architecture for high-quality speech manipulation applications.

Another simple and high-speed F0 extractor with high temporal resolution based on our previous proposal has been developed by adding a higher-order symmetry measure. This extension made the proposed method significantly more robust than the previous one. The proposed method is a detector of the lowest prominent sinusoidal component. It can use several F0 refinement procedures when the signal is the sum of harmonic sinusoidal components. The refinement procedure presented here is based on a stable representation of instantaneous frequency of periodic signals. The whole procedure implemented by Matlab runs faster than realtime on usual PCs for 44,100 Hz sampled sounds. Application of the proposed algorithm revealed that rapid temporal modulations in both F0 trajectory and spectral envelope exist typically in expressive voices such as those those used in lively singing performance.

A new spectral envelope estimation procedure is proposed to recover details beyond band limitation imposed by the Shannon’s sampling theory when interpreting periodic excitation of voiced sounds as the sampling operation in the frequency domain. The proposed procedure is a hybrid of STRAIGHT, a F0-adaptive spectral envelope estimation and the auto regressive model parameter estimation. Wavelet analyses of these spectral models on the frequency domain enabled objective evaluation of this recovery procedure. The proposed procedure provides better speech quality especially when parameter manipulation is introduced. Index Terms: Speech analysis, envelope spectrum, sampling theory, speech modification, transfer function

Realistic reconstruction and manipulation of strong vocal expressions found in singing voices is a challenging and exciting topic. A speech analysis, modification and resynthesis framework based on interference-free power spectral and instantaneous frequency representations for periodic sounds is extended for handling such voices. Strong expressions are typically characterized by rapid variations in excitation timing and strength as well as complex structured excitation. Three types of excitation source extractors are revised and introduced to handle them. Preliminary tests successfully replicated strong vocal expressions. Also, additional attribute representations for modifying excitation and spectral information based on audio texture features are briefly discussed.

The typical problem in LPC-like vocoder is buzzing sound which is mainly due to the simple pulse train or noise excitation model. One way to improve it is to reconstruct the residual obtained from inverse filtering. So a new parametric representation of speech based on pitch-scaled analysis is proposed in this paper. Pitch-scaled analysis is used to extract the periodic spectrum of residual with half pitch period length. Then these periodic spectrums are de-correlated by principal component analysis (PCA) to reduce their dimension. Aperiodic measure is defined as the harmonic-to-noise ratio in the frequency domain where voicing cut-off frequency (VCO) is used to control the smoothness of aperiodicity. Periodic spectrum and aperiodic measure together with F0 are indicated as excitation parameters in the proposed LPC vocoder. Experimental results show that this proposed vocoder can get a mean opinion score (MOS) of 4.1 for a female voice before dimensionality reduction and keep the high-quality property after parameter compression.

An interference-free representation of the instantaneous frequency of constituent harmonic components of periodic signals is introduced. The power weighted average instantaneous frequency of a band-pass filter yields this property when the effective passband of the filter covers up to two harmonic components and the two windows used in averaging are separated by a half pitch period. The proposed representation eliminates the abrupt changes found in usual instantaneous frequency representations and is applicable to any periodic signals consisting of multiple harmonic components. An F0 extractor of voiced sounds based on this representation is introduced as an example of prospective applications.

This paper describes a high-quality manipulation method of voice quality base on the vocal tract area function (VTAF) obtained from sub-band LSP of STRAIGHT spectrum. Our research group had developed the manipulation technique of voice quality based on VTAF that can generate natural formant transition. However, it is observed that the generated sound sometimes results in degradation when the input signal has a high sampling frequency. Therefore, we develop a new method that extracts VTAF properly from such input signal. This method firstly divides the input spectral envelope represented by STRAIGHT spectrum into lower and higher frequency bands, secondly extracts the Line spectrum pair (LSP) in each frequency band after spectral flattening that is appropriate for the frequency band, thirdly concatenates a pair of the sub-band LSP, and finally obtains VTAF from PARCOR coefficients converted from the concatenated LSP. A subjective experiment proved that the proposed method is high quality enough.

A voice transformation method that only relies on vowel information is proposed. The method is based on empirical cumulative distributions of perceptually relevant spectral distances, which are used to design mapping functions from distance to proximity. A set of operators are optimized in the design phase to implement on-the-fly compilation of executable transformations used in the transformation phase. Proximity of the current input parameters to the speaker's own vowel templates is used in this compilation. The proposed method deforms the source speaker's parameter space using a set of monotonic and continuous mapping functions. This smooth and topology-preserving mapping yields high-quality modification of existing speech resources.

