語音功率 的英文怎麼說

中文拼音 [yǔyīngōnglǜ]
語音功率 英文
phonetic speech power
  • 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
  • 功 : 名詞1 （功勞） exploit; merit; meritorious service [deed]: 戰功 military exploits; 立功 render me...
  • 率 : 率名詞（比值） rate; ratio; proportion
  • 語音 : speech sounds; pronunciation; voice
  1. 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。
  2. The system cybernation method is distributed control system. in this way, the audio process computer switch the signal between input and output channels. the main function of audio process computer is general broadcast, service broadcast and emergency broadcast. there is a special control computer. it " s main function is supervise - control the system status such as the amplifier and speaker circuit

    在控制上，採用了分散式控制方式，由語音計算機進行音源輸入和輸出的切換，執行業務性廣播、服務性廣播和火災事故緊急廣播；由控制計算機進行系統的監控，監視功率放大器和揚聲器迴路的工作狀態，並根據當前的設備狀態執行相應的控制任務。
  3. In this paper, on the basis of absorption of achievements of the research on auditory physiology, an auditory model simulationg the peripheral auditory system and part of the central auditory system is set up. the model is made of the fitlters presenting the characteristics of the basilar membrane for analyzing the voice signals, the half wave rectification modeling the inner hair cells and energy transfer of nerve fiber

    在吸收聽覺生理學研究成果基礎上，建立了一個模擬外圍聽覺系統和部分中樞聖經系統功能的聽覺模型。模型由表徵基底膜的頻率分析的帶通濾波器組、內毛細胞的半波整流特性和神經纖維的能量轉換特性組成，該模型可以作為前端處理來提取語音信號的自相關圖譜。
  4. 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。
  5. For the real time performance need of the low speed speech compress algorithm and the asic implement of the transfer process between programs, the design is put forward in the paper, in which state registers control the cross access between operator and memory, register windows are used for the parameters transfer, and the technique of hardware controlling is used to avoid pipeline conflict, so that the main problems of the transfer process in tr600 are solved effectively

    摘要針對低速率語音壓縮演算法對處理器系統實時處理復雜運算的性能要求，就程序調用過程的asic實現問題進行了對比與分析，進而提出了用層次狀態寄存器控制存取運算元對存儲體交叉訪問的方法，並結合運用寄存器窗口傳遞參數的功能，以及利用空指令硬布線處理流水線沖突的方法，有效地解決了tr600晶元中調用過程存在的主要問題。
  6. 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。
  7. The characteristic and key technologies of the system are as follows : ( 1 ) in realizing the live broadcast of audio and video, the problem of immense multimedia data and low networks bandwidth utilization ratio is solved by using mpeg - 4 as format of audio and video data. audio and video data are collected by video card cv500 which developed by beijing sum tone company ; meanwhile, the contradictory between the delay of networks transmitting and the quality of the image is well solved by setting a " bi - buffer area "

    系統實現中解決的關鍵問題和特色主要有以下幾個方面： （ 1 ）在視音頻直播功能的實現中，通過使用北京算通公司的cv500視頻採集卡和cv500sdk進行視音頻數據採集，並採用當今最新的圖像和語音編碼壓縮標準mpeg - 4作為視音頻數據的採集格式，既保證了圖像的質量，又大大縮減了視音頻所佔的帶寬，從而解決了多媒體數據量大、網路帶寬利用率低的問題；同時，通過設置環形緩沖區的辦法來調和網路傳輸延時與圖像質量之間的矛盾，取得了較好的效果。
  8. Depending on the specific application, the enhancement system may be directed at different objectives. what is contained in the thesis is as following : ( 1 ). a variety of methods based on short - time spectrum estimation for speech enhancement are discussed

    系統地研究了基於語音短時譜估計的各種增強方法，包括幅度譜相減法、功率譜相減法、維納濾波法、最小均方誤差法、兩態軟判決等。
  9. The birds correctly identified the english recording 75 percent of the time

    研究發現，文鳥成功分辨出英語錄音的概率為75 。
  10. A major improvement in recognition accuracy has been achieved in dtw algorithm compared to classical methods, eventually a software for speaker - dependent and isolated - word speech recognition is worked out

    其中識別演算法部分本文對經典的dtw進行了改進，成功地提高了識別率。最終，根據所選參數和識別演算法編制了一個特定人、小詞匯量、孤立詞語音識別軟體。
  11. The paper makes great efforts on the software optimization of evrc vocoder. based on the understanding of tms320c64xx cpu structure, we do deeply - optimization on the loop which appear usually in voice signal processing, and this improve the utility ratio of cpu and the parallelity degree of cpu function cell. at the same time, we utilize the bit - exact test to test the fixed - point evrc vocoder with the test vectors of tia / eia / is - 718, which improve the robustness of the vocoder

    本文圍繞定點evrc聲碼器的軟體優化，做了大量的工作，在充分理解tms320c64xxcpu結構的基礎上，針對語音信號處理中大量出現的循環運算進行了深度優化，大大提高了cpu的利用率以及cpu功能單元的并行程度，同時，我們還用tia / eia / is - 718的測試向量對定點evrc聲碼器進行了嚴格比特對準測試，提高了聲碼器的魯棒性。
  12. Bluetooth is a short - range wireless data and voice communications technology, is the ieee802. 15 agreement as a low - power, low - data - rate, low - cost technologies, which is very suitable for home automation, security system and lo5a8w - data - rate transmission between low - cost equipment, bluetooth technology is in line with the development of information appliances optimization technology is well - suited to the requirements of domestic intelligence, will in the future smart home were widely available

    藍牙技術是一種短程無線數據與語音通信技術，屬于ieee802 . 15協議，作為一種低功耗、低數據速率、低成本的技術，非常適合於家庭自動化、安全保障系統及進行低數據率傳輸的低成本設備之間，藍牙技術是很符合信息家電發展的優選技術，很適合家居智能化的要求，必將在未來的智能家居中獲得廣泛應用
  13. Sound system equipment - objective rating of speech intelligibility by speech transmission index

    音響系統設備.用語音傳輸系數測定語音清晰度的目標額定功率
  14. Sound system equipment - part 16 : objective rating of speech intelligibility by speech transmission index

    聲系統設備.第16部分:用語音傳輸系數測定語音清晰度的目標額定功率
  15. Hardware platform includes data communiacation, voice processing and the system of fpga. rs232 and epp mode are used in tte part of the data communiacation. voice signal collection and magnify, ad / da conversion and power magnify circuits are integrated in the part of the voice processing

    其中軟體平臺包括多媒體文件的讀寫、存儲與界面的設計;硬體平臺包括數據通信、語音處理、 fpga系統三個部分，數據通信部分使用rs232和epp通信模式，語音處理部分集成了語音信號採集放大、數模轉換與功率放大電路等。
  16. The human ear is very sensitive to the speech in the frequency band from 2khz to 4khz, but the speech power spectrum in this frequency band is very small

    人耳對2khz ~ 4khz頻段的語音非常敏感，但是此頻段語音的功率譜本身比較小，因而設計出的減參數將比較大。
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