kaz - (@kaz)

FPGA Design Engineer with focus on DSP functionality on FPGA platforms.

Re: Power Spectral Density PSD° of OFDM signals

Reply posted 2 weeks ago (11/17/2022)
since you have matlab(or Octave) then you can experiment with ifft as follows:%%%%%%%%%%%%%%%%%%%%%%%%%x = zeros(1,1024);x(78) = complex(.2,-.6); %populate one subcarrier...

Re: Power Spectral Density PSD° of OFDM signals

Reply posted 2 weeks ago (11/15/2022)
If you want control per sub-carrier then you do that at constellation level. Once ifft is done it maintains the relative power among subcarriers as it is frame base....

Re: math/money reasoning issue, Not a spam Please

Reply posted 2 weeks ago (11/11/2022)
That sounds plausible indeed.The only issue I disagree with is the final conclusion saying "No one produced anything..." they all did their work based on promise.It...

math/money reasoning issue, Not a spam Please

New thread started 2 weeks ago
I am lost in reasoning the following joke:Posted on Quora by Lynda ClarkeTHE £50 pound noteIt's a slow day in the town and the streets are deserted. Times are...
Remember this is fpga platform and main theme is centred on simple resource efficient solution. If I can sample at correct rate and do one fft there is no issue...
That makes some sense if zero crossing is cared for. And is practical for FPGA platforms. It is worth trying but then you might just measure amplitude in time domain...
Since your bandwidth is 0 to 100MHz then you can't undersample and get frequency info.Though you can get amplitude info wherever your frequency aliases to.

Re: #OFDM sample rate

Reply posted 4 months ago (07/18/2022)
If your RF (air) signal is 10MHz and has further guard band carriers to 15.56MHz then your signal is 15.56MHz and so you need to sample it more than 30.72Moreover,...

Re: #OFDM sample rate

Reply posted 4 months ago (07/18/2022)
for LTE ofdm at 30.72 Msps the bandwidth is (1200*15KHz = 18MHz => 9MHz on either side of centre). That implies guard band of 1Mhz (from 20 MHz nominal bandwidth)....

Re: 'near IQ' sampling

Reply posted 4 months ago (07/18/2022)
I will call it sub-sampling i.e. your sample rate is twice bandwidth of signal and not twice max frequency as dictated by Nyquist rule. You get the alias and use...

Re: Having problem in debugging an issue with IIR filter

Reply posted 5 months ago (07/14/2022)
Judging by above scope you seem to have saturation/overflow glitch issue on negative values. can you show the sine case when the glitch occurs near zero.

Re: Weird Signals when acquiring with SDR boards

Reply posted 5 months ago (07/06/2022)
I don't see post DAC filter. Expect images (aliases) in analogue domain that follow sinc function.Even a single tone passed to DAC will have infinite images that...
So your main question is SNR of decimation.If you decimate by half then all the noise floor (minus filter effect) will alias into your band so expect less SNR.Moreover,...
Not clear post but it seems it is about single tone so its alias can be used. signal bandwidth seems small judging by decimation to 2Mhz.sampling at twice bandwidth...

Re: FFT question

Reply posted 5 months ago (06/24/2022)
An RF signal centered on Fc will be real only but signal bandwidth of interest carries info from low end of band to high end and so is expected to be asymmetric...

Re: FFT question

Reply posted 5 months ago (06/23/2022)
That is not IQ demodulator. It just produces delayed copy of same signal. Moreover I suspect your original signal is at dc i.e. real only.You need to clarify your...

Re: FFT question

Reply posted 5 months ago (06/23/2022)
clocks are 90 degrees phase difference???. Never heard of that. You should mix your signal with cos/sin generator on same clock to get I/Q.

Re: How much CompE for DSP?

Reply posted 5 months ago (06/23/2022)
Sorry but I disagree. I have over 25 years FPGA DSP work experience. Never needed to know about computer architecture of my PCs. Just basic file issues, documentation,...

Re: FFT question

Reply posted 5 months ago (06/23/2022)
yes that is basic fft principle. if your signal Bandwidth violates Fs then it is another issue and fft will show it.

Re: FFT question

Reply posted 5 months ago (06/23/2022)
If so then either fft is wrong or your I = Q or the difference is too small to observe. can you prove numerically that it is just mirrored.

Re: FFT question

Reply posted 5 months ago (06/23/2022)
fft at 192KHz can output 0 to 192 or -fs/2 to +fs/2 depending on fftshift. So what exactly is your issue?

Re: Measuring RSRP in 5G

Reply posted 5 months ago (06/21/2022)
In a general sense a digital power needs to be referenced to a given point for example antenna receiving signal. Any gain between antenna and SRS is implementation...

Re: How to separate these orthogonal inputs

Reply posted 5 months ago (06/20/2022)
Thanks Lito844,That makes plenty sense but...we correlate sum of all UEs with each one's cyclic shift or with base preamble(depending on method) rather than auto-correlate...

Re: How to separate these orthogonal inputs

Reply posted 5 months ago (06/20/2022)
Here is another question that occurred to me related to this thread of the two types of orthogonality.For ofdm the concept is explained on the basis that a peak...

Re: How to separate these orthogonal inputs

Reply posted 5 months ago (06/17/2022)
Thanks David and dres,Concluding from your replies and my analysis there seems to be two methods for SRS channel estimates (at least). A time domain based method...

Re: How to separate these orthogonal inputs

Reply posted 6 months ago (06/11/2022)
Thanks dres, that makes more sense.So my question was wrong as I thought we are after recovering the original SRS preamble but as you confirmed we are after channel...

Re: How to separate these orthogonal inputs

Reply posted 6 months ago (06/10/2022)
I am afraid I haven't got what I was looking for...I am simply asking this: I processed Uplink stream, got fft of each symbol, identified the SRS symbol, then...

Re: How to separate these orthogonal inputs

Reply posted 6 months ago (06/08/2022)
Thanks DavidCan you please elaborate on that window...in the following context:Two UEs send same SRS signals on same REs. uplink radio receives them, does FFT and...

How to separate these orthogonal inputs

New thread started 6 months ago
Hi All,A receiver of ofdm separates orthogonal tones by using fft. No problem.But there are some other signals that are also deemed orthogonal yet they are at same...

Re: Applications for a new FFT algorithm

Reply posted 7 months ago (04/28/2022)
For NBIoT we had 16 channels each needed 48 bins to be extracted out of 8K DFT (if we use LTE stream directly). option1: We can use 8k full FFT to be shared by...

Re: Applications for a new FFT algorithm

Reply posted 7 months ago (04/28/2022)
One application area could be LTE (4G/5G radio radio stations) usually on FPGA platforms. The required bins are as follows:uplink/Downlink: 1200 or so from 2k DFTrandom...

Re: fft combinations

Reply posted 7 months ago (04/27/2022)
an old post, 6 years old...8 is bin 8 of 2560 is bin 0 (dc)so 8 is relative to 256 and its tone = 8/256 *sampling rate applied to DFT.

Re: correlation of two complex signals

Reply posted 7 months ago (04/20/2022)
Hi jeff,YesThe reference Zadoff Chu preamble I was using was based on this equation:zd = exp(-1i*r*pi/L*(0:L-1).*(1:L));but its conjugate did the trick:zd = exp(1i*r*pi/L*(0:L-1).*(1:L));so...

Re: Looking for an exotic channel coder

Reply posted 7 months ago (04/20/2022)
Rephrasing your issue:You tolerate bit errors but not denary(decimal value) yet the denary value is transmitted as binary.In principle you can give priority to bits...

Re: correlation of two complex signals

Reply posted 7 months ago (04/18/2022)
First plot is magnitude of output of the two complex inputs.The problem isn't xcorr function missing conjugation but the test vectors used in LTE lab must be based...

Re: correlation of two complex signals

Reply posted 7 months ago (04/18/2022)
after several attempts it turned out to work if I generate the Zaddoff Chu frequency sweep as conjugate of what I assumed:The equation I got from search originally...

Re: correlation of two complex signals

Reply posted 7 months ago (04/18/2022)
I tried autocorrelation many times. It gives nice peaks 

Re: correlation of two complex signals

Reply posted 7 months ago (04/18/2022)
Hi Slartibartfast,That's what I expect but I am not getting it.here are two plots:left is complex vs complex using matlab xcorrright is real vs real using matlab...

Re: correlation of two complex signals

Reply posted 7 months ago (04/18/2022)
Hi dgshaw6,Yes we are on the same boat.I get the fft 839 bins (or 864) no problem. We use decimation method since a 24k fft is not practical on FPGA. These bins...

Re: correlation of two complex signals

Reply posted 7 months ago (04/18/2022)
I believe Matlab xcorr does conjugation. When I use it directly it gets me a messy peak. When I conjugate one input it gives a nice peak. So I assume Matlab may...

Re: correlation of two complex signals

Reply posted 7 months ago (04/18/2022)
Thanks for the link, I will study it. The method seems to view correlation as coarse estimate. My view is that in our case this coarse correlation is enough and...

Re: correlation of two complex signals

Reply posted 7 months ago (04/18/2022)
The Matlab function "xcorr" expects the two vectors directly. I am not sure of the internal details of their function. I expect it is based on either direct convolution...

correlation of two complex signals

New thread started 7 months ago
I have two complex signals (each a frequency sweep). I need to check delay/advance between them.The first signal is a reference I generate in Matlab. The second...

Re: Coherent re-sampling

Reply posted 9 months ago (03/14/2022)
First your 10GH and then 10,000 GHz are a bit scary.The only way to get no fft bin leakage is to choose your frequency to be exact integer number of samples per...
Those are very very high speeds for FPGA. I use 491MHz for Ultrascale and it is a struggle but it could be you are dealing with some special chips or hardcoded...
"My clock frequency is one sixteenth of the sample rate, so resource sharing is certainly not an option." I am lost now. May be you are referring to some base clocks.clocking...
My first answer though was too short, here are some more thoughts:overlap/add for filtering is not needed if you just let data stream through for ever.It seems you...
If it is me I will do fft as I want it then discard those bins I wanted them filtered.

Re: Viva Ukraine!

Reply posted 9 months ago (02/28/2022)
With my full sympathy to all innocent victims. Here is a summary of our double standards:The BBC - “It’s very emotional for me because I see European people...

Re: Viva Ukraine!

Reply posted 9 months ago (02/28/2022)
Sad whenever a stronger army attacks a weaker country, law of jungle. I hope it will be over soon for Ukrainian innocent victims. I remember the days Iraq was...
I 'believe' for perfect reconstruction we need to interpolate from minus inf to plus inf (i.e. as far back as big bang then towards edge of universe). Obviously...
Here is a picture:blue: signal each sample repeated 4 times for plot purposered: same signal upsampled by 4.The second peak unveiled a new maximum due to interpolation....

Re: One of my favorite mathematics jokes

Reply posted 10 months ago (02/01/2022)
Nice one. Seems related to Einstein's story:https://medium.com/quotes-and-thoughts/learning-ho...‘Oh, this question is so easy that even my chauffeur can answer...
Hi DanBoschen,Thanks for correcting my thoughts that mixed up dc bias and error probability.In return I will add that the rule of 6.02N + 1.76 dB can be extended...

Re: Optimal Cascaded FIR Decimator design

Reply posted 11 months ago (01/04/2022)
If your focus is multipliers then a single filter with polyphase structure for 1 bit input might be better than the multiplier-less CIC. But memory requirements...

Re: Compute inverse FIR

Reply posted 11 months ago (12/29/2021)
If by inverting FIR you mean low to high pass (...etc) then you can do that in two ways at least:1) flip spectrum then do ifft2) multiply your LPF by (+1,-1,+1,-1...)...

Re: Compute inverse FIR

Reply posted 11 months ago (12/29/2021)
Already raised by MichaelRW; what is FRF, IRF and inverted IRF. OK we know what is iFFT.What is inverted FIR filter, I assume you can reverse passband gain but...

Re: Delay measurement using single tone

Reply posted 11 months ago (12/29/2021)
...

Re: Delay measurement using single tone

Reply posted 11 months ago (12/29/2021)
Thanks Marek_Klemes,calculating delay from phase at CFO = 0 is no problem but estimating actual CFO is the problem. CFO is not known and is variable so the question...

Re: Delay measurement using single tone

Reply posted 11 months ago (12/29/2021)
Thanks Lito844,I already got that paper and more. I can see different methodologies but all use a string of hard equations with multiple indices that admittedly...

Delay measurement using single tone

New thread started 11 months ago
In Narrow Band IoT (used with 4G...) a device is scheduled once it is detected and its Time of arrival determined. The device sends single tone (in fact several...
Thanks Krasin. That is the proper way to analyse input. I did not consider that a pretty chirp signal will cheat me that much 
I think that explains it. So the tones are not generated from start to end but are time dependent. That makes good sense. My assumption was that each tone is generated...
Further work reveals that if I select first half of vector using full FFT resolution then second half using again full FFT resolution I get only the relevant half...
Here is the phase and amplitude of bad case(blue is ampl, brown is phase):
Thanks, will try see phase.I never thought of undersampling because the signal is actually upsampled by 6 from its generation point at 2.56Msps into 15.36Msps and...
I wouldn't change the fft resolution but keep it same as vector length. Ideally vector is 12288 samples, so that should be fft resolution because the tone spacing...
Hi All,Below is fft of a sweep signal "x"  which is 1.08MHz wide at edge of otherwise empty LTE 10MHz bandwidth. The sweep range is -4.5MHz ~ -3.42, with 839...

Re: Phase brake in tonal signal and frequency estimation

Reply posted 12 months ago (12/10/2021)
still you are not breaking phase. two tones at same frequency with different phases => same tone at another phase.

Re: Phase brake in tonal signal and frequency estimation

Reply posted 12 months ago (12/10/2021)
a tone with broken phase at edge will benefit from window. If phase spikes in the middle it is like high frequencies generated from rapid change. In this case you...

Re: Phase brake in tonal signal and frequency estimation

Reply posted 12 months ago (12/10/2021)
Hi josefsepp,His code adds two tones in such a way as to break phase badly as a spike.Your code is more kind.
If it was me I will not have pwelch centred on dc, do ifft then centre its output. This avoids mistakes.Edit(I now realised that Josefsepp had already coded it...

Re: Using noise to increase resolution of ADC

Reply posted 1 year ago (11/25/2021)
If it applies at ADC chip then why not afterwards as well. But note that decimation of 4 giving 1 extra bit is dependent on filter implementation and may not stay...
You can try white noise. Apply same idea of fft ratio of noise/received signal instead of freqz path. You can apply averaging to fft (e.g. pwelch) to reduce effect...
In that case apply a frequency sweep signal (chirp) instead of random input. The chirp signal to cover your bandwidth. Then check through fft ratio of chirp input...
If you pass real only input through fft then ifft you will get back your real input with a bit of free imaginary computation noise. This may not apply in your example...

Re: Proving Time Invariance of a system

Reply posted 1 year ago (10/20/2021)
I believe the OP was about using matlab as a learning tool.For that Matlab and other tools are absolutely great compared to lectures or books or lengthy pages of...

Re: Nyquist frequency & odd even FFT

Reply posted 1 year ago (10/04/2021)
It could be but I am used to follow Matlab FFT and spacing:even 1000 bins:499 positive bins @ 1/1000 spacing = .4990500 bins @1/1000 = 0.5 (negative side)odd 1001...

Re: Nyquist frequency & odd even FFT

Reply posted 1 year ago (10/04/2021)
That is how even FFT behaves for example for N = 1000: left side bins: dc bin + 499 positive (but no +.5), total 500 binsright side bins:500 bins (includes...

Re: Nyquist frequency & odd even FFT

Reply posted 1 year ago (10/04/2021)
Thanks Kschutz,Just to fix some issues you mentioned for sake of readers.Using "exp" I actually expected real part of x as +1/-1. It should not have imaginary component...

Re: Nyquist frequency & odd even FFT

Reply posted 1 year ago (10/03/2021)
I think I got the answer.It is inherent in fft bin definition. For example  1000 bin is spread as 500 negative bins, one dc and 499 positive bins so knowing that spacing...

Re: Nyquist frequency & odd even FFT

Reply posted 1 year ago (10/03/2021)
I can think anything but my question is why fft is doing that when I know my frequency is real at Nyquist. Why it doesn't show it on +/-Nyquist?? or at +Nyquist...

Re: Nyquist frequency & odd even FFT

Reply posted 1 year ago (10/03/2021)
+1/-1 alternating is real only.It cannot have 90 degrees imaginary member to pair with because it is just two samples per cycle.You can try fft(real(x)) if in doubt.What I...

