An Analysis of the Hoarse Speech Signals by the Three Mass Model of Vocal Cords *

A three mass model of vocal cords as well as mathematical expression of the model are discussed. Different kinds of typical hoarse speech due to laryngeal diseases are simulated on microcomputer and the effects of different pathological factors of vocal cords on model parameters are studied. Some typical spectrum distribution of the simulated speech signals are given. Moreover, hoarse speech signals of some typical cases are analyzed by the methods of digital signal processing, including FFT, LPC, Cepstrum technique, Pseudocolor encoding, etc. The experiment results show that the three mass model analysis of vocal cords is an efficient method for analysis of hoarse speech signals.

Study of engine noise based on independent component analysis

Independent component analysis was applied to analyze the acoustic signals from diesel engine. First the basic prin-ciple of independent component analysis (ICA) was reviewed. Diesel engine acoustic signal was decomposed into several inde-pendent components (ICs); Fourier transform and continuous wavelet transform (CWT) were applied to analyze the independent components. Different noise sources of the diesel engine were separated, based on the characteristics of different component in time-frequency domain.

Study of Sonic hedgehog signaling pathway related molecules in gastric carcinoma

AIM： To study the expression of Sonic hedgehog pathway-related molecules, Sonic hedgehog （Shh） and Glil in gastric carcinoma. METHODS： Expression of Shh in 56 gastric specimens including non-cancerous gastric tissues, gastric adenocarcinoma, gastric squamous cell carcinoma was detected by RT-PCR, in situ hybridization and immunohistochemistry. Expression of Glil was observed by in situ hybridization. RESULTS： The positive rate of Shh and Glil expression was 0.0%, 0.0% in non-cancerous gastric tissues while it was 66.7%, 57.8% respectively in gastric adenocarcinoma, and 100%, 100% respectively in gastric squamous cell carcinoma. There was a significant difference between the non-cancerous gastric tissues and gastric carcinoma （P 〈 0.05）. Elevated expression of Shh and Glil in gastric tubular adenocarcinoma was associated with poorly differentiated tumors while the expression was absent in gastric mucinous adenocarcinoma. CONCLUSION： The elevated expression of Shh and Glil in gastric adenocarcinoma and gastric squamous cell carcinoma shows the involvement of activated Shh signaling in the cellular proliferation of gastric carcinogenesis. It suggests Shh signaling gene may be a new and good target gene for gastric tumor diagnosis and therapy.

Filter algorithm based on cochlear mechanics and neuron filter mechanism and application on enhancement of audio signals

A filter algorithm based on cochlear mechanics and neuron filter mechanism is proposed from the view point of vibration.It helps to solve the problem that the non-linear amplification is rarely considered in studying the auditory filters.A cochlear mechanical transduction model is built to illustrate the audio signals processing procedure in cochlea,and then the neuron filter mechanism is modeled to indirectly obtain the outputs with the cochlear properties of frequency tuning and non-linear amplification.The mathematic description of the proposed algorithm is derived by the two models.The parameter space,the parameter selection rules and the error correction of the proposed algorithm are discussed.The unit impulse responses in the time domain and the frequency domain are simulated and compared to probe into the characteristics of the proposed algorithm.Then a 24-channel filter bank is built based on the proposed algorithm and applied to the enhancements of the audio signals.The experiments and comparisons verify that,the proposed algorithm can effectively divide the audio signals into different frequencies,significantly enhance the high frequency parts,and provide positive impacts on the performance of speech enhancement in different noise environments,especially for the babble noise and the volvo noise.

Extraction of Partial Discharge Acoustic Signal by Wavelet Transform with Teager′s Energy Operator

To develop a measurement system for monitoring partial discharge （PD） without the effect of external interferences,an algorithm of PD signal extraction based on wavelet transform with Teager＇s energy operators was presented. Acoustic signal generated by PD was selected to remove excessive interfering signals and electromagnetic interferences. Acoustic signals were collected and decomposed into I0 levels by wavelet transform into approximation and detail components. “Daubechies 25” was proved to be the most suitable mother wavelet for the extraction of PD acoustic signals. Compared with conventional wavelet denoising method, Teager＇s energy operators were adopted to the PD signal reconstruction and the signal to noise ratio was in creased by 20%-25% inthe experiment,without lost in energy and pulse amplitude.

