coding rate 中文意思是什麼
coding rate
解釋
符號化速度-
Different bit rates are allocated to wavelet blocks according to energy the wavelet blocks include. bit rates in wavelet blocks are adaptively adjusted in coding process. by use of bipartition, the entropy of each wavelet block approximates to the target bit rate of one
該演算法首先根據每個小波塊所含能量的多少和到每個小波塊實際編碼所用的比特數，給其分配不同的碼率；然後根據二分法，通過調整各小波塊的量化因子使得各小波塊的熵逼近它的目標比特率。 -
In this paper, combined with currently voice coding technique, espically with the fabulosity development of the mixed voice coding and the increasingly utility of the digital signals processor. we investigated the voice coding technique and discussed emphasizedly the technology of variable rate voice coding technology
本文結合當前語音編碼技術尤其是混合編碼技術的驚人發展及數字信號處理器的日益實用化，研究了語音編碼技術，並重點討論了變速率語音編碼技術。論文簡要介紹語音編碼技術中的波形編碼和聲碼器的主要性能。 -
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。 -
Rlean meat percentage is one of the most important economic traints in pig breeding programs. myostatin is a negative regulator of skeletal muscle growth. null or low activity of myostatin, individual muscle of mutant amimals would show a large and widespread increase in skeletal mass. myostatin null animals have significantly larger diameter or more quantity of fiber skeletal muscle. the phenotype was termed double muscling. in order to probe the relation between myostatin and high lean meat rate and plump - hipped trait, we sythesized the c ' 80 amino acids coding sequence of porcine myostatin and costructed the cloning and expressing vector of it
肌生成抑制素（ myostatin ，即mstn ）是近幾年來（ mcpherrona . c等， 1997 ）發現的骨骼肌生長的負調控因子，它主要在骨骼肌中表達。其活性的喪失，會引起動物肌肉的過度發育，肌纖維直徑變大或肌纖維數增加，表現為雙肌癥狀。肌生成抑制素研究的突破將對豬、肉雞、肉牛等畜禽生產性能的提高具有特別重要的意義。 -
As a combination of ofdm with space - time coding technique, mimo - ofdm becomes more and more important in the future wireless communication systems, the mimo - ofdm system can not only effectively enhance the transmission rate and capacity of the wireless communication system but also greatly mitigate the effections of mufti - path fading and interfere
Mimo - ofdm技術將ofdm與空時編碼技術有機的結合在一起，能夠大幅度的提高無線通信系統的通道容量和傳輸效率，並能有效的抵抗多徑衰落、抑制干擾和噪聲。 -
The first facet is to put forward four - step block matching algorithm which can deduce the time of motion estimation and improve the coding efficiency, based on the traditional motion estimation algorithms. the second facet is to propose a new rate control algorithm, that is average - reaction rate control algorithm, based on the rate control of mpeg2. the new rate control algorithm can achieve rapid and efficient adaptive coding
首先在對傳統的運動估計演算法進行研究和改進的基礎上，提出了四步搜索塊匹配的運動估計演算法，減少了運動估計的時間，提高了編碼效率；其次在分析mpeg2比特率控制的基礎上，提出了一種新的比特率控制演算法? ?平均響應比特率控制演算法，該演算法能夠快速有效的實現自適應編碼。 -
In this thesis, the research focuses on pitch detection techniques of the low - rate wi speech coding. aimed at the problems of voiced - unvoiced error, pitch doubling and halving, accuracy of pitch detection and pitch quantization, a series of pitch detection techniques including pre - processing, pitch detection and pitch quantization were proposed
本文就低速率wi語音編碼中的基音檢測技術進行了深入研究，針對基音檢測中的清濁誤判、基音加倍減半、基音檢測精度及基音量化問題，提出了包括基音檢測前端處理、基音檢測演算法及基音量化的一整套基音檢測技術。 -
Rate control scheme for base layer in scalable video coding
基於可分級編碼基本層的碼率控制方法 -
In the proposed method, the controller takes the buffer length as congestion indication, takes sources quality and bandwidth utility as object function so as to learn on line. as the controller outputs, the coding rate for input traffic sources and the corresponding user percentage are used to adjust the cells " arrival rate to the multiplexer buffer. compared with the previous method where cells " arrival rate is tuned only by the encoding rate and the encoding rates for all input traffic sources are regulated in a body, the proposed method guarantee that the quality of cells are optimal while cell loss rate is minimized, which means quality of service is guaranteed
在該方法中，擁塞控制器以緩沖區大小信元作為擁塞指示，以信源質量和帶寬利用率作為目標函數進行在線學習，控制器輸出包括信源編碼率及其對應的用戶數在全部用戶中所佔的百分比，即根據信源編碼率及對應的用戶百分數調整信源輸入流，從而克服了以往擁塞控制方法中僅僅調整編碼率帶來的對所有信源進行整體調整的缺陷，使控制系統在信元損失率最小情況下確保信源輸入流質量最高，從而有效地利用了網路帶寬。 -
All of these reduce efficiently the coding bit rate
這些方法有效地降低了編碼位率。 -
