有用聲音信號 的英文怎麼說

中文拼音 [yǒuyòngshēngyīnxìnháo]
有用聲音信號 英文
wanted sound signal
  • 有 : 有副詞[書面語] （表示整數之外再加零數）: 30 有 5 thirty-five; 10 有 5年 fifteen years
  • 用 : Ⅰ動詞1 （使用） use; employ; apply 2 （多用於否定: 需要） need 3 （敬辭: 吃; 喝） eat; drink Ⅱ名...
  • 音 : 名詞1. （聲音） sound 2. （消息） news; tidings 3. [物理學] （音質） tone 4. （姓氏） a surname
  • 號 : 號Ⅰ名1 （名稱） name 2 （別號; 字） assumed name; alternative name3 （商店） business house 4 （...
  • 有用 : useful; serviceable
  1. In actual fact, a suitably modified amateur radio transmitter operating in either the 420 to 450 megahertz band or the 1. 3 gigahertz band, with a highly directional antenna, is capable of transmitting voice to skull signals at less cost than the price of an automobile

    事實上，一個適當限制在420到450兆赫波段或1 . 3千兆赫波段的帶有高度定向天線的業余無線電廣播發射機，能夠把聲音信號傳給頭骨並且所花的費用比一輛汽車的價格還要少。
  2. In accordance with chaotic essence of speech signals, syllable segmentation in continuous speech is researched by fractal theory. an approach of syllable segmentation using variance fractal dimension is proposed, its performance is analyzed in detail. the method can discriminate between voiced and unvoiced, between surd and sonant, but it can hardly discriminate between sonant

    本文根據語音信號的混沌本質，利用分形理論研究了漢語連續語音中的音節分割問題，提出了基於方差分形維數的音節分割方法，並詳細分析了該方法的性能，它能很好地解決了無聲與有聲、濁音與清音間的分割問題，但很難解決濁音間的分割問題，當濁音相連且過渡段較短時，該方法無法實現它們之間的分割。
  3. Theoretical expatiate on general concepts and fundamental principles of information hiding and steganography, also point out possible directions for further research, also analysis the probability of speech as the host carry signal and efficient masking characteristics of psycho - acoustic model, it is shown that : there is an improvement on imperceptibility according to human auditory masking effect

    闡述了信息隱藏技術和隱寫技術的重要概念、基本理論以及廣闊的應用前景。分析了將語音信號作為宿主載體信號的可行性，參考心理聲學模型的特性，得出結論:基於人耳聽覺掩蔽效應的隱寫演算法，在隱蔽性上有很大的提高。
  4. Because the adaptive algorithm of conventional adaptive noise canceller is the least mean squares ( lms ), and the convergence rate of lms is heavily dependent on the eigenvalue distribution of the autocorrelation matrix of the input signal, thus lms converges at unacceptably low rates when the input signal is colored noise or speech

    由於傳統自適應噪聲抵消系統（ anc ）自適應演算法主要採用lms演算法，而lms演算法收斂速度依賴于輸入信號自相關矩陣特徵值的分散程度。因此，當輸入信號是語音或有色噪聲時， lms的收斂速度很慢。
  5. The precise clock source is crystal oscillator made of 74hc04 ; the mute circuit can conceal the error and solve the problem of noise ; the antenna switching circuit in the receiver is to select one antenna from two which receives signal better. it can improve the quality of the receiving audio signal, restrain the noise effectively and promote the system performance

    高精度的時鐘源是由74hc04構成的晶體振蕩器;靜音電路將出錯的音頻信號進行差錯掩蓋，很好地解決了噪聲問題;接收機採用兩副天線切換工作，提高了音頻信號接收質量，有效地抑制干擾，提升了系統的性能。
  6. Once has the bandits and thieves to intrude guards against the place, the detector launches the wireless coded signal immediately, the networking center number which installs when is apart from defense area 150 meter within the main engine to send out the police whistle sound to report to the police immediately, reports to the police dials to establish in advance or reports to the police the telephone, the handset number, answers in the police telephone to return puts user pre - record to report to the police the pronunciation, long - distance reports to the police, simultaneously comes the real - time transmission through the internet to deploy troops for defense, to withdraw from a defended position, to report to the police and so on the condition, inquires the historic record through the computer network

