speech quality 中文意思是什麼

speech quality 解釋
話音質量
  • 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。
  • quality : n 1 質，質量；性質，特質；品質，品位。2 優質，美質，優點。3 才能，能力，技能，素養。4 品種。5 身...
  1. So acoustic echo cancellation ( aec ) technique has become a hotspot of competition in famous communicaton company all over the world. however, it is not always easy to implement acoustic equipments for communication system with satisfactory speech quality

    回波消除技術能有效地解決長距離電話網路、 ip電話、免提電話和視頻會議等通信系統中的回波問題，很好地改善了語音通信質量，具有廣闊的市場前景。
  2. The speech processing module can be divided into two parts, the first part includes speech compression, voice activity detection and echo cancellation module, which improve speech quality ; the other part includes dtmf and cpt module, which generate and detect some necessary telephony signal in the communication. this thesis is organized as follows

    該語音處理模塊由兩部分組成，一方面是對語音的處理，包括語音壓縮模塊、靜音處理模塊和回聲消除模塊，主要為了提高voip的語音質量;另一方面是對電話通信的控制和處理，包括雙音多頻模塊和呼叫進程音模塊，主要為了產生和檢測ip電話通信中一些必須的電話信號。
  3. The adaptation processing includes linear prediction coefficient adaptation and adaptation of quantization step size for residual signals. based on g. 726, we adopt a huffman coder to make use of probability statistic of bit cascade covering n ( n 1 ) samples generated from adpcm, in order to further reduce the bit rate. ng is lossless entropy coding, the speech quality of our improved algorithm should be same as that of g. 726 standard

    我們的研究和改進工作包括:研究最優非均勻自適應量化器，及其自適應演算法;研究波形預測函數，以及函數零點、極點的自適應演算法;基於每n （ n 1 ）個樣本所對應符號的概率統計，對預測殘差量化值再進行huffman編碼，進一步降低比特率。
  4. To improve the speech quality, it is necessary to adopt such speech processing techniques like speech compression, voice activity detection, echo cancellation, jitter buffer and so on

    為了提高語音質量，需要採取一系列的語音處理技術，主要包括語音壓縮技術、靜音處理技術、回聲消除技術、抖動緩沖技術等。
  5. The bandwidth is a principal factor that makes the speech quality bad. it is also the most difficult problem to be resolved for the internet

    帶寬是影響語音質量的一個主要因素，也是因特網當前最難解決的問題，一個解決的辦法就是採用高效的語音壓縮演算法。
  6. Mos method of speech quality assessment for military communication systems

    軍用通信系統音質的mso評價法
  7. Hands - free speech communication is indispensable in audio and video conference systems, hot - line telephones and videophones, mobile radio terminals, digital isdn network etc. however, the control ( cancellation ) of the acoustic echo has always had a strong impact on the transmission quality in hands - free telecommunication ~ [ 1 - 3 ] conventional methods of acoustic echo control ( cancellation ), such as echo suppression or gain control, may lead to the degradations in speech quality or make the speakers feel uncomfortable

    免提式話音通信在移動電話、熱線電話、車載電話及isdn網的電視電話會議等多種領域正得到日益廣泛的應用。但至今的免提式話音通信中，仍免不了受回聲引起的話音失真、甚至嘯叫等干擾，大大降低了免提話音通信的質量。由於聲回波對消問題尚未得到圓滿解決，實現高質量的免提式話音通信仍是一個極賦挑戰性的課題。
  8. Good speech quality is found to have high clarity and naturalness

    系統合成語音的質量較好，具有較高的清晰度和自然度。
  9. 4. aimed at the reduction of bit rate and the improvement of speech quality, a serial of speech coding schemes are studied in a gradual refinement way, and an integrated coding scheme at 1. 5 - 2. 4kbps is presented finally

    圍繞編碼位率的降低和語音質量的提高，以逐步求精層層遞進的方式研究了一系列壓縮編碼方案，並最終提出一個位率在1 . 5 2 . 4kbps的綜合編碼方案。
  10. By utilizing the semi - markov model and the network flow model based on rtp, we can ensure the right operations be done in the correct way. finally, the subjective evaluating method mos ( mean opinion score ) and the objective evaluating method dtw ( dynamic time warping ) has been combined to estimate the speech quality