Re: Nyquist frequency & odd even FFT

Reply posted 1 year ago (10/03/2021)
We can also see that dc itself behaves the same way. It is real and has one bin to go to. The +1/-1 tone seems to go to the -ve Nyquist bin only. I believe this...

Re: Nyquist frequency & odd even FFT

Reply posted 1 year ago (10/03/2021)
Many thanks fred,Let us keep it within the context of my use of Matlab fft as a reliable processing function. I got a stream of even number data points of +1/-1I...

Nyquist frequency & odd even FFT

New thread started 1 year ago
Hi All,I came to realise something odd about Nyquist frequency (a bit late though).When I generate it, it comes out only real. So I expected it to mirror around...

Re: Complex IF

Reply posted 1 year ago (09/17/2021)
This matlab code can help modelling your thoughts:clc;clear;x = randn(1,2^10);x = hilbert(x);fup = exp(j*2*pi*(0:length(x)-1)*.3);xup = x.*fup;fdn = exp(j*2*pi*(0:length(x)-1)*-.3);xdn...
@ Frank_oswhat do you mean by frequency offset? is it frequency residue left over from oscillators between Tx & Rx ? @ Mannai_Murali,Your reply is too involved...

Re: Help with polynomial zeros

Reply posted 1 year ago (08/09/2021)
Remez, firls ...etc are based on iterations to converge and so I expect they have limit on number of taps to handle.Using fir2(frequency grid method, internally...

Re: PAPR vs Input Back-off in LTE

Reply posted 1 year ago (07/15/2021)
For Downlink we used to do an inhouse fancy method of peak reduction and achieved PAR of 7.xx dB on lte having PAR of 12 dB or so (FPGA platform). Followed by Digital...

Re: Digital IIR Parallel Implementation

Reply posted 1 year ago (06/26/2021)
The concept of speed is completely different between hard dsp(ASIC/FPGA) and soft DSP.Being an FPGA designer I relate speed directly to the [combinatorial-registers-combinatorial]...

Re: Digital IIR Parallel Implementation

Reply posted 1 year ago (06/26/2021)
try avoid such slow mult e.g. using power of 2 or table.
You can model that and see directly.So you got x(t) sampled at 20Msps. You do fft to get frequency domain data x(f) then you discard all bins except those of 2-7MHz....

Re: Problem in Beam Steering

Reply posted 1 year ago (06/15/2021)
"fred h" is not a header and is not part of matlab keywords yet. It is a string meaning Fred Harris:https://www.dsprelated.com/user/fharris

Re: Overdriven Sine Wave through DSP Filter

Reply posted 2 years ago (05/28/2021)
I will model it and see directly the results.

Re: Off by one windowing

Reply posted 2 years ago (05/09/2021)
I would say if a new concept (term) helps then that is good idea...otherwise it is adding noise and misleading.A first paper I read on digital filters listed the...

Re: Off by one windowing

Reply posted 2 years ago (05/09/2021)
"symmetric window from periodic window or periodic from symmetric" sounds like chicken/egg puzzle.But I am more unimpressed by the "rectangular window" which is...

Re: DSP principles question

Reply posted 2 years ago (04/27/2021)
What do you mean by "original sampling rate?" A signal is sampled from analogue world at the ADC and reverse at DAC. The minimum sampling rate is determined...

Re: Impulse response in OFDM system using IFFT

Reply posted 2 years ago (04/25/2021)
It looks like you are working on SRS (Sounding Reference Signal). This LTE channel is part of UL and used by a UE so that eNB checks channel for best scheduling...

Re: Impulse response in OFDM system using IFFT

Reply posted 2 years ago (04/25/2021)
When you say channel do you mean eNB to UE channel (air) or let me ask this: who is sending to who (eNB to UE) or (UE or eNB). From your descriptio the 1200 subcarriers...

Re: Recover signal from the real part of IFFT

Reply posted 2 years ago (04/24/2021)
Haven't heard this scheme before but my view is as below:In effect the Tx generates symmetric spectrum since only one side of ifft is populated. Unlike other ofdms...

Re: Impulse response in OFDM system using IFFT

Reply posted 2 years ago (04/22/2021)
Your file has 1200 x 10 complex data, with phase breaking at boundaries between every 1200 samples. This is expected in ofdm symbols.Your 1200 samples imply frequency...

Re: parallel polyphase filters

Reply posted 2 years ago (04/19/2021)
If you were upsampling by 2 then two parallel polyphases will fit nicely one per path. But since you are upsampling by 6/5 then you need to break your prototype...
MLS through DAC/ADC is just pseudo-random sequence of +1/-1 and as such it is one case of BPSK channel. So how come it can be used to get system response. Apart...

Re: time recovery algorithm and CORDIC

Reply posted 2 years ago (03/05/2021)
If your focus is on simulation then the first step is to have your symbols vector as if sampled by ADC. This is the essence of your work. If you don't get this...

Re: Band-variable signals filtration - best approach

Reply posted 2 years ago (02/12/2021)
There are many ways to do the job. If you shift band to dc then I suggested decimating your filter coeffs. Another way is to cascade several copies of a LPF (e.g....
Sorry we are not in sync now...I am proving my point that for same bitwidth FIR filter (input/output) the maximum output is case y3, not y1,y2. Good luck
That brings us to the issue of normalisation/scaling. Let us forget x4 as I just added it as example input. If you do FPGA as opposed to software then you can see...
My test is about what input leads to max output. Let me recode:filter has dc power unity as its sum is 1%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%h = fir1(50,.2);x1 = [1...
If you apply dc input to FIR with same sign as coeffs you get maximum possible output, I am trying to prove that impulse input can't even get FIR max output right.Impulse...

Re: Band-variable signals filtration - best approach

Reply posted 2 years ago (02/04/2021)
yes that is correct. for passband at say [.1 ~ .2] if you decimate by n it will move to n*(.1 ~ .2)

Re: Band-variable signals filtration - best approach

Reply posted 2 years ago (02/04/2021)
Sharpest cutoff with suitable number of coeffs you can afford. Any higher cutoff (but not lower) can be achieved by decimating the coeffs:decimate by 2 or 3 or 4...etc....

Re: Band-variable signals filtration - best approach

Reply posted 2 years ago (02/04/2021)
One way is to have a reference filter cutting off sharpest then decimate coeffs as required. cutoff moves double per decimation by 2
How can we represent an IIR filter as "really" equivalent FIR using the IIR impulse response no matter how long is FIR model. Doesn't IIR mean infinite and has...
Practically I would lower input max and/or add clipping logic to output. Finding out the worst input pattern for a given input/IIR filter is no trivial task and...
In the below filter model you can experiment.But it will be sensitive to internal multiplication/rounding which I haven't modelled.You will see input is power all...
You need to control the gain of your IIR to check maximum dynamic range at output.For example the leaky integrator:y = a.x(n) + (1-a).y(n-1) gives unity dc gain...

Re: Matlab Basic Functions Reference (pdf)

Reply posted 2 years ago (01/07/2021)
True row/column could be confusing. I am not sure what you mean by X-Y, angle...etcHere are notes from an old post of mine:The terminology of rows/columns of a matrix...

Re: Spectrum monitoring - signal detection problem

Reply posted 2 years ago (01/07/2021)
That is ok. The RF side gets the job done for you. From the forum perspective I don't see your RF setup or may be I missed reading details.

Re: Spectrum monitoring - signal detection problem

Reply posted 2 years ago (01/07/2021)
500MHz band is your focus. So I assume you don't want to see any out of band power. Then depending on location of the 500MHz within spectrum it has to be filtered...

Re: Spectrum monitoring - signal detection problem

Reply posted 2 years ago (01/07/2021)
Yes that is what I suggested.step1: detection in time domain of your 500MHz Bandwidth (filtered from either side). Discard anything above random noise (or bursts...

Re: Matlab Basic Functions Reference (pdf)

Reply posted 2 years ago (01/07/2021)
Thanks for the link.After 30 years Mathworks gets it tidy. Better than never. I think they have copied it from a third party user as I have seen similar tables in...

Re: Spectrum monitoring - signal detection problem

Reply posted 2 years ago (01/04/2021)
I assume you can first detect the time domain power on/off state from a threshold using continuous averaging power meter . An "on" segment can then be captured and...

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

Reply posted 2 years ago (01/03/2021)
edit to a previous post: I see you are inverting the imaginary to convert iDFT to DFT.I think you should ask why fft is different from DFT (since you are not rounding...

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

Reply posted 2 years ago (01/03/2021)
First there is syntax error on real_acuum ?Your main issue is that you are doing iDFT correctly (not DFT) so if you invert your imaginary it will be correct.

Re: LTE Uplink Timing

Reply posted 2 years ago (12/31/2020)
I don't have direct experience with DM-RS but I know LTE uses it to track each UE timing offset and frequency offset and I assume that includes tracking a moving...
Hi Alexsunny,As others replied you better go for filter in time domain.Regarding your code; you can use it for cyclic shift provided n0 is integer.At n0 = 1 you...

Re: LTE Uplink Timing

Reply posted 2 years ago (12/30/2020)
I am not sure what is the intention of your suggestion. Are you not happy about 3GPP standard LTE system and want to try "DVB-T...etc" methods to apply them on LTE?The...
Thanks Gordono, yes that is what I see as well by mapping -152 dBm RF to 1 lsb rms digital power. The minimum actual RF in our systems is -105 dBm.What I say is...
-70 dBm is the actual RF mean power of given signal.Yes the -152dBm mapping per bit is meant per lsb bit, a theoretical ratio and not meant to be physical.I just...
Thanks Dr Mike,I plan this method for an actual system case:-70dBm RF gives -34dBFs after ADC. fft maintains 0dB power transfer.But I can't use dBFs across fft...
Hi All, A standard says “-152dBm RF power should map to 1 LSB digital domain (at final stage). In our case by the digital domain I mean the following stages:...

Re: Autocorrelation of OFDM Signal ?..

Reply posted 2 years ago (12/21/2020)
As dgshaw6 replied. The ofdm being a long 2D signal is used like a bucket of salad where they put any useful signals, called reference signals (pilots) on top of...

Re: Low-pass filter image data

Reply posted 2 years ago (12/18/2020)
On a different note I just noticed that DCT was invented by a fellow called Nasir Ahmed in 1972 and should have been named after him to be fair.Just like Costas...

Re: Multi-filter Parks McClellan algorithm

Reply posted 2 years ago (12/17/2020)
If you want a pre-equaliser for the second filter's passand ripple then you need to invert the ripple only as long as your second filter is ok for cutoff. If...

Re: LTE delay measurement resolution?

Reply posted 2 years ago (12/17/2020)
Discussions do help. It seems that we can target better resolution by upsampling before correlation.If we ignore specifics of LTE prach and see it through principles...

Re: Multi-filter Parks McClellan algorithm

Reply posted 2 years ago (12/17/2020)
Hi Woodpecker, I am a bit unclear about your task. How come you know h1 & h and want h2. Is this for some academic purpose...It will help if you give us more...

Re: Multi-filter Parks McClellan algorithm

Reply posted 2 years ago (12/16/2020)
A single rate FIR (designed by any method) can be split up into cascaded filters . I prefer single design rather than multiple design tasks. In principle I can't...

Re: LTE delay measurement resolution?

Reply posted 2 years ago (12/16/2020)
The Prach will be very long in time domain. In practice the 24K fft/ifft is managed in a different way (up/dn sampling) but the idea is simpler to view it as full...

Re: LTE delay measurement resolution?

Reply posted 2 years ago (12/16/2020)
Hi David,This is prach channel, not 2k fft lte data (pusch) case. it has spacing of 1.25KHz instead of 15KHz.Prach is generated as I explained and then added to...

LTE delay measurement resolution?

New thread started 2 years ago
Hi All, I am trying to model a well-known delay measurement algorithm in LTE Prach. Let me put it in simpler terms. The mobile user (UE) generates a signal of length...

Re: hardware FFT coefficients .vs. np.numpy.fft.rfft

Reply posted 2 years ago (12/14/2020)
It is likely your real input changes phase due to oscillators or test setup. Can you confirm that input is at same phase as you expect or is changing?
Hi Rick, As a side note here is my personal view on sinc function: The "fact" that it produces negative amplitude means failure of equation. We then repair it...

Re: Purity > 1?

Reply posted 2 years ago (12/09/2020)
Sorry for dragging this further.firstly, we agree that we need to force total power of time domain & frequency domain to be same, not just to keep to Parseval's...

Re: Purity > 1?

Reply posted 2 years ago (12/08/2020)
Hi, I am sorry to say but I am totally puzzled by your analysis.If I use Octave FFT (basically floating point) I always get 1 or very close irrespective of frequency....

Re: How to synthesize band-limited noise?

Reply posted 2 years ago (11/27/2020)
...

Re: How to synthesize band-limited noise?

Reply posted 2 years ago (11/27/2020)
That takes us to lower level filter design methods. Plenty around.Windowed sinc example:M = 39; %number of tapsFc = .2; %cutoff relative to Fs of 1x = linspace(-(M-1)/2,(M-1)/2,M);h...

Re: How to synthesize band-limited noise?

Reply posted 2 years ago (11/27/2020)
Generate random noise then filter it off to the band limits you target.In Ocatve I can use filter functions(fir1...filter) to generate random complex noise of any...

Re: Polyphase Filter Bank channelizer issue

Reply posted 2 years ago (11/26/2020)
I looked into theory of the Tx channeliser design and here is my initial understanding:I assume our given inputs are some (n) baseband channels to be transmitted...

Re: Polyphase Filter Bank channelizer issue

Reply posted 2 years ago (11/26/2020)
Without knowing what the logic is doing it is hard to tell. Slowing each input 89 clocks demands buffering for all eight channels but it becomes guess work. Do...

Re: Is RMS a good metric to estimate gain

Reply posted 2 years ago (11/26/2020)
RMS reflects power but for gain it is either used as ratio of output/input or adjusted for bitwidth such as dBFs which is 10log(power/fullscale power) and so...

Re: Purity > 1?

Reply posted 2 years ago (11/19/2020)
DFT is defined in math as:Thus there is no down scaling of accumulated sum (hence output is scaled up by N relative to input). For iDFT the accumulator result is...

Re: Constructing vector for IFFT

Reply posted 2 years ago (11/06/2020)
DSP terminology is not standardised and its community follows their corners, tools and function names. The case of resample, upsample, oversample, undersample,...

Re: Constructing vector for IFFT

Reply posted 2 years ago (11/05/2020)
we are talking different reverses(FD=frequency domain,TD=Time domain)insert centre zero in FD => upsampled(interpolated) signal in TDreverses:A) upsample(interpolate)...

Re: Constructing vector for IFFT

Reply posted 2 years ago (11/05/2020)
I don't expect "zero insertion duality" between frequency domain & time domain.For the case of zeroing frequency data (in the middle) it implies wider stop band...

Re: Constructing vector for IFFT

Reply posted 2 years ago (10/26/2020)
If I understood your request:insert [s0 as dc bin, samples 1:280 as positive bins, 560 zeros, samples 280:1 for negative mirror]. do ifft and you will get upsampled...

Re: Will this FPGA be suitable for DSP purposes?

Reply posted 2 years ago (10/04/2020)
If you have access to xilinx compiler then you can start your coding and finish simulation then compile and see for sure if it fits the target fpga before buying...

Re: Clock handling in multi rate systems

Reply posted 2 years ago (09/25/2020)
I do fpga and so familiar with clock issues. DSP engineers are less concerned about clocking details - I assume.clock is system clock and is not sample rate (though...

Re: Matlab/Octave loop vectorization

Reply posted 2 years ago (08/22/2020)
Well I mean just a utility to convert loops to arrays. Surely they can do that just like they have done so many utilities with sub-licences and get more money...

Re: Matlab/Octave loop vectorization

Reply posted 2 years ago (08/20/2020)
Just curious why Octave or Matlab tool makers can't automate that job and convert user slow loops to vectors. I have seen some field matlab designs full of slow...

Re: Tilt correction in modulators

Reply posted 2 years ago (08/15/2020)
cable tilt explained here (cable channel frequency response):https://channelvision.com/wp-content/uploads/Instruction-Manuals/3110-tilt-ins.pdfFrom the context...

Re: IFFT Synthesis Amplitude Estimation

Reply posted 2 years ago (08/14/2020)
For scaling fft/ifft I choose whatever suits my purpose and it doesn't have to be power unity issue but rather aiming at best dynamic range for a given case.The...

