語音頻帶 的英文怎麼說
中文拼音 [yǔyīnbīndài]
語音頻帶
英文
speech band-
Because the speech signal is periodicity at sonant which vocal cords surge in low frequency and similarity to white noises at surd, the pitch can be detected in traditional way through the correlation operation without the speech produce model
在人類語音的濁音段，聲帶發生較低頻率的振蕩，語音信號呈明顯的準周期性，而在清音段，語音信號則類似於白噪聲。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。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。Digital speech has preponderance over analog speech in reliability, robustness and security during communication. however, digital speech needs more bandwidth than the analog signal. especially with the requirement for communication frequency increasing, it ' s necessary to code speech signal at low rates
但是，數字化后的信號所佔的頻帶大幅增加，特別是在帶寬需求日益增長的今天，這個問題尤為突出，因此語音的低速率編碼（即壓縮編碼）成為迫切的要求。Translation interpretation, production, videophotographing, audio dubbing, titling, cd writing and production of corporate and product introduction ; translation interpretation and editing of film and tv scripts, video tape, vcd and other audio video products ; titling of videotapes and films with various languages ; voice over ; production of audio - tapes ; speaking aside for a v products ; professional audio recording and a v engineering ; support for production of cd multimedia with various forms and software platforms
錄音帶錄影帶製作：公司及產品介紹的翻譯製作攝影配音字幕製作光盤刻錄製作等電影電視劇本錄像帶vcd等音頻視頻的翻譯錄制和編輯為錄像帶和電影添加各種語言的專業字幕添加畫外音錄音帶的製作視聽產品的旁白專業錄音及音像工程支持各種格式及軟體平臺的光盤多媒體製作。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。Speech communication is one of the most used modes in the digital trunking communication system. excellent algorithm of speech coding can save the bandwidth resource, improve the utilization of frequency, so it has important value for investigation
語音通信是數字集群通信系統中最常用的通信方式之一，優良的語音編解碼演算法能夠更加有效地節省帶寬資源，提高頻率利用率，因此具有重要的研究價值。A method of pitch mark determination for a speech, includes : acquiring a fundamental frequency point and fundamental frequency passband signals by using an adaptable filter ; detecting a number of passing zero positions of the fundamental frequency passband signals ; and generating at least a set of pitch marks from a number of passing zero positions
一種決定語音音高標記的方法，系藉以找出一語音之一組音高標記，此決定語音音高標記的方法系利用一可適性濾波器取得一基頻點與一基頻帶通訊號；求取基頻帶通訊號之復數個過零點位置；然後經由復數個過零點位置產生至少一組音高標記。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。Recently, there is a great interest in researching 4kb / s toll - quality speech coders and the technology of speech bandwidth extension
擁有長途話音質量的4kb / s語音編碼及其頻帶擴展技術是當前研究的熱點。Enhanced variable rate codec speech service option 3 for wideband spread spectrum digital systems
寬帶頻譜擴展數字系統用增強的可變率編譯碼器語音服務選擇3Firstly, it introduces the development of speech coding, along with the significance of the low bit rate speech coding. it also compares the model of traditional dualistic excitation lpc vocoder and the multi - band excitation vocoder, and lucubrates the analytical method of frequency domain and time domain in the parameter extraction of multi - band excitation vocoding. secondly, based on the parameter extraction operation of keynote cycle, it adopts time domain in rough estimate operation of keynote and frequency domain in fine estimate operation of keynote, in according to the immediacy required in practice, to minish operation amount