    該系統還採用美國進口原裝晶元與先進的無線數字高頻技術微電腦cpu控制器主機組成。在防範地點安裝好主機后，並設置在布防狀態。一旦有盜賊闖入防範地點，探測器立刻發射無線編碼信號，安裝在距防區150米以內的主機立即發出警笛聲報警，報警時撥打預先設定的聯網中心號碼或報警電話手機號碼，接警電話里回放用戶預錄的報警語音，遠程報警，同時通過網際網路來實時傳遞布防撤防報警等狀態，通過電腦網路來查詢歷史記錄。
  7. 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。
  8. The traditional detection algorithm, based on zero - crossing or energy, will not acquire ideal effect when the signal - to - noise is low or the signal is weaker. therefore, to resolve the real problem in the real environment that all kinds of random noise and speech signal exit together, some new algorithm must be put forward. account for the complexity of real noise, we integrate the wavelet transform and high - order statistics and advance a new algorithm ; the algorithm can effectively separate the speech signal and the non - gauss noise

    基於過零率和能量的傳統檢測演算法，在噪聲環境比較復雜的情況下效果很不穩定，尤其是信噪比較低或者語音信號較弱時，檢測效果很不理想，因此，在多種語言和噪聲隨機出現、噪聲和語音強弱不一的實際噪聲環境下，必須利用新的演算法提取有用信號和噪聲信號的有效特徵，才能解決實際的問題。
  9. In the commercial equipment, the audio bandwidth tends to be narrower than our amateur equipment and there are circuits installed to filter out the tones so they are truly subaudible

    在商業設備中，聲音頻寬比業余設備更窄些，並且，配備有專門的電路用於過濾音頻，於是，這些音頻信號真正的成為了「亞音」 ，不會被聽到。
  10. In signal space, speech enhancement is adopted to effectively suppress the noise and increase the discriminative information embedded in noisy speech signal. however, the speech distortion introduced by enhancement, as well as the residual noise, is a very adverse factor for recognition

    在信號空間，利用語音增強有效抑制噪聲，提高輸入信號中的鑒別信息，但增強帶來的語音失真和增強后的剩餘噪聲是對語音識別非常不利的因素。
  11. Cabled distribution systems for television and sound signals - part 3 : active coaxial wideband distribution equipment

    電視和聲音信號用電纜配線系統.第3部分:有效同軸寬頻分配設備
  12. 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
  13. 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。
  14. The hardware capacity for speech in spce061a is used and the function of speech compression is embedded in software

    該系統利用spce061a晶元具有的語音播放的硬體條件，並結合軟體演算法上的語音壓縮函數庫，實現了聲信號的智能化輸出。
  15. In the field of audio recognition, with many mature and creative technologies applying, especially the hidden markov models ( hmm ), the effect and efficient of the audio recognition system have been enhanced. but due to the mismatch between training and testing environment ( such as background, audio transition channel ), the recognition systems based on hmm tends to drastically degrade in performance

    在音頻信號識別特別是語音識別領域內，隨著隱馬爾科夫模型（ hmm ）的應用，使得系統的識別性能有了改進，但是由於訓練和測試環境（背景噪聲、音頻傳輸通道等）的失配常常導致識別性能的嚴重下降。
  16. The simulation result show, this speech enhancement system design method, not only realizes simply, but also saves the running time, the speech enhancement effect is good. thirdly, this article discusses a new method of speech enhance based on neural network. mfcc coefficient of the speech signal can be picked up under noise and

    用於bp神經網路的訓練和學習，利用神經網路系統具有非線性映射和自學習，能夠用於噪聲信號的非線性建模的能力，獲取信號的最佳估計，克服信號處理中存在的不確定性，最終達到語音信號消噪和提高可懂度的目的。
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