Search - Speech filtering

Search - Speech filtering - List

[matlab] speech_enhancement DL : 1: 自己编写的用谱相减，最小均方和维纳滤波实现语音增强的matlab文件-themselves with the preparation of spectral subtraction, the minimum mean square Wiener filtering and enhanced voice document Matlab
Update : 2025-02-17 Size : 2kb Publisher : Richard
[Industry research] AnadaptiveKalmanfilterfortheenhancementofspeechsig DL : 0: This paper deals with the problem of speech enhancement when a corrupted speech signal with an additive colored noise is the only information available for processing. Kalman filtering is known as an effective speech enhancement technique, in which speech signal is usually modeled as autoregressive (AR) process and represented in the state-space domain.-This paper deals with the problem of speech enhancement when a corrupted speech signal wit h an additive colored noise is the only informat ion available for processing. Kalman filterin g is known as an effective speech enhancement te chnique. in which speech signal is usually modeled as aut oregressive (AR) process and represented in th e state-space domain.
Update : 2025-02-17 Size : 100kb Publisher : rifer
[Industry research] RobustadaptiveKalmanfilteringbasedspeechenhancemen DL : 0: This paper deals with the problem of speech enhancement when only a corrupted speech signal is available for processing. Kalman filtering is known as an effective speech enhancement technique, in which speech signal is usually modeled as autoregressive (AR) model and represented in the state-space domain.-This paper deals with the problem of speech enhancement when only a corrupted speech signa l is available for processing. Kalman filterin g is known as an effective speech enhancement te chnique. in which speech signal is usually modeled as aut oregressive (AR) model and represented in the s tate-space domain.
Update : 2025-02-17 Size : 240kb Publisher : rifer
[Speech/Voice recognition/combine] MainScript DL : 0: 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
Update : 2025-02-17 Size : 1kb Publisher : 张洁
[Graph program] SpecSubTD_Boll79 DL : 0: 关于语音增强的一些算法心得，如谱减法自适应滤波以及听觉掩蔽效应等。-on speech enhancement algorithms some experiences, such as spectral subtraction adaptive filtering and auditory masking effect.
Update : 2025-02-17 Size : 2kb Publisher : rube
[matlab] JIN DL : 0: 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.
Update : 2025-02-17 Size : 2kb Publisher : yim
[Speech/Voice recognition/combine] sunday DL : 1: 当把m个信号叠加在一起时，经倒谱分析是能检测出其基频。但倒谱分析主要是用在同态滤波中，可用于某些信号的解褶积。例如提供的语音信号，就能通过倒谱把频谱中的包络和基频频谱分离，-When the m signal superimposed together, by the cepstrum analysis is able to detect its fundamental frequency. However, cepstral analysis is mainly used in homomorphic filtering can be used for the solution of certain signal deconvolution. Such as the provision of the speech signal, we can through the spectrum of cepstral envelope and baseband frequency spectrum separation
Update : 2025-02-17 Size : 1kb Publisher : 小单
[Speech/Voice recognition/combine] winenvar0805 DL : 0: 语音信号处理，matlab文件，维纳滤波程序。08年05月-Speech signal processing, matlab files, Wiener filtering procedures. May 2008
Update : 2025-02-17 Size : 1kb Publisher : 意乱
[matlab] show-new DL : 0: 语音信号处理。例如：滤波、检测、增强、估计、识别、谱分析-Speech Signal Processing. For example: filtering, detection, enhancement, estimation, identification, spectral analysis
Update : 2025-02-17 Size : 13kb Publisher : tangli
[matlab] Wiener DL : 1: 这是一个wiener滤波程序，对语音信号滤波。里面有详细的word文档说明-This is a wiener filtering process to filter the speech signal. There are detailed notes word document
Update : 2025-02-17 Size : 57kb Publisher : mw
[Speech/Voice recognition/combine] kalman DL : 0: 该程序是卡曼滤波法在语音处理上的应用,能有效的去除噪声,达到语音增强的目的!-The program is Kaman filtering method in the voice processing application, can effectively remove the noise, to achieve the purpose of speech enhancement!
