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 Platform: |
Size: 464896 |
Author:旺铺 |
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Description: 自己编写GUI界面实现语音增强，可在main.c中点击菜单debug中的run便可以运行程序，可分别实现谱相减、最小均方和维纳滤波语音增强-GUI interface to prepare themselves to achieve enhanced voice, in main.c which debug menu click the run will be operating procedures, can be achieved spectral subtraction, both the smallest and the Wiener filter speech enhancement Platform: |
Size: 97280 |
Author:Richard |
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Description: 本程序界面实现的是在MATLAB下的语音信号处理,采用是巴特沃斯低通滤波器-The program interface is in the MATLAB realize under the voice signal processing, is the use of Butterworth low-pass filter Platform: |
Size: 4096 |
Author:chengdu |
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Description: 这是本人编写的一个利用维纳滤波实现语音增强的程序.在输入语音信噪比不是很低的情况下,效果不错.-This is one I prepared realize the use of Wiener filter speech enhancement procedure. In the input voice signal to noise ratio is not very low circumstances, good results. Platform: |
Size: 1024 |
Author:李茉 |
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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 |
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Description: 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. Platform: |
Size: 6144 |
Author:张文斌 |
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Description: 本程序是提取语音MFCC参数的必经步骤，是求出mel三角滤波器器组的参数，网上只有matlab程序，没有C程序，本人花费两个星期的时间才转换成功，经DSP 的CCS环境调试成功。-This procedure is the extraction of voice MFCC parameters necessary step is to derive the triangular mel filter device group parameters, on-line only matlab procedures, no C program, I spent two weeks time to convert the success of the CCS environment by DSP debugging success. Platform: |
Size: 1024 |
Author:何正明 |
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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 Platform: |
Size: 72704 |
Author:林霞 |
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Description: ：由于许多传统的去噪方法在强背景噪声情况下提取声音信号的能力变弱甚至失效， 提出
应用独立成分分析( I C A) 方法对声音信号进行特征提取， 并证明了这种 I C A 变换能增强语音和音
乐信号的超高斯性． 在此基础上， 应用 I C A基函数作为滤波器， 通过阈值化的去噪方法对含有强高
斯背景噪声的声音信号进行去噪仿真实验． 结果表明， 本方法明显优于传统的均值滤波和小波去噪
方法， 为强背景噪声下弱信号的检测提供 了新的途径．-: As many of the traditional de-noising method in case of strong background noise, the ability to extract the voice signal even weaker failure, the application of independent component analysis (ICA) method of voice signal feature extraction, and prove that this transformation can be enhanced voice ICA and music of super-Gaussian signals. On this basis, the application of ICA basis function as a filter, through the threshold of the denoising method of Gaussian background noise contains strong voice signal denoising simulation. The results show that this method is obviously superior to the traditional mean filtering and wavelet denoising methods for the strong background noise under the weak signal detection provides a new way. Platform: |
Size: 212992 |
Author:金振东 |
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Description: The purpose of this project is to create a different system for processing voice signals and from which we draw useful conclusions about these brands.
More specifically, the system will first filter the voice signal with a pass bandwidth (chebyshev probably type 2) to reduce the presence of noise in it and then we can calculate:
• what limits the signal word out (segmentation)
• how are these words
• the characteristic frequency of the voice of the speaker of each file (voice pitch). -The purpose of this project is to create a different system for processing voice signals and from which we draw useful conclusions about these brands.
More specifically, the system will first filter the voice signal with a pass bandwidth (chebyshev probably type 2) to reduce the presence of noise in it and then we can calculate:
• what limits the signal word out (segmentation)
• how are these words
• the characteristic frequency of the voice of the speaker of each file (voice pitch). Platform: |
Size: 2048 |
Author:Pastel87 |
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Description: 录制一段个人的语音信号，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，采用双线性变换法和窗函数法设计滤波器，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 personal voice signal, and recording the signal is sampled draw sampled speech signal time-domain waveform and frequency spectrum for a given filter performance, the bilinear transformation and design of filter window functions and draw the frequency response filter then filter of their own design collection of the signal filtering, to draw the filtered signal in time domain waveforms and frequency spectrum, and filtering the signal before and after comparison, analysis of signal changes playback audio signals Finally, a signal processing system interface design. Platform: |
Size: 6144 |
Author:liudan |
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Description: 对录音信号集 中的某一语音，利用BATLAB设计一美尔例谱算法，并实现。
取信号集 中的一个语音信号：“xxxxxx”，将它作为输入的语音信号来为设计一个美尔倒谱算法，在该算法中，主要设计了以下环节：
1.读入一个语音信号；2.对这个信号归一化；3.对归一化的信号进行加窗处理（这里的矩形窗长度必须为257，重帧长64）；4.进行预加重处理，即通过一个高通滤波器： ；5进行512点的FFT；6.分别取模平方得到功率谱；7.在设计的mel滤波器组中，我采用了25个带通滤波器；8.将得到的功率谱信号通过mel滤波器组，得到相应的25个功率值；9.对这些功率值取自然对数；10.对这些值取离散作弦变换；11.将得到的值去掉直流分量，取其它值作为MFCC参数。
-Concentrated on recording a voice signal, using BATLAB designed Mayer cases spectral algorithm, and implementation.
