語音濾波器 的英文怎麼說

中文拼音 [yǔyīnlǜbōqì]
語音濾波器 英文
voice filter
  • 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
  • 濾 : 動詞（除去液體雜質） filter; strain
  • 波 : Ⅰ名詞1 （波浪） wave 2 [物理學] （振動傳播的過程） wave 3 （意外變化） an unexpected turn of even...
  • 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
  • 濾波器 : [電子學] electric filter; （electric） wave filter; filter
  1. 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
  2. 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。
  3. For phonetic signal modulation, if the pass band range of the band pass filter ( bpf ) is 300hz - 3400hz, the anti - noise properties of laser are approximately independent of bias current and parameters of the cavity ; when the pass band range of bpf increases to a certain degree, modulating bias current and parameters of the cavity can improve the anti - noise properties of laser

    對語音調制情況，如帶通濾波器的通帶范圍取為300hz - 3400hz ，則激光器的抗噪聲性能基本不依賴于偏置電流和腔內參數；當帶通濾波器的通帶范圍增大到一定程度，調整偏置電流和腔內參數可以實現半導體激光器的高抗噪聲性能。
  4. 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。
  5. A feature extraction method based on gauss wavelet filter in speech recognition

    基於高斯小波濾波器的語音識別特徵提取方法
  6. We select fpga of type xc3s200 as hardware to design the coder and display the hardware resources inside, moreover study the method and steps of designing dsp, based on fpga, by using system generator, finally, it emphasizes the design process of multi - band excitation vocoder. we can work out the module of high pass filter and the module of low pass filter, module of divide frame, module of keynote rough estimate, module of keynote fine estimate, module of band - separated v / u judgment / verdict and module of band - separated amplitude estimate, by using simulink, ise and system generator

    本文選用型號為xc3s200的fpga作為設計編碼器的核心硬體，介紹了其內部所含的硬體資源，並研究了利用systemgenerator基於fpga設計dsp的方法和步驟，最後，本文把重點放在多帶激勵語音編碼器的設計上，利用simulink ， ise和systemgenerator分別設計其中的高通低通濾波器模塊、分幀疊加模塊、基音粗估模塊、基音精細估計模塊、分帶v / u判決模塊、分帶幅度估計模塊。
  7. 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。
  8. Focused on the sine model of the speech signal creation, and finished the reconstruction by using it. at the same time, based on the knowledge of digital signal processing, reconstruction by harmonic correction and time variant digital filter was proposed

    分析了語音信號產生的正弦模型，並在此基礎上完成了骨導信號的語音重構；與此同時，結合數字信號處理知識，分別用諧波修正和時變數字濾波器的方法，完成基於骨導信號的語音重構。
  9. The neural network for gain filter in speech code algorithm

    語音編碼演算法的神經網路增益濾波器
  10. The quantized lp coefficients are replaced by the unquantized lp coefficients in the frequency domain expression of the feel weighted filter. the error signal has more similar envelope shape, and the hearing effect is better than before because the unquantized lp coefficients have more accuracy than quantized lp coefficients

    由於未量化的線性預測系數具有更高的精度，因此，誤差信號通過修正後的感覺加權濾波器以後，具有與語音信號譜更加相似的包絡形狀，從而更好地利用共振峰對誤差的掩蔽效應，達到更佳的主觀聽覺效果。
  11. 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。
  12. A serial generalized morphological filter with multi - structural element is used suppression white gaussian noise or pulse noise embedded in the speech signal. the paper compares morphological speech enhancement algorithm with classical approach on the feature of speech in the frequency domain and time domain

    本文針對形態學在數字語音信號增強中的應用演算法研究，採用多結構元素的廣義形態濾波器，主要用於對被高斯白噪聲或正負脈沖噪聲污染的語音信號的濾波增強，深入研究形態學濾波的語音增強演算法在語音時域、頻域對語音特徵參數的影響。
  13. The paper discussed the bandpass filters analysis method and the technology of linear prediction code ， then reduced the lpcc and the mfcc parameters

    本文還介紹了語音信號分析方法中的濾波器組分析方法和線性預測編碼技術，並推導了lpcc參數和mfcc參數。
  14. In section two, we introduce some of the main coding theories that was in the g. 729, including the digital model of the production of speech signal, the linear prediction coding, vector quantization, lbg arithmetic and perceptual weighted filtering

    第二章著重討論了g 729語音壓縮協議中所涉及到的語音編碼基礎理論。主要包括語音信號產生的數字模型、語音信號的線性預測分析、矢量量化及其lbg演算法和感知加權濾波器。
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