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Denoise.zip
- Speech Signal Denoising,Speech Signal Denoising
FBS
- A short-time analysis-synthesis system for speech is developed in MATLAB using the Filter Bank Summation (FBS) method. Provision for different sampling rates at analysis and synthesis means rate change can be carried out.
encoder
- Implementation of a speech codec based on coding of speech at 8 kbit/s using conjugate-structure algebraic-code-excited linear-prediction (CS-ACELP) - We took .wav files that is sampled at 8000 Hz using 16-bit linear PCM. The encoding process i
soundchange
- 对原始语音进行上采样和下采样，分析对比他们的时域图和频谱图。最后对比原始语音、75HZ激励、150HZ激励以及噪声激励下的效果-SOME SIMPLE MANIPULATIONS OF SOUND USING DIGITAL SIGNAL PROCESSING The original sound and its spectrogram Downsampling the waveform downsampling Upsampling the waveform Li
fant.tar
- This tool can be used to - filter speech data with a frequency characteristic as defined by ITU for telephone equipment and/or - add noise to speech recordings at a desired SNR (signal-to-noise ratio)
harmonic
- 语音harmonic特征提取，先对语音信号分帧，然后对每一帧使用中心频率维基音频率的倍数的带通滤波器进行滤波，最后滤波结果的傅立叶变换的结果-Voice harmonic feature extraction, the first speech signal sub-frame, and then using the center frequency for each frame of Wiki sound frequency multiple of the band-pass filter f
GUI
- 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 voi
yanshou
- 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
Linear-Predictive-Coder-master
- Linear-Predictive-Coder MATLAB Implementation of LPC algorithm for speech signal # Why LPC? In communication systems it is often necessary to transmit audio(speech) signal in compressed or encoded form because of bandwidth limitation of the
MELBANKM
- 在DTW语音识别算法中经常会用到的在MEL频率下组成的若干个带通滤波器-DTW speech recognition algorithm is often used in the frequency of the MEL composed of a number of bandpass filter
speech-signal-filter
- 使用matlab语言对一段语音信号进行滤波处理，首先分析时域信号，之后进行傅立叶变换，转换成频域，使用巴特沃斯低通滤波器去除高频部分。-Use matlab language of a voice signal is filtered first time domain signal after the Fourier transform is converted into the frequency domain, using a Butterworth low-pass filter to