speech signal 中文意思是什麼

speech signal 解釋
語言信號
  • speech : n. 1. 言語；說話；談話；說話能力（或方式）。2. 民族語言，方言，專門語言；〈罕用語〉流言。3. 演說，演講；發言。4. 【語言學】詞（類）；引語；用語。5. （樂器的）音，音色。
  • signal : n 1 信號，暗號；信號器。2 動機，導火線 （for）。3 預兆，徵象。adj 1 暗號的，作信號用的。2 顯著的...
  1. Because of the quickness in calculating and briefness in algorithm, mm will be a preferable approach to nonlinear speech signal processing

    由於數學形態學并行快速、易於硬體實現，很可能成為一種新的非線性語音處理方法。
  2. The hardware of the ip phone codec to be designed is based on the fixed point digital signal processor ( ti ' s tms320vc5410 ) while the compression and decompression core in the software of dsp is based on the itu - t vg. 729a. ip phone codec carryout the task of collecting / playing - back. coding / decoding of speech signal and communication with embedded cpu. etc

    該語音編解碼器的硬體基於tms320vc5410 ，編解碼演算法遵循itu - tg . 729a協議，能夠實現語音信號的採集/回放、編碼/解碼以及同嵌入式cpu通信等功能，在8kbit / s的碼率下能夠提供獲得良好的語音質量。
  3. The results of simulation indicated speech signal processed by the optimum algorithm presents obvious periodicity in time domain, and effect of the formant is removed or restrained effectively in frequency domain

    處理后的語音信號在時域上表現出明顯的周期性特徵，同時在頻域上也觀察到聲道的共振峰結構影響得到消除或有效的抑制。
  4. Moreover in speech enhancement, especially in reducing the pulse noise, morphological algorithm has its unique advantage. particularly morphological filter may maintain the preferable accurate of the speech signal in speech waveform, and which produces little impairment to the formant of speech. so the spectrum structure of the speech is retained well, and the quality of the speech will not be reduced

    特別是，在時域波形分析中，形態學濾波增強較小波去噪更好地保持語音信號的細節；在頻域分析中，形態學濾波對語音信號的基音頻率、頻譜斜率、共振峰等語音特徵的影響很小，因而能夠較好的保留語音信號的頻譜結構，使語音品質不致降低。
  5. This thesis tries to update the cmdsr system to achieve the characters below : real - time, better robust, higher recognition rate, non - special - man. considering the disadvantages of traditional improved spectrum subtraction speech enhancement, this thesis proposes the theory of fuzzy spectrum subtraction based on the fuzzy theory and improved spectrum subtraction speech enhancement ; as for the difficulties of detecting the endpoint of speech signal, the thesis gives the table of initial and the improved parameters, with which we can confirm the endpoints of mandarin digit speech ; the thesis puts forward two - level digit real - time speech recognition system, the first level is based on discrete hidden markov model which is linear predictive coding cepstrum ( lpcc ) and difference linear predictive coding cepstrum ( dlpcc ), the second level is based on formant parameters ; as for the realization of hardware, the thesis depicts the realization of every part of cmdsr based on the tms320vc5402 in detail ; as for the development of software, the thesis gives the software design flow chart of cmdsr, simulates the basic theory with matlab language and gives the simulation results

    針對傳統的「改進譜相減法語音增強」參數設定單一、環境適應能力差的缺點，提出了一種利用模糊理論和「改進的譜相減法」結合的「模糊譜相減法語音增強」 ;針對語音信號端點檢測困難的特點，通過matlab模擬試驗，給出了能夠準確確定數碼語音端點的初始和改進參數表;提出了利用基於線性預測編碼倒譜參數和差分線性預測編碼倒譜參數相結合的離散隱含馬爾可夫模型進行第一級識別、利用共振峰參數進行第二級識別的兩級漢語數碼語音識別系統，在保證系統實時性的同時，實現連接漢語數碼語音識別系統識別率的提高;在硬體實現上，詳細闡述了基於tms320vc5402的連接漢語數碼語音識別系統各部分硬體設計;在軟體開發上，給出了連接漢語數碼語音識別的軟體設計各部分的流程圖，並對各部分進行了matlab模擬，並給出了模擬結果。
  6. Based on the speech produce model, we find the reason of periodicity disappearance and the extremum number increase by analysing the character of speech signal when the glottal closes

    於是從語音產生模型入手，詳細的分析了聲門閉合時刻語音信號的性質，找到了濁音信號經過小波變換後周期性消失、極值點個數增多的原因。
  7. If the wavelet transform is directly implemented in pitch detection, comparing the glottal closure singularity of speech signal with image grey break, we will not obtain the anticipative result

    將聲門閉合在語音信號中表現出相應的奇異性，與圖像邊緣的灰階突變進行等價對比，直接將小波變換用於聲門閉合奇異型的檢測，並不會得到預期效果。
  8. Speech signal is the most convenient and shortcut way of intercommunion

    語音對話是人們相互通訊和交流最方便快捷的手段。
  9. Because the speech signal is periodicity at sonant which vocal cords surge in low frequency and similarity to white noises at surd, the pitch can be detected in traditional way through the correlation operation without the speech produce model

    在人類語音的濁音段，聲帶發生較低頻率的振蕩，語音信號呈明顯的準周期性，而在清音段，語音信號則類似於白噪聲。
  10. Experimental result shows that for sonant part of speech signal, 3 ~ 5 common ridges is enough to describe the main characteristics. signal compression is achieved by choosing proper way to represent the ridge information and use it to reconstruct the original signal

