fred harris (@fred_harris)
here is myfrf called by channelizer_60_xfunction [DH,DW] = myfrf(N, F, GF, W, A, diff_flag)%---function [DH,DW] = remezfrf(N, F, GF, W, A, diff_flag)%REMEZFRF Frequency...
Try my 60-path, non-maximally decimated, perfect reconstruction filter bank:performs 30-to-1 down-sample, a binary mask (turn off subset of channels) and 1-to-30...
Absurd isn't it? that's why i located my manuscript and added it my last reply...(and why I offered to send a copy in response to a request)fred
here is a paper that guides you to the phase continuity reconstructionUsing the FFT as an Arbitrary Function Generator, harris & LowdermilkAbstract:The inverse...
Hello Ronnu,You are mistaking an artifact for aliasing. The artifact is tied to your thinking that one can approximate a smooth signal by straight lines connecting...
My turn: The Fourier transform of a sampled data time series is a continuous periodic function of frequency. This is the Sampled Data Fourier Transform. We can use...
hello all,the multiple notch filters do leave a residue as the signal plus 50 Hz harmonics go through multiple notch filter. We can do some house cleaning with...
what you want is a periodic notch... this is a comb filter with the periodic zeros accompanied by periodic poles slightly offset from the zeros.Same as zero packing...
A few questions.Is the 50 Hz a power main signal?Are the harmonics large compared to signal?What is sample rate? How many harmonics (1, 2,... all the way out to...
Code_warrior,The book in which my chapter appears is"Orthogonal Waveforms and Filter banks for Future Communication Systems"Editors: Markku Refors, Xavier Mestre,...
Attached is matlab script I used for solution to a HW problem. Compare time interval for convolution of 415 tap filter with 100 K data set using 5 options. 1:Direct...
Hi Mark,It is indeed a small world. Where are you located?We should meet some time and I can show you some really interesting stuff.I just sent out the attached...
This is a small chapter of a book on filter banks.It is a nice introduction to polyphase filters. Also a supplement that contains some matlab code.might also want...
Hello dsavio,You did not describe the BW of the noise. You described its density function but not its spectral spread. Is the noise band-limited to the signal bandwidth...
You didn't tell us some other important things. Is the BW small compared to the sample rate? If it is you can partition the filter to an M-path filter operating...
xilinx_DSP_Small_room.pdf. This is a common problem when you are presenting and accepting data streams with different clock domains: it is called clock domain alignment....
AGC_paper_.pdfThe second log/ln block is the inverse log... you have to undo the input log... you have a feed forward system... might want to consider a feedback...
hello CTLEE,fast_conv_2.mAttached is a matlab HW solution and demonstration I used for my DSP class,,, It performs a convolution of a 100000 data sequence (two sinusoids)...
The problem you have is that the center frequency you want is not in the baseband Nyquist zone. You have to access a higher Nyquist zone. The way you do this is...
The signal presented to the mixer is presented to two mixers, with quadrature versions of the local oscillator. There is an in-phase and quadrature version of the...
Hello ee3sdc,If you can find a copy of my book, Mulirate signal processing, I do an OK job describing the modulator structure which includes a polyphase filter for...
You can follow your shaping filter output with the arbitrary interpolator and go to any sample rate you want and the heterodyne trivially to the quarter sample rate.run...
spawar_modem_2a.mThe standard approach is to use the sqrt Nyquist filter for two tasks: the first is to perform the spectral and temporal shaping and the second...
Hilb_xfrm_options.mThe real to complex transformation of the time series can also be performed by the weaver modulator... Heterodyne the quarter sample rate to DC,...
Don't confuse linear phase with even symmetric. Normally linear phase comes from a zero phase design which builds a non causal impulse response... the linear phase...
PLL_Derivation.pdfThe two signals presented to the product phase detector are sin(w_o t + phi_1) and cos(w_o t + phi_2) ... or sin(w_o t + phi_1) and sin(w_o...
% quad phase FIR filterfor k=1:80H0=zeros(1,64);H0(33+(-7:7))=[0.23 0.77 ones(1,11) 0.77 0.23];H1=[0 H0(2:64).*exp(+1i*2*pi*0.0002*k*(-31:31).^2)];H1=fftshift(H1);h11=fftshift(fft(H1));h1=h11(1:64);h1=h1/sum(h1);figure(22)subplot(2,1,1)stem(-32:31,h1,'linewidth',2);grid...
