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WDS18B20
- DS18B20采样得到的温度值存入72H--75H中,再传送到SAA1064驱动的四位数码管上显示.-DS18B20 temperature value sampled into 72H- 75H, the SAA1064 driver then transmitted to the four digital tube display.
Proceedings_1993
- 进行采样语音信号，计算出基频，从而判断出性别。-Sampled speech signal to calculate the fundamental frequency, which determine gender.
phase_shifted_sampled
- reflectivity group delay and dispersion of a sampled fiber Bragg grating with periodic phase shifts at each sampling period
ditong_fir
- 在2812上调试通过的，绝对好用的fir滤波程序，对采样电流进行滤波，等到理想-Adopted in 2812 on the debugging, the absolute ease of use of fir filtering process to filter the sampled current, until the ideal
clc
- 数字滤波器 ,三级数字滤波器，对采样数据执行加减等操作。-Digital filter, three digital filter, the sampled data implementation of addition and subtraction and other operations.
MATLAB
- 基于 MATLAB 的语音信号分析与处理的课程设计.录制一段自己的语音信号，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，采用窗函数法或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语音信号进行滤波，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-MATLAB-based voice signal analysis and pr
step_10m
- system identification achieved for data sampled at 10 ms through m file
6.10
- Linux DSP Program that treates sampled audio
caiyangdingli
- 使用每秒10个采样的频率 ，对信号 进行采样，画出原始信号的傅立叶变换X(f) 和理想采样信号的傅立叶变换 。假定重构滤波器是带宽为 的理想低通滤波器，具有带通增益 ，画出重构滤波器的输出-Using 10 samples per second frequency, the signal is sampled, to draw the original signal Fourier transform X (f) and the ide
THS1206
- 运用verilog对ad1206芯片进行控制，对信号进行采样的信号，已通过测试-Using verilog on ad1206 chip control, the signal is sampled signals, has been tested
Cwave
- 音频文件处理程序。DOS界面，功能包括打开所给音频文件，显示文件信息，并可对该文件进行采样，插值等，从而改变语音质量。-Audio file processing. DOS interface, features include open the given audio file, display file information, the file can be sampled, interpolation, so as to cha
05-music-coding
- Music Coding LPC-based codecs model the sound source to achieve good compression. Works well for voice. Terrible for music. What if you can’t model the source? Model th
decomp_reconst_W
- Decompose image into subbands, denoise using BLS-GSM method, and recompose again. fh = decomp_reconst(im,Nsc,filter,block,noise,parent,covariance,optim,sig) im: image Nsc: number of scales filter: type
sinfapm
- 该函数sinfap.m评估频率，振幅，相位和平均均匀采样谐波信号值。-The function sinfap.m evaluates frequency, amplitude, phase and mean value of a uniformly sampled harmonic signal
SimulinkFSK
- 题目要求： 1.录制一段自己的语音信号，并对录制的信号进行采样； 2.画出采样后的语音信号的时域波形和频谱图； 3.给定滤波器的性能指标，采用窗函数法和双线性变换法设计滤波器， 并划出滤波器的频域响应； 4.用该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； 5.回放语音信号； 6.设计一个信号处理系统界面。-Subject to
F2812-MixerFIR
- 1．掌握A/D 转换的基本过程和程序处理过程； 2．学习通过对采样值进行计算产生混频波形。 3．熟悉FIR 滤波器及其参数的调整。-1. Master A/D conversion process of the basic processes and procedures 2. Learning through the computed values of the sampled wave mixing. 3. Familia
T-REC-G.711-200911-I!Amd2!SOFT-ZST-E
- G.711使用64Kbps的带宽，可将14bits转换成8bits。目前G.711有两个编码方式，A-law以及Mu-law。-G.711 defines two main compression algorithms, the µ -law algorithm (used in North America & Japan) and A-law algorithm (used in Europe and the rest of
T-REC-G.718-200806-I!!SOFT-ZST-E
- 8-32 kbit/s语音和视频的结构误差不稳定窄带和宽带嵌入式变数率编码-The G.718 encoder can accept wideband sampled signals at 16 kHz, or narrowband signals sampled at either 16 or 8 kHz. Similarly, the decoder output can be 16 kHz wideband, in additio
convtrapz
- 快速卷积算法的matlab实现，以及对应的文章。-(Fast) Approximation of the convolution integral between two sampled functions.
s
- 采样数据导入后，实现信号的频谱分析，采用基2FFT实现，能处理1024点采样数据，界面交互友好-Sampled data import, the realization of the signal spectrum analysis, using the base 2FFT implementation can deal with 1024 points sampled data, interactive and friendly in