Description: Levison-Durbin 语音信号处理中的线性预测编码LPC 理论、格型滤波器以及求解现行预
测方程的算法，可以实现对语音信号重要元素的分析、合成甚至识别。
基于现有的实验平台，我们可以利用 Matlab 函数来获得几个固定语音元素（如元音）
的模型系数，LPC 得到的系数组成 IIR 滤波器。利用冲击脉冲
序列作为输入，我们就可以得到原来的语音。这是一种简单的语音合成功能。-Levison-Durbin speech signal processing in linear predictive coding LPC theory, lattice filters, as well as the current prediction equation solving algorithm, can achieve an important element of the speech signal analysis, synthesis or recognition. Based on the existing experimental platform, we can use Matlab function to obtain the number of fixed-voice elements (such as vowels) model coefficients, LPC coefficients are the composition of IIR filters. Shock pulse sequence used as input, we can get the original voice. This is a simple voice synthesis. Platform: |
Size: 283648 |
Author:Ender Lee |
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Description: 录制自己的一段语音，时间控制在15秒到30秒左右；利用wavread函数对自己的语音进行采样，记住采样频率。
（1）画出原始语音信号的时域波形，而后以1秒为间隔，求出每秒数据的功率谱。
（2）根据语音信号特点，分别设计FIR及IIR滤波器，分别画出滤波器幅频和相频特性曲线。用设计的滤波器对信号滤波，画出滤波后时域波形。用sound函数回放语音信号。
（3）求出特征频段语音信号随时间变化的曲线（每间隔0.05秒求一次功率谱）。
-Record your own voice for some time (15 seconds to 30 seconds) use wavread function of the voice samples, remember the sampling frequency. (1) Draw the original speech signal in time domain waveform, and then to 1 second intervals, the data obtained power spectrum per second. (2) According to the voice signal characteristics, FIR and IIR filters were designed, namely to draw the amplitude and phase frequency characteristic curve. With the design of the filters on the signal filtering, to draw the filtered time-domain waveform. Playback with sound function of the speech signal. (3) Find the characteristic band speech signal curve over time (0.05 seconds intervals seeking a power spectrum.) Platform: |
Size: 2048 |
Author:J |
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Description: 以MATLAB软件为工具，在GUI图形用户界面下针对不同特点的语音信号进行八种不同模式的滤波处理的语音信号处理系统，涉及基于巴特沃思滤波器的IIR滤波器和汉明窗设计的FIR滤波器，能够实现语音文件的打开及自定义路径的存储功能，同时可以实现语音信号加噪和音频倒放功能。-The source code, by means of MATLAB, under the Graphical User Interface, achieves a kind of speech signal processing system with eight different modes of speech signal filters working according to different characteristics of speech signal. The system involves IIR filter based on Butterworth analog filter and FIR filter based on Hamming Window, is able to open and save audio files in the user-defined place and realizes functions of noise addition and audio back-play. Platform: |
Size: 19456 |
Author:李小 |
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Search - filters in speech processing