Faculty at ECE Department at University of California San Diego specialty is DSP and Multirate Signal processing. Author of Book, Multirate Signal processing for Communication Systems.

Re: Non-zero DFT components where zero is expected?

Reply posted 2 weeks ago (01/03/2021)
Hello bittersweet,What you are observing is that the trigonometric functions are transcendental, which means that they need very very wide width representations...
alex,Your attempt to obtain linear phase shift over the entire spectral span is doomed by Gibbs phenomena. You have to select a fractional BW, 80% and then use the...
OOPs, is your transform being implemented in fixed point or in floating point? If fixed point the amplitude should be 1/2 the largest allowable amplitude, such...
Did you do the single sinewave test on the cascade forward and inverse fft?use a single complex sinewave with one cycle per interval exp(j*2*pi*(0:N-1)/N) as the...
dear dszabo,when you perform fast convolution by two forward transforms and one inverse transform, you can save the address reversal process in the three transforms...

Re: PMCW Radar for cars

Reply posted 4 weeks ago (12/16/2020)
Hi mannai,Thanks for pointing out that I misinterpreted the DSP radar question. I am quite familiar with direct sequence spreading sequence radars. I worked for...

Re: PMCW Radar for cars

Reply posted 1 month ago (12/14/2020)
hello adaptivefilter,by phase modulated continuous wave radar, I'm assuming you mean linear FM sweep per transmitted pulse. This is the common wave shape for automotive...
We start with the observation that a sampled data sequence has a continuous periodic Fourier transform, periodic in 2pi, that is observed on the perimeter of the...

Re: How to synthesize band-limited noise?

Reply posted 2 months ago (11/27/2020)
Marcin,The most common approach is to take wide bandwidth white noise sequence and filter its bandwidth down to the desired bandwidth with a, IRR filter. That's...

Re: split Hilberts

Reply posted 3 months ago (10/25/2020)
Exactly,The sampled data domain has access to pure delay lines that the continuous domain does not. That simple access changes the rules and is one of the reason's...

Re: split Hilberts

Reply posted 3 months ago (10/24/2020)
% linear Phase FIR filter Hilbert Transform, two lengthsh=1:2:30;h1=1./h;h2=reshape([h1;zeros(1,15)],1,30);h4=[-fliplr(h2) 0 h2];hh=kaiser(61,8)';h3=(2/pi)*[0 h2...

Re: Anything like a “shift delay”?

Reply posted 3 months ago (10/24/2020)
You should expand the cos(wt+s(t)) to cos(s(t)) cos (wt)-sin(s(t) sin(wt)... what you have ere is QPM (quadrature phase modulation) or CPM (continuous phase modulation)....

Re: split Hilberts

Reply posted 3 months ago (10/23/2020)
in analog phase shifting hilbert transforms the two arms are +/-45 degrees to get a 90 phase difference. Much easier to do in DSP land.... are comfortable there....

Re: CFO correction in OFDM

Reply posted 3 months ago (10/01/2020)
You can correlate the CP with then of the OFDM symbol... same signal except for cannel distortion and phase offset due to residual frequency error... not a good...

Re: CFO correction in OFDM

Reply posted 4 months ago (09/30/2020)
Rob,Now you know why the 802.11 uses pilots. The initial estimate of frequency offset made with the preamble will always (repeat always) have a residual error. You...

Re: Will this FPGA be suitable for DSP purposes?

Reply posted 4 months ago (09/29/2020)
Corn1996,You first have to gather some skills in the design of phase lock loops. From your questions I gather you are trying to align the frequency and phase of...

Re: FSK demodulation with Doppler correction

Reply posted 4 months ago (09/24/2020)
loganathan,the preamble will give you enough information to estimate the dopple and the clock frequency as well as the phase. At the end of the preamble you should...

Re: Tilt correction in modulators

Reply posted 5 months ago (08/14/2020)
A good many modulators in transmitters are implemented in the DSP domain. Often the DSP based signal is interpolated to a much higher sample rate than the Nyquist...
asuchan,If your need is a reduction in BW and sample rate by 8000-to-1 you should be doing this with an 2000-to-1 CIC filter followed by 2- half band filters that...
You have 13 bits of growth per stage in your filter... how may stages do you have? if you have 4 stages you have 52 bits of growth... how many bits wide is the input...
Are you using 2's complement overflow  in your matlab simulation... I'm guessing from your question that you are concerned with your accumulators overflowing......
the ramp is the input-output relationship without quantizing.... the staircase is the instantaneous non linearity of a rounding quantizer... rela quantizers don't...
the adc always estimates the signal from below... the highest discrete level not higher than the measured signal... it performs truncation....successive approximation...
the self mixing properties of the ADC of the analog mixers down converts a DC line on both I and Q... the truncation transfer function of the ADC always injects...

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