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Application of Digital Processing of Heart Sounds To Heart Valve Modelling

IFAC Proceedings Volumes 1982年第1期15卷 639-642页

作者： D. Nandagopal J. Mazumdar R.E. Bogner G. Karolyi Department of Electrical Engineering University of Adelaide G.P.О. Box 498 Adelaide South Australia Australia Department of Applied Mathematics University of Adelaide G.P.O. Box 498 Adelaide South Australia Australia

The application of spectral analysis of heart sound signals using linear prediction coding to determine the exact resonant frequencies corresponding to heart valves vibration is discussed. The resonant frequency of a cardiac valvular structure is an important parameter in determining the in-vivo constitutive properties and normal-pathogenic states of heart valves and ventricular myocardium. A peak augmentation technique for the spectral phonocardiagram is also demonstrated to obtain improved quality of resonance. The analysis is confined mainly to the second heart sound. As a result of such studies, important diagnostic information can be obtained, thereby enhancing the clinical significance of spectral phonocardiagraphy.

关键词： phonocardiagram signal processing linear prediction coding microprocessor modelling

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OBJECTIVE QUALITY ASSESSMENT OF SPEECH ENHANCEMENT ALGORITHMS USING BOOTSTRAP-BASED MULTIPLE HYPOTHESES TESTS

OBJECTIVE QUALITY ASSESSMENT OF SPEECH ENHANCEMENT ALGORITHM...

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IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Zhihua Lu Philipp Heidenreich Abdelhak M. Zoubir Signal Processing Group Technische Universitat Darmstadt Merckstrasse 25 Darmstadt 64283 Germany

ISBN: (纸本)9781424442959

In this paper bootstrap resampling techniques are applied to assess speech quality and thereby evaluate performance of distinct speech enhancement algorithms, under the assumption that the speech segments can be approximated by an autoregressive model. A bootstrap-based multiple hypotheses testing procedure is constructed to test a distance measure based on linear predictive coding, which is the log-likelihood ratio distance. It is shown that the multiple hypotheses test results correlate well with conventional numerical distance measures, which suggests the applicability of the proposed procedure in assessment of speech quality as well as speech enhancement algorithms.

关键词： Bootstrap speech signal processing linear predictive coding multiple hypothesis testing linear prediction coding bootstrapping Hypothesis testing distance measurement Speech Voice Signal Quality assessment Speech enhancement

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Formants and LPC Analysis of Kannada Vowel Speech Signals

Formants and LPC Analysis of Kannada Vowel Speech Signals

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IEEE International Conference on Recent Trends in Electronics, Information & Communication Technology

作者： Sadashiva Chakrasali Umesh Bilembagi K. Indira Dept. of Electronics and Communication Engineering Ramaiah Institute of Technology

ISBN: (数字)9781538624401

ISBN: (纸本)9781538624418

Acoustic phonetics is the study of the physical properties of sounds and provides means to distinguish one sound from another in quality and quantity. A study of acoustic characteristics of Kannada begins with the phonemic analysis of the language. Phonetic analysis of Kannada vowels is presented in this paper. The analysis of speech signal based on formant space provides a method of assessing the influence of each formant on a phoneme across gender and different age groups. PRAAT software is used for the purpose of analysis of speech signals. In this work Kannada vowels speech signals were recorded from different age groups of both male and female. Formant frequencies of corresponding vowels were computed. The analysis is carried out separately for male and female speakers. The preliminary analysis of formants of vowels show significant variations across gender and age groups. In the similar way using the linear Predictive coding (LPC) analysis is done to get in depth understanding of formants by considering different filter orders. Then order of the LPC filter is typically estimated by using information about the formants obtained using PRAAT tool.

关键词： Formants linear predictive coding (LPC) Vowels Vocal tract Inverse filtering and spectrum formant Vocal tract Kannada linear prediction coding PCSK7 gene Voice Signal Speech

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Relation of Resonant Frequency and Appropriate linear Predictive Order for Evaluating Concrete Strength by Quality Factor

Relation of Resonant Frequency and Appropriate Linear Predic...

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IEEE International Ultrasonics Symposium

作者： Ryo Toh Yahiro Mori Seiichi Motooka Department of Electrical Electronics and Computer Engineering Chiba Institute of Technology Department of Architecture and Civil Engineering Chiba Institute of Technology

