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检索条件"主题词=Linear predictive coding"

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Analysis, modelling and synthesis of formants of British, American and Australian accents

Analysis, modelling and synthesis of formants of British, Am...

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： Qin Yan S. Vaseghi Brunei University Middlesex UK

The formant space of three major English accents namely British, American and Australian are modelled and used for accent conversion. Accent synthesis, through modification of the acoustic parameters of speech, provides a means for assessing the perceptual contribution of each parameter on conveying an accent. An improved method based on a linear prediction (LP) model feature analysis and a 2D hidden Markov model (HMM) is employed for estimation of formant trajectories of vowels and diphthongs. Comparative analysis of the formant space of the three accents indicates that these accents are partly conveyed by the fronting and backing of vowels. It is found that the first formants of the vowels of British and American English accents are higher than those in Australian accent while Australians have higher second formants in vowels compared to Americans and British. The estimates of the distributions of formants for each accent are used in a speech synthesis system for accent conversion. Perceptual evaluations of accent conversion results illustrate that formants, in particular the second formant, play an important role in conveying accents.

关键词： Australia Speech synthesis Hidden Markov models Automatic speech recognition Speech analysis linear predictive coding Trajectory Loudspeakers Tongue Natural languages

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Speech recognition using filter-bank features

Speech recognition using filter-bank features

引用

Asilomar Conference on Signals, Systems & Computers

作者： Sourabh Ravindran C. Demirogulu D.V. Anderson School of Electrical and Computer Engineering Georgia Institute of Technology Atlanta GA USA School of Electrical and Computer Engineering Georgia Institute of Technology Atlanta GA

Mel-frequency cepstral coefficients (MFCC) have been shown to be very useful in tasks of speech recognition and are the preferred features in state of the art speech recognition systems. The author present features derived from filter bank outputs whose performance is comparable to that of MFCCs for connected digit recognition using a hidden Markov model (HMM) based speech recognition system. The feature extraction method we present is easily implementable in floating gate analog VLSI circuitry which makes it a viable option for low power speech recognition tasks.

关键词： Speech recognition Finite impulse response filter Filter bank linear predictive coding Decorrelation Channel bank filters Hidden Markov models Very large scale integration Cepstral analysis Nonlinear filters

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Context-adaptive phone boundary refining for a TTS database

Context-adaptive phone boundary refining for a TTS database

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： Ki-Seung Lee JeongSu Kim Department of Electronic Eng Konkuk University Seoul South Korea Human-Computer Interactive Laboratories Samsung Advanced Institute of Technology Gyeonggi South Korea

ISBN: (纸本)0780376633

A method for the automatic segmentation of speech signals is described. The method is dedicated to the construction of a large database for a Text-To-Speech (TTS) synthesis system. The main issue of the work involves the refinement of an initial estimation of phone boundaries which are provided by an alignment, based on a Hidden Markov Model (HMM). Multi-layer perceptron (MLP) was used as a phone boundary detector. To increase the performance of segmentation, a technique which individually trains an MLP according to phonetic transition is proposed. The optimum partitioning of the entire phonetic transition space is constructed from the standpoint of minimizing the overall deviation from hand labelling positions. With single speaker stimuli, the experimental results showed that more than 95% of all phone boundaries have a boundary deviation from the reference position smaller than 20 ms, and the refinement of the boundaries reduces the root mean square error by about 25%.

关键词： Speech synthesis Databases Hidden Markov models Labeling Automatic speech recognition Signal synthesis Multilayer perceptrons Detectors Root mean square linear predictive coding

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A comparison between multi-channel audio equalization filters using warping

A comparison between multi-channel audio equalization filter...

引用

IEEE Workshop on Applications of Signal Processing to Audio and Acoustics

作者： S. Bharitkar C. Kyriakakis Integrated Media Systems Center University of Southern California Los Angeles CA

Typically, room equalization techniques do not focus on designing filters that equalize the room responses at perceptually relevant frequencies. Thus, by performing Bark warping of the room responses and using lower order spectral models, it is possible to design low order psycho-acoustically motivated equalization filters. We compare the performance, through experiments, between the traditional RMS averaging filter (with and without warping to the Bark scale) and our pattern recognition based multiple listener equalization filter with warping (Bharitkar, S. and Kyriakakis, C., Proc. 36th Asilomar Conf. on Signals, Systems and Computers, 2002). It is shown that our pattern recognition filter, using warping, outperforms the RMS averaging filter (with and without warping to the Bark scale).

关键词： linear predictive coding Nonlinear filters Psychoacoustic models Pattern recognition Frequency response predictive models Signal processing Electronic mail Psychology linear systems

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Frequency-domain linear prediction for temporal features

Frequency-domain linear prediction for temporal features

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IEEE Workshop on Automatic Speech Recognition and Understanding

作者： M. Athineos D.P.W. Ellis LabROSA Department of Electrical Engineering Columbia University NY USA

Current speech recognition systems uniformly employ short-time spectral analysis, usually over windows of 10-30 ms, as the basis for their acoustic representations. Any detail below this timescale is lost, and even temporal structures above this level are usually only weakly represented in the form of deltas etc. We address this limitation by proposing a novel representation of the temporal envelope in different frequency bands by exploring the dual of conventional linear prediction (LPC) when applied in the transform domain. With this technique of frequency-domain linear prediction (FDLP), the 'poles' of the model describe temporal, rather than spectral, peaks. By using analysis windows on the order of hundreds of milliseconds, the procedure automatically decides how to distribute poles to model the temporal structure best within the window. While this approach offers many possibilities for novel speech features, we experiment with one particular form, an index describing the 'sharpness' of individual poles within a window, and show a relatively large word error rate improvement from 4.97% to 3.81% in a recognizer trained on general conversational telephone speech and tested on a small-vocabulary spontaneous numbers task. We analyze this improvement in terms of the confusion matrices and suggest how the newly-modeled fine temporal structure may be helping.

