echo speech 中文意思是什麼
echo speech
解釋
模仿言語-
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。 -
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。 -
So acoustic echo cancellation ( aec ) technique has become a hotspot of competition in famous communicaton company all over the world. however, it is not always easy to implement acoustic equipments for communication system with satisfactory speech quality
回波消除技術能有效地解決長距離電話網路、 ip電話、免提電話和視頻會議等通信系統中的回波問題，很好地改善了語音通信質量，具有廣闊的市場前景。 -
The speech processing module can be divided into two parts, the first part includes speech compression, voice activity detection and echo cancellation module, which improve speech quality ; the other part includes dtmf and cpt module, which generate and detect some necessary telephony signal in the communication. this thesis is organized as follows
該語音處理模塊由兩部分組成，一方面是對語音的處理，包括語音壓縮模塊、靜音處理模塊和回聲消除模塊，主要為了提高voip的語音質量;另一方面是對電話通信的控制和處理，包括雙音多頻模塊和呼叫進程音模塊，主要為了產生和檢測ip電話通信中一些必須的電話信號。 -
To improve the speech quality, it is necessary to adopt such speech processing techniques like speech compression, voice activity detection, echo cancellation, jitter buffer and so on
為了提高語音質量，需要採取一系列的語音處理技術，主要包括語音壓縮技術、靜音處理技術、回聲消除技術、抖動緩沖技術等。 -
The primary advances in speech and audio signal processing that contributed to multimedia applications are in the areas of speech and audio signal compression, speech synthesis, acoustic processing, echo control and network echo cancellation
語音和音頻信號處理的改進對多媒體應用的貢獻在下述范圍：語音和音頻信號壓縮、語音合成、聲學處理、回聲控制以及網路回聲消除。 -
Hands - free speech communication is indispensable in audio and video conference systems, hot - line telephones and videophones, mobile radio terminals, digital isdn network etc. however, the control ( cancellation ) of the acoustic echo has always had a strong impact on the transmission quality in hands - free telecommunication ~ [ 1 - 3 ] conventional methods of acoustic echo control ( cancellation ), such as echo suppression or gain control, may lead to the degradations in speech quality or make the speakers feel uncomfortable
免提式話音通信在移動電話、熱線電話、車載電話及isdn網的電視電話會議等多種領域正得到日益廣泛的應用。但至今的免提式話音通信中，仍免不了受回聲引起的話音失真、甚至嘯叫等干擾，大大降低了免提話音通信的質量。由於聲回波對消問題尚未得到圓滿解決，實現高質量的免提式話音通信仍是一個極賦挑戰性的課題。 -
This article mainly discusses the basic theories and related protocol and technology of speech sound communication based on ip. the key discuss is the support of real - time services that is provided by the current ip network, codec and the acoustic - echo cancellation, as well as the dynamic monitor of network communication quality of service. it gains some conclusion by compare the merit and shortcoming between the h. 323 protocol and sip protocol and analyzing their bases : rtp / rtcp protocol
本文主要論述基於ip的語音通信所涉及的基本理論和相關的協議與技術，重點論述當前ip數據網路對實時業務提供的支持、語音編碼、回聲消除和網路通信服務質量的動態監測，在協議的分析上比較h . 323和sip協議之間的優缺點，並具體分析這兩個協議的基礎： rtp rtcp協議。 -
Because ip telephone systems employ the packet switching technology, the echo signals generated by the hybrid due to the impedance mismatch usually experience longer transmitting delay, which significantly decrease the speech communication quality
對于ip電話系統而言，由於採用了分組交換技術，由2 / 4線轉換器阻抗不匹配而產生的回波在系統中傳輸的時延較長，這將加重回波對語音通信的影響。 -
The implementation of such a system is extremely complicated due to three factors : the impulse response of a lrm system have a duration of several hundred milliseconds, the system has to be adaptive, and the adaptation has to be performed with a speech input the traditional adaptive algorithms such as lms and nlms can not obtain the satisfied result in the real - time acoustic echo cancellation processing
但由於聲回授通道的特殊性和復雜性，普通的自適應演算法無法滿足要求，必須採用一些十分復雜的演算法，這樣一來，實時性的問題變得十分突出。快速lms newton演算法的提出，給自適應聲回波對消問題的解決開辟了新的途徑。該演算法集lms演算法的簡單易行性和newton演算法的快收斂性為一體，是一種非常有應用前景的自適應演算法。 -
Based on those process algorithms, the other important applications, such as speech coding in gsm system, echo cancellation, speaker recognition, are discussed
在此基礎上，對gsm系統中的編碼、回聲抵消、說話人識別和交通車輛內部的噪聲抵消應用進行了研究。 -
Tele - conference, video - conference. acoustic echo cancellation is normally achieved by means of an acoustic echo canceller, which, in its simplest form, consists of an adaptive filter which mimics the transfer function of the echo path ( or room acoustic ) to synthesize a replica of the echo, and then subtracts the estimation from the combined echo and near - end speech ( or disturbance ) signal to obtain the near - end signal alone
聲回波抵消通常採用聲回波抵消器來實現，最簡單的聲回波抵消器由自適應濾波器組成。具體方法是用自適應濾波器來估計回波信號，並從麥克風信號中減掉該估計值，從而實現聲回波的抵消。 -
In a telephone circuit controlled by an echo - suppressor, the inability of one or both subscribers to get through because of either excessive local circuit noise or continuous speech from one subscriber
在回波抑制器控制的電話線路中，用戶的一方或雙方因為過量的局部線路噪音或對方連續說話而不能通話的情況。
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