Re: IFFT Synthesis Amplitude Estimation

Reply posted 2 years ago (08/13/2020)
The maximum possible output will be the sum of all sines or all cosine amplitudes coinciding.so is dependent on number of tones you generate and their sin/cos...

Re: IFFT Synthesis Amplitude Estimation

Reply posted 2 years ago (08/13/2020)
Since your platform is fpga I expect you will use fft core from a vendor. I also assume you are using fixed pont processing. vendor fft cores use block floating...
No as far as I know. You need to look after that, sounds like AGC if variable. DC gain serves as good indicator for fir dynamic range and power transfer.
CIC has its own inherent scaling though you can double your input or so but at expense of filling bits without information. FIR can be scaled as you wish by scaling...
Thanks Fred,Based on that I assume all ADCs are not rounding quantisers but truncating quantisers. Yet the SNR of quantisation noise assumes half bit error (i.e....
Hi Fred,Do you imply that ADC error is one lsb max. My understanding it is 1/2 lsb max error and that is the basis of SNRQ formula. Please see attached extract from...
Current digital computations such as fft and any truncation can avoid dc by using dc unbiased rounding. As to ADC I can assume that a good quality ADC will manage...
I am a bit lost here. For ofdm we always expect some analogue mixing as the digital mixing is usually lower down. So what am I missing here. I know 4G/5G uplink...

Re: Has any one seen this window function?

Reply posted 2 years ago (08/01/2020)
And I assume any vector that goes up then comes down is defined as window but it needs a name. The only difference is the boxcar window which is not a window!!

Re: Has any one seen this window function?

Reply posted 2 years ago (08/01/2020)
it looks same (or very close) as hanning window scaled by 4
4G Downlink ofdm keeps dc bin empty.5G fills it. so it is optional
if I do fft shift on your plot A vector then do ifft/fft and finally fftshift I get back your original input.I don't see any issue but your plot B is wrong.This...
I think the answer will be very simple if the OP sends those 64 data over as table or file here to the forum.
I find it vague to agree on index issues.given any frequency domain vector if you do proper ifft then fft you get back to what you started with, without index changing.I...

Re: Moving Average Filter's Magnitude Response

Reply posted 2 years ago (07/27/2020)
you have scaled dc to 60 db. lower it to zero by scaling coeffs by sum of coeffs

Re: Blackman Window FIR Filter Coefficients

Reply posted 2 years ago (07/18/2020)
You can apply a large ifft with resolution of your choice then truncate either by selecting centre bins as you did or by decimating the whole bins to the size of...

Re: Blackman Window FIR Filter Coefficients

Reply posted 2 years ago (07/18/2020)
The matlab function fir2 does that for you. You need to expect that the sharper the cutoff the less efficient a filter is for same number of taps and same methodology....

Re: The size of an FIR filter for PDM-PCM conversion

Reply posted 2 years ago (07/18/2020)
You are pointing to filtering PDM signal in order to get its PCM equivalent. In that case use running average filter, not block based average filtering. And as...
If you started with log(0.54) you get some result.if you scale 0.54 by 2^24 you get a large number and a new log.Thus you need to scale down that result...so what...
This may helpy1 = log(0.54)e^log(y1)y2 = log(0.54*2^24)e^y2/2^24

Re: Autocorrelation of OFDM Signal

Reply posted 2 years ago (06/15/2020)
I can imagine that you may have two cases depending on lag.1) cp aligns with symbol end. Leading to distinct but low peaks regularly2) [cp & symbol] aligns...

Re: Autocorrelation of OFDM Signal

Reply posted 2 years ago (06/15/2020)
If I xcorr any vector with copy of itself I will get a very nice peak.I wouldn't do that as I don't see any benefit.If I xcorr a known tx signal with its Rx received...

Re: Autocorrelation of OFDM Signal

Reply posted 2 years ago (06/15/2020)
Hi,can you tell us how do you do autocorrelation of received ofdm, with what?
Hi,fft then ifft is very expensive for that. I will just use farrow resampler.
Every symbol has same shape in principle no matter what level it has in IQ map. If a symbol in 16QAM is lower than another by design then you want to keep it like...

Re: Parallel CIC filter processing enquiry

Reply posted 2 years ago (06/11/2020)
Hi Rick,Regarding your view "I suggest you apply one of your five input sequences (it doesn't matter which one) to a CIC filter with no decimation..."It seems you...

Re: Parallel CIC filter processing enquiry

Reply posted 2 years ago (06/10/2020)
Hi,If you are concerned with dc component only then consider a simpler approach to get the average (block average) of the 5 streams into one stream. Next you...

Re: Help with precoding (SDR implementation)

Reply posted 2 years ago (06/07/2020)
You haven't told us what is that mystery signal you are sending from Tx. Is it ofdm signal 64 QAM or is it FDM 64QAM? If it is ofdm is it proper ofdm frame/pss/sss/...
Hi Thirunavukkarasu,Just to stimulate discussion. In that case I will send no subcarriers and get zero error. Apologies if it sounds sarcastic but it is not. I was...
Hi Sumit,I think it is too obvious, you must be kidding, sorry.Low subcarriers means you can use PA more efficiently and hence have better power level without saturation...

Re: Coherent Combination and SNR improvement

Reply posted 3 years ago (05/21/2020)
That is not silly question and it reminds me of concept of diversities. I once asked how come we can get multiple/infinite number of copies of a signal without paying...

Re: delay error due to quantization

Reply posted 3 years ago (05/19/2020)
Trying to understand your test:Your x1 is generated from randnyour x2 is generated from FD of x1 (how?, do you apply FD filter).if my samples are 0 and 1 only(1...

Re: Root Raised Cosine Filter

Reply posted 3 years ago (05/18/2020)
That could be misleading. There is one RRC at Tx and another one at Rx.the response of each one is Root raised cosine so in theory the overall response is said...

Re: Root Raised Cosine Filter

Reply posted 3 years ago (05/18/2020)
Any two cascaded fir filters (h1,h2) is same as one filter with h = h1 convolved with h2.But why you want two RRC filters instead of one? 

Re: Root Raised Cosine Filter

Reply posted 3 years ago (05/18/2020)
For symbol rate (Rs) of 4800 (at baseband) you need to pass frequencies from dc to half Rs i.e. 0-2400MHz. If the filter rolls off at 1920 to 2880 I expect it to...

Re: LTE IFFT Sampling and Nyquist Rate

Reply posted 3 years ago (05/14/2020)
To be more specific the orthogonality requirement is more restrictive than sampling rule for ofdm based processing. Orthogonality requires symbol time to be inverse...

Re: Root Raised Cosine Filter

Reply posted 3 years ago (05/14/2020)
Once you map bit stream to symbols (2 bits to 1 symbol) then you choose the four symbol levels (this need to occupy full dynamic range available to you followed...

Re: Root Raised Cosine Filter

Reply posted 3 years ago (05/14/2020)
I can see symbol rate is meant to be 4800/sec. However I am not sure why you repeat the symbol ten times to filter it. If you are targeting upsampling by "repeating"...

Re: Root Raised Cosine Filter

Reply posted 3 years ago (05/14/2020)
I am not familiar with DMR standards but with other comm potocols. Just curious what is exactly dibit which you input to the filter. I am expecting you create four...

Re: LTE IFFT Sampling and Nyquist Rate

Reply posted 3 years ago (05/14/2020)
your understanding needs plenty polishing regarding I/Q and baseband.20MHz LTE generation at eNB: 1200 subcarriers (frequency domain data as I/Q) meant to go...
DUC implies mixer to upconvert to higher frequency. DDC reverses by mixing down to dc. I assume you are using mixers (digital may be)
That is partly correct at least.There are two issues with your ofdm based channel:1) mixer phase (and frequency) must match between Tx & Rx2) filtering delay...

Re: Block Floating Point in FFT.

Reply posted 3 years ago (05/11/2020)
Both Xilinx and Intel have fft versions based on block floating point. You truncate internally at various stages, keep track of scaling and finally output the net...
You need to apply exactly an offset equal to delay.Below my code adjusted for actual filtering, see how delay can make difference of constellations. I am applying...
Resample matlab function is ideal i.e. it removes initial filter samples. In your design using CIC you should discard those first samples. You can check directly...
Let me help with this ideal model. Play with it...it might answer a lot of questions. The resample functions remove initial filter delay but you can try other filters...
It seems to me you are generating ofdm(using ifft) then this is upsampled to DUC and back downsampled through DDC then you do fft then check constellation.If so...

Re: Computing the delay from a room response FIR filter

Reply posted 3 years ago (05/06/2020)
You can use white noise but I prefer a clean frequency sweep for measurement. Here is how I will do example model system (iir) from frequency sweep input and compare...

Re: Computing the delay from a room response FIR filter

Reply posted 3 years ago (05/05/2020)
I will use frequency sweep signal instead of white noise, of bandwidth equal to my system bandwidth. compare DFT of input and output as ratio, get phase of this...

Re: Matlab, properly using IFFT, FIR Filter Desing

Reply posted 3 years ago (04/25/2020)
is this what you want? reverse freqz output back to coeffs, then follow below:f = [0 .1 .2 .3 .4 .5 .6 .7 .8 1];a = [1 1 1 0 0 0 0 0 0 0];b = fir2(29,f,a); %fir2...
I have used resample frequently but I don't get it when you write it as:[yr,tr] = resample(yj,tj,fs);tj should be single value for interpolation rate, fs is integer...
I do FPGA platform which is somehow the basis of dsp as well at low level but normally hidden from you in the decimal point representation.For example assume input...
Why don't you quantise the full multiplication term alfa*(x[n] - y[n-1]) or even after final addition of y(n-1). Of course you can't allow full bit growth but you...

Re: OT: Laughing in the Face of Adversity

Reply posted 3 years ago (04/05/2020)
and to be safe when shopping:https://www.reddit.com/r/CasualUK/comments/fjhq9s/corona_virus_pandemic_what_can_i_wear_to_protect/

Re: OT: Laughing in the Face of Adversity

Reply posted 3 years ago (04/04/2020)
Most efficient masks against male corona, invented some 1500 years ago

Re: OT: Laughing in the Face of Adversity

Reply posted 3 years ago (04/04/2020)
Cleaning properly (Turkey):https://www.youtube.com/watch?v=cevvzpMPEoo
you can do reverse engineering of ifft. start with zeros vector and insert any bandwidth as non zeros, do ifft and you got the signal generated. (do fft and you...

Re: N-point DFt of cosine and sine

Reply posted 3 years ago (04/03/2020)
without diving in math equations what happened to variable (n) that appears in first equations but disappears in the final equation:x(m)  = .5*A*N*[1 + exp(-j*4*pi*k/N)]N...

Re: Covid-19 Stories

Reply posted 3 years ago (03/31/2020)
I am going to be different from many people. Yes the disease is not simple flu but it is not worth all the panic and lock down. If we exclude the elderly - including...

Re: Sampling and using the data

Reply posted 3 years ago (03/20/2020)
I think your issue is a basic tool issue. If audacity has setting for complex input then you are done. If not then pass I only and see spectrum then pass Q only...
Adding my voice to all other good answers:ofdm has long symbols and the ISI is inherently removed by choice of subcarrier spacing to be inverse of symbol time. cyclic...

Re: Sampling and using the data

Reply posted 3 years ago (03/18/2020)
If you want to centre the signal on dc as complex signal (not real) you will see the asymmetry. Your tool seems pushing band to dc as real only.
I will reply with respect to your previous posts as well.if you are using bpsk at say 40 kbits/s rate then your symbol rate is also 40 ksymbols/s.You are expected...

Re: FFT stitching

Reply posted 3 years ago (03/16/2020)
I was at some point excited about doing Altera (Intel) 2k fft from their 2x1k fft as described here(figure 2):https://www.researchgate.net/publication/279753021_Implementing_super-efficient_FFTs_in_Altera_FPGAsHowever...

Re: FFT stitching

Reply posted 3 years ago (03/16/2020)
True, The butterfly is just fft of 2 samples (N = 2) the rest of the design of any N fft is built on this unit.Any modifications such as using 64 N for 1024 N etc....
I don't see any shaping filter in your simulink model. You need to have a shaping filter in DUC followed by matched filter in DDCThen at receiver when you decimate...

Re: FIR allpass with desired group delay

Reply posted 3 years ago (03/14/2020)
A possible method to consider for scaling any filter response is to scale its sampling rate if feasible. if you lower sampling rate of 48KHz by 5.4 ratio to 8.889KHz...
One alternative is to model DUC/DDC using matlab code. You will need very few statements such as one or two for mixing up/dn, filters for interpolation/decimation...
I agree with dudelsound that upconversion/downconversion if done correctly should not affect BER. UPconversion/downconversion imply frequency shifting to some point...

Re: DQPSK - New t this, and want to learn...

Reply posted 3 years ago (02/18/2020)
The front end of demod requires care with symbol timing,carrier tracking and match filtering apart from having sampled proper signal with good snr. The number of...
some issues to consider. 1) With DUC/DDC you likely introduce group delay of various filters.You need to take that into account when deciding ofdm frame boundaries.2)In...

Re: Impulse response of CIC filter

Reply posted 3 years ago (01/28/2020)
Hi ChuckMcM,I managed to model cic as per your setting. I get the following outputs:impulse at 1: 1,18,6,0,impulse at 2: 0,15,10,0,impulse at 3: 0,10,15,0,impulse...

Re: Impulse response of CIC filter

Reply posted 3 years ago (01/27/2020)
Hi ChuckMcM,I see your work interesting in that you use scanning impulse input to work out the original filter response that has been decimated.I used a simpler...

Re: Advice for FM Demodulation Techniques

Reply posted 3 years ago (01/23/2020)
Hi Greg,I didn't mean AC issue but by +/- symbols I meant you need two frequencies in your (s) signal each running for a symbol period, preferably continuous in...

Re: Advice for FM Demodulation Techniques

Reply posted 3 years ago (01/23/2020)
Firstly did your signal of 1000 Hz have alternating symbols (+/-).Secondly why are you filtering with that [1,0,-1]. If you are using atan you don't need it.Thirdly...

Re: Running Sum filter

Reply posted 3 years ago (01/22/2020)
Sorry but your words and description are not helpful. Your filter is meant to filter adc data so must be working at same sampling frequency. The filter frequency...

Re: Running Sum filter

Reply posted 3 years ago (01/22/2020)
ADC samples at 16MHz yet filter cut-off is at 30MHz??? this is just not right

Re: Test signal names

Reply posted 3 years ago (01/19/2020)
Hi ChuckMcM,We need to coin a name for it. "inverted bowl" sounds right to me. The problem is then we need to be fair to other patterns such as [-1,+1,-1, zeros]...

Re: How to parallelize polyphase FIR resampling filters

Reply posted 3 years ago (01/13/2020)
I understand your work in this context. I am not sure about any specific book on this mixed scenario of parallelizing and filtering but will just add the following...

Re: How to parallelize polyphase FIR resampling filters

Reply posted 3 years ago (01/13/2020)
The principle is same as having single rate filter when used to interpolate/decimate directly (without polyphase approach) i.e. insert I-1 zeros then filter then...

Re: How to parallelize polyphase FIR resampling filters

Reply posted 3 years ago (01/13/2020)
I will describe my thoughts (in words rather than picture or formula):your single rate parallelized case of two subfilters is the starting point.1) you can nest...

Re: About timing recovery

Reply posted 3 years ago (01/07/2020)
you don't need to know symbol boundary. You target the peaks and dips and as long as there are plenty transitions you can keep tracking symbol timing

Re: About timing recovery

Reply posted 3 years ago (01/07/2020)
It seems you are asking about the very concept of symbol timing recovery as well as carrier recovery.They are two distinct issues. Though recovery of either assists...
Hi Slartibartfast,Thanks for the extended post. I am not RF engineer and I don't deal with PA stage but it is just to consolidate my understanding... the crucial...
Hi Slartibartfast,Thanks for the reply but you actually did not answer my question, here it is again:**************************************************************************The...

Re: ADC sampling rate matlab/simulink M-PSK

Reply posted 3 years ago (12/25/2019)
At Tx you need two samples per symbol minimum. Why? for shaping because if you send symbols in air without shaping the bandwidth you will get in trouble with law...
With reference to this paper regarding celluar IoT:https://arxiv.org/pdf/1605.05384.pdfThe Prach signal it is referring to is that used in LTE UE attachment to base...

