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[Speech/Voice recognition/combine] FIRDsgn DL : 0: 不错的FIR滤波器源代码！-good FIR filter source code!
Date : 2026-01-08 Size : 135kb User : 刘平
[Speech/Voice recognition/combine] FIRDsgn DL : 0: 有限脉冲响应过滤器。This program designs a Finite Impulse Response (FIR) filter. The window-based method is used to obtain a low-pass, high-pass, band-pass or band-stop FIR filter-finite impulse response filters. This program designs a Finite Impulse Response (FIR) filter. The window-based method is used to obtain a low-pass, high-pass, band-pass or band-stop FIR filter
Date : 2026-01-08 Size : 155kb User : 站长
[Speech/Voice recognition/combine] 接收机的MATLAB仿真程序 DL : 0: Title: MMSE Receiver for DS-SS in AWGN Channel Author: Panson Tantikovit Summary: An adaptive receiver for DS-SS systems MATLAB Release: R12.1 Required Products: Communications Toolbox,Signal Processing Blockset Description: This is an adaptive receiver for a direct-sequence spread spectrum (DS-SS) system over an AWGN channel. The adaptive receiver block is modified from the LMS adaptive filter block in DSP Blockset. For DS-SS signal reception, the adaptive filter needs to have multi-rate operation. The input sample rate is equal to chip rate and the output is at symbol rate. Two rates are related by PG, processing gain. -Title: MMSE Receiver for DS-SS in AWGN Channel Author: Panson Tantikovit Summary: An adaptive receiver for DS-SS systems MATLAB Release: R12.1 Required Products: Communications Toolbox,Signal Processing Blockset Description: This is an adaptive receiver for a direct-sequence spread spectrum (DS-SS) system over an AWGN channel. The adaptive receiver block is modified from the LMS adaptive filter block in DSP Blockset. For DS-SS signal reception, the adaptive filter needs to have multi-rate operation. The input sample rate is equal to chip rate and the output is at symbol rate. Two rates are related by PG, processing gain.
Date : 2026-01-08 Size : 19kb User : zzp
[Speech/Voice recognition/combine] col DL : 0: 一个什么都能做的语音处理软件，Manual segmentation of speech waveforms - creates label files which can be used to train speech recognition systems Waveform editing - cutting, copying or pasting speech segments Formant analysis - displays formant tracks of F1, F2 and F3 Pitch analysis Filter tool - filters speech signal at cutoff frequencies specified by the user Comparison tool - compares two waveforms using several spectral distance measures Speech degradation -can do what a voice processing software, Manual segmentation of speech waveforms- crea tes label files which can be used to train speech recognition systems Waveform editing- cuttin g, copying or pasting speech segments Formant ana P <0.05-displays formant tracks of F1, F2 and F3 Pitch analysis Filter tool-filters sp eech signal at cutoff frequencies specified by the user Comparison tool-compares two wavefor ms spectral distance using several measures Sp eech degradation
Date : 2026-01-08 Size : 403kb User : 威威
[Speech/Voice recognition/combine] recarry DL : 0: fir filter 程序老师上课留的作业，在这里跟大家分享一下，希望能有所帮助-fir filter procedures teacher in the class to stay the operation here to share with you, hope can be helped
Date : 2026-01-08 Size : 1kb User : zb
[Speech/Voice recognition/combine] ECEAdaptiveFilters DL : 0: Adaptive filter designer.LMSNLMS
Date : 2026-01-08 Size : 28kb User : Howard
[Speech/Voice recognition/combine] zishiyinglms DL : 0: 自适应滤波器的LMS算法希望能够对大家有所帮助，这各算法是实现过了的，可是运行，图像还比较令人满意，要是大家下载了，请留下评价-LMS adaptive filter algorithm hope to be helpful to everyone, the algorithm is realized in a, However operation, image is relatively satisfactory, if you download, please leave Evaluation
Date : 2026-01-08 Size : 282kb User : 马小娜
[Speech/Voice recognition/combine] melbankm DL : 0: matlab编写，求mel滤波器矩阵的系数-Matlab prepared for mel filter coefficient matrix
Date : 2026-01-08 Size : 2kb User : wh
[Speech/Voice recognition/combine] 16to8K-Downsample DL : 0: 本程序将指定的16K采样的语音数据文件转换为经G.723编解码后的8K语音数据。降采样前先使用180阶的FIR滤波器对语音数据进行频率压缩，然后进行抽取，并对抽取的数据进行G.723编解码。该程序在非特定语音识别的库文件处理中使用，也可扩展至其他用途。-this procedure will be designated the 16K sampling voice data files converted to G.723 codecs by the 8K words Music data. Sampling down before use 180 bands FIR filter frequency voice data compression, then proceed to collect, also collected data G.723 codecs. The procedures in non-specific voice recognition for the use of document processing, and can be expanded to other uses.
