語音碼 的英文怎麼說
中文拼音 [yǔyīnmǎ]
語音碼
英文
phonetic code-
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。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。Phonological activation of disyllabic compound words in the speech production of chinese
言語產生中雙詞素詞的語音編碼Study of phonological encoding of chinese disyllabic compound words in patients with mild cognitive impairment
輕度認知功能損害患者漢語雙詞素詞的語音編碼研究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。The optimal gain filter of ld - celp
低延遲碼激勵語音編碼演算法的最佳增益濾波器With the developing of vlsi in recent years, high function dsp has been produced ( such as tms320 series dsp produced by ti ) and their cost is dropping. thus, this established the foundation for making complex speech coder practical and producible. the paper researched and discussed the fix - point real implementation of g. 728 by dsp tms320c5402 chip
但是，近幾年來，隨著大規模集成電路（ vlsi ）的發展，已生產出高性能數字信號處理晶元（例如ti的tms320系列dsp晶元） ，而且其成本在不斷降低，這就為復雜的語音編碼器的實用化和產品化奠定了基礎。Supported by superior technologies and system design expertise, dsp has made great development in the markets of education and e - toys, featured products as digital language learner / reciter, e - dictionary, kid - learning player, children - book, al - quran 、 sutra - player, speech responsive toys, all kinds of electronic - gifts and long times speech solutions
公司研發實力雄厚，在承接客戶委託開發項目的同時，專注于語音及音樂播放類的教育電子和玩具電子兩大專業市場，已推出獨具特色的數碼學習機/復讀機、電子詞典、早教機、課本點讀機、古蘭經播放器、念佛機、語音互動玩具、各類電子禮品及長秒數語音產品方案。Real time implementation of g. 723. 1 speech coder based on tm
1語音編解碼器的實時實現After about two years " insisting and hard working, this goal set at the beginning has become true. the developed c54x general assembly program for g. 729 speech signal compressing algorithm has passed the tracking with more than 3, 000 unitary standard measuring vectors. g. 729 speech signal compressing compiler using c54x general assembly program has been accomplished real - timely, and undistorted rebuilt speech signals have been obtained
因此本課題選用c54x的通用匯編語言編程實現g . 729語音壓縮編碼演算法，調試並通過了統一標準測試矢量三千多幀，最終在5402開發實驗板上實時實現了g . 729語音壓縮編碼器，獲得未失真的重建語音信號。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。It synthesizes the excellence of wave coding and parameter coding, adopts vector quantity, analyse - synthesize, perceptual weighting, therefore, gains good speech coding quality at 8kbit / s. cs - acelp can be used in individual telecom, iphone, c / n, microwave telecom and isdn
Cs - acelp演算法綜合了波形編碼和參數編碼的優點，以自適應預測編碼技術為基礎，採用了矢量量化、合成分析和感覺加權等技術，在8kbit / s速率上獲得了較高的語音編碼質量。It is expected to be used for 3g personal handy - phone system as standard algorithms which encode speech signals and decode it. additionally, this kind of algorithms which own excellent quality can be application in viewphone and video order programme etc. the thesis introducethe algorithm structure of g. 729
該協議在可預見的將來可能應用於三代移動通信系統中作為語音編解碼演算法。另外，由於其良好的性能也可應用在多媒體系統中如：可視電話，視頻點播等。本論文概要介紹了g . 729協議的演算法結構。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。At the present time, evrc is the best vocoder in the cdma system when take into account both the voice quality and the encode rate
在目前的cdma系統中，綜合語音質量和編碼速率， evrc是最佳的語音編碼器。It is conceived to introduce barker codes as synchronization preamble and add synchronization signal in front of speech signal to implement speech detection. the principle of this idea is presented
提出了引入barker碼作為同步碼，在語音信號前添加同步信號以實現起點檢測的工作設想，並介紹了其同步原理。Second, we optimize the codebook and choice a part of the codeword which is used most efficiently. the result is not degraded too much while the complexity is reduced. at the end of the paper the development prospect of cs - acelp and speech coding are described
對lsp參數量化中的第一級碼書的128個碼字的使用頻率進行了統計試驗，選用了128個碼字中使用頻率高的112個碼字作為新碼書，語音質量基本不變但降低了碼書搜索的復雜度。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。The basic characteristics of the current data network are point - to - point, connectless, doing one ' s endeavor, no quality of service, etc. these characteristics do not meet the requirement of real - time services, therefore, the realization of voip need support of the some key technology. these technologies includes : speech sound coding and data compression, real - time transmission and control, mute compression and multicast, acoustic - echo cancellation and comfort noise generator, dynamic monitor and guarantee of quality of network service, as well as, the compatible of different network and different protocol with each other
但現有的數據網路的基本特性：點對點的、無連接的、盡力而為的、沒有服務質量保證等特性並不適合與實時的業務要求，因此voip的實現需要一些關鍵技術的支持，這些技術包括：語音編碼和壓縮技術、實時傳輸和控制技術、組播技術、靜音壓縮和舒適噪聲生成技術、回聲消除技術、網路服務質量的動態監測和保證技術、以及不同的網路、不同的協議之間的互連互通等等。分享友人