The problem I was solving was building a control signal with a given power spectra but with a small PAPR. That's why I modified the white noise by a spectral product...
This is an old problem. I would suggest book by Manfred Schroeder,Number Theory in Science and Communication: with applications in Cryptography, Physics, Biology,...
WPMC_2010_BE_Filters_5 .pdfThere are a number of ways to estimate small frequency offsets... most traditional one is to first perform time estimate then sample output...
tony,, the window changed the amount of dc by weighting the envelope... I ran into that years ago when i plated a lot with windows. You can apply the window ad take...
Hello Milad,You are trying to build a biquad low-pass filter with a large ratio of sample rate to bandwidth. The biquad is ill conditioned for this task, see attached...
Your problem is probably the costas loop.. you really want to have a timing recovery loop to identify the correct timing samples... using a polyphase matched filter...
I have written lots and lots of papers on how to design very narrow bandwidth filters.. Have two options IIR and FIR... in IIR you design filter bandwidth at 1/4...
The problem you are describing is not the fault of the cIC. The CIC is a recasting of the N-Tap boxcar integrator which becomes the cascade of an integrator (the...
I use corel-draw... great graphics package ... i get compliments all the time for their quality and often have people switch from their drawing package to corel-draw. A...
simplest possible waveshape, complex sinusoid with a phase angle:x=(1+j1)*exp(j*phi*n)= sqrt(2) exp(j*phi*n+j*pi/4)
Mr E.I ran the coefficients of the A-weighted filters you identified in(http://jenshee.dk/signalprocessing/aweighting.pdf)The coefficients for each biquad form filters...
I couldn't attach matlab file so here it is in textf_h1a=zeros(1,211);f_h1a(107+(-10:10))=ones(1,21);f_h1=fftshift(f_h1a);figure(1)subplot(2,1,1)plot((-0.5-1/422:1/211:0.5-2/422)*211,f_h1a,'linewidth',2)grid...
Here is the problem...The continuous rectangle spanning the interval -NT to +NT has a continuous transform often called the sinc of the form h(f)= 1/(2*NT) *sin(2*pi*f*2NT/2)/(2*pi*f*2NT/2)the...
It sounds as if the I-Q down conversion from the first (digital) IF to second (digital) IF centered at 1 kHz i keeping the signal real unless you are using the...
Hi greg... we need some more information... you said "demodulate and detect tones"... "tones" suggests more than one option. Is the tone a single tone with one of...
Hi Rick,always nice to reach out and touch you (old phone company advertisement)...I always enjoy your contributions too!fred
editing out undesired high sidelobes is a process called excision... it is a messy process that is often applied in frequency domain to remove tonal jammers in...
Hello Imosh,standard response to high spectral side-lobes is to increase "beta", the time bandwidth product of the Kaiser window... doing so of course increases...
last comment... don't use full amplitude in sigma-delta ... scale input to be less than 1 as shownx = 0.9*sin(1*pi*[-n:1:n]*(f1/fs));
I have spent a fair amount of time with your code so i feel ok making some other suggestionsget rid of the window on the data... you want the window in the transform%x1...
One more suggestion, change your filter to have falling sidelobes rather than constant sidelobes.. use my modified remez.. that calls myfrf... see remez_frf_example...
Chalil,I tried running your code... i don't have thd(-,-)Strange as it seems, try changing the frequency of the fundamental so it is no longer harmonically related...
Chalil,I'm suggesting you read a paper i wrote with titleReduce Energy Requirements by Coupling a Poly-Phase Pre-filter and CIC Filter in High-Performance Sigma-Delta...
You can eliminate, or reduce to any desired level, the overshoot in the ring-up transient by reducing the size of the impulse response side lobes... one way to do...
Hi, all,I think the 3% amplitude ripple is simply the starting transient of the 1-to-8 interpolator. If we have a random phase between cosine and sampling process...
your multiplication is performing a circular convolution with the channel power spectrum.. the channel and its matched filter response to account for the spectral...
Hi Dan,The bi-quadratic filter, contrary to its popular usage, is not a good candidate for very narrow bandwidth filters. By narrow bandwidth, I mean a large ratio...
Another comment... the zero-packed CIC that produces multiple spectral copies which are suppressed by a second CIC is reminiscent of a filter design technique...
Hello all,I looked at the spectrum of one of the s-sinc filters: Once you see it, you know how the filter was designed. They are cascades of multiple CIC filters....
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