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Re: About timing recovery

Reply posted 9 months ago (01/07/2020)
Adding some information on top of what kaz has said.The early-late method provides an error signal that indicates if you are in advance or delayed in relation to...

Re: Error correcting codes

Reply posted 10 months ago (11/18/2019)
Yes,I would add that Block codes can be linear, systematic and cyclic. I don't know if non-linear block code are used in practice.

Re: Automatic Frequency Correction AFC in DSP

Reply posted 2 years ago (10/16/2018)
To reduce the computation intensity, I would first do the symbol time recovery using a feedback strategy, because they are usually less computationally intense...

Re: DC Blocking unexpected results

Reply posted 2 years ago (08/14/2018)
The y axe is in logarithm scale. A logarithm function returns a negative value for a lower than 1 input. If you compute the logarithm of a negative number you will...
Forgive me if I am saying things you already known. Group delay is not constant for IIR filter. It is a function of the frequency. Therefore, your filter will produce...
Thank you Slartibartfast,However I am still a little bit confuse. I thought that the single-sided power spectral density definition was$$ S^{'}_{xx}(\omega)=S_{xx}(\omega)+S_{xx}(-\omega)...
Please someone correct me if I am wrong.Baseband and Passband bandwidth definitions are the same for real and complex signal. I mean, baseband bandwidth is the...

Re: Why downsampling with sine and cosine

Reply posted 3 years ago (01/22/2018)
Hi, First, lets fix some terms. The process of converting a signal to a lower frequency is named as downconversion. So, you downconvert you signal, initially...
Do you want to implement a filter with a non-constant group-delay x frequency response?If so, you can use an IIR filter or a non-linear phase FIR filter.To design...

Re: Fixed-point FFT scaling

Reply posted 3 years ago (09/20/2017)
Yes, your right for complex sinusoidal signal or a constant signal. For a real sinusoidal signal, it will be N/2.

Re: Fixed-point FFT scaling

Reply posted 3 years ago (09/19/2017)
It depends on your input signal frequency because of the DFT scalloping loss. For rectangular time window, the scallop loss can be as high as 3.92 dB (or 0.6368...

Re: Fixed-point FFT scaling

Reply posted 3 years ago (09/18/2017)
If we can assume that your real input tone frequency is never close to 0 Hz, we can say that you will not experience gains larger than N/2. The input power will...

Re: Choice between Scilab and Python + numpy

Reply posted 3 years ago (09/12/2017)
MatLab has a home edition now. Current price is U$ 95.https://www.mathworks.com/products/matlab-home.htm...
To reduce LUT length, use abs(x) as LUT address, instead of re(x) and image(x). If you want to further reduce LUT, you can convert abs(x)=M*2^E, where M is a short mantissa...

Re: anti aliasing filter

Reply posted 3 years ago (05/18/2017)
I have never design an analog anti-aliasing filter. However, I know that If you are going to model your input as a white noise, you have to set a band limit to...

Re: anti aliasing filter

Reply posted 3 years ago (05/17/2017)
A white noise has infinite energy, if you do not filter the signal before sampling, you will have infinite noise energy. However, a true white noise do not exist...

Re: Optimal Filter lineup selection

Reply posted 3 years ago (05/16/2017)
Have you already heard the sentence "Perfection is the enemy of done."?If you have a solution that has acceptable performance and resource consumption, why spend...

Re: DFT

Reply posted 3 years ago (05/15/2017)
Am I required to use at least N point DFT? In a way, is the requirement on N point somehow related to sampling frequency?Required is a very strong word. If you...

Re: DFT

Reply posted 3 years ago (05/15/2017)
1) I do not know if this term "observation window" is normally used, but I guess you got the idea correctly here.2) I guess you mean that your tone frequency is...

Re: Can we paralleize the integrator stage of CIC?

Reply posted 3 years ago (04/10/2017)
Accumulation is a linear operation, so it is possible. You can have N parallel accumulators, preceded by a commutator, and followed by a pipeline adder to sum up...

Re: Frequency resolution

Reply posted 4 years ago (03/24/2017)
You can always greatly improve your resolution and noise bandwidth by increasing the window length, but this comes with a cost in computation complexity. Changing the...

Re: Measuring Allan Variance of atomic clock

Reply posted 4 years ago (02/22/2017)
The proposed laser thing is not a clock, since it only generates two pulse. It only generates one time period that has very low variation between experiment realizations.By...

Re: Measuring Allan Variance of atomic clock

Reply posted 4 years ago (02/21/2017)
I have no experience in measuring Allan Variance, but I guess you need a very stable time interval to make the measures for all atomic clocks. So the question is...

Re: ASK BPSK

Reply posted 4 years ago (02/16/2017)
Taking the equation that Tim wrote describing the quadrature modulator output. If you make A(t)=exp(j*2*pi*f1*t) them your output real signal x(t) will have a frequency...

Re: ASK BPSK

Reply posted 4 years ago (02/14/2017)
Yes, a quadrature modulator can be used to generate all kind of modulation, as long as you can control the I and Q signals that feed it. It allows you to convert...
No. They are not equivalent. In $$x(t) \cdot e^{-j w_1 t} * h(t) $$ frequency shift is applied in the input signal, while in $$x(t) * (h(t) \cdot e^{-j w_1 t})$$...
You don't need to know the phase of your income signal to make a frequency shift on it.
Yes, Yes and Yes. But I am not so sure that$$ x(t) \cdot e^{-j w_1 t} * h(t) = x(t) * (h(t) \cdot e^{-j w_1 t} \cdot rect(2\pi/T_h) $$when \(w_1=2\pi/T_h\),I...
So, you want to approximate$$y_1(t) = \left (x(t) \cdot e^{-jw_{1}t} \right ) * h(t)$$where \(x(t)\) is your input signal, * is the convolution operation, \(w_{1}\) is...
The Discrete Fourier Transform (DFT) term is an overloaded term, as it can be a transform applied to periodic sampled signals, or a transform applied to time limited...

Re: Downsampling from 2.0 MHz to 192kHz

Reply posted 4 years ago (10/26/2016)
If you prototype your code using MatLab, see resample and upfirdn functions help for a quickly implementation a fractional decimation.Marcelo

Re: Zero-Padding as scalloping loss attenuator

Reply posted 4 years ago (10/23/2016)
I always think on data flow graph, because I am more used to RTL design than software design. That is why I called mixed-radix, I can not avoid to see the radix-5...

Re: Zero-Padding as scalloping loss attenuator

Reply posted 4 years ago (10/21/2016)
Thank you Jos!This is exactly what I was looking for.Now the procedure to compute the scalloping loss for any combination of window and zero padding is clear...

Re: Zero-Padding as scalloping loss attenuator

Reply posted 4 years ago (10/21/2016)
Hi Lyons,I am aware that i can use a mixed radix fft algorithm, but i'm going to implement this solution in an FPGA, and the FFT IP that I have is for power of 2...

Zero-Padding as scalloping loss attenuator

New thread started 4 years ago
Hello, I am working in a signal detector system, based on DFT, that has as input N=1280 samples from a signal where multiple tones may be present. In order to...

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