The combined time-frequency analysis of the acoustic signals backscattered from ultrasonic contrast agents in the evaluation of the blood perfusion

Objective: To analyze the non-periodic, unstable and even chaotic echoes scattered from microbubbles which are extremely sensitive and may easily collapse, fragment or shrink when ultrasound contrast agents are exposed to ultrasound (US) irradiation. Methods: The combined time-frequency analysis was applied to the original signals instead of the traditional Fourier spectral analysis technique. Results: The results obtained from simulation as well as experiment showed that the subharmonic, 2nd harmonic and ultra harmonic of the microbubbles occurred during the oscillation and varied with time. The dependence on the incident ultrasonic amplitude and microbubble parameters were established. Conclusion: The transient echoes backscattered from the ultrasound agent in the evaluation of the blood perfusion can be analyzed thoroughly by the technique of combined-frequency analysis and the time detail of the frequency contents can be revealed.

Extracting Acoustic Source Information of Shipping Noise for Dynamic Ambient Noise Modelling

Shipping traffic is one of the largest contributors to anthropogenic noise in the ocean. Noise generated by merchant ships elevates natural occurring ambient noise level by 20-30 dB in many areas of the world＇s ocean. In order to model the contributions of the noise generated by merchant ships to underwater ambient noise level correctly, a database that consists of the source levels as a function of frequency for different types of ships is essential. This paper describes the conceptual design, with an emphasis on the characteristics of shipping noise as sound sources, of a marine noise database. It was developed for providing necessary parameters for underwater ambient noise modelling. The parameters relevant to shipping noise modelling are organized in two catalogues： （l） source-receiver geometry related parameters, namely the coordinates of the ships at a given time period, as well as the sizes/types of the ships from which the noise source depths may be derived, and （2） acoustically relevant parameters, i.e., the acoustic SLs （source levels） at given frequencies. An example is presented here to demonstrate the efficacy of this database. The study area is a 117 × 55 km2 region off the coast of La Spezia, Italy, in the Mediterranean Sea.

A Particle Filter Based Compressive Sensing Method for Tracking Moving Wideband Sound Sources

Tracking moving wideband sound sources is one of the most challenging issues in the acoustic array signal processing which is based on the direction of arrival(DOA) estimation. Compressive sensing(CS) is a recent theory exploring the signal sparsity representation, which has been proved to be superior for the DOA estimation. However, the spatial aliasing and the offset at endfire are the main obstacles for CS applied in the wideband DOA estimation. We propose a particle filter based compressive sensing method for tracking moving wideband sound sources. First, the initial DOA estimates are obtained by wideband CS algorithms. Then, the real sources are approximated by a set of particles with different weights assigned. The kernel density estimator is used as the likelihood function of particle filter. We present the results for both uniform and random linear array. Simulation results show that the spatial aliasing is disappeared and the offset at endfire is reduced. We show that the proposed method can achieve satisfactory tracking performance regardless of using uniform or random linear array.

A continuous differentiable wavelet threshold function for speech enhancement

Enhanced speech based on the traditional wavelet threshold function had auditory oscillation distortion and the low signal-to-noise ratio (SNR). In order to solve these problems, a new continuous differentiable threshold function for speech enhancement was presented. Firstly, the function adopted narrow threshold areas, preserved the smaller signal speech, and improved the speech quality; secondly, based on the properties of the continuous differentiable and non-fixed deviation, each area function was attained gradually by using the method of mathematical derivation. It ensured that enhanced speech was continuous and smooth; it removed the auditory oscillation distortion; finally, combined with the Bark wavelet packets, it further improved human auditory perception. Experimental results show that the segmental SNR and PESQ (perceptual evaluation of speech quality) of the enhanced speech using this method increase effectively, compared with the existing speech enhancement algorithms based on wavelet threshold.

