抖音“变身”短视频中的类型化叙事与性别秩序加固

抖音平台上通过化妆术实现"变身"的短视频,通过前后的强烈对比,为观众塑造审美体验。在类型化的叙事和共享的社会经验下,观众与视频创作者共同参与并完成意义的建立与传递。但这种形象塑造神话,看似是赋予女性改造自己的主动权,其实是象征性地抚慰外貌焦虑,是对原有性别秩序的加固。通过文本分析,笔者探讨了在对这些传媒文本进行创作、消费、传播时,女性如何内化并实践这一性别意识形态。

索菲亚·古拜杜丽娜音乐作品中微分音解析

前苏联作曲家古拜杜丽娜(Sofia Gubaidulina1931-)音乐作品中微分音的使用,有其发展历程和作品表现的目的。古拜杜丽娜将微分音作为"变音"即类似经过音与辅助音的方式,以及使用两个十二半音体系进行复合,而形成的二十四音体系的微分音,这对音高使用范围的探索与拓展是一种有益的尝试。

论形声书卷字今音之注中的一类混乱现象

口语异变音和特定的声符条件引发了形声书卷字今音的异变音之注,施注当中的主观随意性和某些客观因素,则导致这种今音之注产生了一类混乱现象。混乱局面应该理顺,具体的实施方案,本文略有探索,还待进一步完善。

Canadian English and its phonetic features

British English and American English are regarded as the two major varieties of English in the world, and Canadian English is often treated as a variation of American English. The paper aims to identify the major phonetic features of Canadian English by addressing its differences from British and American English based on a historical, demographical, and regional and dialectal analysis. Canadian English is a cultural phenomenon that has evolved on the basis of British English and under the heavy influence of American English. As a distinctive variety of English, Canadian English possesses its phonetic uniqueness. And in the context of globalization, Canadian English will remain subject to Americanization.

Study of accent differences among regional varieties of English

The globalization and nativization processes of English resulted in the appearance of a large number of regional varieties of English. Concerning their accent between the single norm theory and the multiple norm differences, there has always been a debate theory. This paper argues against the ＇standard-nonstandard＇ classification of English accents upheld by the former, and revises the concept of ＇the three concentric circles of English＇ proposed by the latter, providing some helpful tips on the current instruction of listening and speaking in China＇s colleges and universities.

Seismic data denoising based on learning-type overcomplete dictionaries

The transform base function method is one of the most commonly used techniques for seismic denoising, which achieves the purpose of removing noise by utilizing the sparseness and separateness of seismic data in the transform base function domain. However, the effect is not satisfactory because it needs to pre-select a set of fixed transform-base functions and process the corresponding transform. In order to find a new approach, we introduce learning-type overcomplete dictionaries, i.e., optimally sparse data representation is achieved through learning and training driven by seismic modeling data, instead of using a single set of fixed transform bases. In this paper, we combine dictionary learning with total variation （TV） minimization to suppress pseudo-Gibbs artifacts and describe the effects of non-uniform dictionary sub-block scale on removing noises. Taking the discrete cosine transform and random noise as an example, we made comparisons between a single transform base, non-learning-type, overcomplete dictionary and a learning-type overcomplete dictionary and also compare the results with uniform and nonuniform size dictionary atoms. The results show that, when seismic data is represented sparsely using the learning-type overcomplete dictionary, noise is also removed and visibility and signal to noise ratio is markedly increased. We also compare the results with uniform and nonuniform size dictionary atoms, which demonstrate that a nonuniform dictionary atom is more suitable for seismic denoising.

Pathological Voice Classification Based on Features Dimension Opti mization

The classification of pathological voice from healthy voice was studied based upon 27 acoustic features derived from a single sound signal of vowel /a:/. First, the feature space was transferred to reduce the data dimension by principle component analysis (PCA). Then the voice samples were classified according to the reduced PCA parameters by support vector machine (SVM) using radial basis function (RBF) as a kernel function. Meanwhile, by changing the ratio of opposite class samples, the accuracy under different features combinations was tested. Experimental data were provided by the voice database of Massachusetts Eye and Ear Infirmary (MEEI) in which 216 vowel /a:/ samples were collected from subjects of healthy and pathological cases, and tested with 5 fold cross-validation method. The result shows the positive rate of pathological voices was improved from 92% to 98% through the PCA method. STD, Fatr, Tasm, NHR, SEG, and PER are pathology sensitive features in illness detection. Using these sensitive features the accuracy of detection of pathological voice from healthy voice can reach 97%.