Re: Guaranteed stable sliding Goertzel implementation

Reply posted 3 years ago (12/07/2019)
My suggestion was not based on your diagram above. Instead it is based on sheer accumulation of the product of input with exponential...at a given bin, there is...

Re: Moving average filter using Blockram

Reply posted 3 years ago (12/06/2019)
Running average filter is just a low pass filter (FIR). However it can be implemented in a recursive structure that looks like IIR but is not as each new sample...

Re: Guaranteed stable sliding Goertzel implementation

Reply posted 3 years ago (12/05/2019)
For an ideal sliding DFT you need a delay pipe to remove first sample of a frame as new sample enters for next frame. What I see as a problem here is this long...

Re: Guaranteed stable sliding Goertzel implementation

Reply posted 3 years ago (12/04/2019)
You may consider direct DFT as follows: multiply input by exp(-j*2pi*k*n/N) , then accumulate to end of a chosen frame size. Then to make it sliding, ideally...

Re: Guaranteed stable sliding Goertzel implementation

Reply posted 3 years ago (12/02/2019)
true, for magnitude of complex signal root square is expensive. I have used cordic before and it is cheap but suffers latency.

Re: channel coding and puncturing

Reply posted 3 years ago (11/24/2019)
Yes. A communication system does not need to put all the burden on receiver to detect parameters automatically. Constants such as FEC rate, bandwidth of signal,...

Re: channel coding and puncturing

Reply posted 3 years ago (11/23/2019)
That is why any user sets FEC rate at receiver as instructed by transmitter

Re: Discrete Time Feedback Control

Reply posted 3 years ago (11/16/2019)
For PID system, a very good explanation from Tim Wescott:https://www.dsprelated.com/showarticle/943/pid-without-a-phd

Re: ADC and Complex Mixer Nyquist question

Reply posted 3 years ago (11/09/2019)
With full respect to all useful replies I will answer with an analogy:The main entrance door(real ADC) allows one slim man to get through but the inner door(complex...

Re: 5G NR, Numerology and bandwidth part

Reply posted 3 years ago (11/01/2019)
5G is pretty new and here are my thoughts thus far:For Question 1 you can research the standards and I expect the answer to be directly available somewhere.My guess...
You have tones from somewhere(signal?) but want to generate/regenerate tones at certain frequencies from fft/ifft. I am lost in the reasoning here, sorry.
Yes amplitude does matter but the op is about how to control filter scaling.The best way and since they do offline fft is to do fft of input and fft of output then...

Re: concept of bandwidth

Reply posted 3 years ago (10/26/2019)
Below is my summary of bit rate/BW relationship:Generation of signal in digital domain(Tx):QAM(FDM based): bits => symbols (DAC) => BW equal symbol rate in...

Re: A question regarding MATLAB's 'invfreqz()' command

Reply posted 3 years ago (10/23/2019)
Hi Rick,I would think the dc gain of your filter is to blame.iir dc gain = sum(b)/sum(a) = 0/0 = NAN because of divide by zero... if I am right...

Re: concept of bandwidth

Reply posted 3 years ago (10/23/2019)
Hi Rick,May I add this interruption regarding confusing terminology:In digital communication systems there are three sub-domains:data bits, symbols, samples for...

Re: concept of bandwidth

Reply posted 3 years ago (10/22/2019)
A starting point is to consider QPSK casedata bits are mapped to symbols, every two bits mapped to one symbol as below example:00 => I/Q of 00, 01 => I/Q...
Amplitude level in dB is only relevant as ratio to stop band. This is purely related to filter performance irrespective of any scaling (within reason). If you mean...

Re: Bin frequency amplitude estimation for convolution

Reply posted 3 years ago (10/02/2019)
Fractional delay is a bit advanced topic. The sample values are changed (per each sample) such that the "signal" shape looks as if it was delayed or advanced relatively...

Re: Bin frequency amplitude estimation for convolution

Reply posted 3 years ago (10/02/2019)
even if you insert the impulse anywhere it is still just an impulse with advance or delay and the fft will always show flat amplitude but different phase.With signal...

Re: Bin frequency amplitude estimation for convolution

Reply posted 3 years ago (10/02/2019)
so basically you want to understand it for your own knowledge.Your comparison between time domain and frequency domain is interesting.in digital time domain an impulse...

Re: Bin frequency amplitude estimation for convolution

Reply posted 3 years ago (10/02/2019)
Well your question is not right if you are already in digital domain. It might make some sense if you are imagining an analogue impulse event that the ADC missed...

Re: Bin frequency amplitude estimation for convolution

Reply posted 3 years ago (10/02/2019)
I still can't see the answers to my questions.If your time domain input is an impulse then its frequency response is same on all bins. You don't need to do anything...

Re: Bin frequency amplitude estimation for convolution

Reply posted 3 years ago (10/01/2019)
There is ambiguity as what is your issue:Let me ask these questions first:1) what is your platform: dsp, or pc matlab or fpga...?2) what is your given input 3)...

Re: Few questions on wireless and LTE

Reply posted 3 years ago (09/29/2019)
The eNB (at base station) generates 20MHz (Downlink) by inserting 1200 subcarriers(tones) into iFFT of 2k then adds cyclic prefix and samples this ifft output on...
I thought what matters for time accuracy is the sampling clock jitter and as long as we can count on this clock using a suitable counter width then the error does...

Re: CMSIS DSP ARM Complex FFT output arrangement

Reply posted 3 years ago (09/04/2019)
Yes Chalil you are missing all the fundamental definitions. I and Q in communications is meant to be Re/Im. Moreover what has mixer got to do here in such definitions....

Re: CMSIS DSP ARM Complex FFT output arrangement

Reply posted 3 years ago (09/04/2019)
adding few more notes.bit reverse order is that for 000,001,010,011 bins as example it outputs bins 000,100,010,110 i.e. reverse index bit order left/right.This...

Re: Designing an IIR comb (peak) filter

Reply posted 3 years ago (09/03/2019)
...

Re: Efficient digital interpolation filter

Reply posted 3 years ago (08/06/2019)
You didn't follow me???L1 = 16;L2 = 8;fp1 = 0.45*2/L1;fs1 = 0.55*2/L1;fp2 = 0.45/(L1*L2);fs2 = (L1-0.55)/(L1*L2);[S2_B_N1, S2_B_Wn1] = buttord(fp1, fs1, .001, 174);[S2_B_N2,...

Re: Efficient digital interpolation filter

Reply posted 3 years ago (08/03/2019)
frequency points might need to be doubled due to convention of matlab for frequency being normalised to Nyquist (not Fs)
There is nothing wrong but the title. Your observation applies to real signal as well.You have not done any fft? if you do fft you will see a line of frequency....

Re: FFT Interpolation

Reply posted 3 years ago (07/27/2019)
Zero padding increases resolution (N) of FFT process at expense of bin shift(bin centers occur at Fs/N). It does not add any info or power to signal though causes...

Re: FFT Interpolation

Reply posted 3 years ago (07/27/2019)
Not me. We had enough terms already. Here is another case where terms cause confusion. LTE uses the word "time domain data" and "frequency domain data" quite frequently......

Re: Frequency estimation in between the bins ...

Reply posted 3 years ago (07/20/2019)
If you are after high resolution fft over some specific limited section of frequency domain then apply high resolution fft on that section only ignoring the rest....

Re: Signal Generation methods and alternatives

Reply posted 3 years ago (07/16/2019)
In that case of few tones with resolution of 1Hz you can do this:generate one frequency (f1) from an NCO.generate another frequency of 1Hz.mutliply f1 by 1Hz to...

Re: Signal Generation methods and alternatives

Reply posted 3 years ago (07/15/2019)
Hi, you didn't say anything about sampling rate and how you will achieve 1G on fpga for the 500MHz tone.If you want to generate all 10000 tones and add them up then...

Re: Clock drift and compensation

Reply posted 3 years ago (06/29/2019)
We have used a technique similar to Farrow for a TV modulator. The input TV video stream was clocked by a source different from our DAC. We inserted a buffering...

Re: Window Impulse Response

Reply posted 3 years ago (06/26/2019)
I will go for "Flirting"

Re: Multistage decimation by two with halfband filters

Reply posted 3 years ago (06/10/2019)
Assuming any LPF (rather than halfband specifically) For interpolation, a multistage approach exploits the gap between first image of interpolation and signal band...

Re: Is this time-domain aliasing?

Reply posted 4 years ago (05/27/2019)
Hi Rick,It certainly is aliasing but above in the guy who plays the strings

Re: Simulating higher order all-pass filters in matlab?

Reply posted 4 years ago (05/18/2019)
Hi Rick,you can plot poles/zeros:zplane(b,a);or read values:[z,p,k]=tf2zp(b,a)z =   0.9946 + 0.5888i   0.9946 - 0.5888i   1.0348 + 0.4498i   1.0348 - 0.4498i  ...

Re: Aliasing in Radix-2 FFT - Decimation in time

Reply posted 4 years ago (05/10/2019)
Consider a mixer that splits up a stream to odd/even (say for the sake of speed it is made into two parallel paths). You multiply odd stream by odd samples of frequency...
you haven't described what your input is apart from saying it is +/- 5V.If you want to estimate noise created by your ADC system then why not just try zero input(i.e....

Re: Phase in a filter

Reply posted 4 years ago (04/10/2019)
Hi AXP,I think Rockymarx meant constant delay when he mentioned phase delay.obviously these terms constant delay,phase delay,group delay are confusing

Re: Phase in a filter

Reply posted 4 years ago (04/10/2019)
fixed delay (system) be it piece of wire or registers as you noticed just causes delay and this changes phase linearly. i.e. if you inject a sine tone through 5...

Re: Phase in a filter

Reply posted 4 years ago (04/10/2019)
I can see your work is correct and the replies given here are useful.To explain it there are two perspectives:1) complicated one say you got two systems:assume a...

Re: Phase in a filter

Reply posted 4 years ago (04/10/2019)
can you show how you enter matlab to get mag/phase please

Re: Time delay

Reply posted 4 years ago (04/08/2019)
You already had plenty on how to design filters. I will add some notes on delay. filter delay is both constant integer (actual delay caused by registers) and group...

Re: Digital Modulator Artefacts

Reply posted 4 years ago (03/28/2019)
You can design RRC upsampling by 32 manually or through ip cores.For core you don't need to tell it for polyphase as it knows how to implement it- hopefully

Re: Digital Modulator Artefacts

Reply posted 4 years ago (03/28/2019)
...

Re: Matlab CIC fixed point modelling

Reply posted 4 years ago (03/26/2019)
Finally I found the reason...fft normally scales its output by sqrt(Resolution). As such normal fir as an example comes correct for gain.However with cic the fft...

Re: Matlab CIC fixed point modelling

Reply posted 4 years ago (03/26/2019)
yes that is what I found but is wrong because time domain direct measurement agrees with formula. I think it is some artefact of fft that is not clear. It is not...

Re: Matlab CIC fixed point modelling

Reply posted 4 years ago (03/26/2019)
I modelled both FIR interpolator and CIC Interpolator for gain analysis in time domain and frequency domain.Time domain: inject step input. compare output to input.Frequency...

Re: Matlab CIC fixed point modelling

Reply posted 4 years ago (03/25/2019)
Use time domain. inject input (x) as all ones (or 2^17) = dc.pass through model and get output y then :20*log10(y(end)/x(end))y will acquire dc value and settle,...

Re: Matlab CIC fixed point modelling

Reply posted 4 years ago (03/25/2019)
your estimate of 4.35 dB is correct(time domain equation). Your model estimate is wrong(frequency domain).

Re: Fixed point filter gain

Reply posted 4 years ago (03/23/2019)
26 lsbs to be discarded, that is what gives dc power unity and we are talking about. the chances of mid point is 1 in 2^26?unless it is meant for moon travel application...

Re: Fixed point filter gain

Reply posted 4 years ago (03/23/2019)
Direct truncation will lead to dc bias but may be tolerated,else basic rounding (floor(x + .5)) is enough in this case. Mid point issues are too trivial as many...

Re: Estimating SNR in the Frequency Domain

Reply posted 4 years ago (03/23/2019)
Hi,full scale power for 16 bits signed as an example is square of 2^15 and is just reference point to measure signal power dynamic range for fixed point.If you are...

Re: Estimating SNR in the Frequency Domain

Reply posted 4 years ago (03/22/2019)
Hi,This issue is complicated. Firstly studying a single tone case is well established within ADC/DAC industry as they use it to measure performance of their devices...

Re: Digital Modulator Artefacts

Reply posted 4 years ago (03/22/2019)
If I(0) to I(3) are the successive parallel four samples per symbol then you need to multiply by : 1,0,-1,0 (not sure why you use 1,0,0,1)and Q(0) to Q(3) by : 0,1,0,-1so...

Re: Digital Modulator Artefacts

Reply posted 4 years ago (03/22/2019)
It could be false alarm if you are mixing after upsampling by4. So ignore my comments otherwise. option A is correct...

Re: Digital Modulator Artefacts

Reply posted 4 years ago (03/22/2019)
up4 is interpolate by4 (RRCos filter module in your case).you can apply mixing either before or after interp4 but affects target frequency value. you will get Fs/4...

Re: Digital Modulator Artefacts

Reply posted 4 years ago (03/22/2019)
it is just mixer (e.g. for quadrature demod) and specifically it generates frequency without LUT as f = Fs/4

Re: Digital Modulator Artefacts

Reply posted 4 years ago (03/22/2019)
I guess you are not that clear. Do you mix before or after up4.either way the stream (be it named samples or symbols) that goes to mixer must mix one sample at...

Re: Digital Modulator Artefacts

Reply posted 4 years ago (03/22/2019)
Your mixer is not crude first. It is well known. You need to mix your signal with frequency sample by sample. one sample of signal multiplies by one sample of frequency....

Re: Digital Modulator Artefacts

Reply posted 4 years ago (03/22/2019)
your mixing as such will cause lobes. You still may demodulate but I see it wrong mixing

Re: Digital Modulator Artefacts

Reply posted 4 years ago (03/22/2019)
That looks wrong to me. Mixer inputs should be at same sampling rate.The mixer must use one sample of each to produce one output. 

Re: Digital Modulator Artefacts

Reply posted 4 years ago (03/22/2019)
Based on your diagram it looks like you are mixing one sample of signal with 4 samples of sin/cos. am I right

Re: Fixed point filter gain

Reply posted 4 years ago (03/19/2019)
if you discard 26 LSBs then gain approaches unity for dc ~= 0.6dB

Re: Fixed point filter gain

Reply posted 4 years ago (03/19/2019)
Hi,CIC interpolator dc gain = (R.M)^N/R [where R = Rate change, M = comb delay stages, N = number of stages]Hence in your case gain = (92 * 1)^5/92 = > 2^26,...
I can look at the dn2 as having three outputs! the full stream before decimation then the decimated stream1 that discards every odd sample and decimated stream2...

Re: Slow fading channel & SNR

Reply posted 4 years ago (03/13/2019)
measure it...

Re: Fixed point filter gain

Reply posted 4 years ago (03/08/2019)
Since fpga is your platform then don't expect useful logic level points here as the mindset of guys here is geared to high level software functions and they can't...

Re: Fixed point filter gain

Reply posted 4 years ago (03/07/2019)
there are two areas to control fir filter gain.1) coeff scaling2) final sum truncation.for single rate low pass fir filter (or fir decimator) power unity is achieved...
As I understand, the idea is to use two band edge filters (one mirroring the other's frequency response and each centred on band edge), compute power output from...

Re: Digital Up conversion of QAM symbols

Reply posted 4 years ago (02/20/2019)
Rereading your post it seems you are not even saying anything about pulse shaping. A classic modulator shapes pulses of symbols and upsamples to DAC speed then the...

Re: Digital Up conversion of QAM symbols

Reply posted 4 years ago (02/20/2019)
Mixing requires input and cos/sin to be at same sample rate.You need to interpolate symbols properly(not repeat).

Re: Major Outage

Reply posted 4 years ago (02/19/2019)
On the positive side: You never realize the value of what you have until you lose it. And one day you will..Murphy's law

Re: Regarding my recent post on Multistage Decimators

Reply posted 4 years ago (02/11/2019)
Hi Neil,OK different perspectives. I am normally interested in output. In effect you are looking ahead expecting decimation further down. 

Re: Regarding my recent post on Multistage Decimators

Reply posted 4 years ago (02/11/2019)
Hi Neil,Nice topic but you might have to tweak it more.Normally I analyse it by direct modelling as follows: b1= [-1 0 9 16 9 0 -1]/32;                 ...