加强集成电路的开发力度,加快我国的手机发展速度

据报道,目前我国的电话用户总数已达4.1299亿户。其中,固定电话用户数为2.1268亿户,移动电话用户数为2.0031亿户。值得注意的是,在过去的2002年中,移动电话用户数的绝对增长数远大于固定电话用户数的增长数:移动电话用户数增加了5,509万户,而固定电话用户数仅增加了3,132万户。再从电话普及率来看,固定电话为:31.99户/百户;而移动电话为14.91个/百人。就上述这些统计数字来说,相对于我国总人口数,差距仍然很大。换句话说,那就意味着。

Lung sounds auscultation technology based on ANC - ICA algorithm in high bat- tlefield noise environment

AIM:To explore the more accurate lung sounds auscultation technology in high battlefield noise environment.METHODS: In this study, we restrain high background noise using a new method-adaptive noise canceling based on independent component analysis (ANC-ICA), the method, by incorporating both second-order and higher-order statistics can remove noise components of the primary input signal based on statistical independence.RESULTS:The algorithm retained the local feature of lung sounds while eliminating high background noise, and performed more effectively than the conventional LMS algorithm.CONCLUSION:This method can cancel high battlefield noise of lung sounds effectively thus can help diagnose lung disease more accurately.

Effects of internal waves on signal temporal correlation length in the South China Sea

We simulated the temporal correlation of sound transmission using a two-dimensional advective frozen-ocean model with temperature data from a temperature sensor array on a propagation path in the South China Sea （SCS） Experiment 2009, and investigated the relationships of temporal correlation length, source-receiver range, and maximal sound speed fluctuation mainly caused by the solitary internal waves. We found that the temporal correlation length is -h2-power dependent on source-receiver range and -0.9-power dependent on maximal sound speed fluctuation. The empirical relationship is deduced from one-day environmental measurements in a limited area, needing more works and verification in the future with more acoustic data. But the relationship is useful in many applications in the area of SCS Experiment 2009.

Realtime robust speech communication based on iterative joint source-channel decoding and demodulation algorithm for MELP vocoder

Realtime speech communications require high efficient compression algorithms to encode speech signals. As the compressed speech parameters are highly sensitive to transmission errors, robust source and channel decoding and demodulation schemes are both important and of practical use. In this paper, an it- erative joint souree-channel decoding and demodulation algorithm is proposed for mixed excited linear pre- diction （MELP） vocoder by both exploiting the residual redundancy and passing soft information through- out the receiver while introducing systematic global iteration process to further enhance the performance. Being fully compatible with existing transmitter structure, the proposed algorithm does not introduce addi- tional bandwidth expansion and transmission delay. Simulations show substantial error correcting perfor- mance and synthesized speech quality improvement over conventional separate designed systems in delay and bandwidth constraint channels by using the joint source-channel decoding and demodulation （JSCCM） algorithm.

音频演示在傅里叶原理教学中的应用

傅里叶原理是"信号与系统"课程重要的教学内容,是信号分析与系统分析的基础理论,傅里叶分析将时间变量变换成频率变量,揭示了信号内在的频率特性以及信号时间特性与其频率特性之间的密切关系,从而导出了信号的频谱、带宽以及滤波、

关于提高初中计算机硬件教学效益的一点体会

在初中信息技术的教学中,有关硬件的基础知识和相关原理这部分内容一直是老师们觉得比较难上的,其主要原因是深浅难以掌握。如果讲深了、讲多了,学生不仅听不懂,而且极易产生对信息技术的厌恶情绪;如果讲浅了又达不到教材的要求。

Ultra-sensitive graphene strain sensor for sound signal acquisition and recognition

