对话、殷勤之意与语篇声音——关于旅游广告语域中人际习语表达的话语分析

本文试图考察“人际习语表达”在话语中的功能。所用语料是The TimesMagazine和Newsweek两种杂志中的旅游广告。在旅游广告中,人际习语表达是构建对话性、语篇声音、礼貌与殷勤的重要手段。旅游广告作为现代社会的“经典话语”,从本质上讲具有对话性、多声性。其主要话语功能是促销商品和服务,其占主导地位的话语气氛是友好而非冲突,因此表达殷勤、好客的人际习语表达在这一语域中备受青睐。用于报道的人际习语表达有利于创造旅游广告话语的多声性。旅游广告话语中习语资源的大量使用,体现了自由经营的基本规律,是“语言资本主义”最新阶段。

从视点视角透析语篇“声音”的生成机制——以自由引语为例

声音是作者运用概念化的语言手段传达非概念意义的情感认知方式。语篇构建过程是一个高级的认知过程,视点作为认知处理的基本要素,不仅影响语篇的表层结构的组织,又制约语篇深层结构。声音是视点认知过程的一种表现形式,体现了作者视点的选择和安排。因此从视点视角解读语篇声音,可以从语篇表层到语篇深层结构全方位地阐释语篇"声音"的生成过程以及产生的感知效果,同时也有助于读者理解和把握语篇主题。

20世纪前期上海流行歌曲的诗学阐释

盛行于20世纪20—30年代的上海流行歌曲不仅是中国现代音乐的重要现象,也是不可轻视的诗学现象,其抒情话语所呈现出来的独特的文本语言、文本结构、审美趣味及隐匿其后的写作策略都在相当程度上拓展了中国现代诗学的阐释空间,丰富了中国现代诗学的内涵。本文以"大众诗学"作为理论的切入点,以接受美学、文化研究理论作为基本阐释方式,发掘上海流行歌曲中的歌词所蕴含的诗学价值。

Whisper intelligibility enhancement based on noise robust feature and SVM

A machine learning based speech enhancement method is proposed to improve the intelligibility of whispered speech. A binary mask estimated by a two-class support vector machine （SVM） classifier is used to synthesize the enhanced whisper. A novel noise robust feature called Gammatone feature cosine coefficients （GFCCs） extracted by an auditory periphery model is derived and used for the binary mask estimation. The intelligibility performance of the proposed method is evaluated and compared with the traditional speech enhancement methods. Objective and subjective evaluation results indicate that the proposed method can effectively improve the intelligibility of whispered speech which is contaminated by noise. Compared with the power subtract algorithm and the log-MMSE algorithm, both of which do not improve the intelligibility in lower signal-to-noise ratio （SNR） environments, the proposed method has good performance in improving the intelligibility of noisy whisper. Additionally, the intelligibility of the enhanced whispered speech using the proposed method also outperforms that of the corresponding unprocessed noisy whispered speech.

Discriminative tone model training and optimal integration for Mandarin speech recognition

Two discriminative methods for solving tone problems in Mandarin speech recognition are presented. First, discriminative training on the HMM （hidden Markov model） based tone models is proposed. Then an integration technique of tone models into a large vocabulary continuous speech recognition system is presented. Discriminative model weight training based on minimum phone error criteria is adopted aiming at optimal integration of the tone models. The extended Baum Welch algorithm is applied to find the model-dependent weights to scale the acoustic scores and tone scores. Experimental results show that tone recognition rates and continuous speech recognition accuracy can be improved by the discriminatively trained tone model. Performance of a large vocabulary continuous Mandarin speech recognition system can be further enhanced by the discriminatively trained weight combinations due to a better interpolation of the given models.

SPEECH ENHANCEMENT METHOD FOR LPC AUTOREGRESSIVE MODEL AND SYSTEM IMPLEMENTATION OF COMMAND WORD RECOGNITION USED IN NOISY ENVIRONMENT

At present, almost all the systems and products for speech recognition are working in quiet environment and their performances are degraded or even can′t work when they are operated in high noisy environment. In this paper, after analyzing the features of speech and noise, a speech enhancement method for LPC autoregressive model for command words recognition used in noisy environment is proposed, and an experimental system is realized. In different background noisy environments, we conduct experiments about SNR, basic accuracy, noise resistant ability and system environment adaptability with different microphones. The experimental results show that the system has good recognition performance in high noisy environments. The system can resist many kinds of noises and meet the needs of application areas on the whole such as military, traffic, marketplace and factory etc.

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.

