語音解碼器 的英文怎麼說

中文拼音 [yǔyīnjiěmǎqì]
語音解碼器 英文
decoder, tone
  • 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
  • 解 : 解動詞（解送） send under guard
  • 碼 : Ⅰ名詞（表示數目的符號或用具） a sign or object indicating number; code Ⅱ量詞1 （指一件事或一類的...
  • 器 : 名詞1. （器具） implement; utensil; ware 2. （器官） organ 3. （度量; 才能） capacity; talent 4. （姓氏） a surname
  • 語音 : speech sounds; pronunciation; voice
  • 解碼器 : codec
  • 解碼 : decoding; decipher; decode
  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. 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. Real time implementation of g. 723. 1 speech coder based on tm

    1語音編解碼器的實時實現
  4. The testing results show that the vocoder can synthesize high - quality speech when transmission rate is set to be 2. 4kbps

    非正式編解碼測試表明，本文的聲碼器能在2 . 4kbps的編碼速率上合成出較好的語音。
  5. Section ii describes the design approach and implementation of speech module on mcf5249 coldfire core. the speech codec optimizes g. 729a codes and added voice activity detection of g. 729b to save bandwidth ; the implementation of acoustic echo cancellation uses nlms algorithm and it can reduce echo though designing adaptive fir filter and speech detector ; the dtmf and cpt generate signal using two second order digital sinusoidal oscillators and detect signal by picking up the frequency information. but only get the frequency information is not enough in cpt detector, this thesis introduces a method

    其中對語音編解碼器的設計採用優化g . 729a代碼達到設計要求，並在此基礎上加入g . 729b的靜音檢測模塊，以進一步降低網路傳輸帶寬;對回聲消除器的設計採用nlms演算法，通過設計自適應fir濾波器和語音檢測器達到回聲消除目的;對雙音多頻設計，信號發生端採用構造靜態參數表並通過二階正弦振蕩器產生信號，信號檢測端提取頻率信息以檢測信號;對呼叫進程音設計，除了類似雙音多頻的信號發生及頻率檢測設計外，還需要檢測信號持續時間，作者設計了一種基於匹配狀態表的方法以檢測信號持續時間。
  6. The system is consist of the main data processing board which is based onthe fpga device and fast ethernet phyceiver rtl8201l and a - law pcm data encoder and decorder chip msm7702 - 3, and the dial - up and display board which is based on mcu. the main board would carry out the core task of data processing, such as voice data packing and unpacking, the ethernet frame processing, protocol processing, call processing, etc. the dial - up and display board would carry out the task of display the ip address which is input by consumer and status of network during talk period from the main board, and so on. in the paper the system of lan ip telephone and the tcp / ip protocol is introduced firstly, then the fpga device is stated. after that the fpga - based hardware scheme is introduced in detail in chapter four

    系統以altera公司的acex1k系列的fpga和快速以太網控制器rtl8201l和語音編解碼晶元msm7702 - 3為核心構建了數據處理主板和以單片機為控制器的撥號顯示子板組成。數據處理主板的核心任務，包括語音數據處理、以太網幀處理、協議處理、呼叫處理等。撥號顯示子板則完成通話前的顯示用戶所撥過的ip地址，通話期間網路狀態的顯示等等。
  7. Speech coding systems are generally based on narrowband speech at present, but its frequency is restricted in 200hz ~ 3400hz and its sample rate is 8khz. with the development of the wideband speech, its bandwidth which is from 50hz to 7khz causes the quality of speech communication to approach in the feeling of face - to - face conversation, and makes the speech natural, expressive and comfortable. hence, it ’ s quite significant in researching on wideband speech coding system. in recent 20 years, the dsp and its software development kit has improved greatly, but the price has fallen sharply, thus it has more and more widespread applications now

    隨著當今世界的飛速發展，寬帶語音越來越受到人們的青睞，因為它的50hz 7khz的帶寬使得語音通訊質量接近於面對面交流的感覺，大大提高了語音的自然度、表現力和舒適度。因此，開發研製基於寬帶語音的編解碼系統具有十分重要的意義。在過去的短短二十年裡， dsp處理器的性能得到很大的改善，軟體和開發工具也得到相應的發展，價格卻大幅度地下降，從而得到越來越廣泛的應用。
  8. Per channel, and the result from ti ' s software simulator in ccs ( code composer studio ) is given, putting forward the principles and difficulties of realization of this algorithm. the recovered music signal is very close to the original signal and they are difficult to tell apart. in this paper a scheme of real - time implementation for this algorithm is discussed

    本文敘述了mpeglayer音頻壓縮編解碼器的演算法及模擬實現研究，用c語言實現了的高保真音樂信號64kbps每聲道的非實時解碼，並在ti的ccs （ codecomposerstudio ）系統中的軟體模擬器上進行實時研究，提出了該演算法在具體實現中的要點和難點。
  9. The valid approach to solve this problem is low rate, delay, loss and high quality codec

    低速率、高質量和低成本的語音編解碼器成為解決這些問題的有效途徑。
  10. And the main focus is the design, implementation and testing of an adaptive line echo canceller and a vocoder based on multi - band excitation ( mbe ) model

    主要工作是設計並實現電話線路自適應回波抵消器和基於多帶激勵模型的語音編解碼器。
  11. Based on the analysis of current situation and the development trend of ip phone, this paper has put forward the solution of a kind of new - type embedded ip phone terminal and has finished designing the ip phone codec which is the key part of ip phone terminal

