Multiple ADCs and aliasing

Started by Piotr Wyderski in comp.dsp5 months ago 71 replies

I have two 1MHz ADCs and would like to interleave them by a half of the sampling cycle in order to get a single ADC with effective sampling...

I have two 1MHz ADCs and would like to interleave them by a half of the sampling cycle in order to get a single ADC with effective sampling frequency of 2MHz (I want to oversample the signal by as much as possible). But the aliasing properties of the data stream will be as if the signal was sampled at 1MHz because the ADCs are independent blocks and don't influence each other, right? Bes...


Is 'Bessel' digital lowpass filter standard?

Started by gretzteam in comp.dsp5 months ago 64 replies

Hi, I'm trying to design a simple IIR lowpass filter with the following specs: fs = 2.4kHz fc = 150Hz order = 3 or 4 The most important...

Hi, I'm trying to design a simple IIR lowpass filter with the following specs: fs = 2.4kHz fc = 150Hz order = 3 or 4 The most important aspect is that the ringing (overshoot) shouldn't be higher than about 3%. When using a 3rd order Butterworth, I get about 9% overshoot, which is too much. Someone hinted my that a Bessel filter would be better suited. This program is a good start http:...


Simple Integration

Started by Anonymous in comp.dsp5 months ago 3 replies

I have an integral 1/(a^2-x^2) wrt x. If you use partial fractions we get (1/2a) * ln [ (x+a)/(x-a) ] +c and zero initial conditions...

I have an integral 1/(a^2-x^2) wrt x. If you use partial fractions we get (1/2a) * ln [ (x+a)/(x-a) ] +c and zero initial conditions gives c=0 however, this is a standard integral and should be (1/a)arctanh(x/a) How to get to that stage. I know of course arctanh(x)=(1+x)/(1-x) when I use this the sign is wrong.


Real Square Waves

Started by Richard (Rick) Lyons in comp.dsp6 months ago 1 reply

Hi Guys. Want to see some real square waves? Check this out: https://shareably.co/dangerous-square-waves/ [-Rick Lyons-]

Hi Guys. Want to see some real square waves? Check this out: https://shareably.co/dangerous-square-waves/ [-Rick Lyons-]


How to synchronize ADC input to reference pattern?

Started by Marcel Mueller in comp.dsp6 months ago 2 replies

Hi, I have some (quite old) network analyzer software source code which creates amazingly accurate results with PC sound devices. Basically...

Hi, I have some (quite old) network analyzer software source code which creates amazingly accurate results with PC sound devices. Basically it outputs some reference to line out while sampling its own output at the right channel and the two port response at the left channel. When using (cyclic) noise patterns as reference everything is fine because any latency is cancelled by the sam...


Matched filter chiprate issue

Started by Enda McKenna in comp.dsp6 months ago 8 replies

Hi all, I have received a transmission which consists of a series of PN sequences. I recorded at 8*10^6 SPS (8 MSPS) and wish to perform a...

Hi all, I have received a transmission which consists of a series of PN sequences. I recorded at 8*10^6 SPS (8 MSPS) and wish to perform a matched filter on it. The transmission binary sequence is 1*10^6 bits long and I have just been informed that it was transmitted at a rate of 1,016,118.24682 bits/sec which seems like a very strange chip rate to pick but unfortunately this is beyond my ...


Doppler correction, SatCom channel.

Started by Zoul Zubazz in comp.dsp6 months ago 3 replies

Hi, I am looking at a Goldcode transmitted from a LEO satellite and want to perform a matched filter on it. The problem is there is a...

Hi, I am looking at a Goldcode transmitted from a LEO satellite and want to perform a matched filter on it. The problem is there is a significant amount of doppler (20kHz) which degrades the performance of the matched filter significantly. If I have a chunk of this signal (quadrature sampled) what would be a good approach to negate the affects of the doppler? Any input will be much apprec...


Hilbert transforms and sampling rate?

Started by MatthewA in comp.dsp6 months ago

Hi all, My knowledge of biquad filter coefficients isn't great but from what I've experienced, they change based on sampling rate. but I've...

Hi all, My knowledge of biquad filter coefficients isn't great but from what I've experienced, they change based on sampling rate. but I've bumped into these two examples of hilbert transforms online and neither takes sampling rate into account. Are they okay to use or are they just based on 44.1kHz? Thanks, Matt // hilbert quadrature filter hilbertBiquad(x0) { x1 = biq


Shifting the Sinc filter in the frequency domain

Started by runinrainy in comp.dsp6 months ago 1 reply

Hello, The following example from dsprelated.com shows the design of Sinc filter with the following parameters. The link of the example as...

Hello, The following example from dsprelated.com shows the design of Sinc filter with the following parameters. The link of the example as below; https://www.dsprelated.com/freebooks/sasp/Example_1_Low_Pass_Filtering.html With the Matlab command fftshift the frequency response of the filter is centered at 0 frequency which is 256 the frequency bin. My question is, how can I move the c...


wow. it starting to look like USENET is getting pretty dead.

Started by robert bristow-johnson in comp.dsp7 months ago 20 replies

this used to be a very active newsgroup. nothing happening since Oct 17 and now it's Nov 2. sorta gets eclipsed by Stack Exchange (which...

this used to be a very active newsgroup. nothing happening since Oct 17 and now it's Nov 2. sorta gets eclipsed by Stack Exchange (which allows for formatting math equations and pasting in graphics). r.i.p. comp.dsp i guess we all die sometime or 'nother. r b-j


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