Description: 语音信号进行瞬时维纳滤波的程序,进行信号的去噪处理以便于识别或后续过程-speech signals instantaneous Wiener filtering procedures, signal denoising for identification or follow-up in the process Platform: |
Size: 1304 |
Author:张洁 |
Hits:
Description: 语音信号进行瞬时维纳滤波的程序,进行信号的去噪处理以便于识别或后续过程-speech signals instantaneous Wiener filtering procedures, signal denoising for identification or follow-up in the process Platform: |
Size: 1024 |
Author:张洁 |
Hits:
Description: 录制一段个人自己的语音信号，并对录制的信号进行采样；画出采样后语音信号的时域波形和频谱图；给定滤波器的性能指标，采用窗函数法和双线性变换设计滤波器，并画出滤波器的频率响应；然后用自己设计的滤波器对采集的信号进行滤波，画出滤波后信号的时域波形和频谱，并对滤波前后的信号进行对比，分析信号的变化；回放语音信号；最后，设计一个信号处理系统界面。-Record a person s own voice signal, and recording the signal sampling draw sample after the speech signal time-domain waveform and frequency spectrum given filter performance indicators, using a window function method and the bilinear transform filter design and draw the frequency response filter then use the filter of their own design for the signal acquisition filtering, after filtering the signal to draw the time-domain waveform and spectrum, and filtering of signals before and after contrast, analysis of signal changes playback of voice signal Finally, the design of a signal processing system interface. Platform: |
Size: 2048 |
Author:yim |
Hits:
Description: 录制一段自己的语音信号，并对录制的信号进行采样；画出采样后语音信号的时域波形和频谱图；给定滤波器的性能指标，采用窗函数法和双线性变换设计滤波器，并画出滤波器的频率响应；然后用自己设计的滤波器对采集的信号进行滤波，画出滤波后信号的时域波形和频谱，并对滤波前后的信号进行对比，分析信号的变化；回放语音信号；最后，设计一个信号处理系统界面。-Section of their own voice recording signal, and recording the signal sampling draw sample after the speech signal time-domain waveform and frequency spectrum given filter performance indicators, using a window function method and bilinear transformation design of filters, and draw the frequency response filter then use the filter of their own design for the signal acquisition filtering, after filtering the signal to draw the time-domain waveform and spectrum, and filtering of signals before and after contrast, analysis of signal changes playback voice signal Finally, the design of a signal processing system interface. Platform: |
Size: 143360 |
Author:joe |
Hits:
Description: MATLAB代码,利用卡尔曼滤波实现加入白色噪声后的语音信号的增强.效果不错.-MATLAB code, the use of Kalman filter to achieve by adding white noise to enhance the voice signals. Good results. Platform: |
Size: 1024 |
Author:李茉 |
Hits:
Description: 录制一段个人自己的语音信号。对录制的信号进行采样；画出采样后语音信号的时域波形和频谱图；对语音信号进行加噪和去噪处理，画出滤波后信号的时域波形和频谱，并对滤波前后的信号进行对比，分析信号的变化；回放语音信号；实现快录慢放、慢录快放等功能。-Record a person' s own voice signal. Of the recorded signal sampling draw sample after the speech signal time-domain waveform and frequency spectrum of voice signals, noise and de-noising processing, draw Filtered time-domain signal waveform and spectrum, and filtering the signal before and after comparative analysis of signal changes playback voice signal realize quickly recorded slow release, slow release recorded faster functions. Platform: |
Size: 243712 |
Author:或或 |
Hits:
Description: 数字信号处理的课程设计。题目是语音信号滤波去噪——使用双线性变换法设计的切比雪夫II型滤波器-Digital signal processing design curriculum. Topics are filtering denoising speech signals- the use of bilinear transform design Chebyshev Type II Filter Platform: |
Size: 474112 |
Author:阳鹏 |
Hits:
Description: 本 文 首先 介绍了语音识别的研究和发展状况，然后循着语音识别系统的
处理过程，介绍了语音识别的各个步骤，并对每个步骤可用的几种方法在实
验基础上进行了分析对比。研究了语音信号的预处理和特征参数提取，包括
语音信号的数字化、分帧加窗、预加重滤波、端点检测及时域特征向量和变
换域特征向量.其中端点检测采用双门限法.通过实验比对特征参数的选取，
采用12阶线性预测倒谱系数作为识别参数。详细分析了特定人孤立词识别算
法，选定动态时间弯折为识别算法，并重点介绍其设计实现。
在 Vi su alC++环境下，设计并实现一个特定人、孤立词语音识别系统，
系统可以识别数字0-9等简单指令。该系统还具备演示、学习功能，可以演
示语音处理的各个步骤，还可以根据需要添加新的指令。
最 后 ， 重点从端点检测算法和动态时间弯折识别算法对系统进行改进。
实验表明，改进后的系统识别率有很大提高，达到95 ，为进一步开发实用
性语音识别系统产品打下了基础。-This article introduced the first speech recognition research and development, and then follow the voice recognition system
Processing, speech recognition, introduced the various steps, each step of the methods available in the real
A post-mortem conducted on the basis of the analysis and comparison. Research on the speech signal pre-processing and feature extraction, including
Digitized voice signals, sub-frame window, pre-emphasis filtering, endpoint detection feature vector in time domain and variable
Eigenvector for the domain. One endpoint detection method using dual-threshold. Through experiments over the selection of characteristic parameters,
The use of 12-order linear prediction cepstral coefficients as recognition parameters. Detailed analysis of the specific operator who isolated word recognition
Law, selected Dynamic Time Warping Algorithm for identifying and focusing on the achievement of its design.