Re: how would I model my data?

Reply posted 4 years ago (02/02/2019)
pseudo random from two sources? nothing special about a given source. if you know the seed and start then you might check the sequence directly and see which source...

Re: how would I model my data?

Reply posted 4 years ago (02/02/2019)
when you say frequency do you mean pulse rate or actual frequency domain. For random pulses(unshaped) you are likely to get frequencies all over the digital domain,...
Well this one explains it better(scroll down to middle of page), sorry I am not myself focused on details of preamble detection but I do implement Prach extraction...
This might help:https://www.mymowireless.com/wp-content/uploads/20...

Re: Help on Noise Correlation Matrix

Reply posted 4 years ago (01/28/2019)
consider using:corrMatrix = [ r1 ; r2 ]*[ r1.' , r2.' ]./N;

Re: Understanding the Comb Filter Frequency Response

Reply posted 4 years ago (01/21/2019)
I am a bit confused. Is this comb or Integrator. How come you add such long delay to feedback and why. also you rightly say about impulse input but then move to...
Freqz is built on fft. In the case of FIR both will quickly show same behavior with impulse input as long as enough of fir is at output. In case of IIR it gets more...

Re: PSD estimation when I have oversampled data

Reply posted 4 years ago (01/19/2019)
Adding to the useful contributions from Shafie and Fred .DFT frequency resolution = sampling rate/resolution thus I can safely assume that oversampling at ADC will...
For single filter approach as polyphase structure here is what I do:prototype filter cutoff: either I or D decides lowest cutoff. In this case I will design cutoff...
option 1: breaking down the design into a cascade can indeed help since at each stage you cut off just as required by that stage and let next filter or final filter...
In all cases you are interpolating/decimating to achieve fractional rate conversion.Your second three ratios look better for filters size but requires three designs.One...

Re: Negative Latency DSP Methods

Reply posted 4 years ago (01/17/2019)
Thanks for clarification.I am sure removing delay in feedback loop could make loop control more efficient and excessive delay is avoided. As such it may apply to...

Re: Negative Latency DSP Methods

Reply posted 4 years ago (01/16/2019)
1000s of mobile phones in a cell have to send their signals in sync. delay of each is measured by station and the phone is asked to advance accordingly

Re: Folded FIR filter in Matlab or Octave

Reply posted 4 years ago (01/15/2019)
You mean pre-add structure. Nothing folded about, just misnomer.matlab filter function is much faster than loops of convolution as it uses vector computations. so...

Re: Overlap-add with window

Reply posted 4 years ago (01/11/2019)
The link does say:Output windows are not used in simple FFT convolution processors

Re: Overlap-add with window

Reply posted 4 years ago (01/10/2019)
I tried this with some resultn = 100;m = 200;a = randn(n,1);b = randn(m,1);% for the simplest%w = hanning(m);w1 = sin((pi/(m))*(linspace(0,(m),(m))+1/2))';w2 = sin((pi/(m+n-1))*(linspace(0,(m+n-1),(m+n-1))+1/2))';y1...

Re: Overlap-add with window

Reply posted 4 years ago (01/10/2019)
but what stops you using correct size window to match ifft output size?

Re: Phase Locked Loop Books in a time of DSP

Reply posted 4 years ago (01/10/2019)
Is the issue symbol timing recovery or carrier tracking (derotating). For fully digital implementation of Rx front end there are four distinct issues:1) carrier...

Re: Overlap-add with window

Reply posted 4 years ago (01/10/2019)
if you are after fft convolution versus time domain convolution here is my go:as an example of fft based convolution targeting equivalent sample values(bit-true)...

Re: SDR Direct-RF User Symbol Rate

Reply posted 4 years ago (01/03/2019)
800 MHz on fpga doesn't sound right unless you are breaking signal into parallel streams. In this case rate change will get hard. I suggest you post on fpg forum...

Re: SDR Direct-RF User Symbol Rate

Reply posted 4 years ago (12/30/2018)
You are mixing up between timing of registers and rate conversion. They are two different issues. You need a rate converter. Either FIR based or cic.That can be fed...

Re: SDR Direct-RF User Symbol Rate

Reply posted 4 years ago (12/30/2018)
I suggest using variable rate converter that accepts any data rate but outputs at DAC rate
I agree you can design a complex filter with bias towards amplitude over phase response but the original post is looking for practical applications such as some...
as an example of fft based convolution targeting equivalent sample values(bit-true) here is what I will do. There is some rounding issue but you can remove it.n...

Re: Real to Complex conversion

Reply posted 4 years ago (12/15/2018)
Your thoughts are not clear here. You may be getting confused with odd/even split of signal stream to two parallel halves at half clock rate (same sample rate).Regarding...
please ignore my post above. can't delete it
It is common problem with internet articles (and many books) to spend more time decoding the mindset of writer than focus on the subject.This guy just means if phase...

Re: Gain Control

Reply posted 4 years ago (12/11/2018)
...

Re: Gain Control

Reply posted 4 years ago (12/11/2018)
doubling amplitude leads to 6dB mean power increasepeak is amplitude unless you mean peak power. fft has its funny gain (user defined). so you got to see what...
Reading back your various posts here your issue gets unclear. If you are targeting a real fir at the end then it has to be non-symmetric as otherwise it has linear...
I am afraid I disagree here. The Vector Amp may be viewed as points of amplitude with same phase. But multiplying it with random numbers will distort the result...I...
I just played with ifft of Matlab and the way one might design example LPF FIR based on that, here is my code, starting from required amplitude:%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%Amp...
I am trying to clarify your question first...It seems you are simply asking how to design a FIR digital filter optimised for amplitude response assuming it doesn't...

Re: OFDM Cyclic Prefix Understanding

Reply posted 4 years ago (12/06/2018)
The way I see it is that the issue is about multipath. If we can identify and grab the main signal then yes all other signal paths are delayed. But we can't and...

Re: OFDM Cyclic Prefix Understanding

Reply posted 4 years ago (12/05/2018)
My guess is that from ISI perspective the optimum section is the middle as both ends are too close to noisy neighbours.

Re: OFDM Cyclic Prefix Understanding

Reply posted 4 years ago (12/05/2018)
In your case there might be a different definition. I expect with a cp offset of 0.75 of 16(=12) cp samples be ignored from beginning so symbol starts from 13 and...

Re: Fixed Point vs Floating Point

Reply posted 4 years ago (11/30/2018)
Most ADCs that I heard of are fixed point but floating point may exist. At ADC level the number of bits does have effect on quantisation noise but any external noise...
One other option to try is to use correlation with a frequency sweep of practical range and find out maximum peak location.
yes I am sure. The usual belief is that if you apply a signal at full dc then that determines max possible values of sum of products i.e. forx(n-2)*h1 + x(n-1)*h2...etc...

Re: PRISM - A new type of filter ?

Reply posted 4 years ago (11/24/2018)
Using single tone input I can see that the output peak gain goes down with increase of input frequency. However the output is at different frequency (altogether)...
True, practically I will never restrict coeffs for those extreme case. I will target unity power across a function or move gain up/down as required in various scenarios....
For FIR case yes there is math for clipping case.convolution max output = multiply magnitude of each coeff (without sign) by peak of your input value and accumulate....
Salam Morteza,Yes for first statement. FD creates new samples at any point you choose between two samples. so at d of .01 it creates new samples at value of 1/100...
I wouldn't worry about those matlab functions you used. All you need is first order FD (as first stage).here is what fd function should do under the bonnet:x =...
sorry to say but I am lost... is this task about clock crossing or is it about resampling input to different rate or just synchronizer.if it is clock crossing then...
Hi Morteza,I don't drink apart from water.
Hi David ,This to me looks like this simulink fractional delay example(end of page).https://uk.mathworks.com/help/dsp/examples/fractio...
Hi,From where do you get delay value(FD)? 

Re: Automatic Frequency Correction AFC in DSP

Reply posted 4 years ago (10/22/2018)
Hi Fred,Thanks for the post. I got quite interested in your idea of sideband filters for frequency offset and timing and went through some work using my own models...

Re: LTE WiFi FFT Problem

Reply posted 4 years ago (10/20/2018)
I think there must be a stage of zero padding:64 points ifft to 2048 fft requires zero adding in center2048 fft to 2048 ifft, no zero padding2048 ifft to 64 fft,...

Re: LTE WiFi FFT Problem

Reply posted 4 years ago (10/20/2018)
Well I tried in Matlab:a = randn(1,64);b = fft(ifft(fft(ifft(a,64),2048),2048),64);plot(a); hold;plot(real(b),'r--');result is that a = real(b) apart from rounding...

Re: LTE WiFi FFT Problem

Reply posted 4 years ago (10/19/2018)
May be my case could help.Using 2048 ifft for 10MHz lte I pad extra 1024 zeros in the middle of input then decimate the ifft output by 2.For 2048 fft I pad input...
Yes that is true. I should have said that:"It is only valid to say that this matlab exercise indicates that -ideally- all these frequencies will have same samples...
I am afraid I do not see any reason to go to dsp principles and aliasing in this example. The above matlab code generates same and exactly same three sets of data....
May be you can use the cyclic prefix as your flag. For example correlate the stream in a sliding manner with a sliding delayed version and see if you get peaks of...

Re: Down sampling an iq signal

Reply posted 4 years ago (10/07/2018)
Hi Rick,That is a valid point if the OP did not take it into account. Your comments however raised some thoughts with me. The fft in my opinion, should not scale...

Re: Down sampling an iq signal

Reply posted 4 years ago (10/06/2018)
Just adding few more notes. I think aliasing is always mentioned in terms of any power beyond intended Nyquist be it signal or noise. You may allow aliasing as...

Re: Windowing before FFT on packets of IQ datastream

Reply posted 4 years ago (10/04/2018)
The use of fft/ifft for equivalent correlation or convolution as that of time domain is only equivalent (mathematically sample by sample) if you use the proper...

Re: Windowing before FFT on packets of IQ datastream

Reply posted 4 years ago (10/04/2018)
This raises the issue of continuous stream processing Vs packet based processing.If your system is continuous stream then I would not go for fft but do filtering...

Re: Cyclic prefix concept in LTE

Reply posted 4 years ago (09/30/2018)
I basically don't disagree. We can fft a frame starting anywhere [from within cp section right to end of cp] for one complete symbol but not exceed that to next...

Re: Cyclic prefix concept in LTE

Reply posted 4 years ago (09/29/2018)
But I read code from Intel design directly and they chop off exact cp length literally (normal or extended). However recently I was told to add variable offset...

Re: Cyclic prefix concept in LTE

Reply posted 4 years ago (09/29/2018)
But I am following standards or so and I got design examples from Intel and from various other industries. Once I receive the Uplink stream from radio through cpri,...

Re: Cyclic prefix concept in LTE

Reply posted 4 years ago (09/29/2018)
Back to this thread to find some answer. I thought I knew about cyclic prefix (CP) and its significance, after all I am myself engaged in both 3GPP uplink and downlink...
Firstly SNR depends on inband noise and out of band noise. With filtering you can reduce out of band noise or leave to alias and so reduce SNR relative to ADC output....
The rule of decimation (assuming Fs_in, Fs_out) is that:1) If there is no power beyond Fs_out/2 then you can decimate directly. This doesn't apply to adc acquired...

Re: Learning FPGA

Reply posted 4 years ago (09/12/2018)
I started fpga design and stayed only within that boundary since 1998. I am sorry to say that I wish I never started. It is interesting and nice when you design...
You need to double the gain of your filter to get unity power.As to invert spectrum then add ...etc. that can't be done. 
So essentially you have a QAM signal centred at 1MH at Tx then you receive it at 1KHz. First you don't need Hilbert. Second you just need mixing with -1KHz. Third...
When reducing bit-width from 31 bits to 16 bits, yes we can call it "quantisation". If we want rounding to nearest integer then all the fraction values above .5...
Rounding can be of several types:1) direct truncation: just discard LSBs. This leads to dc bias (floor in matlab) and can build up in accumulators.2) rounding to...
Hi Rick, Paul,With due respect for your useful posts I am a bit unsure about some issues discussed here.Firstly: The issue of truncation dc bias is better avoided...
Apologies If I going off track but an alternative that we use in FPGAs to avoid dc of truncation is to round up using "unbiased rounding" rather than pass the error...

Re: Removing DC Spike from an FFT

Reply posted 4 years ago (08/16/2018)
Regarding weights, it is really meaningless jargon with averaging as you just add n samples and divide by n to get mean i.e. each sample has weight of 1.You should...

Re: Removing DC Spike from an FFT

Reply posted 4 years ago (08/16/2018)
In the time domain, dc can be two types:1) dc offset of all values2) dc bias within some signal values, unrelated to offset.removing dc offset is easy by subtracting...

Re: DC Blocking unexpected results

Reply posted 4 years ago (08/12/2018)
Do you mean you apply dc blocker before downconverting centre frequency to zero?
Hi Rick,I meant normal interaction, not the one some may think...Another analogy is that of young people driving their cars fast, reckless to arrive quickly at the...
IMHO the difference is in the amount of typing today. Everywhere people type on everything but ASA they "socialize" they stay quiet or just shout/swear at each...
In various work places I see a variety of network set-up scenarios usually undocumented and impossible to Google. Apart from that I see people using PC/Server for...
Thanks Slartibartfast,No, simulation is not available yet until late in project development.What specifically I wanted to know is the difference between the two...
In 3GPP (LTE) an uplink channel (PRACH) sent by handset has to be processed at baseband. The Prach signal is extracted from its given location and fft (e.g. 2K)...
Clearly it is no more than rubbish distortion of the meaning of revolution of circle A. Rev of 1 of circle A is relative to starting point on surface B.What is...
what if B  is same size as A? would the answer be 1+1 = 2 ?

Re: filtering followed by fft a given frame

Reply posted 4 years ago (07/02/2018)
Hi Slartibartfast,The lte uplink channel goes from radio to cpri then to BBU(baseband unit).I am working on BBU side.From cpri the same stream (30.72 Msps for 20MHz...

filtering followed by fft a given frame

New thread started 4 years ago
The lte Prach channel requires fft at a minimum of 2k resolution. To achieve that some decimation is done. This means the stream will go through some filters then...
Thanks all for the encouragement. Yes it works now!!The bug was in the phase accumulator wrap up (positive and negative directions).My work proved one point about...
I have finished a qpsk/16QAM carrier recovery on fpga based on Costas loop and is relatively tracking well across a good range of frequency shifting tests from a...

Re: Implementing octave algorithm in FPGA using c++

Reply posted 4 years ago (06/18/2018)
Apologies I am quite off topic and wish you find your way but we can nowadays:convert vhdl to circuit..convert Verilog to circuitconvert System Verilog to circuitconvert...

Re: Carrier Recovery, Frequency and Phase in (CP)FSK

Reply posted 4 years ago (06/09/2018)
That is clear enough...to me now, thanksYour system is not familiar to me but I can suggest:1) check that you are indeed sending chirp then signal regularly as described,...

Re: Carrier Recovery, Frequency and Phase in (CP)FSK

Reply posted 4 years ago (06/09/2018)
Hi,I think We are going sideways. Fixed time delay of a signal leads to delay of any frequencies in a linear way and so is not a problem for "mixed frequency" signals...

Re: Inverse filter

Reply posted 4 years ago (06/08/2018)
well with fixed point data that is impossible in my opinion. But may be it is true for floating point.

Re: Inverse filter

Reply posted 4 years ago (06/08/2018)
you mean get LPF to filter a signal that has only power left after HPF.Are you sure as I imagine nothing is left? and How can we recover filtered section

Re: Carrier Recovery, Frequency and Phase in (CP)FSK

Reply posted 4 years ago (06/08/2018)
I am referring to general digital principles. So if you are using tools and their own setup then you are best to hear from people who are familiar with your setup.1)...

Re: Carrier Recovery, Frequency and Phase in (CP)FSK

Reply posted 4 years ago (06/08/2018)
you have two areas to check. (1)is the sampling rate matching between Tx at the module you generate your signals and Rx at the point you sample it back.(2) is there...

Re: Carrier Recovery, Frequency and Phase in (CP)FSK

Reply posted 4 years ago (06/07/2018)
For wireless comms only data is sent. for wired systems data only or data and its clock are sent.If clock is not sent, it has to be recovered. If it is sent you...

Re: Carrier Recovery, Frequency and Phase in (CP)FSK

Reply posted 4 years ago (06/07/2018)
You seem to have experimental setup where Tx and Rx are clocked by same clock.Then you can -I assume- ignore all remote comms issues as your case becomes a case...

