語音分析合成 的英文怎麼說

中文拼音 [yǔyīnfēnxīgěchéng]
語音分析合成 英文
speech analysis by synthesis
  • 語 : 語動詞[書面語] （告訴） tell; inform
  • 音 : 名詞1. （聲音） sound 2. （消息） news; tidings 3. [物理學] （音質） tone 4. （姓氏） a surname
  • 分 : 分Ⅰ名詞1. （成分） component 2. （職責和權利的限度） what is within one's duty or rights Ⅱ同 「份」Ⅲ動詞[書面語] （料想） judge
  • 析 : Ⅰ動詞1. （分開; 散開） divide; separate 2. （分析） analyse; dissect; resolve Ⅱ名詞（姓氏） a surname
  • 合 : 合量詞（容量單位） ge, a unit of dry measure for grain （=1 decilitre）
  • 成 : Ⅰ動詞1 （完成; 成功） accomplish; succeed 2 （成為; 變為） become; turn into 3 （成全） help comp...
  • 語音 : speech sounds; pronunciation; voice
  1. On the basis of the study on the speech coder algorithms, paper describe an advanced method of developing dsp system software, and as the guidlines, we developed the programme of whole decoder unit. paper stress on analysis of the ecu in decoder unit. aiming at amr algorithms disadvantage of angularity of synthetical speech, paper study on the specutral extrapolation which apply to extrapolate reflect coefficient of track model to make error conceal processing of amr. at last paper analyze existing echo cancellation algorithms using on mobile communication system

    在此基礎上，描述了一種較為先進的大型dsp系統程序開發策略，並以此為指導思想，以美國ti公司c6000dsp開發平臺開發出了整個amr解碼器單元的系統程序。論文對amr解碼器的誤碼隱藏處理單元進行了重點分析，針對原有演算法合成語音自然度不好的缺點，論文研究了將譜外推法應用到amr演算法中外推出聲道模型反射系數參數進行誤碼消除處理。
  2. It synthesizes the excellence of wave coding and parameter coding, adopts vector quantity, analyse - synthesize, perceptual weighting, therefore, gains good speech coding quality at 8kbit / s. cs - acelp can be used in individual telecom, iphone, c / n, microwave telecom and isdn

    Cs - acelp演算法綜合了波形編碼和參數編碼的優點，以自適應預測編碼技術為基礎，採用了矢量量化、合成分析和感覺加權等技術，在8kbit / s速率上獲得了較高的語音編碼質量。
  3. Professor ching s research interests include adaptive digital signal processing, time delay estimation and target localization, wavelets, blind signal estimation and separation, automatic speech recognition for cantonese, speech modeling and speech synthesis, hands - free communications and advanced signal processing techniques for communications

    程教授從事的研究包括信息技術、數字信號處理、語音分析合成、識別，以及陣列處理及通信等。他出版了一百六十多篇專文，先後于國際性期刊刊登及於國際會議中發表。
  4. Carry on emulation to melp standard, realize that the compression of the pronunciation file is solved and pressed. first this thesis sample to wav file, carry on the speech to analyze and draws with the parameter to the speech data of every frame. these parameter include pitch, bpvc, jitter, lpc, etc. then, these parameters will be quantized by msvq technology

    該系統首先對語音信號進行采樣；按幀對語音數據進行語音分析和參數提取，提取的參數包括基音周期（ pitch ） 、多帶清濁音判別、非周期抖動標志、線性預測參數（ lpc ）等語音生成模型參數；接著對這些參數進行了量化，量化採用了多級矢量量化技術；最後在解碼端對各個量化參數進行解碼，利用這些參數結合語音合成模型重構語音。
  5. Focused on the sine model of the speech signal creation, and finished the reconstruction by using it. at the same time, based on the knowledge of digital signal processing, reconstruction by harmonic correction and time variant digital filter was proposed

    分析了語音信號產生的正弦模型，並在此基礎上完成了骨導信號的語音重構；與此同時，結合數字信號處理知識，分別用諧波修正和時變數字濾波器的方法，完成基於骨導信號的語音重構。
  6. A voice hiding algorithm that can be used in global system for mobile communication was proposed. it is speech - hiding algorithm based on the analysisbysynthesis energy ratio

    提出了一種可在全球移動通信系統gsm中使用的語音信息隱藏演算法，即基於分析合成的能量比調整演算法。
  7. Firstly, this article solves problems in the uighur text analysis, such as syllable separation, and summarize stress ? pause and tone rules of prosodic based on the features of the uighur language and phonetics. then propose the design of ? context vector ?, and uses greedy algorithm to optimize corpus. at last, this article introduces synthesis method using variable - length concatenating units

    首先根據維語的語音和語言特徵，解決了音節劃分等有關文本分析的問題，並總結了重音、停頓、語氣等韻律規則；然後採用「語境矢量」的設計，用greedy演算法優化語料庫；最後採用不定長單元的拼接合成方法，首先選擇較大單元合成，當拼接單元為音節時，用viterbi演算法，基於語境挑選出最優的單元合成語音。
  8. Following the direction of sinusoidal modeling and sinusoidal analysis, this thesis adopted the matching pursuit techniques along with the psychoacoustic model, explored some novel methods for sinusoidal modeling as well as the quantization of model parameters, and discussed the low bit rate speech coding and its related problems. the major contributions of this thesis are included in the following : 1