SINGLE TONE JAMMING ELIMINATION SCHEME OF MULTICARRIER SPREAD SPECTRUM SYSTEM

The proposed scheme is based on Discrete Fourier Transform (DFT) domain processing. The key technology of this scheme is jamming parameters' accurate estimation and jamming reconstruction. Compared with the 'threshold exciser' scheme the proposed scheme can eliminate more jamming energy on the whole frequency band with the minimum loss of useful signal energy. As shown in the research and simulation, the proposed scheme is much better than the 'threshold exciser' scheme, especially in the case of high power jamming whereas the 'threshold exciser' scheme might be invalid.

Analytical Study of Nonlinear Dust Acoustic Waves in Two-Dimensional Dust Plasma with Dust Charge Variation

The nonlinear dust acoustic waves in two-dimensional dust plasma with dust charge variation is analytically investigated by using the formally variable separation approach. New analytical solutions for the governing equation of this system have been obtained for dust acoustic waves in a dust plasma for the first time. We derive exact analytical expressions for the general case of the nonlinear dust acoustic waves in two-dimensional dust plasma with dust charge variation.

A new formant feature and its application in Mandarin vowel pronunciation quality assessment

In order to improve the Mandarin vowel pronunciation quality assessment, a nox/el formant feature was proposed and applied to formant classification for Chinese Mandarin vowel pronunciation quality evaluation. Formant candidates of each frame were plotted on the time-frequency plane to form a bitmap, and its Gabor feature was extracted to represent the formant trajectory. The feature was then classified by using GMM model and the classification posterior probability was mapped to pronunciation quality grade. The experiments of comparing the Gabor transformation based formant trajectory feature with several other kinds of traditionally used features show that with this method, a human-machine scoring correlation coefficient （CC） of 0.842 can be achieved, which is better than the result of 0.832 by traditional speech recognition techniques. At the same time, considering that the long-term information of formant classification and the short-term information of speech recognition technique are complementary to each other, it is investigated to combine their results with linear or nonlinear methods to further improve the evaluation performance. As a result, experiments on PSK show that the best CC of 0.913, which is very close to the correlation of inter-human rating of 0.94, is gotten by using neural network.

Implementation of a Novel Algorithm on Acoustic Doppler Current Profiler Signal Processing System

Acoustic Doppler current profiler （ADCP） uses acoustic energy directed along narrow beams for current measurement. In conventional method, the quantity of sampling affects the precision of fast Fourier transform （FFT） algorithm, and the algorithm needs a large amount of data to process. A novel frequency estimator.enhanced least mean square （ELMS） algorithm for a single complex sinusoid in complex white Gaussian noise, is proposed in ADCP system. As sampling frequency equals 120 krad/s and the sampling number equals 240. the minimum resolving is 0. 5 krad/s. All variances keep 11.11%. ELMS algorithm needs less data than FFT. And the robust algorithm can estimate the spectrum true value to 99.9% when the signal to noise ratio （SNR） is equal to 0 dB. Experiments prove that the estimation values will diverge much from the ideal when SNR is less than -6 dB.

Fission of the End of the Century——Analysis on the Non-Rationalism Ethics Thoughts in Late Nineteenth Century European Music

In the creation works of European musicians in late nineteenth Century, it has more profound philosophical and ethical foundation, and its production and development has a profound reflection of music art on social life, but it is also a response to contemporary European romanticism non-rationalism ethics.

Some Remarks Regarding Originality and Historical Transformations of Chopin＇s Mazurkas, as Based on the Mathematical Systematic Method of the Tonal Structure Analysis

Musical analysis enables us to study the various elements of a musical work, for example texture, m dynamics, or harmony （including tonality）. Thanks to musical analysis we are able to trace the compos technique, or assess the artistic value of given work. In addition to traditional, descriptive methods of e it ody, onal music analysis, we can observe in 20th century the development of methods based on science （mathematics, otherwise statistics）. Such methodologies allow us to obtain a strict understanding of the structure of musical pieces Applying mathematical methods for musical analysis seems very reasonable in relation to the output of Frederic Chopin, due to large originality of the elements of musical pieces of this composer. Present paper was written as a result of research related to the applying of numerical method of analysis for exploring the tonal structure of Chopin＇s works. The author＇s research method enables a strict analysis of the chord＇s domination that can be classified under given key （range）--taking into consideration the harmonic functions. By the method of analysis we can get charts that depict whole diatonic vertical music material of the given piece. The main objective of the present study is to describe observations on the high uniqueness in the tonal construction of selected Chopin＇s mazurkas, especially, when compared to the miniatures of other composers of the first half of the 19th century and previously analyzed Chopin＇s pieces： etudes, preludes, and songs.