A wearable and high-precision sensor for sound signal acquisition and recognition was fabricated from thin films of specially designed graphene woven fabrics （GWFs）. Upon being stretched, a high density of random cracks appears in the network, which decreases the current pathways, thereby increasing the resistance. Therefore, the film could act as a strain sensor on the human throat in order to measure one＇s speech through muscle movement, regardless of whether or not a sound is produced. The ultra-high sensitivity allows for the realization of rapid and low-frequency speech sampling by extracting the signature characteristics of sound waves. In this study, representative signals of 26 English letters, typical Chinese characters and tones, and even phrases and sentences were tested, revealing obvious and characteristic changes in resistance. Furthermore, resistance changes of the graphene sensor responded perfectly with pre-recorded sounds. By combining artificial intelligence with digital signal processing, we expect that, in the future, this graphene sensor will be able to successfully negotiate complex acoustic systems and large quantities of audio data.

How small could a pup sound？ The physical bases of signaling body size in harbor seals

Vocal communication is a crucial aspect of animal behavior. The mechanism which most mam- mals use to vocalize relies on three anatomical components. First, air overpressure is generated in- side the lower vocal tract. Second, as the airstream goes through the glottis, sound is produced via vocal fold vibration. Third, this sound is further filtered by the geometry and length of the upper vocal tract. Evidence from mammalian anatomy and bioacoustics suggests that some of these three components may covary with an animal＇s body size. The framework provided by acoustic al- Iometry suggests that, because vocal tract length （VTL） is more strongly constrained by the growth of the body than vocal fold length （VFL）, VTL generates more reliable acoustic cues to an animal＇s size. This hypothesis is often tested acoustically but rarely anatomically, especially in pinnipeds. Here, we test the anatomical bases of the acoustic allometry hypothesis in harbor seal pups Phoca vitulina. We dissected and measured vocal tract, vocal folds, and other anatomical features of 15 harbor seals post-mortem. We found that, while VTL correlates with body size, VFL does not. This suggests that, while body growth puts anatomical constraints on how vocalizations are filtered by harbor seals＇ vocal tract, no such constraints appear to exist on vocal folds, at least during puppy- hood. It is particularly interesting to find anatomical constraints on harbor seals＇ vocal tracts, the same anatomical region partially enabling pups to produce individually distinctive vocalizations.

Automatic Artist Recognition of Songs for Advanced Retrieval

Automatic recognition of artists is very important in acoustic music indexing, browsing, and contentbased acoustic music retrieving, but synchronously it is still a challenging errand to extract the most representative and salient attributes to depict diversiform artists. In this paper, we developed a novel system to complete the reorganization of artist automatically. The proposed system can efficiently identify the artist's voice of a raw song by analyzing substantive features extracted from both pure music and singing song mixed with accompanying music. The experiments on different genres of songs illustrate that the proposed system is possible.

Sound propagation from the shelfbreak to deep water

Motivated by a phenomenon in an experiment conducted in the Northwestern Pacific indicating that the energy of the received signal around the sound channel axis is much greater than that at shallower depths,we study sound propagation from the transitional area(shelfbreak)to deep water.Numerical simulations with different source depths are first performed,from which we reach the following conclusions.When the source is located near the sea surface,sound will be strongly attenuated by bottom losses in a range-independent oceanic environment,whereas it can propagate to a very long range because of the continental slope.When the source is mounted on the bottom in shallow water,acoustic energy will be trapped near the sound channel axis,and it converges more evidently than the case where the source is located near the sea surface.Then,numerical simulations with different source ranges are performed.By comparing the relative energy level in the vertical direction between the numerical simulations and the experimental data,the range of the air-gun source can be approximated.

New progress on acoustic communication in the concave-eared torrent frog and its revelation

Animals have special solution to the problem of communication in high levels of background noise. A small group of vertebrates (bats,dolphins and whales,and some rodents) that use ultrasound for communication.Our research first demonstrated that the concave-eared torrent frog is the first non- mammalian vertebrate found to be capable of pro- ducing and detecting ultrasounds for communica- tion.This study may provide a clue for understand- ing why humans have ear canals and how animals auditory systems have evolved,and inspire in de- veloping bionic tecnology for improving hearing in noise.