    針對第三代ip電話的技術現狀和發展趨勢，本論文提出了一種新型的嵌入式ip電話終端解決方案，並完成了該終端的核心部分，語音編解碼器的設計與實現。
  12. The fourth chapter is about how to realtime implement of audio codec in tm1300. after we optimize the code, we real time implement g. 723. 1 and g. 729a low bitrate voice codec. the fifth chapter discusses another important technology : real time transmit a - v data over ip network. we use c program language implemented rtf and we discusses the performance of rtp

    經優化后，我們實時實現了兩個低速率的語音編解碼器g . 723 . 1和g . 729a 。第五章是本文的另一個重點：多媒體數據的實時傳輸，介紹了我們用c語言實現rtp的方法及具體程序。並且討論了rtp對傳輸性能的影響。
  13. In this paper, we give a detail discussion on the key technology, including software and hardware designing of g. 729a multi - channel speech codec realtime implemention on a simple dsp processor - tms320c6202. in combination with the requirements of a military communication network, the atm adaptation solution of g. 729 coder bit stream is analyzed. a kind of new atm adaptation technology - aal2 is introduced. the analyse and research of aal2 are provided

    本文詳細討論了多路g . 729a語音編解碼器在一片dsp處理器tms320c6202上實時實現的軟硬體設計和關鍵技術。結合某軍事通信網設備的需要，進而對g . 729語音編碼的碼流的atm適配方案進行了分析。提出了用一種新的atm適配技術- - aal2進行適配的方案。
  14. It adapts to the cdma system and achieves multi - rate speech coding and decoding. source and mode control are combines in smv for rate selection, so it improves the flexibility of cdma system, it will allow cdma subscribers to enjoy superior quality while allowing service providers to increase capacity as needed. smv is regarded as a breakthrough technology that provides significant capacity and quality gains on cdma systems, so the researching of smv is of great practical value

    可選模式聲碼器（ smv ? selectablemodevocoder ）是3gpp2最新的用於寬帶擴頻cdma通信系統的變速率語音編碼標準，它實現了語音的多種低速編碼和解碼，在速率選擇上將源控和模式控制相結合，提高了cdma系統的靈活性，可以在保證高質量語音的同時盡可能增加系統的容量，被認為是變速率語音編碼在cdma系統中應用的「突破性」技術，代表了當前語音編碼發展的方向和潮流，因此smv的研究具有很大的價值。
  15. Recognition engine the high speed speech recognition engine can reach the speed of 1. 1x real time, thanks to the high performance speech recognition decoders

    快速語音識別引擎採用高效的語音識別解碼器，構建快速語音識別引擎，識別速度達到1 . 1倍實時。
  16. Sound classification technology high performance sound classification technology is the cornerstone of speech recognition, which can automatically segment the speech, music and noise, mark the recording environment, distinguish different speakers, and apply corresponding acoustic models

    快速語音識別引擎採用高效的語音識別解碼器，構建快速語音識別引擎，識別速度達到1 . 1倍實時。
  17. Subjective testing indicates that the quality of cs - acelp is equivalent to that of the 32kbit / s adaptive differential pulse code modulation ( adpcm ) under error - free conditions and it outperforms g. 726 under error condition. in this paper, standard c is adopted in realization of the algorithm, presents program strategies and steps of algorithm of each module. the coder and decoder is tested by utterances with noise

    用標準c語言模擬實現了該演算法，計算了mos分值，女聲： 4 . 180497 ，男聲： 4 . 199782 ，並在相同的測試語句中加入噪聲進行測試，含噪語句通過該編解碼器，輸出的合成語音用主、客觀評價標準評價，與原始不含噪語音效果差別不大，平均mos分值為：女聲4 . 1375 ，男聲4 . 1668 ，說明該演算法是優秀的編解碼演算法。
  18. The computing complexity of wi decoder is reduced greatly by this method, and the reconstructed speech quality keeps invariable meanwhile

    該方法大大降低了wi解碼器的復雜度，同時保證了合成語音質量沒有變化。
  19. This thesis has put forward ip phone terminal solution of dsp + embedded cpu ". on the basis of this scheme, this text introuduces the ip telephone systematic framework and the selection of chips for the codec

    提出了「 dsp +嵌入式cpu 」的ip語音終端解決方案，並給出了ip電話編解碼器的系統構架和晶元選型。
  20. During the process of voice terminal design, the author firstly presents its hardware structure, including the coding and encoding module of ip, the interface to former access network equipment and etc, and then the author illustrates in detail the software realization of g723. 1 & g711 coder and encoder, then introduces how to realize the application of rtp / rtcp protocol in h. 323 gateway by openh323, including the design of the process and the realization of the data package format and partial code

    在語音終端設計中，首先提出了其硬體構架，包括ip語音編解碼模塊，與原接入網設備的介面等；然後重點討論了g . 723 . 1和g . 711編解碼器的軟體實現。在h . 323網關模塊中，首先討論了ip電話網關的系統架構；接下來介紹了openh323軟體開發包，並詳細介紹了利用openh323實現rtp / rtcp協議在h . 323網關中的應用，包括流程設計，數據包格式和部分代碼實現。
分享友人
分享到 Line

分享到臉書