关键词： Speech recognition Automatic speech recognition predictive models Frequency domain analysis Spectral analysis Discrete cosine transforms linear predictive coding Error analysis Telephony Acoustic testing

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Forward and backward LP ELS-based time-varying complex speech analysis based on output error method

Forward and backward LP ELS-based time-varying complex speec...

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IEEE International Symposium on Signal Processing and Information Technology (ISSPIT)

作者： K. Funaki Computing and Networking Center University of Ryukyus Okinawa Japan

We have already proposed the ELS-based time-varying complex AR (TV-CAR) speech analysis based on forward LP as well as forward and backward LP in which the equation error is modeled by an AR model to whiten the error. The methods are based on an equation error method and can estimate unbiased speech spectrum due to the whitened equation error. It can be considered that these speech analysis methods may be suitable for a front-end of robust speech recognition and packet loss concealment on VoIP. This paper presents output error based ELS TV-CAR speech analysis algorithm and compares the performance with the equation error based method.

关键词： Speech analysis Equations Signal analysis Robustness Speech processing linear predictive coding Speech synthesis Mean square error methods Least squares methods Filters

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Estimation of short term prediction parameters under lossy conditions

Estimation of short term prediction parameters under lossy c...

引用

IEEE International Symposium on Signal Processing and Information Technology (ISSPIT)

作者： D. Black M. Sandler Department EE. DSP and Multimedia Group Queen Mary University of London UK

The inconsistencies inherent in packet switched network delivery can be seriously detrimental to the quality of a real-time speech transmission. This paper places its emphasis on the importance of the short term prediction (STP) filter parameters as these are perceptually important to intelligible speech. We introduces several novel schemes for the recovery of lost STP parameters represented as line spectral frequencies (LSFs) based on extrapolation and interpolation techniques. The unique inclusion of a number of past and/or future frames further commends this work. Methods which outperforms the traditional frame repetition and linear interpolation in terms of accuracy are presented and evaluated.

关键词： Interpolation Filters Phase change materials Extrapolation Speech coding Packet switching Pulse modulation linear predictive coding Digital signal processing Frequency

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Performance of hybrid link power control using wideband gapfiller satellite-based automatic level control (ALC) and ground-based link power control (LPC)

Performance of hybrid link power control using wideband gapf...

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MILCOM, Military Communications Conference

作者： L.N. Nguyen G.V. Kinal H.I. Paul Femme Comp Inc. Chantilly VA USA

The wideband gapfiller satellite (WGS) is planned for launch in 2005, Features include the use of phased array antennas to form eight shapeable beams in X band; and mechanically-steered dish antennas to form ten beams in Ka band. Automatic level control (ALC) for each channel may optionally be used to maintain total channel power at the ALC output at a constant level. Digital signal processing techniques are used to implement switchable and gain-adjustable 2.6 MHz subchannels. This paper examines the use of ALC on the WGS satellite; and considers interactions between the ALC (which has variable gain) and link power control (LPC), when both are employed. Simultaneous use of ALC and LPC, when suitably configured, results in a more efficient utilization of the WGS system.

关键词： Power control Wideband Level control Automatic logic units linear predictive coding Broadband antennas Phased arrays Satellite antennas Antenna arrays Digital signal processing

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Teaching trade-offs in system-level design methodologies

Teaching trade-offs in system-level design methodologies

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IEEE International Conference on Microelectronic Systems Education

作者： K. Sakiyama P. Schaumont D. Hwang I. Verbauwhede Electrical Engineering Department University of California Los Angeles CA USA

This paper summarizes two graduate-level class projects in EE201A/EE298 (VLSI Architectures and Design Methods) at the University of California, Los Angeles (UCLA). The purpose of the class is to explore the impact of... 详细信息

ISBN: (纸本)0769519733

关键词： Education System-level design linear predictive coding Digital signal processing Speech synthesis Design methodology Springs Signal processing Signal design Signal synthesis

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An expectation maximization approach for formant tracking using a parameter-free non-linear predictor

An expectation maximization approach for formant tracking us...

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International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： I. Bazzi A. Acero Li Deng Microsoft Research Limited Redmond WA USA

This paper presents a new approach for formant tracking using a parameter-free non-linear predictor that maps formant frequencies and bandwidths into the acoustic feature space. The approach relies on decomposing the speech signal into two components: the first component captures the mapping between formants and acoustic observations, while the second component is intended to capture the residual in the signal. We build the mapping by quantizing the formant space and creating a predictor codebook. Formant tracking is achieved by: (1) EM training of the parameters of the residual component, and (2) searching the predictor codebook for the best formant values. We explore both MAP and MMSE methods for performing formant tracking with the proposed approach. Furthermore, we impose first order continuity constraints on formant trajectories, and use Viterbi search to perform formant tracking. We present formant tracking results on data from the Switchboard corpus.

关键词： Bandwidth linear predictive coding Video recording Speech recognition Speech analysis Mel frequency cepstral coefficient predictive models Nonlinear acoustics Signal mapping Trajectory

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