A generalized framework of auditory morphing based on the speech analysis, modification and resynthesis system STRAIGHT is proposed that enables each morphing rate of representational aspects to be a function of time, including the temporal axis itself. Two types of algorithms were derived: an incremental algorithm for real-time manipulation of morphing rates and a batch processing algorithm for off-line post-production applications. By defining morphing in terms of the derivative of mapping functions in the logarithmic domain, breakdown of morphing resynthesis found in the previous formulation in the case of extrapolations was eliminated. A method to alleviate perceptual defects in extrapolation is also introduced.

A simple and fast voice conversion method based only on vowel information is proposed. The proposed method relies on empirical distribution of perceptual spectral distances between representative examples of each vowel segment extracted using TANDEM-STRAIGHT spectral envelope estimation procedure [1]. Mapping functions of vowel spectra are designed to preserve vowel space structure defined by the observed empirical distribution while transforming position and orientation of the structure in an abstract vowel spectral space. By introducing physiological constraints in vocal tract shapes and vocal tract length normalization, difficulties in careful frequency alignment between vowel template spectra of the source and the target speakers can be alleviated without significant degradations in converted speech. The proposed method is a framebased instantaneous method and is relevant for real-time processing. Applications of the proposed method in-cross language voice conversion are also discussed. Index Terms: voice conversion, STRAIGHT, vowel structure, line spectral pair, vocal tract length

A simple new method for estimating temporally stable power spectra is introduced to provide a unified basis for computing an interference-free spectrum, the fundamental frequency (F0), as well as aperiodicity estimation. F0 adaptive spectral smoothing and cepstral liftering based on consistent sampling theory are employed for interference-free spectral estimation. A perturbation spectrum, calculated from temporally stable power and interference-free spectra, provides the basis for both F0 and aperiodicity estimation. The proposed approach eliminates ad-hoc parameter tuning and the heavy demand on computational power, from which STRAIGHT has suffered in the past.

We have developed a speech-to-text input method for web systems. The system is provided as a JavaScript library including an Ajax-like mechanism based on a Java applet, CGI programs, and dynamic HTML documents. It allows users to access voice-enabled web pages without requiring special browsers. Web developers can embed it on their web page by inserting only one line in the header field of an HTML document. This study also aims at observing natural spoken interactions in personal environments. We have succeeded in collecting 4,003 inputs during a period of seven months via our public Japanese ASR server. In order to cover out-of-vocabulary words to cope with some proper nouns, a web page to register new words into the language model are developed. As a result, we could obtain an improvement of 0.8% in the recognition accuracy. With regard to the acoustical conditions, an SNR of 25.3 dB was observed.

New design procedures of time-frequency alignment for automatic speech morphing are proposed. The frequency alignment function at a specific frame is represented as a weighted average of vowel alignment functions based on similarity to each vowel. Julian, an open source speech recognition system, was used to design a time alignment function. Objective and subjective tests were conducted to evaluate the proposed method, and test results indicated that the proposed method yields comparable naturalness to the manually morphed samples in terms of time alignment. The results also illustrated that the proposed frequency alignment provides significantly better naturalness than morphed samples without frequency alignment.

Smith et al. [2] and Ives et al. [3] demonstrated that humans could extract information about the size of a speaker's vocal tract from speech sounds (vowels and syllables, respectively). We have extended their discrimination and recognition experiments to naturally pronounced words. The Just Noticeable Difference (JND) for size discrimination was between 5.5% and 19% depending on the listener. The smallest JND is comparable to that of the syllable experiments; the average JND is comparable to that of the vowel experiments. The word recognition scores remain above 50% for speaker sizes beyond the normal range for humans. The fact that good performance extends over such a large range of acoustic scales supports Irino and Patterson’s hypothesis [1] that the auditory system segregates size and shape information at an early stage in the processing.

