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a
- 对语音信号进行处理，做fft，语谱图，短时fft，并进行对比-The speech signal processing, do fft, spectrogram, short fft, and compare
fft
- fft exemple. sistem speech recognition
meierdaopufadematlabshixian
- 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
sound_analysis
- 语音信号的频谱分析,包括采样语音的FFT变换,及倒频谱分析.-Speech signal spectrum analysis, including FFT, voice samples, and cepstrum analysis.
Matlab-spectrumananalysis
- 该文给出了一种利用matlab系统实现频谱分析与显示的方法。该方法对语音信号进行基于FFT的短时频谱分析，频谱图的伪彩色映射及显示。-This paper presents a system implementation using matlab spectrum analysis and display methods. The method is based on the speech signal short-time FFT spectrum analysis, spectrum and
fft
- 介绍FFT算法，里面是源代码，还包括语音处理其他算法-Introduction FFT algorithm, which is the source code, including other speech processing algorithms
idbm
- Descr iption Applies ideal binary mask to noise corrupted speech signal. The processing takes place within an FFT based short-time spectral analysis-modification-synthesis framework. The ideal binary mask is computed from an oracle (true) sign
yuyinjiazhongFFT
- 语音信号处理来的预加重随后进行FFT变换，同时呈现结果图。-Speech signal processing preemphasis subsequent FFT, simultaneous presentation of the results in Fig.
shiyupinyu
- 语音信号的时域波形和频域波形，简单易懂，利用FFT实现。另加语音端点检测文献一篇-Speech signal time domain waveform and frequency domain waveform, straightforward, using FFT to realize
yuyinxinhaochuli
- 录制一段语音 （1）对其进行时频域分析 （2）加随机噪声，并对含噪语音进行时频域分析 （3）设计滤波器对含噪语音进行滤波（wavread,fft,awgn或randn, filter等） -Frequency domain analysis (2) plus random noise, and frequency-domain analysis with noisy speech (3) design filters for filtering noisy speech (wav
FFT
- 在语音识别，雷达信号处理生物医学信号检测与识别等应用领域广泛使用基于离散傅里叶变换（FFT快速算法）的频谱分析技术。-Widespread use of spectral analysis technique based on discrete Fourier transform (FFT fast algorithm) in speech recognition, radar signal processing biomedical signal detection and recogniti
fanling
- ofdm系统仿真 含16qam调制 fft 加窗 加cp等模块，完整的基于HMM的语音识别系统，使用起来非常方便。- ofdm system simulation including 16qam modulation fft windowing modules plus cp, Complete HMM-based speech recognition system, Very convenient to use.
speech-enhanceme
- 谱减法语音增强的仿真实现：读取语音文件，产生随机白噪声，将两者联合起来，加入汉明窗，帧间重叠50 ，短时FFT，短时相位谱，频域中合成语音，短时IFFT并各帧重叠相加，去除汉明窗引起的增益，最后得到增强后的语音。-Spectral subtraction speech enhancement Simulation: reading voice files, generate random white noise, the two unite Jiaruhanming window, 50 ov
qanjeng
- 使用起来非常方便，基于kaiser窗的双谱线插值FFT谐波分析，语音信号的采集与处理，数字信号处理课设。- Very convenient to use, Dual-line interpolation FFT harmonic analysis kaiser windows, Acquisition and Processing of the speech signal, digital signal processing class-based.
taiseng
- 语音信号的采集与处理，数字信号处理课设，matlab程序运行时导入数据文件作为输入参数，基于kaiser窗的双谱线插值FFT谐波分析。- Acquisition and Processing of the speech signal, digital signal processing class-based, Import data files as input parameters matlab program is running, Dual-line interpolation FFT
code
- Cep strum Computation This MATLAB exercise compares two different methods of cepstrum analysis of a finite duration frame of speech, namely the conventional method based on FFT
qaopui
- ofdm系统仿真 含16qam调制 fft 加窗 加cp等模块，语音信号的采集与处理，数字信号处理课设，对信号进行频谱分析及滤波。- ofdm system simulation including 16qam modulation fft windowing modules plus cp, Acquisition and Processing of the speech signal, digital signal processing class-based, The signal spe
heikiu
- 基于kaiser窗的双谱线插值FFT谐波分析，完整的基于HMM的语音识别系统，数值分析的EULER法。- Dual-line interpolation FFT harmonic analysis kaiser windows, Complete HMM-based speech recognition system, EULER numerical analysis method.
MATLAB? – A Tutorial [FFT]
- MATLAB features a family of application-specific collections of functions called toolboxes. These extend the MATLAB environment in order to solve particular classes of problems. Areas in which toolboxes are available include signal processing, con
Desktop
- 通过matlab设计的GUI界面实现语音信号的实时录入和频谱分析和滤波等处理，并能够显示各阶段时域波形及频谱图，滤波器种类及参数可调(Through the GUI interface designed by matlab, the real-time input of speech signal, spectrum analysis and filtering are realized, and the time-domain waveforms and spectrograms of eac