Frame-bitrate-change based steganography for voice-over-IP

Steganography based on bits-modification of speech frames is a kind of commonly used method, which targets at RTP payloads and offers covert communications over voice-over-IP(Vo IP). However, direct modification on frames is often independent of the inherent speech features, which may lead to great degradation of speech quality. A novel frame-bitrate-change based steganography is proposed in this work, which discovers a novel covert channel for Vo IP and introduces less distortion. This method exploits the feature of multi-rate speech codecs that the practical bitrate of speech frame is identified only by speech decoder at receiving end. Based on this characteristic, two steganography strategies called bitrate downgrading(BD) and bitrate switching(BS)are provided. The first strategy substitutes high bit-rate speech frames with lower ones to embed secret message, which introduces very low distortion in practice, and much less than other bits-modification based methods with the same embedding capacity. The second one encodes secret message bits into different types of speech frames, which is an alternative choice for supplement. The two strategies are implemented and tested on our covert communication system Steg Vo IP. The experiment results show that our proposed method is effective and fulfills the real-time requirement of Vo IP communication.

Improved speech absence probability estimation based on environmental noise classification

An improved speech absence probability estimation was proposed using environmental noise classification for speech enhancement.A relevant noise estimation approach,known as the speech presence uncertainty tracking method,requires seeking the "a priori" probability of speech absence that is derived by applying microphone input signal and the noise signal based on the estimated value of the "a posteriori" signal-to-noise ratio(SNR).To overcome this problem,first,the optimal values in terms of the perceived speech quality of a variety of noise types are derived.Second,the estimated optimal values are assigned according to the determined noise type which is classified by a real-time noise classification algorithm based on the Gaussian mixture model(GMM).The proposed algorithm estimates the speech absence probability using a noise classification algorithm which is based on GMM to apply the optimal parameter of each noise type,unlike the conventional approach which uses a fixed threshold and smoothing parameter.The performance of the proposed method was evaluated by objective tests,such as the perceptual evaluation of speech quality(PESQ) and composite measure.Performance was then evaluated by a subjective test,namely,mean opinion scores(MOS) under various noise environments.The proposed method show better results than existing methods.

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.

A speech enhancement algorithm to reduce noise and compensate for partial masking effect

To enhance the speech quality that is degraded by environmental noise,an algorithm was proposed to reduce the noise and reinforce the speech.The minima controlled recursive averaging(MCRA) algorithm was used to estimate the noise spectrum and the partial masking effect which is one of the psychoacoustic properties was introduced to reinforce speech.The performance evaluation was performed by comparing the PESQ(perceptual evaluation of speech quality) and segSNR(segmental signal to noise ratio) by the proposed algorithm with the conventional algorithm.As a result,average PESQ by the proposed algorithm was higher than the average PESQ by the conventional noise reduction algorithm and segSNR was higher as much as 3.2 dB in average than that of the noise reduction algorithm.

Speech enhancement through voice activity detection using speech absence probability based on Teager energy

In this work, a novel voice activity detection （VAD） algorithm that uses speech absence probability （SAP） based on Teager energy （TE） was proposed for speech enhancement. The proposed method employs local SAP （LSAP） based on the TE of noisy speech as a feature parameter for voice activity detection （VAD） in each frequency subband, rather than conventional LSAP. Results show that the TE operator can enhance the abiTity to discriminate speech and noise and further suppress noise components. Therefore, TE-based LSAP provides a better representation of LSAP, resulting in improved VAD for estimating noise power in a speech enhancement algorithm. In addition, the presented method utilizes TE-based global SAP （GSAP） derived in each frame as the weighting parameter for modifying the adopted TE operator and improving its performance. The proposed algorithm was evaluated by objective and subjective quality tests under various environments, and was shown to produce better results than the conventional method.

Study on Acoustic Modeling in a Mandarin Continuous Speech Recognition

The design of acoustic models is of vital importance to build a reliable connection between acoustic wave-form and linguistic messages in terms of individual speech units. According to the characteristic of Chinese phonemes, the base acoustic phoneme units set is decided and refined and a decision tree based state tying approach is explored. Since one of the advantages of top-down tying method is flexibility in maintaining a balance between model accuracy and complexity, relevant adjustments are conducted, such as the stopping criterion of decision tree node splitting, during which optimal thresholds are captured. Better results are achieved in improving acoustic modeling accuracy as well as minimizing the scale of the model to a trainable extent.

AN IMPROVED 2.4kb/s MIXED EXCITATION LINEAR PREDICTION VOCODER

This letter presents two improvements on 2.4 kb/s Mixed-Excitation Linear Prediction (MELP) vocoder. The one is a new parameter Redzc named energy to differential zerocrossing rate which is used in adaptation of V/UV decision of transitional segments and low energy level speech segments. The other is a multi-path searching method for Multi-Stage Vector Quantization (MSVQ) of line spectral frequency. Subjective tests show that the intelligiblity and naturallity of improved MELP vocoder are preferable to those of the original one.

