Description: 1．了解数字信号处理系统的一般构成；
2．掌握奈奎斯特抽样定理。
为方便实现，实验中更换了一种表现形式，即抽样频率固定（10KHz），通过改变输入模拟信号的频率来展示低通抽样定理。我们可以通过研究抽样频率和模拟信号最高频率分量的频率之间的关系，来验证低通抽样定理。-1. Understanding digital signal processing system of the general composition; 2. Master Nyquist Sampling Theorem. To facilitate the realization of the experiment to replace a form of expression, that is, fixed sampling frequency (10KHz). By changing the input signal frequency to demonstrate low-pass sampling theorem. We can, through research and sampling frequency of the highest frequency analog signal frequency components of the relationship between to verify low-pass sampling theorem. Platform: |
Size: 3072 |
Author:深蓝 |
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Description: 本程序将指定的16K采样的语音数据文件转换为经G.723编解码后的8K语音数据。降采样前先使用180阶的FIR滤波器对语音数据进行频率压缩，然后进行抽取，并对抽取的数据进行G.723编解码。该程序在非特定语音识别的库文件处理中使用，也可扩展至其他用途。-this procedure will be designated the 16K sampling voice data files converted to G.723 codecs by the 8K words Music data. Sampling down before use 180 bands FIR filter frequency voice data compression, then proceed to collect, also collected data G.723 codecs. The procedures in non-specific voice recognition for the use of document processing, and can be expanded to other uses. Platform: |
Size: 1675264 |
Author:王小飞 |
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Description: 虚拟声卡频率计：
1.请使用支持44100Hz采样频率的声卡。 2.频率测量范围：1Hz—16384Hz。 3.采样时间：1秒（S）。 4.被测信号从声卡线路输入端的左或右声道，或两个声道同时输入，也可从话筒端输入。-Virtual sound card frequency meter: 1. Please use the support 44100Hz sampling frequency of the sound card. 2. Frequency measuring range: 1Hz-16384Hz. 3. Sampling time: 1 seconds (S). 4. The measured signal from the sound card line input left or right channel, or two-channel input, or from the microphone input. Platform: |
Size: 304128 |
Author:daki |
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Description: 演示采样定理，时域采样，频谱周期延拓同时演示采样频率小于2*fc时，产生的混叠效应。可作函数应用-Demonstration sampling theorem, time-domain sampling, the spectrum at the same time cycle extension demonstration sampling frequency is less than 2* fc, the resulting aliasing effect. Function can be used for applications Platform: |
Size: 2048 |
Author:eachgear |
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Description: 本文首先概述了OFDM 的原理、特征和关键技术，介绍了系统需要的几种同步。然后就同步误差对系统性能的影响进行了简单的分析，表明定时偏差和采样频率偏差将会引入码间干扰，载波频率偏差导致载波间干扰的发生。因此，在OFDM系统接收机中定时的确定、载波频率偏差和采样频率偏差的纠正是信号恢复的关键。之后对OFDM系统同步的几种实现方法进行了介绍，分析了各同步方法的性能。-The paper first outlines the principles of OFDM, features and key technologies, the system needs to introduce several synchronization. Synchronization error and then the impact on system performance for a simple analysis, show that the timing deviation and the sampling frequency deviation will introduce inter-symbol interference, the carrier frequency deviation led to the occurrence of ICI. Therefore, in OFDM system receiver to determine the timing, carrier frequency offset and sampling frequency offset correction signal to restore the key. After the OFDM system to achieve synchronization of several methods of introduction, an analysis of the performance of the synchronization method. Platform: |
Size: 268288 |
Author:dajinyu |
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Description: this program uses an impulse train, takes formant frequencies and filters them. output signal is synthesized sound.
