音頻濾波器 的英文怎麼說

中文拼音 [yīnbīnlǜbōqì]
音頻濾波器 英文
tone equalizer
  • 音 : 名詞1. （聲音） sound 2. （消息） news; tidings 3. [物理學] （音質） tone 4. （姓氏） a surname
  • 頻 : Ⅰ形容詞（次數多） frequent Ⅱ副詞（屢次） frequently; repeatedly Ⅲ名詞1 [物理學] （物體每秒鐘振動...
  • 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
  • 音頻 : [物理學] [電學] audio frequency; vf （voice frequency）音頻電路 voice frequency circuit; 音頻振蕩...
  • 濾波器 : [電子學] electric filter; （electric） wave filter; filter
  1. The focus is placed on the investigation of the standard of the encoding algorithm for mpeg audio layer iii, and the analysis of the major four modules in the compression algorithm, including encoding of subband filter bank, psychoacoustics model, quantification and huffman coding, frame packing

    重點研究了mpeg音頻第層編碼的演算法標準。詳細分析了壓縮演算法中的四個主要功能模塊：子帶濾波器組編碼，心理聲學模型，比特流量化與霍夫曼編碼，幀數據流格式化。
  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. 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。
  4. The digital audio processors tas300x series produced by ti have powerful ability to handle the audio. this paper introduces the method of designing the eq filter while introduce the digital audio processors tas300x series. about the control of the system, i2c bus and the msp430 mcu are introduced

    德州儀器公司（ ti ）出品的專用dsp數字音頻處理器tas300x系列，有著強大的音頻處理能力，性價比很高，本文在簡介tas300x系列數字音頻處理器的同時，還介紹了數字均衡濾波器的設計方法。
  5. Analog synthesis often use low - pass filter to remove electronic audio generator generated by high - frequency noise

    模擬合成器常常使用低通濾波器來去除電子音頻發生器所產生的高頻噪聲。
  6. Dsp digital shift frequency sires is new generation conference system and teaching equipment, which adopts dsp technology. it can lock the point of howl a automatically, dsp digital shift frequency and aptitude mix sound of transmitter

    數字自動濾波器，數字話筒智能混音器，數字高低音調節器等多項數字化音頻處理模塊的技術優勢，解決了會議教學擴聲中的聲反饋問題，可提升話筒靈敏度
  7. Microfilter is mainly used to separate the data and voice transmission on the same phone line

    每一個電話分機已安裝了寬頻濾波器確保資料傳輸不被話音傳輸所影響。
  8. 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。
  9. 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。
  10. 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。
  11. The advantage of a sharper filter is that you will have less noise in the audio signal

    較尖銳特性濾波器的好處是音頻（輸出）信號有較少的噪聲。
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