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A method for pitch estimation from noisy speech signals based on a pitch-harmonic extraction

A method for pitch estimation from noisy speech signals base...

引用

International Conference on Neural Networks and signal processing

作者： C. Shahnaz W. -P. Zhu M. O. Ahmad Centre for Signal Processing and Communications Department of Electrical and Computer Engineering Concordia University Montreal QUE Canada

ISBN: (纸本)9781424423101

A new method for pitch estimation from noisy speech signals based on a pitch-harmonic extraction is presented in this paper. At first, a variable-length average magnitude difference function (VLAMDF) that is able to alleviate the adverse effects of intrinsic falling minima of the conventional AMDF, has been proposed. We argue that the fast Fourier transform (FFT) power spectrum of the VLAMDF is capable of revealing an estimate of a pitch-harmonic (PH) more accurately even in a heavy noisy scenario. Then, exploiting the extracted PH, a two-fold criterion is developed in the FFT domain to acquire the corresponding harmonic number for the determination of the pitch. The superior efficacy of the proposed pitch estimator relative to some of the existing ones is confirmed through simulation results using the Keele database.

关键词： Pitch estimation coordinated mining Voice signal Noise Fast Fourier transform Speech PH signals PITCHES pitch

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Identification of autoregressive moving average systems from noise-corrupted observations

Identification of autoregressive moving average systems from...

引用

Annual IEEE Northeast Workshop on Circuits and Systems (NEWCAS)

作者： S. A. Fattah W.-P. Zhu M. O. Ahmad Centre for Signal Processing and Communications Department of Electrical and Computer Engineering Concordia University Montreal QUE Canada

In this paper, a new scheme for the identification of minimum-phase autoregressive moving average (ARMA) systems from noise-corrupted observations is presented. Analyzing the characteristics of the autocorrelation function (ACF) of the observed data in the presence of noise, a set of equations has been developed which is capable of estimating the AR parameters of the ARMA system as well as the noise variance. In order to estimate the MA parameters, first, a residual signal is obtained by filtering the noisy observations via the estimated AR parameters. Utilizing the estimated noise variance and the AR parameters, a noise-subtraction algorithm is proposed to reduce the effect of noise from the ACF of the residual signal. The MA parameters are then estimated employing the spectral factorization corresponding to the noise-compensated ACF of the residual signal. computer simulations on different ARMA systems demonstrate a superior identification results in terms of estimation accuracy and consistency even under a heavy noisy condition.

关键词： Noise Estimation Noise measurement Accuracy Correlation signal to noise ratio System identification

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On extracting pitch from noisy speech signals based on spectral and temporal enhancement

On extracting pitch from noisy speech signals based on spect...

引用

Annual IEEE Northeast Workshop on Circuits and Systems (NEWCAS)

作者： C. Shahnaz W. -P. Zhu M. O. Ahmad Centre for Signal Processing and Communications Department of Electrical and Computer Engineering Concordia University Montreal QUE Canada

A new pitch extractor based on spectral and temporal enhancement of noisy speech signals is presented in this paper. A discrete cosine transform based modified power spectral subtraction scheme is developed and employed prior to pitch extraction in order to suppress the underlying non-stationary noise. The de-noised speech thus obtained is then passed through an inverse filter, whose parameters are derived from the linear prediction (LP) analysis, yielding an output referred to as the LP residual. Since the LP residual is capable of delivering the knowledge of Glottal Closure instants, exploiting its high correlation property, an average magnitude sum function as well as an average magnitude difference function are introduced. The periodicity property of both the functions is argued to be integrated and an enhanced temporal function is put forward for robust pitch extraction in a multi-talker babble noise scenario. The superior efficacy of the proposed pitch extractor relative to some of the existing ones is confirmed through simulation results using the Keele database.

关键词： Speech Noise Discrete cosine transforms Noise measurement signal to noise ratio Speech enhancement Databases

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An Algorithm for ARMA Model Parameter Estimation from Noisy Observations

An Algorithm for ARMA Model Parameter Estimation from Noisy ...