Get a voice signal signal focus: "xxxxxx", will it as input for the design of a voice signal to Mel Cepstrum algorithm in the algorithm, the main design of the following links:
1. Read a speech signal 2. On the signal normalized 3. Of the normalized windowed signal processing (here, rectangular window length must be 257, re-frame size 64) 4. For Pre-emphasis processing through a high-pass filter: 5 for 512 points of FFT 6. modulo square were obtained spectrum 7. in the design of mel filter banks, I use a 25 band-pass filter 8 . will be the power spectrum signal by mel filter bank, corresponding to the 25 power values 9. on the power to take the natural logarithm value 10. to take these values as strings of discrete transform 11. will be of value to out DC component, to take other values as MFCC parameters. Platform: |
Size: 13312 |
Author:赵欣 |
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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-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 |
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Description: 粒子，粒子滤波，并且又在matlab中进行语音识别，-Particles, particle filter, and another for voice recognition in matlab, Platform: |
Size: 10240 |
Author:张丽 |
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Description: Developing an Isolated Word Recognition System
in MATLAB.
Speech-recognition technology is embedded in voice-activated routing systems
at customer call centres, voice dialling on mobile phones, and many other
everyday applications. A robust speech-recognition system combines accuracy
of identification with the ability to filter out noise and adapt to other acoustic
conditions, such as the speaker’s speech rate and accent. Designing a robust
speech-recognition algorithm is a complex task requiring detailed knowledge of
signal processing and statistical modeling. Platform: |
Size: 670720 |
Author:fasterfox |
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Description: 基于MATLAB的语音信号分析及滤波(DSP课程设计说明书) 设计题目、内容及要求 课程设计的题目：基于MATLAB的语音信号分析及滤波。 课程设计的内容：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-Design based on MATLAB voice signal analysis and filtering (DSP curriculum design manual) subject, content and requirements for curriculum design Title: MATLAB-based analysis and filtering of the voice signal. Curriculum design: recording period of the individual' s own voice signal, and the recorded signal is sampled draw the sampled speech signal time-domain waveform and frequency spectrum to set the filter performance, using a window function method and double transform the design of filters, and draw the frequency response of the filter signal acquisition, filtering, and then use the filter of their own design to draw the waveform and spectrum of the filtered signals in time domain and filtering Platform: |
Size: 1449984 |
Author:Rokey_Niu |
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Description: 1）选择一个语音信号作为分析对象，或录制一段语音信号； 2）对语音信号进行采样，画出采样前后语音信号的时域波形和频谱图； 3）利用MATLAB中的随机函数产生噪声加入到语音信号中，使语音信号被污染，然后进行频谱分析； 4）设计用于处理该语音信号的数字滤波器，给出滤波器的性能指标，画出滤波器的频率响应； 5）对被噪声污染的语音信号进行滤波，画出滤波前后信号的时域波形和频谱，并对滤波前后的信号进行比较和分析； 6）回放各步骤的语音信号，给出相应处理程序及运行结果分析。-1) Select a voice signal as an analysis object, or record a voice signal 2) sampling the voice signal, draw the waveform and frequency spectrum of the time domain before and after sampling the speech signal 3) using the random function in MATLAB generated noise was added to the speech signal, the speech signal to be contaminated, and then spectrum analysis 4) for processing the speech signal, the digital filter design, given the performance of the filter to draw the filter' s frequency response 5) on the noise pollution of the speech signal is filtered, time-domain waveform and spectrum draw before and after filtering the signal before and after filtering, and the signal for comparison and analysis 6) playback of the speech signal for each step, given the results of the corresponding processing procedures and run analysis. Platform: |
Size: 2048 |
Author:张三 |
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Description: 1. 给一段原始的语音信号（可以是自己录制的一段语音），加上一频率为3.8kHz的高频余弦噪声和频率为3.6kHz的高频正弦噪声（幅度自己可以选择），用窗函数设计一滤波器（要求最小阻带衰减为50dB）对加噪后的语音信号进行滤波，画出滤波器的频率响应曲线，画出滤波前后的时域图和频谱图。
需要用到的函数：
fir1 用窗函数设计FIR滤波器的函数
2. 用GUI设计一界面（如图1所示）完成如下功能：
1） 输入一语音信号，画出语音信号的时域图和频谱图；
2） 对语音信号加噪处理，画出加噪后的时域图和频谱图；
3） 给定滤波器的性能指标，采用几种不同的方法设计滤波器对加噪信后进行滤波，画出各滤波器的频率响应曲线，画出滤波后的时域图和频谱图。
-1. to some of the original speech signal (which can be a voice you record yourself), plus a cosine frequency of 3.8kHz frequency noise and frequency of 3.6kHz frequency sinusoidal noise (in the range that they can choose), with a window function to design a filter (requires a minimum stop band attenuation is 50dB) after adding noise to the speech signal is filtered, the filter frequency response curve to draw, draw diagrams and time-domain spectrum before and after filtering. Need to use the function: function fir1 with window function design FIR filter design 2. Using a GUI interface (Figure 1) complete the following functions: 1) an input speech signal, the speech signal in the time domain to draw graphs and spectrum 2) adding noise to the speech signal processing, draw diagrams and time-domain plus noise spectrum after 3) to set the filter performance, using several different methods designed to filter letter after adding noise filtering, draw the frequency response curve of each Platform: |
Size: 1024 |
Author:longteng |
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Search - filter voice in matlab