coding filter 中文意思是什麼
coding filter
解釋
編碼濾波器-
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。 -
Image coding based on wavelet and filter banks
基於小波和濾波器組的圖像編碼方法 -
In the second chapter, wavelet theory, which is the theoretic basis of embedded zero wavelet coding is introduced briefly, including wavelet transform " s local characteristics, multi - resolution analysis, discrete signal ' s fast decomposition and construction, wavelet and filter group
第二章，對小波零樹編碼演算法的理論基礎? ?小波理論作了簡要的介紹，包括小波變換的時頻局部特性、多解析度分析、離散信號的快速分解與重構以及小波與濾波器組的關系。 -
Video coding using m - channel filter banks
帶濾波器組的視頻編碼方法 -
Using the run - length coding method, implementing the synchronization of volume date - recording and laplace difference, the filter keeps a high real time capability
運用遊程編碼的思想作為樹集設計逐行掃描的數據錄入方法，使數據錄入與拉普拉斯差分運葬同步進行，以保證高實時性。 -
The paper studies the theory and coding methods of the low bit rate video recommendation h. 263 + detailedly, realizes real - time software codec based on h. 263 + under linux os, furthermore, finishes some negotiable coding options which can observably improve the coding efficiency, such as advanced prediction mode, advanced intra coding mode, deblocking filter mode and modified quantization mode
263 +的原理和編碼方法做了深入的研究，軟體實現了基於h 263 +標準的視頻編譯碼器在linux操作系統下的實時編解碼，並實現了能顯著提高編碼效率的幾種可選編碼模式，主要包括先進預測模式、先進幀內編碼模式、塊邊界濾波模式和改進量化模式等。 -
The main aim of speech signal processing is speech coding, speech recognition and speech understanding by automaton. in this paper, firstly, some basic signal process are discussed, such as signal filter, sampling, fft, pitch detection, then, they are tested on the testing board designed by the author
論文中首先對語音信號的基本處理問題進行了分析和對比，然後在自己設計的基於tms320vc5402的dsp實際系統上，進行了語音處理過程的濾波、采樣、傅立葉變換和譜包絡提取的演算法實現研究，討論了在演算法的dsp實現方法，分析了運行實驗結果。 -
The method includes split originality image region, filter, separately coding the lowest frequency sub - band coefficient, adaptively setting threshold of different image region, setting up flag matrix, and filtering after decoding to eliminate domino effect of split
這種編碼方法主要包括：對原始圖像進行分塊、濾波；對最低頻子帶系數單獨編碼；對各塊圖像自適應地進行閾值設置；設立標志矩陣；解碼后進行濾波消除分塊效應。 -
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。 -
In this dissertation, psychoacoustic models filter banks quantization and bit allocation predicate coding and temporary noise shaping is analyzed
本文首先對心理聲學模型、濾波器組、量化和比特分配、預測編碼、時域噪聲成型等技術進行研究。 -
A deblocking filter with two separate modes in block - based video coding, i. ieee trans. circuits syst
在這一節里，首先介紹了直接幀內預測技術提出的技術背景。 -
This paper introduce the adapive deblocking filter used in the h. 264 / avc video coding standard, analyse the theory of the system and its advantages comparing with similar systems in other video coding standards
介紹了h . 264視頻編碼標準中的自適應去塊濾波系統，分析了該系統原理及其相對于以往去塊效應濾波系統的改進。 -
The standard is designed to give a new technical solution for a broad range of applications, such as very low bit rate conversational services and entertainment quality broadcast, interactive video - on - demand services etc. for the enhancement of the coding efficiencies, the new standard adopts new tools as following : multiple reference pictures, variable block - size with seven block sizes in motion prediction, quarter - pixel accuracy for motion vector, short word - length integer transform, context - adaptive entropy coding and loop filter deblocking
H . 264 / avc標準支持從低帶寬、高誤碼率的無線移動視頻通信到高帶寬、低誤碼率的有線視頻廣播等多種應用，因此，日益受到業界的關注。為了實現更高的編碼效率， h . 264 / avc標準採用了很多新的編碼技術，如多參考幀預測、多尺寸編碼塊模式、 1 / 4像素精度運動矢量、整數變換量化、基於內容的熵編碼、新型幀內預測、去除方塊效應的濾波器等。 -
The use of multiple sampling rates offers many advantages, such as reducing computational complexity for a given task, speeding up transmission rate, and / or raising storage capacity. so the applications of multirate digital filter technology have been found in subband coding of images and speeches, communications, and in many other fields
由於多速率數字濾波技術可有效地降低信號的處理復雜度、加快數據的傳輸率和提高存儲量等，所以它在圖像編碼、語音編碼及通信等眾多領域具有廣泛的應用。 -
In this paper, at first, we analyzed the feasibility of dynamic focusing by interpolation method, and then introduced our particular design of digital - coding emission, dynamic filter, dynamic focusing and digital i / q demodulation in digital front - end based on fpga
論文首先在理論上分析了內插式數字動態聚焦實現的可行性，然後對數字前端的關鍵技術:數字編碼發射、動態濾波、動態接收聚焦、變跡變孔徑、正交包絡檢波等的fpga實現進行了詳細的闡述並給出了實現方案。
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