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Description: Sox是最为著名的Open Source声音文件格式转换工具。已经被广泛移植到Dos、windows、OS2、Sun、Next、Unix、Linux等多个操作系统平台。Sox项目是由Lance Norskog创立的，后来被众多的开发者逐步完善，现在已经能够支持等多种声音文件格式和声音处理效果。基本上常见的声音格式都能够支持。更加有用的是，Sox能够进行声音滤波、采样频率转换，这对那些从事声讯平台开发或维护的朋友非常有用。当然，Sox里面也包括一些DSP算法，有兴趣的朋友可以下载回去研究。Sox可以用于任何用途。但是发布源代码时必须包括版权声明，发布二进制代码必须声明作者。-Sox is the most famous Open Source voice file format conversion tools. Has been widely transplant Dos, windows, OS2, Sun, Next, Unix, Linux and other operating system platforms. Sox was founded by Lance Norskog, later many developers have gradually improved. can now support multiple audio files formats and audio processing effects. Basically common formats can support. More useful is that the Sox to a voice filter, the sampling frequency conversion, This voice of those engaged in the development or maintenance of platforms friends very useful. Of course, the Sox also including some inside DSP Algorithm, interested friends can be downloaded to go back and study. Sox can be used for any other purpose. However, the distribution of source code must include the copyright statement released bina
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Size: 312208 |
Author: 李增军 |
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Description: 论文关于语音滤波的IEEE文章，不错大家好好学习以下吧-voice filter papers on the IEEE article, it is true we can learn from it the following
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Size: 342652 |
Author: zhangjunyan |
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Description: Sox是最为著名的Open Source声音文件格式转换工具。已经被广泛移植到Dos、windows、OS2、Sun、Next、Unix、Linux等多个操作系统平台。Sox项目是由Lance Norskog创立的，后来被众多的开发者逐步完善，现在已经能够支持等多种声音文件格式和声音处理效果。基本上常见的声音格式都能够支持。更加有用的是，Sox能够进行声音滤波、采样频率转换，这对那些从事声讯平台开发或维护的朋友非常有用。当然，Sox里面也包括一些DSP算法，有兴趣的朋友可以下载回去研究。Sox可以用于任何用途。但是发布源代码时必须包括版权声明，发布二进制代码必须声明作者。-Sox is the most famous Open Source voice file format conversion tools. Has been widely transplant Dos, windows, OS2, Sun, Next, Unix, Linux and other operating system platforms. Sox was founded by Lance Norskog, later many developers have gradually improved. can now support multiple audio files formats and audio processing effects. Basically common formats can support. More useful is that the Sox to a voice filter, the sampling frequency conversion, This voice of those engaged in the development or maintenance of platforms friends very useful. Of course, the Sox also including some inside DSP Algorithm, interested friends can be downloaded to go back and study. Sox can be used for any other purpose. However, the distribution of source code must include the copyright statement released bina
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Size: 312320 |
Author: 李增军 |
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Description: 不错的FIR滤波器源代码！-good FIR filter source code!
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Size: 138240 |
Author: 刘平 |
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Description: 用LPC213X系列的PWM来产生语音。语音数据来源于PC机转换后得到的。最大可以是16位的。
由于是PWM所以输出端口需要低通滤波才能够得到比较干净的声音。
可以经过修改使用LPC2132以上的ARM的D/A来产生。
This project outputs audio on PWM0 using wave samples that are stored in the on-chip Flash ROM. -LPC213X series used to generate the PWM voice. Voice data from the PC after the conversion. It is the largest of 16. Because there is PWM output ports need low-pass filter can be relatively clean voice. It has been modified over the use LPC2132 ARM D/A to produce. This project outputs audio on PWM0 using wave samples that are stored in the on-chip Flash ROM.