    在信號重建過程中，選擇合適的方法用少量數據來描述起關鍵作用的參數，並用這些參數來重建信號，可以達到信號壓縮的目的。
  11. After about two years " insisting and hard working, this goal set at the beginning has become true. the developed c54x general assembly program for g. 729 speech signal compressing algorithm has passed the tracking with more than 3, 000 unitary standard measuring vectors. g. 729 speech signal compressing compiler using c54x general assembly program has been accomplished real - timely, and undistorted rebuilt speech signals have been obtained

    因此本課題選用c54x的通用匯編語言編程實現g . 729語音壓縮編碼演算法，調試並通過了統一標準測試矢量三千多幀，最終在5402開發實驗板上實時實現了g . 729語音壓縮編碼器，獲得未失真的重建語音信號。
  12. Annex b introduce a voice activity decision ( vad ) algorithm which class speech signal as voice signal and background noise signal

    Annexb提出了一種靜音壓縮演算法（ vad ） ，它將語音信號分為話音信號和背景噪聲信號。
  13. It is conceived to introduce barker codes as synchronization preamble and add synchronization signal in front of speech signal to implement speech detection. the principle of this idea is presented

    提出了引入barker碼作為同步碼，在語音信號前添加同步信號以實現起點檢測的工作設想，並介紹了其同步原理。
  14. As for the feature of mandarin digit speech, the existing arithmetic is cited to design the software system, and the design process is described in the part. here, the shore - time ^ relative efp ( energy - frequency - product ) is used to make the capsheaf of chinese speech signal, and the short - time relative efq ( energy - frequency - quotient ) is used to separate its syllable and consonant - vowel segment, and it improves the correct rate

    本文採用的漢語語音的端點信號的檢測和清濁音信號切分方法是：短時相對能頻積的方法對漢語語音信號的端點進行檢測；短時相對能頻比的方法對語音信號的清濁音進行切分，提高漢語語音信號切分的成功率。
  15. The characteristics of speech signal have caused the difficulty of speech recognition, these characteristics include changefulness, dynamism, instantance and continuity etc. the process of computer recognition for speech and the identification process for speech by a person are basically consistent

    語音信號本身的特點造成了語音識別的困難，這些特點包括多變性、動態性、瞬時性和連續性等。計算機對語音識別的過程和人對語音的識別過程基本上一致。
  16. Digital speech has preponderance over analog speech in reliability, robustness and security during communication. however, digital speech needs more bandwidth than the analog signal. especially with the requirement for communication frequency increasing, it ' s necessary to code speech signal at low rates

    但是，數字化后的信號所佔的頻帶大幅增加，特別是在帶寬需求日益增長的今天，這個問題尤為突出，因此語音的低速率編碼（即壓縮編碼）成為迫切的要求。
  17. Then this spectral subtraction method is applied to noise speech recognition system as the front - end processing. noise speech signal are processed to improve its snr before recognition. so the recognition rate can be improved in noise environments

    並將改進譜減演算法作為噪聲下語音識別系統的前端處理過程，即通過對含噪的語音進行語音增強以提高信號的信噪比，從而提高語音識別系統的抗噪聲性能。
  18. At the receiving end, a inverse process is performed. the system receives low rate data and the fpga reorganizes a frame of data which is decoded by the compression chip every 20 ms. the obtained pcm signal is performed d / a to restore the analog speech signal

    在收端進行相反的過程，接收低碼率數據，並由fpga重新組幀，送至主晶元解碼得到pcm信號，再作d / a變換，恢復出模擬語音，系統是全雙工的。
  19. This dissertation is different from traditional speech enhancement methods which are based on noise characteristic such as adaptive noise cancellation or spectral subtraction processing. in this dissertation the speech signal conducted by bone was taken as the object to be studied and the exploitive study on the acoustical characteristic of speech signal conducted by bone was performed by the method of theory combined with experiment. then a proposition about speech reconstruction based on speech signal conducted by bone was presented, and the design of software and hardware was completed

    本文與傳統的基於噪聲特性的自適應噪聲抵消法、頻譜減法等語音增強降噪技術不同，是以骨導語言為研究對象，採用理論與實驗相結合的方法對骨導信號的聲學特性進行了探索性研究，進而提出了基於骨導信號的語音重構技術，並完成了相應的軟硬體開發。
  20. The statistic of wavelet transform coefficient algorithm can solve the periodic noise, high - energy noise and some non - gauss noise simply and effectively ; bi - spectrum can acquire more information from the original signal than power - spectrum, detect more information except from range and restrain the gauss noise. short - time speech signal can be considered as stationary and with periodic non - gauss signal, so we can make use of bi - spectrum to obtain the speech character and separate the speech and noise and detect morse telegraph signal ; complex number spectrum variance algorithm is put forward based on the deeply observing speech data, it is a new algorithm, experiment show that it is simple, effective

    統計演算法在解決周期信號、高能噪聲和高斯信號方面有獨特之處，能簡單有效提取以上噪聲的特徵;雙譜能夠提供比功率譜更多的有用信息，有效地檢測信號幅度之外的其它信息，並能有效抑制高斯噪聲，短時語音信號一般認為是平穩且有一定的周期性的非高斯信號，因而可以利用雙譜來提取語音信號特性並實現信噪分離;復數譜方差演算法是在對語音信號進行深入觀察和分析的基礎上而提出來的一種全新的語音特徵提取方法，此方法簡單而有效的提取了語音、噪聲的特徵以及檢測莫爾斯信號，基於實驗表明，該演算法取得了很好的效果。
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