Update : 2025-02-17 Size : 1kb Publisher : zhjuna
[Speech/Voice recognition/combine] mpsound DL : 0: 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.
Update : 2025-02-17 Size : 238kb Publisher : 或或
[Documents] liguyue DL : 0: 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
Update : 2025-02-17 Size : 463kb Publisher : 阳鹏
[Speech/Voice recognition/combine] DTWspeech DL : 0: 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
Update : 2025-02-17 Size : 2.38mb Publisher : 周文超
[Audio program] Residual_Excited_Linear_Prediction_with_MATLAB.ra DL : 0: Simulation of Residual Excited Linear Prediction (RELP) coding for speech: This simulation give your voice or available clear wav file.This encoder have linear predictor that decreases signal s dynamic (lower quantization level). This technique also use lowpass filtering to make lower bandwidth and reconstruct it with three methods that is chosen in decoder. For example duplicate same spectrum of encoded signal that filtered in upper frequency.-Simulation of Residual Excited Linear Prediction (RELP) coding for speech: This simulation give your voice or available clear wav file.This encoder have linear predictor that decreases signal s dynamic (lower quantization level). This technique also use lowpass filtering to make lower bandwidth and reconstruct it with three methods that is chosen in decoder. For example duplicate same spectrum of encoded signal that filtered in upper frequency.
Update : 2025-02-17 Size : 835kb Publisher : Ardalan
[Speech/Voice recognition/combine] voice DL : 0: 本程序为一个用matlab实现的语音滤波程序，能计算并显示语音信号的幅频特性，巴特沃斯低通滤波器幅度和相位响应，引入噪声后的幅频特性，以及滤波后的DFT幅频特性。本程序附有语音文件以供试验。-This program is an implementation using matlab voice filtering procedures, can calculate and display the speech signal amplitude-frequency characteristics of Butterworth low-pass filter magnitude and phase response, after the introduction of noise, amplitude-frequency characteristics, as well as the filtered DFT amplitude-frequency characteristics. This procedure for testing with voice file.
Update : 2025-02-17 Size : 4kb Publisher : 周峰
[matlab] BF_wiener_g DL : 1: 麦克风阵语音增强中波束形成后维纳滤波的MATLAB代码-Microphone Array Speech Enhancement Beamforming MATLAB code after the Wiener filtering
Update : 2025-02-17 Size : 1kb Publisher : 黄邦奉
[Other] colea DL : 0: COLEA is a Matlab Speech Processing Toolkit with a graphical user interface. This program can be used to edit speech waveforms (cut, copy or paste selected speech segments) and compute spectrogramss. It includes several speech related tools including a filter tool for speech filtering, and a comparison tool for comparing two waveforms using a variety of spectral distance measures.COLEA can be used to display time-aligned phonetic transcriptions (e.g., TIMIT s .phn files) and for manual segmentation of speech waveforms. It can also perform formant analysis and pitch analysis. A manual describing COLEA is included. For more information about this program visit the web site: http://www.utdallas.edu/~loizou/speech
Update : 2025-02-17 Size : 348kb Publisher : Gurkan Acı r
[Software Engineering] lunwen DL : 0: 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.
Update : 2025-02-17 Size : 192kb Publisher : Conan King
[Speech/Voice recognition/combine] speech-enhancement DL : 0: 本资料涵盖了几乎所有的语音增强方面的方法，主要有谱减法，听觉掩蔽，最小均方误差，维纳滤波以及一些非主流的方法，这些对于研究语音增强的人来说是很有帮助的-The data cover almost all aspects of speech enhancement methods, the main spectral subtraction, auditory masking, minimum mean square error, Wiener filtering as well as some non-mainstream approach, which for the study of speech enhancement is helpful for people who
Update : 2025-02-17 Size : 60.74mb Publisher : luzhiqiang

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