commented part ,controls time=d, sampling frequency=fs
normal frequency=f
and bits=n, then synhesizes the output sound Platform: |
Size: 1024 |
Author:G眉ven |
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Description: Script for performance measurements for Carrier and Sampling Frequency
Offset Correction in OFDM Systems Platform: |
Size: 1024 |
Author:Ali |
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Description: We have to design a second order Butterworth filter having a sampling frequency of 20 kHz and -3db cut of frequency at 2 kHz in visual C. Only low frequency signals can pass through this filter and the signals having the frequency higher than frequency called cut off frequency get attenuated by this filter. we will use bilinear transform which is one of the most efficient way of converting a analogue filter to digital filter, we will convert the given transfer function from Laplace transform to z-transform and then we will realize poles and zeroes in the z-domain. And their corresponding frequency response.-We have to design a second order Butterworth filter having a sampling frequency of 20 kHz and-3db cut of frequency at 2 kHz in visual C. Only low frequency signals can pass through this filter and the signals having the frequency higher than frequency called cut off frequency get attenuated by this filter. we will use bilinear transform which is one of the most efficient way of converting a analogue filter to digital filter, we will convert the given transfer function from Laplace transform to z-transform and then we will realize poles and zeroes in the z-domain. And their corresponding frequency response. Platform: |
Size: 447488 |
Author:Mak_m |
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Description: 巴特沃斯低通滤波器的MFC程序，FS是采样频率，fp是截止频率，大家-Butterworth low-pass filter MFC process, FS is the sampling frequency, fp is the cut-off frequency, we take a look at Platform: |
Size: 40960 |
Author:金隆 |
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Description: 改变原始wav音频信号的抽样频率，并且显示处理前后声音信号的频谱.有助于帮助大家理解奈圭斯特抽样定理。-Change the original wav audio signal sampling frequency, and shows the spectrum of sound signal before and after treatment. Helps to help you understand the Nyquist sampling theorem. Platform: |
Size: 143360 |
Author:leo |
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Description: 设计一个IIR数字低通滤波器， 逼近一组模拟滤波器的指标参数（通带截止频率Wp=2*pi*2000rad/s,阻带边界频率Ws=2*pi*3000rad/s,通带波纹 Rp=3db, 阻带衰减Rs=15db, 采样频率f=10000Hz）； 分别用脉冲响应不变法和双线性变换法实现设计，列出传递函数并描绘模拟和数字滤波器的幅频和相频响应曲线。用上述设计滤波器完成几组给定信号的滤波，证明滤波器的有效性和滤波范围限制.-Design a IIR digital low-pass filter, approaching a group of analog filters indicator parameters (pass-band cut-off frequency Wp = 2* pi* 2000rad/s, stop-band edge frequency Ws = 2* pi* 3000rad/s, pass-band ripple Rp = 3db, stop-band attenuation Rs = 15db, the sampling frequency f = 10000Hz) were used to change the impulse response method and bilinear transformation method for design, list and describe the transfer function of analog and digital filter amplitude-frequency and phase-frequency response curve. A filter was designed with the completion of the above-mentioned groups of a given signal filtering, to prove the effectiveness of filters and filter limit Platform: |
Size: 71680 |
Author:刘文珍 |
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Description: Multirate滤波器设计。模拟带通信号频率范围1KHz—1.1KHz，设计一个采样频率8KHz数字频谱分析系统，要求频率分辨率0.1Hz。-Multirate filter design. Analog bandpass signal frequency range 1KHz-1.1KHz, design a sampling frequency of 8KHz digital spectrum analysis system, required frequency resolution of 0.1Hz. Platform: |
Size: 37888 |
Author:曾茉 |
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Description: 通过AD采样测量频率，输出到数码管显示，测量低频效果并不好-Measured by AD sampling frequency, the output to a digital display, measuring low-frequency effect is not good Platform: |
Size: 334848 |
Author:ljx |
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Description: ECGSYN generates a synthesized ECG signal with user-settable mean heart rate, number of beats, sampling frequency, waveform morphology (P, Q, R, S, and T timing, amplitude,and duration), standard deviation of the RR interval, and LF/HF ratio (a measure of the relative contributions of the low and high frequency components of the RR time series to total heart rate variability). Platform: |
Size: 4096 |
Author:Ma |
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Description: For a function that varying with time s(t) be a continuous function to be sampled, and sampling be performed by measuring the value of continuous function every T seconds, which is called sampling interval. This sampled function is given by S(nT),for integer values of n. Sampling frequency or sampling rate fs is defined as number of samples obtained in one second,fs=1/T. Nyquist sampling theorem provides a description for nominal sampling interval required to avoid aliasing. It may be stated simply as sampling frequency should be at least twice the highest frequency contained in a signal. Platform: |
Size: 10240 |
Author:Shahbaz Badr |
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Description: 在不同采样频率下，对已知信号进行采样和滤波，本文附录了MATLAB程序及仿真结果。-At different sampling frequency, the sampled signal is sampled and analyzed, and the MATLAB program is introduced in this paper. Platform: |
Size: 129024 |
Author:刘麒麟 |
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Description: 给出原始信号，用不同采样频率对信号进行采样，得出波形图-Given the original signal, with a different sampling frequency signal is sampled waveform obtained Platform: |
Size: 97280 |
Author:zzm |
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Description: 分析带通采样定理的原理和实现过程，说明带通信号重构原始信号的方法，仿真给出不同采样频率和低通采样性能对比-By analyzing the principle and implementation process of the bandpass sampling theorem, the method of reconstructing the original signal with the signal is explained, and the performance comparison of different sampling frequency and low pass sampling is given. Platform: |
Size: 808960 |
Author:金南城 |
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Search - sampling frequency