    在控管結果的評測方面，本文引入了主客觀評測相融合的話音質量評測方法，通過將改進的mos主觀話音評測方法和基於動態時間規正（ dynamictimewarping ， dtw ）的客觀話音評測方法相結合，給出了較為準確的評測結果。
  11. We made an improvement in overcoming the defects in speech signal adaptive delta modulation ( abbr. adm ), such as slope overloading and grain noise. in this method, numerical sliding average filtering was used for filtering decoding speech signal. experiments and analyses indicate that the method makes waveforms in good agreement between the decoding of adm and the original pulse coding modulation ( abbr. pcm ) signal, and considerably improves, the playback speech quality in naturalness, legibility and under standability

    針對語音信號自適應增量調制（ adm ）方式中斜率過載和顆粒噪聲缺點，提出了一種改進方法，它利用滑動平均方法對解碼后的信號進行數字濾波.試驗和分析表明，該方法使解碼后的信號波形與原脈沖編碼調制（ pcm ）波形具有很好的一致性，使再生語音質量在自然度、清晰度和可懂度方面比改進前均有較大提高
  12. The informal subjective test in real - time processing indicates that the synthetic speech quality of amr vocoder is better than it of rpe - ltp in gsm system, achieving toll quality, that can apply to devices employing the amr vocoder within the 3gpp system in terms of software and hardware

    實時處理的非正式主觀測試表明，合成語音質量優于gsm的rpe - ltp的語音質量，達到長途語音質量，完全可以實際應用，為第三代移動通信中聲碼器設備的研製奠定了良好的軟體和硬體基礎。
  13. Cs - acelp speech coder use a special codebook structure to simplify the search procedure of codebook, has the advantage of low bit - rate, low complexity, and high speech quality

    它採用了特殊的碼本結構，簡化了碼本的搜索過程，具有低延遲，低計算復雜度和高語音質量的優點。
  14. Informal subjective a / b listening tests indicated that the reconstructed speech quality of the improved wi codec with unquantized parameters is a little better than that of itu g. 726 32kbps adpcm coder

    非正式的a b測試表明，改進的wi演算法的模型質量好於itu32kbpsg . 726標準。
  15. The speech quality produced by g. 729a is equivalent to that of 32kbits / s adpcm for most operating conditions, achieving the request of long - distance telephone

    G . 729a具有非常好的性能，其語音質量與32kbit / s自適應差分脈沖編碼調制（ adpcm ）演算法相當，達到了長途電話質量要求。
  16. To meet the needs of the third mobile communication system for higher speech quality and greater system capacity, the cdma systems employ variable rate speech coding technologies, which select the encode rate dynamically according to the signal energy and background noise. this method insures the speech quality as well as reduces, the average encode rate, so the system capacity is increased

    為了滿足第三代移動通信系統對高質量語音和大容量系統的需要，在cdma系統中普遍採用了變速率語音編碼技術，根據話音的信號能量和背景噪聲動態地決定編碼速率，這種方法既保證了話音的質量，又降低了平均編碼速率，從而增加了系統容量。
  17. Now, the standards of speech compression coding provide a way of transporting speech signals efficiently. in fact, all of them are to reduce the baud rate of data under definite speech quality

    相繼出現的語音信號壓縮標準為語音信號的高效傳輸提供了一種有效方法，其實質就是在相當的語音質量指標下，降低數字化語音的數碼率。
  18. H. 323 is an itu standard which provides several services including speech, data and multimedia etc. being a speech compression coder protocol supported by h. 323, g. 729 has the advantages of low bit - rate and high speech quality and been selected by itu - t as 8kb / s standard

    G . 729做為h . 323支持的語音壓縮編碼協議，具有低延遲，高語音質量的優點，被itu確定為8kb / s語音編碼標準。
  19. Simulation methodology for speech quality of voip in mobile communication networks

    電話語音質量的模擬方法
  20. Vocoders based on the mbe model are popular due to their good synthesis speech quality

    基於多帶激勵模型的語音壓縮編碼演算法以其較好的合成語音質量而受到廣泛的重視。
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