ISBN: (纸本)9781467345613

Concrete is a popular construction material used broadly in many kinds of structures. The compressive strength is the most important performance of the concrete responsible to the safety of such structures. Although method of sampling a specimen is commonly employed, it makes damage to the structures. While Schmidt Hammer measures the hardness of the surface, but it relates to the compressive strength with poor accuracy owing to the influence of the conditions of the surface. Considering that the ultrasound propagating inside the concrete with higher strength will decrease less, the quality factor of the ultrasound signals should be higher, we have proposed a new method that evaluates the concrete strength non-destructively using the quality factor of resonant peak of multi-reflected ultrasound propagated inside the concrete. The quality factor is calculated by a linear predictive coefficient (LPC) processing on the ultrasonic signals, where the order of the LPC is an important parameter, by which the number of the poles of the LPC model is determined. Neither too many poles, which include too much details of the original signal with noise, nor too few poles, which lack of the proprieties at the resonant peaks, may express the propriety of ultrasonic propagation clearly. In this paper, as one of the approach of optimizing the parameters employed in measurement and signal processing, the relation between the resonant frequency and the appropriate order of LPC is discussed by numerical simulation and experiment. In the experiment, three types of concrete specimen with identical dimensions but different strengths (48.4 N/mm~2, 37.1 N/mm~2 and 27.9 N/mm~2 respectively) and four kinds of sound sources and receivers with different resonant frequency (28 kHz, 46 kHz, 68 kHz and 105 kHz, respectively) are employed. In the numerical simulation, a low-pass IIR filter is introduced to simulate the absorption of ultrasonic propagation in the concrete. Though the absorption is v

关键词： Quality factor resonant frequencies linear prediction coding PCSK7 gene concrete strength Infinite impulse response filters Concrete numerical simulation Ultrasonic propagation linear prediction Construction Materials Compressive strength Pole

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IMPROVINGWYNER-ZIV VIDEO coding BY BLOCK-BASED DISTORTION ESTIMATION

IMPROVINGWYNER-ZIV VIDEO CODING BY BLOCK-BASED DISTORTION ES...

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European Signal Processing Conference

作者： Zouhair M. Belkoura Thomas Sikora Communication Systems Group Technische Universit¨at Berlin Einsteinufer 17 10587 Berlin Germany phone: + (49) *** fax: + (49) *** email: belkourasikora@nue.tu-berlin.de web: www.nue.tu-berlin.de

Conventional video coding uses motion estimation to perform adaptive linear predictive encoding. Wyner-Ziv coding does not use predictive coding but performs motion estimation at the decoder. Recent work uses a difference signal at the encoder to estimate the prediction quality at the decoder. In this paper, we recognise that this operation constitutes a step in the motion estimation process. We exploit this information by omitting suitable blocks, effectively implementing linear predictive coding with deadzone quantisation for parts of the input signal. This modified Wyner-Ziv coding results in large bitrate reductions as well as significant decoding complexity decrease. At certain bitrates, our modified Wyner-Ziv codec outperforms conventional hybrid coding in an I-B-I-B setup.

关键词： motion estimation hybrid coding decoders linear prediction coding prediction coding Video coding ENCODER distortion coding

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Automatic Speech Recognition bay System MFCC-LPC-SVM for energy optimization in intelligent building

Automatic Speech Recognition bay System MFCC-LPC-SVM for ene...

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International Conference on Image and Signal Processing and their Applications (ISPA)

作者： Henni Sid Ahmed Caelen Jean dept. Geni Electrica faculty science and technology University Abdelhamid Ibn Badis–Mostaganemm Route nationale Mostaganemmm Algeria LSI Carnot Institute (Software an Intelligent Systems Laboratory (LIG)CTL – Center for Software Technologies University Joseph Fourier – MultiCom Gières Grenoble France

ISBN: (纸本)9781728131573

the problem of smart buildings is their overconsumption of energy; in this article we propose a solution to this problem. This solution is to give optimum comfort to the people who occupy the smart buildings, but sometimes these people are not satisfied with this comfort, so we realized a system of automatic speech recognition, to give the possibility for these people to change the parameters of comfort, and to obtain the desired comfort. This solution allows to optimize the smart building energy, for example if a room is empty, then the person who has been in this room must say off when leaving this room, in this case the automatic recognition system of speech, instructs the system that managed the energy to turn off the lamps and lower the heater or air conditioner. This automatic speech recognition system using the MFCC method (Mel frequency cepstral coefficient) and a LPC method (linear prediction coding) for the representation of the speech signal, and SVM (support vector machine), for speech recognition. The recognition will be of global type, the speech signal will be parameterized using the MFCC coefficients and LPC coefficients to form the input vector of the recognition system, the SVM will be used for the learning phase and recognition phase. An experiment will be implemented using an input corpus, the vocabulary used in this corpus consists of 10 words in English language. The results obtained will be analyzed and compared with other similar works that use other techniques of automatic speech recognition.

关键词： MFCC LPC Speech processing SVM Classification Energy Intelligent building intelligent buildings Speech recognition Mel frequency cepstral coefficient linear prediction coding PCSK7 gene Support Vector Network Speech processing Energy sources Voice Signal COMFORT Taxonomy recognition system Persons

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Audio modeling and loudness estimation with IJDSP mobile simulations

Audio modeling and loudness estimation with IJDSP mobile sim...

