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I absolutely *LOVED* this presentation! I graduated in 1976 with a BSEE, but all of this was a practical impossibility due to the relative slowness of existing compute power. Imagine trying to implement these ideas with an 8 bit microprocessor (think Z-80) running at a _”leisurely”_ 4 mHz.! And yet … all of the *COMPLEX MATH* was possible, just nothing fast enough to run it! Perhaps we ought to think ahead and create (theoretical) concepts, even if they have to wait *DECADES* to be practical. You know, the old humorous chemical equations that had a block labeled *_”An then a miracle occurs…”_* 😀
Yeah, sorry if it's there and I've just missed it, but but how seems like you left out something major, i.e. HOW does this "digital complex mixer" get I and Q from just the sampled amplitude values? From what I could gather from the source code of uSDX (seemingly the only source on the subject), what it does is sample at 2 * Fs, treating even samples as I, odd as Q, but linearly interpolates the Is, replacing each one with the average of it and the next one. What this clearly does do is shift one of the streams in time by one sample period, so that the synthetic Is coincide with Qs. While the thing does work in the end, - the device does seem to receive SSB, which is the only use for the I/Q data there, as far as I can tell, - I can't possibly see a) how can those I values be correct beyond the simple assumption that both domains are (-1; 1), and, more importantly, b) any basis for the downright strange assumption that a shift by one sample in the array is equivalent to a 90 degree phase shift. This technique, which quacks like a dirty hack for the anemic 8-bit MCU it's written for, is called a "Hilbert transform" in the comments. Whatever. Correct or not, the whole transition from amplitude samples to I, Q pairs doesn't seem like an "implementation detail" you can just omit here, especially when the end result of the formulation is supposed to be a general SDR, and not some special case transceiver.
Excellent presentation. But the devil is in the details. Where do I get an ADC and a DAC breakout board at a cheap enough price( < $20) that will handle 200 MSPS ?
Now we also have a different kind of SERDES protocols, such as ESI stream that seems to be having same charecteristics of JESD with deterministic latency.
I think even if we consider the clock cycles of 1600, and the JESD204 works happily at 4.5Ghz it is at 355ns, and for the LVDS it is around 99ns considering the 1Ghz sample. So yep 4 times slower, so detection capability goes down accordingly.
What is the benchmark for an acceptable synchronous delay? Is it measured w.r.t to the sample rate? In your measurement, 27 ps time delay is 10% of the sample period, how do we know that's good enough. What are the results of out of sync data in time and frequency domain. Thanks
Looking for a detailed overview of the Tyloe quadrature demodulator that shows how the recombination of the 4 samples, 0/270, 90/180, results in the I and Q signals.
Great video, its incredible how a deep understanding of a complex system by a skilled operator can do wonders. Check out the story of Zoltan Dani who modified an outdated radar system to "see" the stealth aircraft.
Great presentation! As a front end guy I got roped into helping with the digital side of things. Your talk really helped to clarify a few things. Question though: On slide 11. If the situation were revered and we were working with a D/A, would the exact same thing happen? The D/A Fs = 40 Mhz, and we are trying to produce 30 Mhz, we would see an alias at 10 MHz?
Hi, everyone just a question regarding the SDR transmission, where will modulation occur? I presume in the DSP stage but at what frequency? Why cant it be directly modulated onto the IF frequency avoiding the need for the digital upconverter? Any response appreciated
Very nice presentation but... 5:12: there is a mistake in the first formula. You say: cos(f1)sin(f2) = 1/2*{sin(f1-f2) + sin(f1+f2)} which is wrong. It should be: cos(f1)sin(f2) = 1/2{sin(f2-f1) + sin(f1+f2)}. Second one is OK. Both formula should be presented in this way: cos(f1)sin(f2) = 1/2{sin(f2-f1) + sin(f2-f1)} cos(f1)cos(f2)=1/2{cos(f2-f1) + cos(f2+f1)}
Actually this is not really wrong since a negative frequency in the cos is the same as a cos with the positive frequency. Negative frequencies get "mirrored" up in the spektrum. Its only better to undertand in this case.
5:35 I don’t understand what you mean on simplify it by removing the high frequency component. Why don’t you simplify it by multiplying by 0? That would be more simple.