引用

2008 IEEE International Symposium on Circuits and Systems (ISCAS 2008), vol.9

作者： S. A. Fattah W.-P. Zhu M. O. Ahmad Centre for Signal Processing and Communications Department of Electrical and Computer Engineering Concordia University Montreal QUE Canada

This paper presents a new algorithm for the parameter estimation of minimum-phase autoregressive moving average (ARMA) systems from noise-corrupted observations. In order to estimate the AR parameters of the ARMA system, based on a repeated autocorrelation function (ACF) of the observed data, a set of zero lag compensated equations has been developed. For the estimation of the MA parameters, first, a noise-subtraction algorithm is proposed to reduce the effect of noise from the ACF of the residual signal which is obtained by filtering the noisy ARMA signal via the estimated AR parameters. The MA parameters are then estimated by using a spectral factorization corresponding to the noise-compensated ACF of the residual signal. computer simulations are carried out for ARMA systems of different orders under noisy environments and simulation results demonstrate a superior identification performance in terms of estimation accuracy and consistency.

关键词： Parameter estimation Working environment noise signal processing algorithms Noise reduction Autoregressive processes Autocorrelation Equations signal processing computer simulation Filters

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A pitch extraction algorithm in noise based on temporal and spectral representations

A pitch extraction algorithm in noise based on temporal and ...

引用

International Conference on Acoustics, Speech, and signal processing (ICASSP)

作者： C. Shahnaz W. -P. Zhu M. O. Ahmad Centre for Signal Processing and Communications Department of Electrical and Computer Engineering Concordia University Montreal QUE Canada

In this paper, a new algorithm for pitch extraction from noisy speech signals based on both temporal and spectral representations is presented. We derive a harmonic sinusoidal correlation (HSC) model of clean speech as a temporal representation. Given only a noisy speech frame, a noise-robust least-squares minimization technique is proposed to acquire the parameters of the HSC model which are directly employed for the accurate estimation of a pitch-harmonic (PH). Exploiting the extracted PH and based on a spectral representation which is an enhanced spectrum in the discrete cosine transform domain, a two-fold criterion is developed in order to achieve the true consecutive number corresponding to PH that is finally adopted for pitch detection in the presence of noise. Simulation results using the Keele pitch extraction reference database manifest that combining the multi cues obtained from the temporal as well as spectral representations, the proposed algorithm is able to achieve a superior efficacy in comparison to some of the existing methods from high to very low signal-to-noise ratio (SNR) levels.

关键词： Speech enhancement signal processing algorithms Speech processing Discrete cosine transforms signal to noise ratio Noise robustness Power harmonic filters Autocorrelation White noise Speech analysis

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A signal Perturbation Free Transmit Scheme for MIMO Channel Estimation

A Signal Perturbation Free Transmit Scheme for MIMO Channel ...

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IEEE Conference on Vehicular Technology (VTC)

作者： Feng Wan Wei-Ping Zhu M. N. S. Swamy Centre for Signal Processing and Communications Department of Electrical and Computer Engineering Concordia University Montreal QUE Canada

In this paper, a novel signal perturbation free transmit scheme is proposed for MIMO channel estimation. A perturbation analysis of the WR-based method is first conducted, showing that the method is subject to a signal perturbation error and therefore, its performance is very poor under the moderate to high signal-to-noise ratios (SNRs). A new transmit structure is then proposed to cancel the signal perturbation error at the receiver in order to improve the performance of the WR-based method in the high SNR case. computer simulations show that the WR-based method with the proposed signal perturbation free transmit scheme significantly outperforms the original WR-based method as well as the training-based LS method in terms of the MSE of the channel estimate.

关键词： MIMO Channel estimation signal to noise ratio Noise Algorithm design and analysis Matrix decomposition Receiving antennas

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A Frequency-Domain Correlation Matrix Estimation Algorithm for MIMO-OFDM Channel Estimation

A Frequency-Domain Correlation Matrix Estimation Algorithm f...

引用

IEEE Conference on Vehicular Technology (VTC)

作者： Feng Wan Wei-Ping Zhu M. N. S. Swamy Centre for Signal Processing and Communications Department of Electrical and Computer Engineering Concordia University Montreal QUE Canada

The second-order statistics of a time-domain signal are very often used in blind and semi-blind channel estimation. Considering that the received signal in MIMO-OFDM systems might be corrupted in the time-domain due to some adverse factors such as frequency offset and large peak-to-average power ratio (PAPR), an IFFT processor is required in the receiver to achieve a good-quality time-domain signal. This additional IFFT incurs extra computational complexity and probably a long time delay as well in real-time communication systems. In this paper, we propose a new algorithm for the computation of the time-domain correlation matrix directly from the received frequency- domain signal. The proposed frequency-domain correlation matrix estimation method is then used to develop a new semi-blind MIMO-OFDM channel estimation approach. A number of computer simulation based experiments are conducted, confirming the effectiveness of the proposed method.