A CODEBOOK COMPENSATIVE VOICE MORPHING ALGORITHM BASED ON MAXIMUM LIKELIHOOD ESTIMATION

This paper presents an improved voice morphing algorithm based on Gaussian Mixture Model(GMM) which overcomes the traditional one in the terms of overly smoothed problems of the converted spectral and discontinuities between frames.Firstly, a maximum likelihood estimation for the model is introduced for the alleviation of the inversion of high dimension matrixes caused by traditional conversion function.Then, in order to resolve the two problems associated with the baseline, a codebook compensation technique and a time domain medial filter are applied.The results of listening evaluations show that the quality of the speech converted by the proposed method is significantly better than that by the traditional GMM method, and the Mean Opinion Score(MOS) of the converted speech is improved from 2.5 to 3.1 and ABX score from 38% to 75%.

Stochastic Resonance of Instantaneous-State in Linear Regime of Single-Mode Lasers Driven by Colored Noises

According to a linear equation of the laser intensity of single-mode laser with input signal, we compute the generalized signal-to-noise ratio （GSNR） in the instantaneous-state of the single-mode laser, which is driven by two colored noises and correlated in the form of an e-exponential function. We detect that the stochastic resonance （SR） also occurs on instantaneous state at any time. Furthermore we discuss the GSNR trend to stable state in three different forms when taking different signal frequencies.

An Audio Zero-Watermark Scheme Based on Energy Comparing

Zero-watermark technique,embedding watermark without modifying carriers,has been broadly applied for copyright protection of images.However,there is little research on audio zerowatermark.This paper proposes an audio zero-watermark scheme based on energy relationship between adjacent audio sections.Taking use of discrete wavelet transformation(DWT),it gets power approximations,or energies,of audio segments.Then,it extracts the audio profile,i.e.the zero-watermark,according to the relative size of energies of consecutive fragments.The experimental results demonstrate that the proposed scheme is robust against general malicious attacks including noise addition,resampling,low-pass filtering,etc.,and this approach effectively solves the contradiction between inaudibility and robustness.

Spatially Periodic System with Infinite Globally Coupled Oscillators Driven by Temporal-Spatial Noise

In this paper, we study spatially periodic system with infinite globally coupled oscillators driven by temporal-spatial noise and subject to a constant force. The results show that the system exhibits the phenomena of the non-equilibrium phase transition, transport of particles, and the anomalous hysteresis cycle for the mean field and the probability current.

An Intelligent Household Heart Rate Meter

The measuring instrument which uses Sunplus SPCE061A MCU as the controller is a practical intelligent household heart rate meter.It can serve all people timely and effectively to detect the heart rate.It rings some voice alarm if the HRV is abnormal.Then it is decided whether to see a doctor at a convenient time for further diagnosis.The instrument has a feature of voice guidance that can save keyboard steps,which extends the scope of use.All people,old and young,the blind and the deaf,can use it by themselves.

An Optimality-theoretic Account of Tone Sandhi in Disyllabic Sequences in Chengdu Dialect

This paper aims to offer an optimality-theoretic account of citation tones and tone sandhi in disyllabic sequences in Chengdu Dialect （CD henceforth）. It shows that the constraint ranking ＊[H,hl].T〉〉IDENT-LT〉〉 ＊[H,lh],＊[L,1h]〉〉 IDENT-RT〉〉 ＊[H,hl] is able to account for the tone sandhi phenomena of disyllabic sequences in CD.

Audio Zero-Watermark Scheme Based on Discrete Cosine Transform-Discrete Wavelet TransformSingular Value Decomposition

Traditional watermark embedding schemes inevitably modify the data in a host audio signal and lead to the degradation of the host signal.In this paper,a novel audio zero-watermarking algorithm based on discrete wavelet transform(DWT),discrete cosine transform(DCT),and singular value decomposition(SVD) is presented.The watermark is registered by performing SVD on the coefficients generated through DWT and DCT to avoid data modification and host signal degradation.Simulation results show that the proposed zero-watermarking algorithm is strongly robust to common signal processing methods such as requantization,MP3 compression,resampling,addition of white Gaussian noise,and low-pass filtering.