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Size: 63488 |
Author: 徐勇 |
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Description: EVM板话音扰码器，使用-D命令展开 C54X带通滤波器汇编文件 -EVM voice scrambler, use-D start C54X order band-pass filter compilation
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Size: 21504 |
Author: 张智 |
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Description: 关于带通带阻滤波器的设计，巴特沃思滤波器
压缩包里有：
bendpass.m 为带通程序
bendblock.m 为带阻程序
SE144.wav 采用的声音文件，请放在 C:\下
bendpass.wav 通过带通滤波器之后的声音，保存在C:\ bendpass.wav
bendblock.wav 通过带阻滤波器之后的声音，保存在C:\ bendblock.wav
-on Bandpass Filter with the design, Butterworth filter bag is compressed : bendpass.m for band-pass procedures for bandstop bendblock.m procedures SE144.wav the voice file, on the C : \ under bendpass.wav through bandpass filter after the voice preserved in C : \ bendpass.wav bendblock.wav band stop filter through after the voice preserved in C : \ bendblock.wav
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Size: 464896 |
Author: 旺铺 |
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Description: 自适应滤波器
滤除声音回波
起到声音回波抵消作用-adaptive filter to filter voice echoes voices echo canceller play role
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Size: 102400 |
Author: |
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Description: 语音信号通过滤波器分析，matlab程序编写-voice signal through the filter, Matlab programming
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Size: 1057792 |
Author: jingjingbit |
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Description: matlab编写，求mel滤波器矩阵的系数-Matlab prepared for mel filter coefficient matrix
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Size: 2048 |
Author: wh |
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Description: 语音滤波源代码，基于视频会议中语音编码前滤波，可以有效地消除噪声-voice filtering source code, based on video conference speech coding filter can effectively eliminate noise
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Size: 384000 |
Author: 黄河 |
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Description: 声音过滤器工具大集合.-Voice filter large collection of tools.
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Size: 660480 |
Author: 不告诉 |
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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.
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Size: 2048 |
Author: yim |
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Description: Using the function to sample the voice signal and achieve fast Fourier transform in MATLAB, and then get the signal characteristics of the spectrum.Filtering the signal from the filter,and then playback the signal of voice
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Size: 72704 |
Author: 林霞 |
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Description: 本次课程设计是用麦克风采集一段语音信号并对该语音信号进行滤波去噪处理。课程设计平台为MATLAB7.1。设计步骤为：首先采集一段语音信号并观察其频谱，然后设计一个汉明窗FIR滤波器，最后对该信号进行滤波。信号在进行滤波处理后，能够听到滤波后的语音信号和滤波前相比明显的变得清晰，基本达到了设计目的。-The curriculum design is a section of a microphone speech signal acquisition and filtering of the speech signal de-noising processing. Curriculum design platform MATLAB7.1. Design steps: First, a section of speech signal acquisition and to observe its spectrum, and then the design of a Hamming window FIR filter, the final filtering of the signal. Signal during the filtering processing, to hear the voice signal after filtering and filtering obvious ago became clear, the basic design to achieve the purpose.
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Size: 96256 |
Author: 球球 |
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Description: 分析了IP 电话系统中回声的特点, 并针对其回声具有路径延迟、延迟抖动大等特点, 提出
了一种基于延迟分析的自适应回声消除器设计的新思想和相关算法。介绍了自适应回声消除器的
基本结构及各个功能模块, 例如: 变阶数自适应滤波器, 语音状态检测器, 延迟分析器和NLM S 算
法控制器. 本文还详细论述了以TM S320C6201DSP 为硬件平台来实现能运用在IP 电话网关中的
自适应回声消除器及其在IP 电话系统中的连接关系。在文章的最后, 列举了对自适应回声消除器
的性能测试结果, 并分析了自适应回声消除器的优点, 证明了自适应回声消除器能很好地解决IP
电话系统中的回声消除问题。-Analysis of the IP telephone system in the characteristics of echo, and echo with a path for its delay, delay jitter features and so on, the
A delay based on the analysis of adaptive echo cancellation design of the new ideas and related algorithms. Introduced the adaptive echo cancellation device
The basic structure and the various functional modules, such as: change the order of adaptive filter, voice status detector, the delay count analyzers and NLM S
Law controller. This article also discussed in detail in TM S320C6201DSP to achieve for the hardware platform to use in the IP Phone Gateway
Adaptive echo cancellation and its IP phone system in the connections. Finally, listed on the adaptive echo canceller
The performance test results and analysis of the adaptive advantages of echo cancellation devices to prove that the adaptive echo canceller can be a good solution to IP
Telephone system in the echo cancellation problem.
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Size: 259072 |
Author: wlch |
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Description: 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.
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Size: 4096 |
Author: 周峰 |
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Description: 语音信号的数字滤波 fir数字滤波器的海明窗函数设计-Voice signal digital filter fir digital filter design of the Hamming window function
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Size: 393216 |
Author: 周小川 |
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Description: 基于MATLAB的语音滤波器，通过MATLAB使原来的语音信号分别进行带通，高通，低通滤波，得到更好的效果-MATLAB-based voice filter through the MATLAB so that the original speech signal, respectively, for band-pass, high pass, low pass filter, to get better results
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Size: 2048 |
Author: 柳沐璇 |
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Description: 实现语音信号通过低通滤波器后，得到频谱，并且输出滤波前后的图形。-Voice signal through a low pass filter to obtain a frequency spectrum, and outputs the graphic before and after filtering.
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Size: 77824 |
Author: ant |
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