本文闡述了一種基於fpga的多帶激勵語音編碼器的研究與設計，首先介紹語音編碼研究的發展狀況以及低速率語音編碼研究的意義，接著對比分析了傳統二元激勵lpc聲碼器模型和多帶激勵編碼器模型，並深入研究了多帶激勵語音編碼參數提取的頻域和時域分析法，然後根據實際應用的實時性要求，為了減小運算量，在基音周期參數的提取的演算法實現上，本文採用在時域進行基音粗估運算，在頻域進行基音精細估計運算。In the next, we discuss the system of the meg - 1 layer i. the paper centers on the two kernel sub - parts : filtering coding and psychoacoustic model, do some research work in sub - band coding ( cbc ) theory and the relate theory such as quadrature mirror filter ( qmf ) and analyse sub - band filter ; also do research work in psychoacoustic theory especially the part related to the mpeg - 1 layer i. in the third chapter, introduce the ti tms320c6000 series dsps and their characteristics, also about the software development flow and the ti dsp / bios operating system of it. the forth chapter is the most important, firstly, according the algorithm flow in protocol, using c language validate the algorithm ; then, transplant and optimize the coding in dsp. in the processing of optimize, acording the assembler program characteristic of ti dsp, the paper put forward the analyse sub - band filter dsp optimization algorithm base on the eight spot idct. the algorithm has been optimize have greatly improved the work efficiency. make use of the technology of the dsp / bios host channels, data io pipe, software interrupt, we implement the musicam algorithm base on dsp / bios
論文首先對當前語音編碼技術的發展、分類以及mpeg系列音頻標準作了介紹；接著在第二章，給出了layer的musicam （ masking - patternuniversalsubbandintegratedcodingandmultiplexing ）演算法的系統組成，圍繞分析子帶濾波器和心理聲學模型兩個核心模塊，深入研究了子帶編碼工作原理、比特分配及子帶編碼中用到的正交鏡像濾波器和分析子帶濾波器；探討了心理聲學基本原理和mpeg . 1layer所用到的心理聲學模型。第三章對titms320c6000系列dsp作了簡介，介紹了6000系列dsp結構特點、 c6000dsp軟體開發流程和tidsp / bios操作系統。第四章是本文的重點，首先根據協議給出的演算法用標準c語言編程實現並調試通過。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。After having the keynote cycle, it separately processes harmonic waves of speech band ; whereafter takes v / u judgement / verdict and amplitude estimate to each band
得到基音周期后，對語音幀頻帶按基音頻率的諧波進行分帶處理，並對每個帶進行v / u判決和幅度估計。At first the article puts emphasis on analyzing those current network ip technology, various audio codec algorithms, realtime stream medium transmit technology and those process mechanism of realtime low bandwidth audio stream medium, etc. in allusion to high requirement of system, resulted from so many terminations of attending a lecture, rate of flow, bad situation of network and realtime interactive voice, a new algorithm and the relevant project of processing low bit - rate audio stream was brought forward
本文著重分析了當前網路方面的ip技術，各種音頻編碼演算法，實時流媒體傳輸技術，實時低帶寬音頻流媒體的處理機制等等。針對網路實時應用中諸如客戶端眾多，各種多媒體數據流量大，網路狀況差，語音交互實時性等方面較高的要求，提出一種新的實時低帶寬音頻流處理演算法及相應的處理方案。Pay - per - view movie channel, voice mail, data port, in - room personal safe box, and tea / coffee making facilities, non - smoking floors, an executive floor
每間客房裝有因特網/傳真介面、帶語音信箱的國際國內直撥電話、調控自如的冷暖空調、國際衛星電視頻道、電熱茶壺和電子保險箱。With advanced audio coding at 14khz, sony vc brings audio to another level of clear sound to the videoconference creating a more conversational tone
Superb音頻品質14khzh的音頻編碼， sony視像會議系統把音頻帶到令一個水平，亦令到會議的語音更暢順Essential requirements for terminal equipment intended for connection to 4 - wire analogue speechband leased lines of the public telecommunications network
公共電信網路4條模擬語音頻帶租賃線連接終端設備的基本要求Essential requirements for terminal equipment intended for connection to 2 - wire analogue speechband leased lines of the public telecommunications network
公共通信網路雙線模擬語音頻帶租賃線連接終端設備的基本要求分享友人