Re: Arbitrary Resampling issue : 30.72 MHz to 20 MHz

Reply posted 4 years ago (06/07/2018)
Thanks, that reminds me of a qpsk/16QAM demodulator clock recovery with free running ADC oscillator. To hit the peaks/troughs of symbols, either the adc clock phase...

Re: Arbitrary Resampling issue : 30.72 MHz to 20 MHz

Reply posted 4 years ago (06/07/2018)
Hi Dave,Yes I agree it happens in any platform. I managed that with feedback arrangement and never thought there is an alternative. This feedback controlled the...

Re: Arbitrary Resampling issue : 30.72 MHz to 20 MHz

Reply posted 4 years ago (06/06/2018)
Thanks Dave. That seems interesting but I think it is not the way we work in FPGAs. All Clocks have to be tied up together or prevented from drift. setup/hold Timing...

Re: Arbitrary Resampling issue : 30.72 MHz to 20 MHz

Reply posted 4 years ago (06/06/2018)
Gretzteam,Thanks for the link but I am lost. It talks about three types of rate conversion:(1) integer(2) fractional(3) irrationalI have never heard of (3) and where/how...

Re: Arbitrary Resampling issue : 30.72 MHz to 20 MHz

Reply posted 4 years ago (06/03/2018)
Olivier,You can do it in 3 stages. But for fpga one design is easier and quicker than 3 designs I believe.The one design approach is the well known fractional rate...

Re: Arbitrary Resampling issue : 30.72 MHz to 20 MHz

Reply posted 5 years ago (05/29/2018)
yes, 125 polyphases in LUT then you just calculate those samples that are not going to be discarded. 

Re: Arbitrary Resampling issue : 30.72 MHz to 20 MHz

Reply posted 5 years ago (05/28/2018)
125/192 may not be that costly, you may need a large lookup table for coeffs but few multipliers. You can run an accumumulator as +192 modulo 125 then address the...

Re: Interpolated FIR filter group delay

Reply posted 5 years ago (05/18/2018)
I am not sure about IFIR but in your case I would try decimation to a low sampling rate then apply final shaping.
From where did you get your example of 1/(z - 0.9z)It doesn't look right to me and reduces to 1/0.1z
MATLAB allows you to enter both forward and reverse coeffs.e.g. freqz(b,a);b = forward coeffs, i.e. 1 in your case of numeratora = feedback coeffs i.e. [1, -.9]...
I think the question is not right. You can filter a signal without any upsampling.But for shaping pulses in comms you must upsample by a minimum of 2 since you cannot...

Re: CIC filters with decimation ratio jitter

Reply posted 5 years ago (05/06/2018)
One possible solution is for Ned to relax and be tolerant. But seriously it is normally the responsibility of input side to eject the right outputs and not expect...
I have this quick code to study phase computation from fft. As far as I notice I can detect phase within 1 cycle ratio(with some wrap up issue) but understandably...
some other issues:you are entering phase in time domain as samples but fft outputs phase in radians.The tone is going to leak to adjacent bins unless you choose...
Just a quick guess:signal_2=sin(2*pi*f0*chunk_t-100*pi/180);shoud besignal_2=sin(2*pi*f0*(chunk_t-100*pi/180));
Time domain:you filter in time domain by applying convolution.reversal is deconvolution??frequency domain:you first move to this domain then you can filter in frequency...
Hi Nelson,convolution is dead easy. you get sum of products as you slide your audio signal "underneath" the fixed IR.Deconvolution is not that easy as we have lost...
Indeed Audio signal is 1D but an image is 2D array of pixels1D signals can be conveniently processed in some tools as 2D array with each channel independent of others.Matlab...
Hi Nelson,Your thoughts on signal streams as matrix seems wrong. Signal streams are 2D arrays and this is not same as matrix. I have never used matrix computations...

Re: CIC Filter "zero-crossing" Distortion

Reply posted 5 years ago (04/18/2018)
well your integrator was wrongly written with regard to index of samples.As for comb I will do this:-----------------------------%single comb model%m = delay stagesfunction...

Re: CIC Filter "zero-crossing" Distortion

Reply posted 5 years ago (04/16/2018)
Hi Jing,If it was me I will just write integrator as follows:--------------------------------------------------%single integrator model (accumulator)function y =...
By the way in -say- QPSK modulator the I & Q are generated from stream by some arbitrary selection e.g. even bits seen as I and odd as Q. In other words there...
If your baseband signal is upconverted it just changes frequency centre from dc to new centre. This does not change its symmetry. Your baseband signal must be asymmetric...
Not sure what you are asking but:Spectrum of (I) alone is always symmetric. Of (Q) alone is always symmetric. Of (I +jQ) as one signal could be asymmetric.I alone...

Re: Phase and Gain alignment

Reply posted 5 years ago (04/08/2018)
I suggest you calibrate by applying frequency sweep covering your signal range then do fft on input and each of three outputs. Each of three fft ratios of output...

Re: Matlab USRP toolbox / Interpolation question

Reply posted 5 years ago (03/17/2018)
away from tool specifics, in general you can and better upsample at DAC to lessen the burden on analogue anti-image filter and to reduce complexity of analogue...

Re: spectrum of rectangular pulse using two methods

Reply posted 5 years ago (03/03/2018)
yes that was true in my first code but corrected later in subsequent post. The corrected code is given in that post.

Re: spectrum of rectangular pulse using two methods

Reply posted 5 years ago (03/03/2018)
Many Thanks I now got it right. It seemed there are different versions of rectangular pulse equations. The one that works is sin(T*pi*f)./(T*sin(pi*f));But strangely...

Re: spectrum of rectangular pulse using two methods

Reply posted 5 years ago (03/03/2018)
Hi again,I am a bit clueless as what has averaging filter got to do here. Do you mean applying it to the pulse before sinc frmula or applying it to the pulse before...

Re: spectrum of rectangular pulse using two methods

Reply posted 5 years ago (03/03/2018)
Hi dszabo,Thanks. It does make sense regarding aliasing and may explain the difference but that is if the rectangular pulse is in analogue world. I thought the...
Hi all,I am trying to model spectrum of a single rectangular pulse in matlab. I am using sinc formula versus fft and got some two issues that I can't understand.. issue...

Re: Currently at Embedded World

Reply posted 5 years ago (02/28/2018)
Hi BV Ramesh,just clicked on that website and it says website for sale.

Re: Digital Image Processing

Reply posted 5 years ago (02/28/2018)
Well let us try:Fs = 1;N = 1024;x = sin(2*pi*3.5*Fs*(0:N-1)/N);y = fft(x,N);plot(20*log10(abs(y))); 

Re: Digital Image Processing

Reply posted 5 years ago (02/28/2018)
Your equation suggests frequency of s to be:3.5*Fs/Nif it is normalised as f/Fs it becomes (3.5*Fs/N)/Fs = 3.5/Nsince N is unknown then f is unknown...

Re: Track delays between two speech signals over time

Reply posted 5 years ago (02/27/2018)
I am not sure about your setup details but if you compare it to radar pulses then you must make sure you are correlating the relevant chunks of your signals, otherwise...

Re: Understanding Subband Filtering?

Reply posted 5 years ago (02/25/2018)
If you have to use 512 resolution then lower Fs well below 44100Hz so that you get the required resolution. That is decimate using appropriate filtering and then...

Re: Understanding Subband Filtering?

Reply posted 5 years ago (02/25/2018)
your f1 and f2 are too low and their separation is too low for 512 fft resolution. a resolution of 512 means bin resolution of 44100/512 = 86 Hz yet your f1-f2...

Re: Understanding Subband Filtering?

Reply posted 5 years ago (02/19/2018)
One other way is to repeat the downsampled section twice. This should give the same frequencies as one section with a bit of windowing tricks at the edges if too...

Re: Understanding Subband Filtering?

Reply posted 5 years ago (02/19/2018)
You downsample to half rate but still should have enough of signal to go for 512 samples. Otherwise your case is too restrictive and you may try insert zeros but...

Re: Understanding Subband Filtering?

Reply posted 5 years ago (02/18/2018)
Hi,Without going into details of your case I would like to comment on your assumption that"but since the signal is already in discrete/digital form, lowering the...

Re: A Sinusoid with Missing Periods

Reply posted 5 years ago (02/16/2018)
Not quite a PSK case to me because the missing value section is empty

Re: Frequency Shifting Confusion

Reply posted 5 years ago (02/08/2018)
I notice the signal shape at baseband is slightly different when frequency shifted. This may give a clue

Re: Frequency Shifting Confusion

Reply posted 5 years ago (02/08/2018)
Why is there minus at -1/fst = (0:1/fs:((1/fs)*(zb_packet_len))-1/fs)

Re: Multiplying the input by the frequency response

Reply posted 5 years ago (01/04/2018)
A picture is worth thousand wordsrun the following code in Matlab, here I check your filter system in two ways and compare, I then apply an input in three ways...

Re: Spread Sinc (ssinc) filter

Reply posted 5 years ago (01/03/2018)
In FPGA (and same applies to ASIC like this chip) classic FIR are common and resource is not a problem. Though CIC is favored for high rate changes. I don't think...

Re: Spread Sinc (ssinc) filter

Reply posted 5 years ago (01/03/2018)
I just checked one ssinc filter and it is FIR with multiple coeffs. I don't see any CIC connection here.

Re: correlation for complex number

Reply posted 5 years ago (01/03/2018)
I think it is now clear how best to correlate two complex vectors. The horse has been killed.I have been doing this sort of correlation for years in Matlab without...

Re: correlation for complex number

Reply posted 5 years ago (01/02/2018)
48 minutes Video here!! can anybody confirm I am not hallucinating.

Re: correlation for complex number

Reply posted 5 years ago (01/02/2018)
Are you talking about correlation in frequency domain? if so the conjugate means reversal of stream (compared to time domain).

Re: Multiplying the input by the frequency response

Reply posted 5 years ago (12/30/2017)
Another observation of your statements is that related to 10Hz generation:You assume 3*cos((20*pi )/40) is a frequency but is it?It should be a vector of sinusoidal...

Re: Multiplying the input by the frequency response

Reply posted 5 years ago (12/30/2017)
If you are saying I got two coeffs 0.5, 0.5 for fir i.e. Hz = 0.5 + 0.5z-1I get the frequency response from z transfer function => H(e^jw)= 0.5(1+e^-jw)then...

Re: How to downconvert a Complex RF signal in Matlab?

Reply posted 5 years ago (12/28/2017)
Since you have complex input then apply tone of -10Hz (complex) i.e. do complex mixing.You can view what is going here as follows:(1) bits => repeated 100 times...

Re: How to downconvert a Complex RF signal in Matlab?

Reply posted 5 years ago (12/28/2017)
ok that is in effect repeat of each bit 100 times then mixing with 10Hz tone.It should do though in practical systems pulse shaping may be considered to avoid high...

Re: How to downconvert a Complex RF signal in Matlab?

Reply posted 5 years ago (12/28/2017)
bitrate = 1Hz, then suddenly Fs = 100???How did move from 1Hz to 100. Did you do upsampling or repeated samples 100 times?

Re: How to downconvert a Complex RF signal in Matlab?

Reply posted 5 years ago (12/27/2017)
you got complex input and you want complex baseband. just downconvert to dc by mixing with complex cos/sin at -f of your signal centre frequency(f). So what is stopping...

Re: How to downconvert a Complex RF signal in Matlab?

Reply posted 5 years ago (12/26/2017)
You have already been answered but I will just confirm that in modelling tx/Rx chains in matlab you can:1) model DAC by just truncating or rounding your tx output...

Re: Help on Demodulating an 8-PSK Signal

Reply posted 5 years ago (12/17/2017)
yes for proper system design the adc speed and RF design are to be considered. But the original post is more of a modelling student task who is learning the very...

Re: Help on Demodulating an 8-PSK Signal

Reply posted 5 years ago (12/17/2017)
No need for Hilbert. Just multiply your signal with cos/sin at same frequency as Tx. You get I & Q. That is all.

Re: Help on Demodulating an 8-PSK Signal

Reply posted 5 years ago (12/17/2017)
you can apply I/Q demod on your real only signal definitely. You will get constants for I & Q separately if when doing that you down convert to dc. Then you...

Re: Help on Demodulating an 8-PSK Signal

Reply posted 5 years ago (12/17/2017)
option 1: If phase varies, and it will, then correlation loses accuracy. To keep up you will need initially, at least, to find a way to track it.Correlation will...

Re: Help on Demodulating an 8-PSK Signal

Reply posted 5 years ago (12/17/2017)
In that case just use 8 phase correlator. Correlate rx signal with each in parallel and decide which phase is it per symbol period. But that is going to be pain...

Re: Help on Demodulating an 8-PSK Signal

Reply posted 5 years ago (12/16/2017)
agreed with Mukul.The 8 equi-distant phases of the OP vector gives rise to equal amplitude and in theory is a form of 8 PSK but the popular 8 psk should have four...

Re: FIR Amplitude

Reply posted 5 years ago (12/13/2017)
the sum of your FIR coeffs is 15763 yet you divide by 2^15. That is a gain of 0.48you should scale them so that sum is 2^15

Re: FIR Amplitude

Reply posted 5 years ago (12/12/2017)
you should consider power gain rather than amplitude gain and then at some frequency e.g. dc gain or your passband gain. I assume you mean you are not getting any...

Re: fft/ifft scaling revisited

Reply posted 5 years ago (11/28/2017)
Thanks all for the replies. All make sense. In fact I work with my own scaling well away from unity issues and more to do with signal dynamic range control. But...

fft/ifft scaling revisited

New thread started 5 years ago
Hi All,Judging by Matlab or Octave fft/ifft the scaling applied for (n) resolution is:fft output is scaled up by (n) for powerifft is scaled by 1/n for powerObviously...

Re: Off Topic: A geometry problem

Reply posted 5 years ago (11/16/2017)
The best solution is agreement of the two people and availability of parking spaces. Is this Forum alright? We might need to see Forum psychiatrist.

Re: Off Topic: A geometry problem

Reply posted 5 years ago (11/16/2017)
...

Re: AGC on FPGA

Reply posted 5 years ago (11/14/2017)
Hi Neil,Any chance you or Rick explaining "in baby steps for dummies" how log/antilog approach sorts out the time constant problem. ThanksKaz

Re: AGC on FPGA

Reply posted 5 years ago (11/14/2017)
I am not clear why AGC loop should work on sample by sample basis without any delay. Normally we check signal power over some time window, possibly then filter the...

Re: IIR in FPGA

Reply posted 5 years ago (11/10/2017)
IIR in FPGA are very simple computations. No need for state machine or any waits. That could cause latency issues on Y output stage.Just use input delay line, multiple/add/truncate...

Re: IIR in FPGA

Reply posted 5 years ago (11/10/2017)
Just make one version that works. To prove it works use a math model as your reference. Inject same input into model and design and compare outputs.

Re: What is Windowing and when/why do we need it?

Reply posted 5 years ago (11/06/2017)
To keep my life simple I first avoid analogy with my bedroom window and second I view it as nothing more than a scale factor over a finite given samples. This scale...

Re: DDS jitter

Reply posted 5 years ago (11/03/2017)
I assume jitter of sine wave equals that of its clock which may be reported by the fitter tool.

Re: Narrow band filter at high rate

Reply posted 5 years ago (10/24/2017)
I am doubtful about your case. 5MHz band requires at least 10Msps.and 20KHz x 250 = 5MHz implies channels with no spacing?

Re: Narrow band filter at high rate

Reply posted 5 years ago (10/23/2017)
FDM channel(s) or channel??sounds tough to me. what is your sample rate for this 5MHz band
you can decimate as long as the result contains the information you are after

Re: Gapped-signals

Reply posted 5 years ago (10/02/2017)
A gap means "no information" and there is no way to get it back but you can either suppress the phase discontinuity by windowing or create false samples by extrapolation...

Re: Gapped-signals

Reply posted 5 years ago (09/29/2017)
My short answer is that you can't chop off part of a signal then stitch back without phase break...unless it is made up of few preknown discrete sinusoids.
IIR has nonlinear delay if that is what you are after

Re: SNR measurement with fixed-point FFT

Reply posted 5 years ago (09/22/2017)
I have no idea about how the fft ip core works internally 

Re: SNR measurement with fixed-point FFT

Reply posted 5 years ago (09/22/2017)
Can we share your fixed point code? I am interested because I am just migrating an Altera fpga design from 32 bits floating point(single precision) to fixed point...