In order to obtain an understanding of the user’s emotion in human-machine dialogues, an analysis of dialogical utterances in the real world was performed. This work comprises three major steps. (1) The actual conditions of 16 basic emotions were evaluated using Japanese child voices, which were collected through the field test of the public spoken dialogue system. (2) Two factors were derived by a factor analysis. The factors were defined as fundamental psychological factors representing “delightful” and “hateable” emotions. (3) The relationships between the factors and the physical acoustic features were investigated to establish a capability to sense a user’s mental state for the dialogue system. In the experimental discriminations between the delightful and hateable emotions, a correct rate of 98.8% was achieved in classifying child’s utterances by the SVM (Support Vector Machine) with 11 acoustic features. Index Terms: emotional speech, classification, factor analysis, dialogue system, real world

We propose a new method to segregate concurrent speech sounds using an auditory version of a channel vocoder. The auditory representation of sound, referred to as an "auditory image," preserves fine temporal information, unlike conventional window-based processing systems. This makes it possible to segregate speech sources with an event synchronous procedure. Fundamental frequency information is used to estimate the sequence of glottal pulse times for a target speaker, and to repress the glottal events of other speakers. The procedure leads to robust extraction of the target speech and effective segregation even when the signal-to-noise ratio is as low as 0 dB. Moreover, the segregation performance remains high when the speech contains jitter, or when the estimate of the fundamental frequency F0 is inaccurate. This contrasts with conventional comb-filter methods where errors in F0 estimation produce a marked reduction in performance. We compared the new method to a comb-filter method using a cross-correlation measure and perceptual recognition experiments. The results suggest that the new method has the potential to supplant comb-filter and harmonic-selection methods for speech enhancement.

A tool to investigate an important fundamental question in speech processing is proposed aiming to promote research on voice quality and para and non linguistic aspects of speech. The proposed method effectively emulates waveform-based methods, sinusoidal models and the high quality source filter model STRAIGHT The key idea that enables blending these seemingly disjoint algorithms is a group delay based representation of signal excitation. By using a STRAIGHT-based smoothed time-frequency representation that is shared by these three types of speech processing methods, a unified source representation is used to implement the proposed system. Informal listening tests using the proposed system indicated that phase manipulation introduces different timbre, but it does not need to reproduce the exact waveform to reproduce the same timbre. This may suggest that the possibility of further information reduction exists in synthesizing close to natural quality speech.

Intelligibility of degraded speech sounds has been investigated based on a new signal processing technique using a high-quality vocoder, STRAIGHT. This enables us to manipulate essential speech parameters for vocal tract filtering and glottal excitation. We report that the effect of spectral smearing on the intelligibility of Japanese fourmora words as an initial study. Results reveal that the intelligibility decreases as the degree of smearing increases. We also investigated the relationship between the phonetic and word intelligibilities and found that the word identification score was predicted as the power function of the phonetic score when the power value was about 5. This implies that the mora structure and prosodic information such as F0, timing, and duration also play an important role in speech perception.

A procedure to remove vibrato from singing voice was proposed to enable auditory morphing between musical performances played under different conditions. Analyses of singing samples in the RWC music database using a speech analysis, modification and synthesis system STRAIGHT provided necessary information to implement “senza vibrato,” the procedure that removes vibrato. A preliminary subjective evaluation for artificially adding and removing vibrato indicated that the proposed procedure effectively control perceived vibrato while preserving naturalness of the original singing.

It is very important to capture the distant-talking speech with high-quality for voice-controlled systems or teleconferencing systems. A microphone array steering is an ideal candidate for this purpose. However, for the microphone array steering, it is necessary to track the target talker. Conventional talker tracking algorithms with audio signal only (ex. CSP (cross-power spectrum phase) analysis) have a difficulty estimating the target talker direction accurately in higher noisy environments. To overcome this problem, we propose a new target talker tracking algorithm that not only utilize the audio signal, but also utilize the visual signal. The proposed algorithm is based on integration of CSP analysis with audio signal and frame difference with visual signal. As a result of evaluation experiments in a real room, we confirmed that the proposed algorithm could track the target talker accurately than the conventional algorithm.

We present a new auditory method to segregate concurrent speech sounds. The system is based on an auditory vocoder developed to resynthesize speech from an auditory Mellin representation using the vocoder STRAIGHT (Kawahara, H. et al., Speech Communication, vol.27, p.187-207, 1999). The quality of the transmitted sound is improved by introducing an event synchronous procedure to estimate glottal pulse times. The auditory representation preserves fine temporal information, unlike conventional window-based processing, which makes it possible to segregate the speech synchronously. The results show that the segregation is good even when the SNR is 0 dB; the extracted target speech was a little distorted but entirely intelligible (like telephone speech), whereas the distracter speech was reduced to a non-speech sound that was not perceptually disturbing. This auditory vocoder has potential for speech enhancement in applications such as hearing aids.

