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i have one doubt. when A and B system are cascade, then if the system is linear time invariant system even A and B are interchanged, no change in result. quantizer is a nonlinear system. but even the sampling and quantization is interchanged the entire result is unchanged and same result (...)
When you sample a continous signal (a continous signal is something like sin(w*t), you can find its value for any time t), the signal exists only at the sampling instant (like t=1s, 2s, 3s, ...). Else the signal is not zero, but doesnt exist. The signal can be quantized, which means that the signal is now restricted to have a few voltage level
There are a few problems with your code... The best way to define a sinewave is: A=1; n=7; N=2^n; % simulation length fs=1e2; % sampling frequency t=(0:N-1)/fs; % time (sampling points) f=3*fs/N; % sinewave frequency x=A*sin(2*pi*f*t); % cw sinewave r=10; % number of bits of resolution R=2^R; % number of quantization levels q=fl
hi all i just need any one know a link for a site descripe natural sampling and flat topped sampling in details specially the equations thanks for your help
you have already implented an aliasing example. now you are sampling every 10th zero crossing of the sinusoid signal. for instance set interval now to 950 and you will see a sinus with a frequency of 50.
why do you want exactly to do? are you using a software ? Added after 7 minutes: why we need to sample a signal, conv,fft,and filter.. if anybody knows please tell me step wise why we need these parameters in signals,during signal analysis.... best regards if we need to
hi What is different sampling and modulation ?
Hi, all When I was trying to design a sampling and hold circuit as shown in the following figure, I met a problem about the opamp in the circuit. The question is that, is that ok if there is no input DC voltage bias for the opamp in the hold period? When I was simulating the circuit, the output is ok as expected, but is the simulation really cor
I think you need some basic understanding of ADC. First sampling and quantization then switch to concrete ADC architecture
hi friends. i am doing my final year project titled "FPGA Implementation of Pipelined 2D-DCT and quantization Architecture for JPEG Image Compression". I have completed the coding in verilog. I need help from you for preparing my project report. I am not able to find enough materials on internet. can any one who has done a similar project mail me s
Hai, i have one doubt. what is difference between normal sampling and area sampling. why we need area sampling?
Dear friends, I would like to ask for you help to come out with an accurate answer for the following two (2) tasks. 1- System analysis 2- sampling and reconstruction Thanks in advance!
Hi I need a simple code for sampling and reconstruction the signal with matlab Tnx
Hi, I hope you are not looking for DIRECT digital down conversion. I had used digital downconversion for a GPS receiver. It is fairly easy. you follow in digital all the process of analog. However you need to be careful about sampling and quantization levels. B R Madhukar
Dear all I need the following the paper, but i don't access to it, if you can ,pls send it for me. thanks you for your help. Chang,H., ?Presampling filtering,sampling and quantization effects on the digital matched filter performance? Proceeding of International telemetering conference,pp.889-915,San (...)
I need mathematical examples about the subjects in image processing: -filtering in the spatial domain. -filtering in the frequency domain. - Histogram equilization. -Image sampling and quantization. -Gray Level Transformations. Anyone can help me in that?
To convert an analog signal into digital signal two steps are required 1. sampling 2. quantization We can perform these operations in any order to get a digital output.But why it is common to do sampling first? Looking forward for reply. Thanx in advance.
Hi all What is the relationship between "sampling frequency" and "quantization noise" in audio signals?
such maximum value must be specified at the same time with the resolution. different resolution have different maximum speed achievble. In CMOS, 8-10bit pipelined ADC fs,max is around 200 MHz. Maximum bit per stage depends on the typical applications, but usually maximum 3.5b per stage. more bit per stage leads to slower sampling and large po
Correlated double sampling (CDS) is a particular case of auto-zeroing. You can refer to this paper: Enz, C.C.; Temes, G.C. ?Circuit techniques for reducing the effects of op-amp imperfections: autozeroing, correlated double sampling, and chopper stabilization? Proceedings of the IEEE Volume 84, Issue 11, Nov. 1996 Page(s):1584 - 1614
Hi all, Can someone please advise me, what does it mean by normalised sampling? How to perform sampling in the first place? I've tried to perform FFT in Excel. In Column A, I have input value time 't'=0, 1, 2, 3, 4, 5...., 62, 63 then in column B, i have rect(t)=1, 1, 1, 1, 1, 1,..., 0, 0 Next, i perform FFT based on rect(t) and (...)