Re: SNR measurement with fixed-point FFT

Reply posted 5 years ago (09/22/2017)
Use "randn" instead of "rand" to get zero mean. By the way do you mean "matlab fft function" when you say full precision and do you mean your own scripting when...
Hi John,Regarding frequency synthesis a third approach is as follows: In FPGA (and ASICs) frequency resolutions in MicroHz are possible using phase accumulators...
you might use CIC filter. It requires very low resource if correction filter is not added.My understanding is that variable sampling from analogue side had been analysed,...
It will have repeated phase pattern in your case with regard to duplication of a sample but I see it equivalent to introducing random amplitude error at well spaced...
Neither me a dsp half guru but I feel you are distorting your signal then analysing the result of distortion based on "what if" approach. You should create new samples...

Re: Extract a period from a unkown signal

Reply posted 5 years ago (09/12/2017)
I got the data from your own previous thread @:https://www.dsprelated.com/thread/3616/visualizing...The sine sections are obvious from initial plot then just zoom...

Re: Extract a period from a unkown signal

Reply posted 5 years ago (09/12/2017)
you seem looking at different signal from the one posted or may be you are looking at correlation output. The sine wave are there in your original signal and should...

Re: Extract a period from a unkown signal

Reply posted 5 years ago (09/12/2017)
for the 100 samples you need to zoom in well and you will see sine waves. I find them everywhere in clean sections of your signal.For your final xcorr results you...

Re: Extract a period from a unkown signal

Reply posted 5 years ago (09/12/2017)
and this may help filtering using running average:c = xcorr(yr,sine_data);c_filtered = filter(ones(1,100)/100,1,abs(c));

Re: Extract a period from a unkown signal

Reply posted 5 years ago (09/12/2017)
The 100 samples I suggested must be from a clean sinusoid area and not from just any area of your signal as there is plenty of noise sections.xcorr can be applied...

Re: Extract a period from a unkown signal

Reply posted 5 years ago (09/12/2017)
Sorry for jumping into this but I had accessed your mat file before from previous post and the way I will do it is matlab xcorr:get a segment from your signal when...

Re: Calculating Noise Level of a Pool

Reply posted 5 years ago (09/11/2017)
What do you mean by noise charactersitic? You don't seem to mean SNR as there is no mention of your signal.

Re: Extract a period from a unkown signal

Reply posted 5 years ago (09/09/2017)
Thanks David for the Goertzel filter.I just learned that. Here is my practice in matlab code:%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%clear all; clc;freq = 1/153;x...
I see Rayleigh in my simulation:%%%%%%%%%%%%%%%%%%%%%%%clear all;n = 10000;f = zeros(n,1024);for i = 1:n   x=randn(1,1024);   f(i,:) = abs(fft(x));endtest =...

Re: Extract a period from a unkown signal

Reply posted 5 years ago (09/05/2017)
If you just want to read your frequency (as a figure, as opposed to decoding signal) you either check zero crossings (as I suggested) or just read the peaks of...

Re: Extract a period from a unkown signal

Reply posted 5 years ago (09/03/2017)
it seems you just want to find out frequency value of a detected sinusoid without eye inspection. Well first apply a filter to remove noise. Then detect/measure...

Re: Extract a period from a unkown signal

Reply posted 5 years ago (09/02/2017)
One "easy" way used in FSK is this:m = diff(unwrap(atan2(I,Q)));try it on your signal

Re: Extract a period from a unkown signal

Reply posted 5 years ago (09/01/2017)
if you know the expected frequency and want to extract it then you can generate that tone and correlate with signal to get peaks indicating detection.The other way...
according to rules of filter function you need to add initial 1 for IIR as below:I also realised you are using b,a possibly meaning forward/reverse coeffb1 = [1...

Re: Narrow band filter at high rate

Reply posted 5 years ago (08/31/2017)
Ok but where is the 20KHz in relation to the user signal. I mean there must be user signal to pass through the filter and the filter is meant to get rid of 20KHz...

Re: Narrow band filter at high rate

Reply posted 5 years ago (08/31/2017)
With decimation more filters are needed and so more delay. I am not sure I understand the case raised originally:signal bandwidth is 5MHz, sampling rate is 7.68Msps...
I think you don't mean downsampling but rather finding a time unit (interval) where a pattern repeats then send pattern only and tell Rx start/end of pattern. Obviously...

Re: Narrow band filter at high rate

Reply posted 5 years ago (08/26/2017)
The 7.68 msps is standard lte sampling rate for lte5 which is 2.4MHz but nominally 5MHz complex. Assuming I am right then the 20KHz is outside band of interest and...

Re: Narrow band filter at high rate

Reply posted 5 years ago (08/26/2017)
You can use a notch IIR filter. Sounds you want to remove lte antenna test tone.As to DDC/DUC delay that is a separate issue. 

Re: Downsampling from an FFT

Reply posted 5 years ago (08/24/2017)
note that your fft output is 8192 complex samples and not as 16384.Each Re/Im pair represents one sample 

Re: Downsampling from an FFT

Reply posted 5 years ago (08/24/2017)
Your fft produces Re/Im (complex output) so the bins for your signal at baseband would e symmetrical around dc. you need to zero on either side of dc away from your...

Re: Downsampling from an FFT

Reply posted 5 years ago (08/24/2017)
1) correct, zero both +/- frequencies2) correct3) correctas to which 3? it does affect phase of signal(delay) but otherwise I will discard regularly as you described....

Re: Downsampling from an FFT

Reply posted 5 years ago (08/24/2017)
Since your signal is at FFT output then pass it to IFFT directly then decimate by direct discarding. To avoid aliasing discard (by zeroing), when at FFT output...
I meant magnitude in time domain. I don't see why you want to see spectrum as it is useful for initial checks on moving from RF to baseband. The signal in the...
I will focus on time domain. You already have ON/OFF states in your signal with short and wider durations of sine wave ON followed by OFF noise. (Plot magnitude...
OK I understand RF centre frequency is 433MHz. This is centre of signal at capture but what is the signal meant to be? is it meant to be a sine wave generated...
I can't help with your tool setup. You haven't explained your expectations or nature of capture and you seem trying study a signal by just eyeballing (not best way)....
The data you pass to fft must represent your signal in a sufficient way and you may discard repeated patterns or any section that is not relevant to your observation.Concerning...
Actually I added to above post:You may just use 1/abs(x) as precomputed content to be multiplied at run time by re(x),im(x) & d
In FPGAs we sometimes resort to lookup tables(LUT) for all or part of equations at the expense of memory.In your case you can do all in LUT i.e.you use re(x) and...
In modern FPGAs we can run digital systems @ clock rates of 400~500MHz. ASICs can do more.But from DSP perspective (For Fs) you need to take into account upsampling...

Re: Aliasing effect in RRC filter

Reply posted 5 years ago (06/06/2017)
I suggest that the Rx filter must cut off sharp at f = half symbol rate then decimate to Fs = symbol rate.The question why use 100% roll off at Tx?edit: I also believe...

Re: cross correlation of two images

Reply posted 6 years ago (05/23/2017)
Thanks bmoers,image size is 160x288fft resolution is 256 x 512, padded in mirrored pattern across two images.ideally fft should be (160+160-1) x (288+288-1) using...

Re: cross correlation of two images

Reply posted 6 years ago (05/23/2017)
A follow up.I compared the classic time domain correlation(sliding) with fft based correlation using same input image. Sliding correlation result gets better if feature...

Re: cross correlation of two images

Reply posted 6 years ago (05/22/2017)
Thanks bmoers & DaniloDara,Indeed I tested random 2D data set to various ratios of greyscale values (0~255) & binary (0,1) and I can see the peak quality...

Re: cross correlation of two images

Reply posted 6 years ago (05/21/2017)
It is real time and we have full cross correlation logic available. I am in fact comparing two scenarios, both use correlation:1) direct cross correlation2) feature...

cross correlation of two images

New thread started 6 years ago
Hi All,I am designing cross correlation of two images to detect the offset between them. According to my model it looks like correlation works best if I correlate...

Re: Optimal Filter lineup selection

Reply posted 6 years ago (05/16/2017)
Away from wisdom remember unlike software each fpga module takes effort and time to design and test. less modules less work. I will go for cic (decimating by 1300)...

Re: Optimal Filter lineup selection

Reply posted 6 years ago (05/16/2017)
Time for wisdomTwo opposing forces in nature:==> "Perfection is the enemy of done."==><== "There is a better way to design it, find out" <==

Re: Average FFT Magnitude in bins

Reply posted 6 years ago (05/15/2017)
You are best to research that and tell us. Certainly if you can identify the frequency content (power and its bin location) you can see if there is some pattern...

Re: Average FFT Magnitude in bins

Reply posted 6 years ago (05/15/2017)
...

Re: Average FFT Magnitude in bins

Reply posted 6 years ago (05/15/2017)
pwelch does not just use one fft, it applies several ffts , windowing, scaling and averaging.If you do all that you may get equality.  

Re: Average FFT Magnitude in bins

Reply posted 6 years ago (05/15/2017)
pwelch outputs power estimate (i.e. square of abs of fft). To convert power to db use 10*log10. If it was abs(fft) to db use 20*log10

Re: Average FFT Magnitude in bins

Reply posted 6 years ago (05/15/2017)
yes pwelch can do it all nicely but is not intuitive.

Re: Average FFT Magnitude in bins

Reply posted 6 years ago (05/15/2017)
if your Fs = 44100 Hz and you want fft resolution of 10 Hz then use fft resolution N = 44100/10 = 4410. You can then take just 4410 samples of signal or split the...

Re: Average FFT Magnitude in bins

Reply posted 6 years ago (05/15/2017)
can you explain what set of values you want to average. you say you want to "calculate the average FFT magnitude in each of these bins". but each bin will have...

Re: DFT

Reply posted 6 years ago (05/15/2017)
You can find answers to most of your questions by working on tools like matlab.1) Do not use the term "observation window". That does not make any sense and even...

Re: DTFT of a signal

Reply posted 6 years ago (05/13/2017)
My guess is your code is generating single tone (x) & base functions wrong and your output (X) does not compare well with matlab fft. This apart from wrong frequency...

Re: DTFT of a signal

Reply posted 6 years ago (05/13/2017)
It is DFT. DFT is based on correlating input with cos/sin base functions that are generated over the chosen frequency grid.There are three steps in this exercise:1)...

Re: DTFT of a signal

Reply posted 6 years ago (05/13/2017)
The line: x = (0.9*exp(j*pi/3)).^n;generates a distorted single tone test vectorThe line: b_var = (exp(-j*pi/100)).^(n'*k);generates base tones across frequency...
I did xcorr on your data and found out delay = 176 samples on 8MHz.Then I tried interpolate by 25 (as 125 is too much) and got 4400 samples. I used first 100000...
It is not just large delay but also negative i.e. the receiver gets it before it is sent by tx. This possible if time is reversed & Only God knows!Work out my...

Re: OFDM sampling

Reply posted 6 years ago (05/12/2017)
"Fs is only resident of digital domain and is lost in space"  That is meant to be metaphor. as Fs is applied through a digital clock that ends at DAC. What goes...

Re: OFDM sampling

Reply posted 6 years ago (05/12/2017)
The actual bw in tx signal is 5MHz. Fs is only resident of digital domain and is lost in space. The signal inside digital domain in fact will be upsampled to whatever...

Re: OFDM sampling

Reply posted 6 years ago (05/12/2017)
The connection of sampling rate(Fs) is that it physically decides the bw as follows:signal bw = Fs/nFFT * Number of active carriers.where Fs/nFFT = bw of one binFor...
Here is my example using interp then xcorr:tx = randn(1,1000);rx = [0 0 0 tx(1:end-3)];tx = interp(tx,125);rx = interp(rx,125);y = xcorr(tx,rx);L = length(tx);m...
an interpolation rate of 125 will take you to 1 ns accuracy and I suggest you do that on signal1 and signal2 instead of doing it on xcorr result and find delay from...
Sounds a familiar professor mystery problem/solution.What I can tell is that xcorr of a pair of tx data/rx data helps find out time delay from start of tx data...

Re: Magnitude/phase vs real/imaginary

Reply posted 6 years ago (05/11/2017)
you can plot Re/Im separately or one against the other but plotting both values as pair against time requires 3D and is pointless.
if you want avoid complex filter you can downconvert somewhere near dc apply ordinary filter on both Re/Im to remove one sideband then downconvert to dc
As any filter, yes you will have to cut sharp enough and tolerate latency. You will need really good filter with brickwall cutoff to do what you target.  You might...
Have you thought of downconverting to baseband followed by complex filter?
I understand you are doing demodulation but you still need to start from a properly SSB modulated signal for your demod testing.
Just a thought, my understanding of USB & LSB modulation is that you have to modulate by removing one sideband before transmission. You are not doing that, you...
you are not just measuring air delay but DAC/ADC/antennas/Rf/ and any digital processing involved and you also need to have some trigger so that you measure the...
check the code works by directly counting samples and you might try inserting zeros at end of short vector. Then getting microseconds is not surprising especially...
well according to your diagram they correlate. You should use one burst of tx and capture Rx with some sort of synchronised trigger. then Rx vector would be different...
it seems your time scale is not helpful as such. So you have two bursts.you only need to correlate once with some sort of sync as to which rx relates to tx.e.g.if...
I don't see how you can correlate these two signals. As far as see they are so different. Apart from noise, they could be on different sampling rate or different...
You have it all wrong!!I assume you have signal1(tx) and signal2(Rx) both at same sampling rate and same centre frequency. Then you only need to correlate the pair once,...

Re: Magnitude/phase vs real/imaginary

Reply posted 6 years ago (05/10/2017)
Not much, I will leave rectangular form for computations. I and you as "human GUI" will look at amplitude/phase.for single tone you can infer some conclusions

Re: Magnitude/phase vs real/imaginary

Reply posted 6 years ago (05/10/2017)
Not just DFT but in general it is rectangular Versus polar versions of same vector.They carry same info. For us amplitude & phase is more visual but for computations...

Re: DSP Filter Verification in FPGA

Reply posted 6 years ago (05/10/2017)
Once your coeff sum is 1 then any pre-scaling of coeffs by 2^n should be followed by descaling by /2^n of final sum. Using power of 2 is convenient instead of using...

Re: DSP Filter Verification in FPGA

Reply posted 6 years ago (05/08/2017)
so your conclusion is wrong...

Re: DSP Filter Verification in FPGA

Reply posted 6 years ago (05/08/2017)
As suggested by Fred, one easy quick test (but partial) is to inject a constant input e.g. your maximum of +32767 for 16 bits signed input for one sample only....

Re: DSP Filter Verification in FPGA

Reply posted 6 years ago (05/08/2017)
possibility 1: your observation is wrongpossibility2: your conclusion is wrongpossibility3: your platform is doing something wrongare you checking in simulation...

Re: DSP Filter Verification in FPGA

Reply posted 6 years ago (05/08/2017)
srid, Though your post is mixing between FIR (coeffs) & CIC but I guess you are using both.Regarding FIR scaling (without upsampling by zero insertion), you...

Re: DSP Filter Verification in FPGA

Reply posted 6 years ago (05/08/2017)
thanks dudelsound, got your perspective.

Re: DSP Filter Verification in FPGA

Reply posted 6 years ago (05/08/2017)
you truncate as required by how you pre-scaled coeffs. This is unrelated to issue of 16 x 16 => 31 bits. In fact it should be 32 bits since the max negative value...

Re: DSP Filter Verification in FPGA

Reply posted 6 years ago (05/08/2017)
dudelsound,If you truncate (n) LSBS from a data bus you are dividing its value by 2^n.That applies always and irrespective of any other issues.

Re: DSP Filter Verification in FPGA

Reply posted 6 years ago (05/06/2017)
srid, don't get it wrong. The interpolation effect on gain relates to classic use of FIR as interpolator by zero insertion. It does not apply to CIC. I have posted...

Re: DSP Filter Verification in FPGA

Reply posted 6 years ago (05/06/2017)
For interpolator CIC each comb stage requires input width plus oneextra bit while the bit growth at each integrator depends on stageand the final integrator gain...

Re: DSP Filter Verification in FPGA

Reply posted 6 years ago (05/05/2017)
CIC filter is unique and has its own rules of gain control. There is plenty literature around. 

Re: DSP Filter Verification in FPGA

Reply posted 6 years ago (05/05/2017)
No need to re-multiply. If you truncate 15 LSBs it gives a gain of 2.In this case you need to discard one MSB and may need clipping but you can pre-compute that...

