psychoacoustic model 中文意思是什麼

Main ideas as follows : research of audio digital watermark algorithm based on fast fourier transforms and psychoacoustic auditory model

主要工作有:基於快速傅立葉變換與聲學模型的音頻數字水印演算法研究。

A parametric psychoacoustic model based on critical band rate, level - dependent spreading function, non - linear superposition and forward masking was proposed

摘要提出了基於臨界頻帶率、依賴于強度的擴展函數、非線性迭加和前向掩蔽效應的參數心理聲學模型。

Study on algorithms of low bit rate hi - fi audio coding based on wavelet transform and psychoacoustic model

基於小波變換和音質模型的音頻編碼演算法研究

The frequency analysis of the psychoacoustic model can give the inaudible, redundan elements of audio. in this paper we employ it to guarantee the inaudibility of the embedded watermark

心理聲學模型的時頻域分析主要用於音頻數據的壓縮編碼，這里主要考慮將其用於音頻水印演算法中去，以便從理論的角度來考慮嵌入水印的不可感知性。

The newly proposed method makes good use of the psychoacoustic model. and the perceptual information contained in the synthesized signal is increased in a furthest way during the modeling procedure

感知梯度正弦建模比較好地利用了心理聲學模型，在建模過程中最大限度地增加合成信號的感知信息，提高了建模效率。

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。

Second, we analysis the character of the human auditory system, especially the psychoacoustic model. then we analyze the principle of the audio part of mpeg - 1 which apply the psychoacoustic model to compressing the audio data

本文總結了人類聽覺系統的特點，特別是人類聽覺的感知特性，分析了mpeg標準音頻部分所使用的心理聲學模型。

Wth regard to the threshold which used in the algoritlun based on wavelat transform and are determined by experience, we put forward a automatic and adaptive algorithm that can calculate the suitable threshold. this suggested algorithm makes use of the frequency analysis of the psychoacoustic model to make the embedded watermark inaudible. this idea can be used in such sort of watermark embedded algorithins based on wavelet transform

對于使用量化小波系數的加水印的演算法，對于其中要由經驗來確定的量化閾值，本文結合心理聲學模型的頻域分析，提出了一種自適應的求量化閾值的方法，這一方法對于其它使用小波變換加水印的演算法求閾值都有一定的參考價值。

Following the direction of sinusoidal modeling and sinusoidal analysis, this thesis adopted the matching pursuit techniques along with the psychoacoustic model, explored some novel methods for sinusoidal modeling as well as the quantization of model parameters, and discussed the low bit rate speech coding and its related problems. the major contributions of this thesis are included in the following : 1

本文正是沿著正弦建模正弦分析的方向，採用匹配跟蹤技術，結合心理聲學模型，研究了新的建模方法以及模型參數的量化編碼，對低位率語音編碼及相關問題進行了有益的探索，並取得了如下創新性研究成果： 1