An elastic spectral distance measure based on a F0 adaptive pitch synchronous spectral estimation and selective elimination of periodicity interferences, that was developed for a high-quality speech modification procedure STRAIGHT [1], is introduced to provide a basis for auditory morphing. The proposed measure is implemented on a low dimensional piecewise bilinear time-frequency mapping between the target and the original speech representations. A preliminary test results of morphing emotional speech samples indicated that proposed procedure provides perceptually monotonic and high-quality interpolation and extrapolation of CD quality speech samples.

We present a new auditory method to segregate concurrent speech sounds. The system is based on an auditory vocoder developed to resynthesize speech from an auditory Mellin representation using the vocoder STRAIGHT. The auditory representation preserves fine temporal information, unlike conventional window-based processing, and this makes it possible to segregate speech sources with an event synchronous procedure. We developed a method to convert fundamental frequency information to estimate glottal pulse times so as to facilitate robust extraction of the target speech. The results show that the segregation is good even when the SNR is 0 dB; the extracted target speech was a little distorted but entirely intelligible, whereas the distracter speech was reduced to a non-speech sound that was not perceptually disturbing. So, this auditory vocoder has potential for speech enhancement in applications such as hearing aids.

We assume that speech rnorphing, noise suppression, and speech segregation would improve if they were more accurately based on human perception. Accordingly, an Auditory VOCODER was developed to resynthesize speech from an auditory Mellin representation used to explain human perception. The Auditory VOCODER has three modules: an Auditory Mellin Image model [9,10], a STRAIGHT VOCODER [2], and a mapping module consisting of warped-frequency cepstral analysis and nonlinear, multivariate regression analysis (MRA). We describe the modules and an evaluation of the system. Informal listening indicates that the sound quality is reasonable.

cote interne IRCAM: Kawahara02a

This paper reports the comparative evaluation of several speech F0 evaluation algorithms over a wide database of laryngographlabeled speech. Included are several classic algorithms that are available in software on the net, as well as two new algorithms that offer greatly reduced error rates. Particular attention is given to the methodology of evaluation.

High-resolution F0 analysis using a speech database with simultaneously recorded EGG (Electroglottogram) signals indicated that there are systematic F0 glitches around nasal-vowel transitions. The durations of the glitches are 10 to 20 ms and they introduce 5 to 10 Hz F0 shifts. A detailed series of analyses of these glitches indicated that the major contributing factor of these glitches is sudden changes of group delay values of the vocal tract transfer function in the vicinity of the fundamental frequency at nasal-vowel transitions. It is also suggested that the Doppler effects due to apparent changes of vocal tract length are marginal, even if they exist. Finally, issues in evaluating high resolution F0 extraction algorithms and applications to high quality speech manipulation methods are discussed.

This paper proposes a noise-tolerant method for fundamental frequency (F0) extraction. This method includes several new ideas, including the estimation of the instantaneous frequencies of the higher harmonic components, and the design of an adaptive weighting function based on a bandwidth equation that combines the F0 information in the harmonic components. To evaluate the proposed method, we constructed a relatively large database of simultaneous recordings of speech waveforms and EGG (Electro Glotto Graphy). The database consists of 30 sentences pronounced by 14 male and 14 female normal subjects, i.e., 840 sentences in total. The duration of the sound is about 35 minutes including about 20 minutes of voicing. The experiments were performed with additive noise for four pitch extraction methods, i.e., the proposed method, the original TEMPO, an improved cepstrum method, and a common F0 extraction program in ESPS. The results were as follows: 1) the proposed method is always better than any of the other methods when the SNR is greater than about 2 dB; 2) for high SNR values (> 15 dB), the correct rates of the proposed method and the original TEMPO are about 95% and much better than the improved cepstrum method (92%) and the ESPS function (89%); and 3) all of the methods degrade to less than 62% when the SNR is 0 dB. As a result, the proposed method improves the performance for low SNR values and also maintains high accuracy inherent from the original TEMPO for high SNR values.

A new procedure for event detection and characterization is proposed based on group delay and fixed point analysis. This method enables the detection of precise timing and spread of speech events such as a vocal fold closure. A mapping from the center of a Gaussian time window to the mean time provides event locations as its fixed points. Refining these initial estimates using minimum phase group delay functions derived from the amplitude spectra provides accurate estimates of event locations and durations of excitations of each event. The proposed algorithm was tested using synthetic speech samples and natural speech database of simultaneously recorded sound waveforms and EGG signals. These tests revealed that the proposed method provides estimates of vocal fold closure instants with timing accuracy within 60 µ st o 210µs standard deviations. This algorithm is implemented to be suitable for real-time operation by making extensive use of FFTs without introducing any iterative procedures. It is potentially a very powerful tool for speech diagnosis and construction of very high quality speech manipulation systems.