Hi all, I am wondering for a 12-bit resolution ADC, how big of a sampling capacitor at the front end should be in order to surpress the kT/C noise and opamp thermal noise from the later stages? Peoples used 6pf before, I think it is way too big. Any ideas?
Hi all, What are the differences between bit rate and Max IF sampling in A/D? and how it is settle with the Nyquist criteria? Because I have seen a TI ADS5545 with 170Msps bit rate and 500MHz IF sampling frequency :?::!: Thank you all :D
Hi friends, I need to remember the answer of a question. But I cant handle the equation now. At BSC side of GSM speech transmits to BTS side at 260 bits over E1 and at BTS it has been coded to 456 bits and transmit over 20ms blocks so gross total we have a 22 kb/s data rate. We assume the sampling frequency (...)
Hi guys, I was wondering when you use jpeg or mpeg encoding softwares for encoding video, you vary two factors qf and qp. you write something like this -qf 10 .... and -qp 23 either you vary quantization parameter(qp) or quality factor(qf) My ? is is quality factor of 5 equal to quantization parameter of (...)
I am building an optical OOK (On off Key Ring) system without a carrier frequency. . Essentially, the presence of signal indicates one and lack of it indicates zero. I have a precise clock that is synced with transmitter. The
Hi dear all: it is because that periodical sampling would aliasing the high frequency noise into low frequency, so that the thermal noise should be integrated from 0 to unlimited high frequency, isn't it? But now my ADC is used in an image sensor and the sampling operation is not periodical. So: (1)what is the proper integration frequency (...)
CW measurements are long term averages. For constant amplitude signals the peak and average are the same. Pulse measurements give the power during the pulse. This can be peak during the pulse or average during the pulse. If you put a pulse signal to an average reading meter you will get a very low reading of approximately the duty factor multip
hi, Does someone know how to generate reentrant interrupt code in C with code composer 2.2 or 4.12 on EZDSP2407 and how to write the code. i would like to do this because during a 40kHz interrupt i have to make calculus that are longer than the interrupt period and to sample some analog signal at this frequency, the long calculus are made every
Hi all, I'm interfacing a thermocouple (type K)to a CY8C26443 chip. Thermocouple to the INSAMP (Ref = AGND) then to the ADCINC12 (REFMUX = 2BG +/- BG). I've applied the (Correlated double sampling and IIR software filter) to nuteralize the offset voltages and signal noise. The main problem is: i get a correct readings out of the ADC (-800 (...)
I'd like to design DSM. While I study the basic principals in DSM design, there are some questions. which is the relationship bet. "loop filter gain" and "quantization noise"?
Hi, there are some techniques that can be used to reduce flicker noise of opamp. For example, autozeroing, correlated double sampling and chopper stablization techniques.
Hi all, To suppress the distortion of the sampling switches that needed to pass 500MHz bandwidth signal, some techniques has been done it before: 1) Increase the transistor size to reduce the on resistance, so that the distortion is suppressed. However, a big transistor also makes a bit parasitic capacitor. I tried this technique, the transmi
in general: A preamble is used in Digital Communication Systems to train the VCO of the receiver to the incoming signals's clock so as to produce a clocking in the reciver that is synchronized to the received signal, and so a perfect sampling and/or demodulation can be done. Secondly: The pilot symbols are used in wireless communication (...)