关键词： Channel estimation Frequency domain analysis Correlation Matched filters Time domain analysis Estimation Finite impulse response filter

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On the estimation of pitch of noisy speech based on time and frequency domain representations

On the estimation of pitch of noisy speech based on time and...

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Canadian Conference on electrical and computer engineering (CCECE)

作者： C. Shahnaz W. -P. Zhu M. O. Ahmad Centre for Signal Processing and Communications Department of Electrical and Computer Engineering Concordia University Montreal QUE Canada

In this paper, we propose a new algorithm for pitch estimation from speech signals heavily degraded by additive noise based on both time and frequency domain representations. A least-squares minimization technique is first developed for the accurate estimation of a pitch-harmonic (PH) wherein a harmonic sinusoidal model of clean speech is exploited as a time domain representation. Then, relying on a power spectrum in the Fast Fourier Transform domain which is a frequency domain representation, a two-step criterion is formulated in order to acquire a true harmonic number corresponding to the extracted PH for robust pitch detection. Extensive simulations have been carried out to demonstrate the effectiveness of the proposed methodology as compared to some of the existing techniques in literature. It has been shown that our new approach consistently outperforms the other methods especially at low levels of signal-to-noise ratio (SNR).

关键词： Frequency estimation Frequency domain analysis Speech enhancement Fast Fourier transforms signal processing algorithms Additive noise signal to noise ratio Personal digital assistants Autocorrelation Noise robustness

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Evaluation of voice activity and voicing detection

Evaluation of voice activity and voicing detection

引用

INTERSPEECH 2008 - 9th Annual Conference of the International Speech Communication Association

作者： Kotnik, Bojan Sendorek, Pierre Astrov, Sergey Koc, Turgay Ciloglu, Tolga Fernández, Laura Docío Banga, Eduardo Rodríguez Höge, Harald Kačič, Zdravko Faculty of Electrical Engineering and Computer Science University of Maribor Slovenia Department TSI Signal-Images Telecom Paristech France Corporate Technology Siemens AG Munich Germany Electrical and Electronics Eng. Dept Middle East Technical University Ankara Turkey Signal Processing and Communications Dept. University of Vigo Spain

This paper describes the ECESS evaluation campaign of voice activity and voicing detection. Standard VAD classifies signal into speech and non-speech, we extend it to VAD+ so that it classifies a signal as a sequence of non-speech, voiced and unvoiced segments. The evaluation is performed on a portion of the Spanish SPEECON database with manually labeled segmentation. To avoid errors caused by the limited precision of manual labeling we introduce "dead zones" -tolerance intervals ±5 ms around label changes in the data set. In these tolerance intervals we don't evaluate the signal. Copyright © 2008 ISCA.

关键词： Speech communication

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On the fast generation of long-period pseudorandom number sequences

On the fast generation of long-period pseudorandom number se...

引用

IEEE Long Island Systems, Applications and Technology Conference (LISAT)

作者： Ishaan L. Dalal Jared Harwayne-Gidansky Deian Stefan SProCom2 Department of Electrical Engineering Center for Signal Processing Communications and Computer Engineering Research The Cooper Union for the Advancement of Science and Art New York NY USA

Monte Carlo simulations and other scientific applications that depend on random numbers are increasingly implemented in parallel configurations in programmable hardware. High-quality pseudo-random number generators (PRNGs), such as the Mersenne Twister, are based on binary linear recurrence equations. They have extremely long periods (more than 2 1024 numbers generated before the entire sequence repeats) and well-proven statistical properties. Many software implementations of such dasialong-periodpsila PRNGs exist, but hardware implementations are rare. We develop optimized, resource-efficient parallel architectures for long-period PRNGs that generate multiple independent streams by exploiting the underlying algorithm as well as hardware-specific architectural features.

关键词： Hardware Random access memory Field programmable gate arrays Throughput Silicon Mathematical model Equations

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