Re: DSP Filter Verification in FPGA

Reply posted 6 years ago (05/05/2017)
if coeff sum is 2^16 and you truncate 16 LSBs off sum then that should be ok unless you are upsampling in which case you need to increase gain by the upsampling...

Re: DFT of a signal and system

Reply posted 6 years ago (05/05/2017)
when you filter a signal you just filter it. Your question is not applicable.My point of windowing before DFT applies if you want to assess frequency spectrum...

Re: DFT of a signal and system

Reply posted 6 years ago (05/05/2017)
One issue is that you are free to window a signal vector to avoid sharp phase discontinuity and so get rid of false high frequencies but you should not window an...

Re: sum of sinusoids

Reply posted 6 years ago (04/27/2017)
I prefer to say:"The addition of two sinusoids, having different frequencies generates a repeating pattern whose amplitude is sinusoidally modulated but is never...

Re: sum of sinusoids

Reply posted 6 years ago (04/26/2017)
Hi RickYou don't really mean it in frequency domain[or in time domain]! 

Re: PAPR (peak to average power ratio)

Reply posted 6 years ago (04/14/2017)
ofdm signal being the sum of sine waves at different phases & amplitudes has inherently high PAR. Reduction methods are possible using some inhouse techniques...

Re: PAPR (peak to average power ratio)

Reply posted 6 years ago (04/14/2017)
I agree with Tim on his best ever practical reply:"From the perspective of the PA, the preferred PAPR is 1:1So from the ivory-tower comms mathematician perspective,...

Re: PAPR (peak to average power ratio)

Reply posted 6 years ago (04/13/2017)
I guess you mean LTE. It Depends on type of PA & any predistortion. We used to target PAR of 6 dB ish

Re: DDC in FPGA with high speed ADC

Reply posted 6 years ago (04/10/2017)
I would rather go for direct down conversion to dc by rotating by 1000MHz then apply one fractional decimator to downsample from 3200MHz to 500MHz. This is so since...

Re: Deriving FIR coefficients for a higher sample rate

Reply posted 6 years ago (04/10/2017)
If your filter has n taps and you use fft method I suggest: you use n fft resolution (you may use high resolution but then it requires truncation).Pad the fft bins...

Re: DDC in FPGA with high speed ADC

Reply posted 6 years ago (04/09/2017)
I think both approaches to down conversion are possible i.e to 200MHz then dc or direct to dc. A parallel mixer is not a problem.However the downsampling filter(s)...

Re: Deriving FIR coefficients for a higher sample rate

Reply posted 6 years ago (04/06/2017)
Thanks David,It works. Interesting method but hard for me to understand its principles.  

Re: Deriving FIR coefficients for a higher sample rate

Reply posted 6 years ago (04/06/2017)
Hi David,Not sure about your method so I tried in Matlab:%%%%%%%%%%%%%%%%%h1 = fir1(20,.25);h2 = conv(h1(1:2:end),h1(2:2:end));freqz(h1);freqz(h2);%%%%%%%%%%%%%%%%%h2...

Re: Deriving FIR coefficients for a higher sample rate

Reply posted 6 years ago (04/06/2017)
I suggest using fft on your given filter then zero pad the bins either side to get twice frequency axis then back to time domain. Just a thought.

Re: Can we paralleize the integrator stage of CIC?

Reply posted 6 years ago (04/06/2017)
Yes pipeline the adder. This introduces one delay stage but to all stream. So I believe you can then match this delay at next stage adders.
this blog might helphttps://www.dsprelated.com/showarticle/965.php

Re: Happy Pi Day

Reply posted 6 years ago (03/14/2017)
you missed (2 + 15 = 17), (1 +16 = 17) ...

Re: Bit width in a digital delta sigma

Reply posted 6 years ago (03/13/2017)
...

Re: Standard deviation in DSP

Reply posted 6 years ago (03/12/2017)
The way I visualise this issue (for electronic signals at least) is that the mean is the "dc" offset while standard of deviation is the "AC" mean. Thus if mean...

Re: measure signal power

Reply posted 6 years ago (03/07/2017)
I mean that when it comes to power addition you can add power of two or more signals if they are not correlated else the sum will vary depending on how much they...

Re: measure signal power

Reply posted 6 years ago (03/07/2017)
Well have you considered correlation issues. To explain that what is the power of two sine waves at same frequency but opposite in phase (zero isn't it) 
taking a simple example (cos) its analytic is(cos+jsin).any scaling is matter of design but matlab keeps unity of amplitude:x = cos(2*pi*(0:1023)*.01);y = hilbert(x);plot(x);holdplot(real(y));...

Re: measure signal power

Reply posted 6 years ago (03/05/2017)
There are some funny scaling issues.you can avoid all by just this command:freqz(h);

Re: measure signal power

Reply posted 6 years ago (03/04/2017)
As explained by Rick you look at input/output spectrum.For unit impulse input, the spectrum will be a nice flat top all over frequencies. The filtered output of...

Re: measure signal power

Reply posted 6 years ago (03/04/2017)
I think your purpose is to measure effect of filter on power rather than measure power per se.The best way (in software) is to inject an impulse input of unity power...

Re: measure signal power

Reply posted 6 years ago (03/03/2017)
rms = square root of mean of squared values.so square each sample, get the mean of squares (this is mean power)get square root of that if you wish
If I want to measure delay between two signals (especially sinusoids) I just do correlation and find out index of maximum correlation. This works as long as the...

Re: Cyclic prefix concept in LTE

Reply posted 6 years ago (03/01/2017)
All the replies in this thread are useful and I will add this link:http://www.telecomhall.com/what-is-cp-cyclic-prefi...for basic visualisation of cyclic prefix...

Re: On-Off Keying Communications

Reply posted 6 years ago (02/27/2017)
read about timing and symbol recovery in BPSK/QPSK receivers and in particular fractional interpolation

Re: On-Off Keying Communications

Reply posted 6 years ago (02/27/2017)
The post is not about Fs at Tx but BW. Normally pulse shaping filter is used which implies upsampling.At Rx still the minimum is 2B but a higher Fs means less...

Re: On-Off Keying Communications

Reply posted 6 years ago (02/27/2017)
bandwidth of 250MHz is enough to carry info. actual bandwidth is higher but you filter it to 250 (pulse shaping).minimum Fs at receiver is 2B if signal is centred...

Re: Filter design

Reply posted 6 years ago (02/27/2017)
There are two perspectives (and this can cause confusion)1) you are in digital domain(Matlab) then you apply Fs on your digital stream (x input of a filter) then...
Indeed a precision running RMS algorithm as required by op is very extreme case but it could be part of academic research or some unknown corners of a control...
Hi Fred,you are right about rms formulas predefined for some known ideal waveforms.The OP - I believe - wants to measure it in running mode rather than calculate...
I am thinking of continuous phase detection using an NCO to decompose signal to I/Q (relative to a fixed nco frequency) then use the I/Q result to adjust window...
so basically you are concerned about the transition section say from f1 to f2.In that case I assume some feedback mechanism may do. I am not sure how but will think about...

Re: Filter design

Reply posted 6 years ago (02/24/2017)
Higher Fs but same absolute cutoff means sharper filter and hence larger NKaz
If you know your new frequency (after deviation) then resize window.If you don't know about new frequency then I believe you have to increase window size to an acceptable...

Re: Filter design

Reply posted 6 years ago (02/24/2017)
f1 or f2 also change with Fs and proportionallykaz

Re: Filter design

Reply posted 6 years ago (02/22/2017)
Any digital system has no sense of time until clock is applied to sample the signal.So is matlab, DSP, FPGA, ASIC etc. cutoff is relative to Fs as ratio.Unfortunately...

Re: Amplitude calculation from cross correlation data

Reply posted 6 years ago (02/20/2017)
Hi Charansai, that looks good idea but to get correlation equivalent mathematically between time domain and frequency domain care is needed regarding Fourier resolution,...

Re: Any fancy idea for an algorithm ?

Reply posted 6 years ago (02/20/2017)
as far as I know Fourier transform followed by multiplication (plus reversal) is equivalent to time domain correlation if done with correct resolution.Kaz

Re: Any fancy idea for an algorithm ?

Reply posted 6 years ago (02/20/2017)
indeed. You may also correlate each and every tile with each of all other tiles just to get more info.Kaz

Re: Any fancy idea for an algorithm ?

Reply posted 6 years ago (02/20/2017)
I think the request of the op is clear. I have no idea apart from correlation but you can also divide into tiles and correlate tile by tile as well then sort out...

Re: Frequency multiplication in the digital domain

Reply posted 6 years ago (02/06/2017)
with more effort to understand your post I think two NCOs may do.one NCO use to lock in closed loop to input frequency(e.g. 10Hz). second NCO open loop to generate...

Re: Frequency multiplication in the digital domain

Reply posted 6 years ago (02/05/2017)
if I understood you just generate in an NCO 32,64,128 Hz and multiply by input.Kaz
the concepts must first be explained without any numbers then fortified/quantified with equations. Not the other way around.Kaz

Re: Spurious when Fs/F is not an integer

Reply posted 6 years ago (01/24/2017)
Indeed it is the other way round. if a frequency correlates with Fs it will result in spikes and dither helps here.Kaz

Re: Spurious when Fs/F is not an integer

Reply posted 6 years ago (01/23/2017)
A common practice for NCOs is phase dither. You can model that in Matlab to find optimum ditherKaz
Hi KevinjkriegerIf FIR is designed in baseband and then frequency shifted to 12MHz then it is no problem and it will filter the signal around 12MHz but will not...
I personally don't get it. Though it is just my immediate thoughts. If you multiply filter by frequency(f) it will move the filter centre to (f) frequency before...

Re: 3D fft Matlab

Reply posted 6 years ago (12/15/2016)
£D fft is not a problem in Matlab, here is how Matlab suggests:X = randn(5,5,5);Y = X;for p = 1:length(size(X))    Y = fft(Y,[],p);endThe OP is not having difficulty...

Re: Matlab Home-use version

Reply posted 6 years ago (11/11/2016)
Matlab uses fliplr & flipud on sets so does Kardashians

Re: Understanding maths behind DFT?

Reply posted 6 years ago (11/10/2016)
If you drive a car and think how it works you know what is going to happen.If I use a resistor do I need to know all about electron and hole currents ...etc??It...

Re: two image correlation

Reply posted 6 years ago (11/07/2016)
Thanks Andy, your reply is what I needed to hear.

Re: two image correlation

Reply posted 6 years ago (11/07/2016)
x/y offset no problem in principle. I am asking how best to have a measure of confidence in the result in a running system when noise is also likely. Using software...

two image correlation

New thread started 6 years ago
We use 2D cross correlation in fpga platform to find x/y offset of two images. The images are meant to be same but with minor differences due to small movement...

Re: Downsampling from 2.0 MHz to 192kHz

Reply posted 6 years ago (10/28/2016)
A sampling conversion of I/D (in FPGAs) is conveniently implemented as follows:a filter is designed with cuoff for most restrictive case. for 12/125 it is 0.008 of...
I am looking at the cascade from frequency perspective. We Interpolate => filter => decimate. So how can we decimate before filtering? or am I am in a different...

Re: Sampling frequency in Baseband

Reply posted 6 years ago (10/12/2016)
yes, one ofdm symbol takes 2048 data points @30.72MHz (1/30.72*2048) = 66.67 us.This should never change. Upsampling (when chosen) increases the number of data points by...

Re: Sampling frequency in Baseband

Reply posted 6 years ago (10/11/2016)
...

Re: LTE Downlink TX understanding

Reply posted 6 years ago (09/26/2016)
For OFDM I think the concept of time domain/frequency domain is confusing to beginners.I will just say I got symbols (zeros/ones or if you want +/-1 and want a frequency(sine/cos)...

Re: LTE Downlink TX understanding

Reply posted 6 years ago (09/24/2016)
The 20MHz LTE must be sampled physically at 30.72MHz then it results in 18.0MHz bandwidth (nominally known as 20MHz LTE), you can work out bandwidth as follows: 30.72*1200/2048...

Re: Invert Phase Response

Reply posted 6 years ago (09/19/2016)
apologies, the terms are indeed normalised for dc gain since sum(num)/sum(den) = 1.However matlab convention is that the first recursion term is set to unity....

Re: Invert Phase Response

Reply posted 6 years ago (09/19/2016)
I don't see dc gain normalised. Moreover we need to add 1 to first den(once normalised)Kaz

Re: Upper bound on the decimation factor M

Reply posted 6 years ago (08/18/2016)
Correction: B = 0.5(M = 1, no decimation possible) B = 0.25Fs(M = 2, not efficient relative to single rate filter) B = 0.125Fs (M = 4, efficiency starts here)...

Re: Upper bound on the decimation factor M

Reply posted 6 years ago (08/18/2016)
The document(page 9) says B to be Fs/4 for design to be more efficient than using a single rate filter.imagine B = 0.5 Fs (M = 1, i.e. no decimation)B = 0.4Fs(M...

Re: Removal of DC Spike from quadrature receiver

Reply posted 6 years ago (08/15/2016)
so random data with zero mean has got zero dc component??Kaz

Re: Removal of DC Spike from quadrature receiver

Reply posted 6 years ago (08/14/2016)
I believe that dc component is either dc offset which will become zero if this offset is removed by subtracting mean or dc component is part of a flat topped spectrum...

Re: How to estimate the SNR

Reply posted 6 years ago (08/12/2016)
When using peak of single tone/spectrum floor as SNR you need to take into account that it will vary with fft resolution and this artefact need to be subtracted....

Re: How to estimate the SNR

Reply posted 6 years ago (08/12/2016)
If your fft method worked for single tone SNR then it can work for your band limited signal provided you target out-of-band noise.Using fft method, you must subtract...

Re: Matlab - How to plot magnitude error

Reply posted 6 years ago (08/01/2016)
or just use freqz to plot in one single line:freqz(num,den,-50:0.01:50,100);dc centred, fs of 100, range -50 ~ + 50, resolution of 100/.01Kaz 

Re: fft combinations

Reply posted 6 years ago (07/14/2016)
Thanks LabPE43,May I ask about the two cases of "OR", how could both cases be correct when case 1 asks for extra multiplications (exp(-1*j*2*m...). And if you...

Re: fft combinations

Reply posted 6 years ago (07/10/2016)
Thanks Rick,In my case I have "complex" input of 512 points and so it does not seem your diagram applies. Moreover I need do ifft of 512 using 256 ifft. And in...

Re: Zero IF vs Low IF receivers

Reply posted 6 years ago (07/10/2016)
Not sure I can understand that if signal is centred on 8MHz then downconverted to 200KHz or to zero or anywhere the bleed through of carrier will always be there...

Re: fft combinations

Reply posted 6 years ago (07/09/2016)
In  my case zero padding does the job. we are using fft/ifft to implement cross correlation of two images by multiplying in frequency domain instead of sliding...

Re: fft combinations

Reply posted 6 years ago (07/09/2016)
apologies Raph for misspelling your name.Also the word column in my above post should be dropped.Kaz

Re: fft combinations

Reply posted 6 years ago (07/09/2016)
Thanks Ralph,That is an option on the table but we need the fft to finish early enough for 256 case. Duration of 512 is too much as we are doing column 2D fft on...

Re: fft combinations

Reply posted 6 years ago (07/09/2016)
Thanks Rick,Just to make sure you are aware of my setup.I use FPGA platform(not software) and use Altera's fft ip core set to 256 points on complex input and get 256...

fft combinations

New thread started 6 years ago
Hi all,I got an FPGA design running #FFT at 256 points. Our system requires fft be configured as either 256 or 512 points on the run. I can use variable fft that...

Re: A digital filter question

Reply posted 6 years ago (07/06/2016)
Looking at dc removal case there are two ways to remove dc:1) forward subtraction of running average from input.2) feedback subtraction of running average from incoming...

Re: Restorative Upsampling

Reply posted 6 years ago (07/02/2016)
The terminology of rows/columns of a matrix can be confusing.  I want to add these notes:Matlab(and hence Octave) views x(r,c) as r = number of rows, c =  number...
when you apply shaping filter you need to upsample by 2(minimum). when you receive upsampled shaped stream you apply matched filter then downsample and choose correct...

Re: Pass band in digital up conversion of LTE signals

Reply posted 6 years ago (06/12/2016)
no the filter need not be complex.I and Q are filtered separately with a filter of passband = half of lte bandwidth in air. This was already explained by Tim....

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