The simulation results show that the algorithm have not only cut down the average coding time, but also reduced the average coding bit rate
測試結果表明該演算法在減少平均編碼時間的同時，也降低了平均編碼比特率。 -
The current researches include how to cut down the computation complexity, how to reduce the average coding bit rate, how to improve the quality of reconstructed image, and which algorithm to be suitable to vlsi implementation
在該技術中，減少運算復雜度、降低平均編碼比特率、提高恢復圖像的質量和便於硬體實現等方面是當前研究的主要方向。 -
Experimental results show that the proposed method can reduce the computational complexity significantly compared with the full search method, and maintain similar decode visual quality and coding bit rate, then improve the ray - space data coding efficiency
實驗結果表明，與全搜索方法相比，該方法計算復雜度明顯降低，同時保持了近似的解碼圖像質量和編碼碼率，進一步提高光線空間數據的編碼效率。 -
In the past few years, there has been significant interest in digital video applications like video conferencing and video e - mailing over internet, video telephony over public switched telephone networks ( pstn ) and wireless networks, etc. by removing spatio - temporal redundancies existing in adjacent frames, motion estimation can reduce the coding bit - rate significantly
低速率視頻編碼技術已在許多領域得到了應用，如在pstn （公共電話網） 、無線網、 n - isdn （窄帶綜合業務數字網） 、 b - isdn （寬帶綜合業務數字網）上傳輸視頻會議、可視電話和無線手持終端等。 -
The project uses for reference the algorithm thought of sbc ( subband coding ) to measure off the audio to the corresponding frequency width and encode it by the different sensitivity of human hearing, which results in the lower coding rate and bearable voice quality. the algorithm processing low bit - rate audio is designed to be self - adaptive by the situation of network. the component developped by that algorithm and project has already been used in the realtime interactive educational system
該方案借鑒sbc （ subbandcoding ）子帶編碼演算法思想，將音頻按對人聽覺敏感程度不同劃分為相應的頻帶並進行相應的編碼，從而得到較低的編碼率和較好的語音質量，設計了可根據網路狀況進行自適應的低帶寬音頻處理演算法。 -
On the basis of designing the serial structure of mq encoder, parallel structure of mq encoder is designed using pipelining technique and the coding rate is approximately 1bit / cycle
為了得到更高速率的mq編碼器，採用流水線結構設計了并行的mq編碼器。模擬結果表明mq編碼器的編碼吞吐量明顯提高，達到了硬體規模和編碼效率的平衡。 -
Subsequently, taking into consideration the characteristics of audio data over internet including delay, jitter, packet loss and etc., we propose a series of methods for solving this above problems, such as pre - storage technology, buffer technology, dynamic adjustment of the voice - coding rate to the state of network and integrated media synchronization playing mechanism, and etc. in the end, simulation on 10 / 100m lan is made using the above methods, and the result of the experiment demonstrates the method has good performance and can improve the quality of the audio data transmission
其次本文還深入研究了語音數據在非實時的internet數據網上的傳輸特性，這些特性包括延時、延時抖動、數據包丟失等。在本文的設計方案中提出了針對這些問題的解決方法，包括預取機制、設置緩沖區技術、動態速率調節技術以及媒體綜合同步播放機制等。最後採用這些方法在10 100m局域網上做了模擬實驗，實驗結果表明本文提出的方法是有效的，在網路狀況惡劣的情況下能夠改善語音播放質量。 -
By reducing coding rate, more speech signals can be transferred in the same channel. so, low bit rate speech coding has especially important significance when the transmission rate is limited very strictly
通過降低編碼速率，可以使同樣的通道容量能夠傳輸更多路的語音信號，在傳輸比特限制十分嚴格的場合，低速率語音編碼具有特別重要的意義。 -
This paper gives the shannon limit analysis and numerical results while the modulation is bpsk. the numerical results of different coding rate are shown
本文對bpsk信號調制格式下的香農限進行了理論分析和數值求解。 -
Future mobile communications have to support the transmission of high rate data and multimedia applications in the radio spectrum, which is already extremely scarce. the basic idea of adaptive coded modulation is to maintain a constant throughput by varying the transmitted power level, symbol transmission rate, constellation size, coding rate or any combination of these parameters. thus, without sacrificing bit - error rate ( ber ), these schemes provide high average spectral efficiency by transmitting at high speeds under favorable channel conditions, and reducing throughput as the channel degrades
自適應編碼調制（ adaptivecodedmodulation ， acm ）的基本思想是在不犧牲系統傳輸性能（比如ber ）為代價的前提下，通過單獨改變發送功率、波特率、編碼方案、碼率、調制方式，或者是綜合改變前面所述的各種參數，在有利的通道條件下，獲得較大的吞吐量，當通道質量下將時，相應地降低傳輸速率，最終達到提高平均頻譜效率的目的。
分享友人