The goal of this paper is to locate and understand the fine fundamental problems that exist in the representation of speech sounds by a very high quality speech analysis/synthesis engine namely STRAIGHT. The approach followed here is the evaluation of this system using subjective measures. We use the diagnostic rhyme test (DRT) to evaluate the intelligibility of speech analysed and synthesised by this system for various analysis frame-rates. Consequently we catagorise the fine problems and suggest possible improvements. The results from the DRT have indicated that STRAIGHT can produce speech with an average DRT score of 95 between 1-5 ms analysis frame-rate. In addition, a set of subjective quality measures using MOS and MNRU tests have been conducted. These tests have been carried out for three different versions of the STRAIGHT system: versions 17, 23 and 30. The DRT has been carried out using version 23 only. Based on the subjective evaluation results, a discussion of possible improvements to the STRAIGHT system is given.

An efficient representation of short-time phase characteristics of speech sounds is proposed, based on findings which suggest the perceptual importance of phase characteristics. Subjective tests indicated that the synthesized speech sounds by the proposed method are indistinguishable from the original speech sounds with a moderate data compression. The proposed representation uses lower-order coefficients of the inverse Fourier transform of the group delay of speech. It also alleviates the voiced/unvoiced decision, which is an indispensable part in conventional speech coding algorithms. These features make our method potentially very useful in many applications like speech morphing.

How can we provide feedback to second language (L2) learners about the goodness of their productions in an automatic way? In this paper, we introduce our attempts to provide effective feedback when we train native speakers of Japanese to produce English /r/ and /l/. First, we adopted spectrographic representation overlayed with formant frequencies as feedback. Second, we investigated the correlation between human judgments of L2 production quality and acoustic scores produced by an HMM-based speech recognition system. We also adopted the HMM-based scores as feedback in the production training. Evaluation of the preand post-training productions by human judges showed that production abilities of the trainees improved in both training groups, suggesting that both spectrographic representation and HMM-based scores were useful and meaningful as feedback. These results are discussed in the context of optimizing L2 speech training.

A simple new procedure called STRAIGHT (speech transformation and representation using adaptive interpolation of weighted spectrum) has been developed. STRAIGHT uses pitch-adaptive spectral analysis combined with a surface reconstruction method in the time-frequency region, and an excitation source design based on phase manipulation. It preserves the bilinear surface in the time-frequency region and allows for over 600% manipulation of such speech parameters as pitch, vocal tract length, and speaking rate, without further degradation due to the parameter manipulation.

In this paper, a method is proposed to evaluate contributions of auditory feedback to speech F0 trajectory generation. This method is based on data obtained in a series of new auditory feedback experiments (TAF: transformed auditory feedback) in which quantitative measurements were taken of interactions between speech perception and production under natural speech conditions. Experimental results revealed that the effects on power spectra of F0 trajectories vary among subjects and that the maximum magnitude exceeds 10 dB.

A novel method of signal reconstruction from a modified auditory representation is presented. This consists of three parts: (1) an algorithm to reconstruct a signal from its modified wavelet transform with a general wavelet; (2) obtaining an auditory representation using an auditory wavelet transform whose analyzing wavelet is the impulse response of an auditory peripheral model; and (3) estimating the reconstruction algorithm both with and without data reduction. An example of its application to the time-scale modification of speech is presented. High-quality speech successfully generated by time-scale modification shows that the reconstruction method is suitable for various applications as well as making experimental auditory stimuli.>

A nonlinear multiple logistic model and multiple regression analysis are described as a method for determining the weights for two-layer networks and are compared to error backpropagation. We also provide a method for constructing a three-layer network whose semilinear middle units are primarily provided to discriminate two categories. Experimental results on speaker-independent vowel recognition show that both multivariate methods provide stable weights with fewer iterations than backpropagation training started with random initial weights, but with slightly inferior performance. Backpropagation training with initial weights determined by a multiple logistic model after introduction of data distribution information gives a recognition rate of 98.2%, which is significantly better than average backpropagation with random initial weights.