Learn a rich foreign language research on teaching voice Chinese language of minority

Foreign teaching and Chinese language teaching of minority are two different forms of teaching in nature, both the nature and the system of teaching are different, but in the teaching objectives, content, principles, methods, and therefore there are many similarities, so two studies outcomes can learn from each other. In this paper, with voice teaching as an example, we will talk about foreign language studies in ethnic minority Chinese Teaching. I think, in the teaching program design, the teaching Chinese minorities can learn from the simplified idea of teaching phonological Mr. Zhao Jin ruing, which is proposed foreign teaching Chinese, to simplify Voice teaching. In teaching principles, we should emphasize specific principles, instructional design based on students＇ different characteristics. In the teaching content, teaching should not only pay attention phonetic syllables within the system, but also more emphasis on syllables voice level teaching, especially teaching speech flow. In teaching methods, we should learn teaching, research tone, softly and so on.

Enhancement of Voice Quality and Bit-rate Reduction by Prosodic Control in G.723.1 Vocoder

Speech coding techniques have been studied not truly to reduce the complexity and bit rate but also to improve the sound quality. CELP type vocoder, used as standard, supports the great stead quality even low bit rate. In this paper, the preprocessing of input speech to reduce the bit rate is different from the conventional vocoder. Different kinds of parameter are used for the preprocessing compared with the other parameters to t＇md the more appropriate parameter for the vocoder. The Parameters are used to synthesize the speech not to encode or decode for coding technique so we proposed the simple algorithm not to have the influence on the processing time or the computation time. The parameters in the preprocessing step are speaking rate, duration, and PSOLA technique.

Lexical Tone Perception in Mandarin Chinese Speakers With Aphasia

The brain localization debate of lexical tone processing concerns functional hypothesis that lexical tone, owing to its strong linguistic features, is dominant in the left hemisphere, and acoustic hypothesis that all pitch patterns, including lexical tone, are dominant in the right hemisphere due to their acoustic features. Lexical tone as a complex signal contains acoustic components that carry linguistic, paralinguistic, and nonlinguistic information. To examine these two hypotheses, the current study adopted triplet stimuli including Chinese characters, their corresponding pinyin with a diacritic, and the four diacritics representing Chinese lexical tones. The stimuli represent the variation of lexical tone for its linguistic and acoustic features. The results of a listening task by Mandarin Chinese speakers with and without aphasia support the functional hypothesis that pitch patterns are lateralized to different hemispheres of the brain depending on their functions, with lexical tone to the left hemisphere as a function of linguistic features.

A Network Model on the Processing of Sound Wave

On the base of auditory neural system, the network model on the processing of the sound wave is presented. The mathematic equation of the network is also discussed. In the network model, in addition to the negative feedback of the neural cell in the output layer, the cell in the input layer excites the corresponding cell in the ontput layer meanwhile it inhibits the lateral cells. The network has its advantage on the processing of sound wave. In addition to filter the noise, it can search the significance frequency segments （Barks）. The ＂channel suppresser＂ feature, the special phenomena of the human ear, is explained based on the model. The learning algorithm of the network model is discussed, too. In the end, an example is introduced about the application of the network.

Translator＇s Voice in Translated Texts

Ever since Venuti put forward the concept of translator＇s invisibility in 1995, studies have been conducted on the discursive presence of translators in the translated texts. The translator, as the receiver of the source text and m the meantime the producer of the target text, is sure to leave his/her voice traceable in the translated texts throughout the whole translating process. This paper aims to present an overview of the conceptual development of the translator＇s voice in translation studies from different perspectives like narratology, stylistics, socio-narrative theory, speech-act theory etc.

The Free-Flowing Voice and its Connection With the Mother Tongue

Does the native tongue confer greater authenticity and connection？ And how does this connect with languages acquired later in life？ From thirty years of directing, training, and auditioning actors from a range of ethnicities, I have believed that the mother-tongue has a particular and organic connection for an actor, one difficult to achieve in any other language. This belief was confounded in a laboratory conducted with Romanian actors, March 2013. The work was performed in both English and Romanian and it was with a sense of shock that I observed that the work was more vital, compelling, and physically and vocally engaged when they spoke in English. What were the factors at play here and what are the implications for future work？ Patsy Rodenburghas written of the giddy delight children find in language. Under what conditions does the native tongue evoke that ＂giddy delight＂ and where and when does it become an obstacle to such pleasure？