    本文正是沿著正弦建模正弦分析的方向，採用匹配跟蹤技術，結合心理聲學模型，研究了新的建模方法以及模型參數的量化編碼，對低位率語音編碼及相關問題進行了有益的探索，並取得了如下創新性研究成果： 1
  9. And even more incredible is the young brain ' s ability to pick out an order in language from the mixture of sound around him, to analyze, to combine and recombine the parts of a language in new ways

    甚至更令人難以置信的是，幼兒的大腦能夠從周圍的嘈雜的聲音中把話語的條理理出，並且能夠以新的方式對語言成分進行分析、組合以及重新組合。
  10. Considering the main problems of traditional mandarin text - to - speech system, in - depth research was conducted on a series of key techniques such as text prosodic level marking, corpus analysis and design, unit selection strategy and etc. we firstly take a glance back at the history of mandarin speech synthesis technology whose defects is also indicated

    針對傳統的漢語文語轉換系統存在的主要問題，採用基於語料庫的語音合成方法，在文本韻律層級標注、語料庫分析與設計、合成單元挑選策略等關鍵技術上做了一系列研究。
  11. System scheme of speech coding plus spread spectrum communication was presented based on a full analysis of noise characteristic, attenuation characteristic and impedance characteristic of low - voltage power line. spread spectrum carrier ( abbreviated as ssc ) technology is adopted to overcome problems existing in signal transmission over power line. high quality, low rate mbe compression algorithm was used to complete speech encoding and decoding

    在對低壓電力線路的噪聲特性、衰減特性和阻抗特性三個方面充分分析的基礎上，本文提出一種語音編碼+擴頻傳輸的系統總體方案，採用擴頻載波（ spreadspectrumcarrier ，縮寫為ssc ）技術克服電力線傳輸信號存在的問題，採用語音合成質量高並具有較低碼率的mbe壓縮演算法完成語音信號的編解碼。
  12. The wavelet analysis is applying widely to pure mathematics, applied mathematics, signal processing, speech recognition and synthesis, automation processing and image analysis etc. the key advantage of the wavelet analysis is that, in contrast to the conventional fourier analysis, it can be localized in the time and frequency domain simultaneously

    小波分析已經廣泛應用於理論數學、應用數學、信號處理、語音識別與合成、自動控制和圖像處理與分析等領域。同傳統的傅立葉分析相比，小波分析的最大優勢在於可以同時在時頻兩方面實現局部化分析。
  13. Meanwhile, the telephone gateway in tetra system is introduced. in further research, the principle of tetra speech coding algorithm ? algebraic codebook excitation linear prediction ( acelp ) is introduced and analysed in detail, which is a advanced codebook excitation linear prediction ( celp ). acelp algorithm replaces the excitation signals with algebraic codebook and uses some technique such as minimizing the mean square error ( mse ) and the analysis - synthesis method to obtain characteristic parameters of speech

    同時，介紹tetra系統的市話網關，並在接下來的研究中詳細介紹tetra電話網關中應用到的語音編解碼演算法? ?代數碼本激勵線性預測碼（ acelp ）的基本原理，它是一種簡化了的碼本激勵線性預測碼（ celp ） ，它把激勵信號用代數碼本代替，並且運用了均方誤差最小、分析?合成等技術提取出語音的特徵參數，極大地降低了比特率，而且具有較好的重建語音質量。
  14. Based on that, two key questions are proposed : one of them is constructing the visual speech representation model, the other, audio / visual mapping model

    在基於對可視語音合成問題分析的基礎上，提出了可視語音合成系統研究方法中首先要解決的2個問題：視覺語音特徵模型的構建和聲視頻映射模型的構建。
  15. And the paper expatiates on the further research on the optimum algorithm for solving the problem of noise disturbance, and experiments to examine its effect to remove noise. thus it discusses how to select the width of numerical value filter on the basis of a compromise in pitch detection and reconstructed speech quality

    最後本文得出結論認為選擇數值濾波演算法分析窗的寬度需要綜合優化演算法中數值濾波對基音檢測和合成語音質量這兩方面不同的影響，而不單單只是考慮它對其中一方面的影響。
  16. In this paper, the author studies computer telephone integration and call center technology at first, analyzes the unicall integration platform, and then presents a detailed voice query system design and implementation solution which adopt the pc and voice card on the unicall platform

    論文首先研究了計算機電話集成技術cti和其上的呼叫中心，分析了unicall綜合平臺，然後在該綜合平臺上採用pc語音板卡的技術方案實現了語音查詢系統。
  17. This paper describes a method of realizing voice application system of atc ( air traffic control ) training simulator. through study on speech recognition and text - to - speech, this system introduces ibm ' s newly techniques - viavoice speech recognition and text - to - speech system, makes an organic combination between voice application system and atc training system, realizes automatic voice control of atc training simulator, which can be programmable and self - educated

    本文通過對語音識別技術和語音合成技術的研究，在引進ibmviavoice語音識別與語音合成系統的基礎上，根據空管模擬訓練機的功能需求分析，將語音識別技術與管制訓練控制系統有機結合，實現了計算機系統自動對非特定管制學員、連續管制指令語言的識別和模擬模擬雷達顯示控制。
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