Date : 2026-01-08 Size : 1.6mb User : 王小飞
[Speech/Voice recognition/combine] LMSandRLS DL : 0: 用于数字信号处理中的LMS和RLS自适应滤波器实现的算法源码-for digital signal processing LMS and RLS Adaptive Filter algorithm source code
Date : 2026-01-08 Size : 4kb User : ouxianfeng
[Speech/Voice recognition/combine] matlab_bp DL : 0: 关于bp算法来逼近一个函数，还有一些关于一些简单的matlab实验程序，包括切比雪的滤波-On the bp algorithm to approximate a function, there are some simple matlab on experimental procedures, including snow Chebyshev filter
Date : 2026-01-08 Size : 233kb User : rch
[Speech/Voice recognition/combine] matlab_frequency_analyse DL : 0: 1．产生一个连续信号，包含低频，中频，高频分量，对其进行采样，进行频谱分析，分别设计三种高通，低通，带通滤波器对信号进行滤波处理，观察滤波后信号的频谱。 2.采集一段含有噪音的语音信号（可以录制含有噪音的信号，或者录制语音后再加进噪音信号），对其进行采样和频谱分析，根据分析结果设计出一合适的滤波器滤除噪音信号-1. Produce a continuous signal, including low-frequency, medium frequency, high-frequency components, its sampling, spectral analysis, are designed for three high-pass, low pass, band-pass filter the signal filtering treatment to observe the signal spectrum after filtering. 2. Acquisition section of the speech signal contains noise (can record a signal containing noise, or voice recording, additional noise after the signal), its sampling and spectral analysis, based on an analysis of the results to design a suitable filter to filter out one noise signal
Date : 2026-01-08 Size : 1kb User :
[Speech/Voice recognition/combine] lmsfilter DL : 0: LMS自适应滤波器的matlab源码文件大小不够，所以加了个图-LMS adaptive filter matlab source code file size not enough, so the addition of a map
Date : 2026-01-08 Size : 14kb User : yuefeng
[Speech/Voice recognition/combine] identificy_fir_with_iir22 DL : 0: matlab 仿真 lai 实现用自适应iir滤波器用lms算法-matlab simulation lai using adaptive IIR filter using LMS algorithm
Date : 2026-01-08 Size : 1kb User : shanghai
[Speech/Voice recognition/combine] LMS-C DL : 0: LMS滤波器示例程序，在TURBOC中运行这是一个简单的可图形显示的C程序输入信号是一个被噪声污染了的sin信号。 */ /* 运行后，屏幕的上方是输入信号，下方是经过LMS滤波后的输出信号 -LMS filter sample programs, run in TurboC This is a simple graphical display can process the C input signal is a noise contaminated signal sin.*//* Running, the top of the screen is the input signal, the bottom is the result after the LMS filter output signal
Date : 2026-01-08 Size : 4kb User : 蜗牛
[Speech/Voice recognition/combine] work DL : 0: 一种简单的cic filter matlab实现方法-A simple way to realize cic filter matlab
Date : 2026-01-08 Size : 10kb User : lvxiaobing
[Speech/Voice recognition/combine] LMS DL : 0: LMS自适应滤波算法是很常见的滤波算法。本代码包括常用的LMS自适应滤波算法，如基本LMS算法，解相关LMS算法，滤波型LMS算法，变换域LMS算法等。-LMS adaptive filtering algorithm is a common filtering algorithm. This code, including commonly used LMS adaptive filtering algorithms, such as the basic LMS algorithm, decorrelation LMS algorithm, filter-based LMS algorithm, transform domain LMS algorithm.
Date : 2026-01-08 Size : 5kb User : 阿华
[Speech/Voice recognition/combine] matlab_reduce_noise DL : 0: matlab去除50hz噪声。我用电脑录了一段声音，里面有50hz的周期噪声（因为受交流电干扰）。而我自己的声音频率最低是90hz。我使用了一个10阶butterworth高通滤波器，边带是70hz(介于50跟90之间)。问题是，这不能直接用。因为声音文件的采样率是22k，70相对于22k来说太小了。所以我得先把我的声音欠采样，然后再滤波，然后再插值。-matlab remove 50hz noise. I used the computer recorded a voice, there are 50hz cycle noise (due to AC interference). I own voice is the lowest frequency of 90hz. I use a 10-order Butterworth high-pass filter, edge belt is 70hz (the range between 50 to 90). The problem is that this can not be directly used. Because the sound files of the sampling rate is 22k, 70 compared with the 22k run too small. So I have to put my voice due to sampling, and then filtering, and then interpolation.
Date : 2026-01-08 Size : 6kb User : 张文斌
[Speech/Voice recognition/combine] audio_noise DL : 0: 语音降噪。从Codec AD50采集话筒语音，通过DSP TMS320vc5402处理，在送到AD50输出降噪后语音，涉及加汉宁窗，切比雪夫滤波器，快速傅立叶变换和反FFT，有声无声判断谱分解，谱合成等功能-Voice noise. Codec AD50 collected from the microphone voice, through the DSP TMS320vc5402 treatment, in AD50 to the output noise after the voice, involve Hanning windows, Chebyshev filter, Fast Fourier Transform and anti-FFT, audio silent judge spectral decomposition, spectral synthesis and other functions
Date : 2026-01-08 Size : 43kb User : 黄胜华
[Speech/Voice recognition/combine] wiener-filter DL : 0: 维纳滤波器实现语音信号降噪，语音增强。压缩包中已经包含测试语音片段(wiener filter for speech enhancement, speech denoising)
Date : 2026-01-08 Size : 159kb User : 俊俊zgj

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