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...

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.

To reduce the computation intensity, I would first do the symbol time recovery using a feedback strategy, because they are usually less computationally intense...

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...

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...

Yes, your right for complex sinusoidal signal or a constant signal. For a real sinusoidal signal, it will be N/2.

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...

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...

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...

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...

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...

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...

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...

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...

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...

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...

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...

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...

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...

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...

**Re:** 'Phase drift' if oscillator doesn't go through full 360 deg rotation when mixing FIR to bandpass

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 $$x(t) * (h(t) \cdot e^{-j w_1 t})$$...

**Re:** 'Phase drift' if oscillator doesn't go through full 360 deg rotation when mixing FIR to bandpass

You don't need to know the phase of your income signal to make a frequency shift on it.

**Re:** 'Phase drift' if oscillator doesn't go through full 360 deg rotation when mixing FIR to bandpass

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...

**Re:** 'Phase drift' if oscillator doesn't go through full 360 deg rotation when mixing FIR to bandpass

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...

If you prototype your code using MatLab, see resample and upfirdn functions help for a quickly implementation a fractional decimation.Marcelo

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...

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...

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...

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