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IEEE International Conference on Acoustics, Speech and Signal Processing

作者： G. Kalyanasundaram M. Banavar A. Spanias SenSIP Center Arizona State Univ. Tempe AZ USA

ISBN: (纸本)9781467369985

Audio signal modeling and simulation is important in several coding, noise removal, and recognition applications. This paper focuses on implementing models for loudness estimation and their use in estimating parameters on iOS mobile devices (iPhones and iPads). We briefly address estimating excitation patterns and loudness through auditory models. These loudness estimation and other algorithms were implemented in the award winning educational iOS app iJDSP for performing DSP simulations on mobile devices. The modules were introduced to graduate students in the general signal processing area, to evaluate their effectiveness as teaching tools. The evaluation process involved giving the students a pre-quiz, guiding them through hands-on activities on the iOS app, and finally, a post-quiz. Assessments results were positive with noticeable improvement of student understanding of topics such as spectrograms and linear predictive coding.

关键词： Loudness estimation auditory pattern iJDSP iOS mobile education loudness IO linear prediction coding Spectrogram Noise cancellation simulating Mobile devices Students Audio

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Compressive Sensing Framework for Speech Signal Synthesis Using a Hybrid Dictionary

Compressive Sensing Framework for Speech Signal Synthesis Us...

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2011 4th International Congress on Image and Signal Processing(第四届图像与信号处理国际学术会议 CISP 2011)

作者： Yue Wang Zhixing Xu Gang Li Liping Chang Chuanrong Hong College of Information Engineering Zhejiang University of Technology 310014 Hangzhou Zhejiang P.R.China

Compressive sensing (CS) is a promising focus in signal processing field, which offers a novel view of simultaneous compression and sampling. In this framework a sparse approximated signal is obtained with samples much less than that required by the Nyquist sampling theorem if the signal is sparse on one basis. Encouraged by its exciting potential application in signal compression, we use CS framework for speech synthesis problems. The linear prediction coding (LPC) is an efficient tool for speech compression, as the speech is considered to be an AR process. It is also known that a speech signal is quasi-periodic in its voiced parts, hence a discrete fourier transform (DFT) basis will provide a better approximation. Thus we propose a hybrid dictionary combined with the LPC model and the DFT model as the basis of speech signal. The orthogonal matching pursuit (OMP) is employed in our simulations to compute the sparse representation in the hybrid dictionary domain. The results indicate good performance with our proposed scheme, offering a satisfactory perceptual quality.

关键词： speech synthesis compressive sensing linear prediction coding DFT basis

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ESTIMATION OF FRAME INDEPENDENT AND ENHANCEMENT COMPONENTS FOR SPEECH COMMUNICATION OVER PACKET NETWORKS

ESTIMATION OF FRAME INDEPENDENT AND ENHANCEMENT COMPONENTS F...

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IEEE International Conference on Acoustics, Speech, and Signal Processing

作者： Daniele Giacobello Manohar N. Murthi Mads Graesboll Christensen Soren Holdt Jensen Marc Moonen Dept. of Electronic Systems Aalborg Universitet Aalborg Denmark Dept. of Electrical and Computer Engineering University of Miami USA Dept. of Electrical Engineering Katholieke Universiteit Leuven Leuven Belgium

ISBN: (纸本)9781424442959

In this paper, we describe a new approach to cope with packet loss in speech coders. The idea is to split the information present in each speech packet into two components, one to independently decode the given speech frame and one to enhance it by exploiting inter-frame dependencies. The scheme is based on sparse linear prediction and a redefinition of the analysis-by-synthesis process. We present Mean Opinion Scores for the presented coder with different degrees of packet loss and show that it performs similarly to frame dependent coders for low packet loss probability and similarly to frame independent coders for high packet loss probability. We also present ideas on how to make the coder work synergistically with the channel loss estimate.

关键词： Speech coding Voice over IP (VoIP) linear predictive coding analysis-by-synthesis linear prediction coding speech coding Packet loss ENCODER Packet switched networks Packet FRAMES linear prediction loss probability redefinition Speech Frame

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Extraction of pitch, duration and formant frequencies for emotion recognition system

Extraction of pitch, duration and formant frequencies for em...

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Fourth International Conference on Advances in Recent Technologies in Communication and Computing (ARTCom2012)

作者： A. A. Khulage Cummins college of Engineering Pune India

ISBN: (纸本)9781634396226

Analysis of speech for recognition of emotion can be done using important characteristics such as pitch, intensity, vocal tract spectrum, duration, formant frequencies etc. Out of these characteristics, pitch, duration and formant frequencies estimation method are used for identification of angry, disgust, sad, boredom and neutral emotion identification. Analysis of above mentioned characteristics is done using TU-Berlin database (Technical University of Berlin). Pitch is determined using cepstral analysis and formant frequencies are using LPC analysis. Total duration is calculated by the extraction of vowel, semivowel, and consonant containing in that particular speech signal.

关键词： Pitch Formant frequencies LPC Cepstral analysis Duration PITCHES pitch PCSK7 gene linear prediction coding Cepstral analysis emotion recognition formant Voice Signal Ship motion

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