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sox-12.17.1.tar
- 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
DigitalEqualizer
- 圖形調整式數位音訊均衡器，使用matlab gui介面寫成。自動畫出頻率響應圖，可讀取.wav檔。濾波器由fdatools設計而成。-Digital audio graphic equalizer adjustments, and the use matlab gui interface language. Automatically draw the frequency response can be read. Wav file. Filter designed with the fdato
音频波形显示和滤波
- 音频波形显示和滤波,1、录音功能：CWaveIn类2、播放功能:CWaveOut类3、波形显示和滤波：CFilter类4、语音数据压缩解压：CG729aCompress、CGsmCompress 因为继承至同一个父类，所以它们的接口是一样的, Audio frequency profile demonstration and filter,1, recording function: CWaveIn kind 2, broadcast function: CWaveOut kind 3, pr
batfilter
- 0.4db 截止频率为10khz, 30khz处最小阻带衰减为50db, 抽样频率为100khz的巴特沃兹滤波器.-0.4dB cut-off frequency of 10khz, 30khz Department minimum stopband attenuation for 50db, sampling frequency of 100kHz the巴特沃兹filter.
winsinc
- The windowed sinc filter uses the inverse fourier transform of ideal low - pass filter frequency response. The Inverse transform of the ideal low - pass response is the sinc function (sinx/x). The inverse fourier transform of the required frequency r
movavg
- A moving average filter averages a number of input samples and produce a single output sample. This averaging action removes the high frequency components present in the signal. Moving average filters are normally used as low pass filters. In recursi
adaptivefilter
- 利用自适应陷波器对引入单频干扰的原始正弦信号进行恢复-The use of adaptive notch filter for the introduction of single-frequency interference for the restoration of the original sinusoidal signal
alie4
- This appendix gives the waveforms and FFTs of spectral content of the he file alien_voices.c. The frequency response curves were produced using the MATLAB Filter Design Toolbox
imcorrelation_33
- this s the code for high frequency emphasis filter
p02
- digital band-pass filter with a bandwidth of 100 Hz, mean frequency 250 Hz, 0.1 db
audio
- 对一段音频文件进行了一系列的操作，诸如设置一个低频滤波。-processing an audio signal, such as filtering it with a Lowpass frequency filter.
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
digitalfilter
- 设计产生一个连续信号，包含低频，中频，高频分量，对其进行采样，进行频谱分析，使用矩形窗设计不同特性的数字滤波器对信号进行滤波处理，分析所设计滤波器(画出了频率特性曲线)，并对信号进行滤波，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，同时设计出了一个友好的人机交互界面。-Designed to produce a continuous signal, including low frequency, medium frequency, high frequency componen
noise_filter
- its a training for students to add a noise in any audio and then filter that specific frequency.
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
imageConvertExNet
- KEY FEATURES Resolution of 230,000 Counts (Peak-to-Peak) Offset Drift: 5 nV/8C Gain Drift: 2 ppm/8C Line Frequency Rejection: >150 dB Buffered Differential Inputs Programmable Filter Cutoffs Specified for Drift Over Time Operogramm
Subband
- subband is band the frequency of particular filter it pass some desire frequency ,-subband is band the frequency of particular filter it pass some desire frequency ,
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