In Vi su alC++ environment, design and realization of a s Platform: |
Size: 2491392 |
Author:周文超 |
Hits:
Description: 本文简要介绍了语音信号采集与分析的发展史以及语音信号的特征、采集与分析方法，并通过PC机录制自己的一段声音，运用Matlab进行仿真分析，最后加入噪声进行滤波处理，比较滤波前后的变化。-This paper introduces the voice signal acquisition and analysis of the history and characteristics of speech signals, sampling and analysis methods, and through a PC, record your own voice, use of Matlab simulation and analysis, and finally adding noise filtering, comparing before and after filtering change. Platform: |
Size: 196608 |
Author:Conan King |
Hits:
Description: 录制一段个人自己的语音信号，并对录制的信号进行采样；画出采样后语音信号的时域波形和频谱图；给定滤波器的性能指标，采用窗函数法和双线性变换设计滤波器，并画出滤波器的频率响应；然后用自己设计的滤波器对采集的信号进行滤波，画出滤波后信号的时域波形和频谱，并对滤波前后的信号进行对比，分析信号的变化；回放语音信号-The individual' s own record a voice signal, and the recorded signal is sampled draw sampled speech signal time-domain waveform and frequency spectrum filter performance given by the window function method and bilinear transformation design a filter and draw the filter frequency response then use their own filters designed to filter the collected signals, to draw the filtered time domain waveform and frequency spectrum, and comparing the signal before and after filtering, analysis of signal changes playback of the speech signal Platform: |
Size: 12288 |
Author:姚湘陵 |
Hits:
Description: 语音的IIR滤波程序，可以实现对语音信号的IIR整型滤波-Voice of the IIR filter program, you can achieve integer IIR filtering of speech signals Platform: |
Size: 1024 |
Author:张少川 |
Hits:
Description: 该程序实现对混有噪声的语音信号进行处理，滤除噪声，得到清晰的语音信号-The program for processing speech signals mixed with noise, noise filtering to give a clear voice signal Platform: |
Size: 7612416 |
Author:乔静萍 |
Hits:
Description: 在本文中，我们将提出一种分离语音组成部分的程序进行讨论。该程序是基于对非加性信号的非线性滤波的方法，也被称作广义线性滤波。- In this paper, a procedure for separating the components of speech is proposed and discussed. The procedure is based on an approach to nonlinear filtering of signals which have been nonadditively combined, that has been termed generalized linear filtering. Platform: |
Size: 627712 |
Author:李经纬 |
Hits:
Description: Normally speech signals are contaminated with noise and interference that reduces the intelligibility of speech during communication. In order to make speech
signals e ective and useful, they need to be enhanced from the noisy speech signal.
In speech processing eld many speech enhancement techniques are developed and
are providing very good results.
Multichannel microphone array is also one of the techniques used for speech
enhancement, that provides better results than the single channel speech enhancement. Moreover, Wiener ltering is the most commonly used technique for
multichannel microphone array for speech enhancement. The main focus of this
thesis is to implement multichannel microphone array using Wiener ltering in
the modulation domain system and also in the time domain system to enhance
the speech. Platform: |
Size: 914432 |
Author:an mchol |
Hits:
Description: 用于混响背景语音分离的结构稀疏模型(Strutured sparisty model)方法-To further tackle the ambiguity
of the reflection ratios, we propose a novel formulation of the
reverberation model and estimate the absorption coefficients
through a convex optimization exploiting joint sparsity model
formulated upon spatio-spectral sparsity of concurrent speech
representation. The acoustic parameters are then incorporated
for separating individual speech signals through either structured
sparse recovery or inverse filtering the acoustic channels.
The experiments conducted on real data recordings of spatially
stationary sources demonstrate the effectiveness of the proposed
approach for speech separation and recognition. Platform: |
Size: 1575936 |
Author:bigbigtom |
Hits:
Search - filtering speech signals