I have seen some echo and reverb solutions, but all have some complexity. Mechanical: use of a delay line (coil) and piezo transducers in each tip. Magnetic: a closed short magnetic tape (ring) with two heads (one for recording and one for reading). sampling and memory: a special IC converting (...)
Dear hebu, This paper maybe useful for you. 1. Circuit techniques for resucing the effects of op-amp imperfections: autozeroing, correlated double sampling, and chopper stabilization, IEEEE, invited paper Best regards,
Designing a 10Mhz analogic bandwidth DSO is not a simple thing even with FPGA. An old and reference design could be found here: Remember that 10Mhz analogic bandwidth means at least 20MHz Nyquist sampling, and an oversampling up to 40MHz will probably be necessary for aquiring all kind (...)
Hi, I am engaged in a project which need to process digital IF signal. The passband sampling theory shows that when we sample a signal based on the theory, we can get the signal in the base band. But when I read some references , they show me that the digtal down converter prevail against the passband (...)
what is sampling? and question related to it. as this is the very first step. Questions like sampling theorm. sampling error noise etc..
The sampling theorem was implied by the work of Harry Nyquist in 1928 ("Certain topics in telegraph transmission theory"), in which he showed that up to 2B independent pulse samples could be sent through a system of bandwidth B; but he did not explicitly consider the problem of sampling and reconstruction of continuous (...)
I have a question about frequency spectrum (components) after sampling. Lets assume that the frequency content before sampling is: -a band of frequencyes between 5kHz~10kHz -399kHz -402kHz -803kHz -807kHz sampling frequency is: Fsampling=800kHz and what interests us are frequencies (...)
hi guys , I want to ask something . How does the CD work ? I am doing the digital class D amplifier, in this small part digital PWM . My system will receive the PCM signal and then transfer it to the PWM before put it through the LPF to get the analog signal . So that is why i need to know the format of the output from the CDs , Is it the sequ
Hello, i was wondering if someone can provide any data/sources/links about non-uniform sampling and how to reduce its effect using signal processing techniques thanks, Safwat
Hello, I'm interested in variants of time equivalent sampling and how to choose the parameters, time delays neccessary in a typical application, like an oscilloscope. Most of what i seen only present briefly the principle, if you know a paper/tutorial preferably with examples, please post. Thanks
a bandlimit white noise x(t) with PSD of S0 is sampled (no aliasing) to produce x. The PSD of x is calculated to be S0/Ts (Ts is the sample period). Now I just reconstruct the continuous noise xr(t) by passing x impulses to the ideal reconstruction filter (gain=Ts, -fs
hi all what is ment by "saturation" and "rounding" thanks
Hi! What type of resolution: * in amplitude, in phase, in frequency, etc.? Normally you should know at least: * the signal sampling factor (SSF) N = Fdigitalization /F signal * the number of the accurate bits n * the moment (angle) of the first sample * the total number of bits * the frequency band of the signal... BR
Hi friends, I am working to realize HSP50110 / 50210 architecture, used for demodulation in MATLAB and FPGA. I started with MATLAB's inbuilt example "symbol timing recovery with fixed sampling" and extended the design for HSP50110. I tried to replace Squaring loop method with either Gardner or Early-late Gate algorithm, but no success. (...)
hi one query for 10bit Nyquist Rate ADC . i am giving 1k signal input and sampling frequency is 4MHz. then what snr i expect therotically. snr = 6.02N+1.76 ==> 61.96dB but i have seen the snr equation is SNR=6.02N+1.76+10log(OSR) here i am applying the input fin=1k and fsamp= 4M so the OSR = fsamp/2*fin = 4M/2K = 2000 hence total (...)
dear members can any one help me in implementing this project plssssssss i had my 2d-dct block part completed and i need help in designing of quantizer and zigzager blocks in that or i request u to give me codes for that so that i vl try my level best to do that kindly help me out i requesting all kind hearts or rather plesing u to atmost