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…”_* 😀
Good explanation to the basics. i would like to add that you also need another band pass filter at the RF stage to counteract the signals at image frequencies
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.
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.
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
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 ?
So the reason you multiply/mix the input by an analog RF tuner, is only to bring the signal to the area of the band-pass hardware filter? Would Direct-Sampling remove this Intermediate-Frequency requirement, and if so, why isn't it the preferred approach? However, it still feels like a free lunch that you can sample and get a *single* ADC value, but then digitally mix (DDC) to get 2 baseband samples (the cosine (Q) and sine data (I))! After-all, couldn't more operations be done on the same sample to get even more bandwidth?
At 4:45 you say sin of f2 is real and cosin of f2 is imaginary, but in the previous slide this was reversed; sin was imaginary and cosin was real. Can you explain this discrepancy?
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.
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.
It's a most perfect fundamental SDR presentation I've ever seen. Thanks for your contribution.
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…”_* 😀
Excellent presentation of software Radio with much needed basics
Thank you for this clear presentation. It was very helpful!
Hello. This video has been one of the best videos on IQ modulation and demodulation. Thank you.
Thx for the explanation. It REALLY helps me a lot
very amazing explanation, now i understand about SDR include I/Q Signal
The ultimate presentation I have ever seen on SDR. Thanks for your such a nice presentation.
great explanation! clear and persuasive. Many thanks for sharing and making understanding simpler ;)
It's crisp and clear explanation of SDR and any lay man can also understand the modules required for SDR.
Outstanding presentation! Presents a complex topic very clearly and in an easy to understand manner! Thank you!
Thanks a lot. This is the best explanations on SDR that I've ever seen. This is the real deal.
Thank you for a clear explanation.
This was extremely helpful. Thank you.
This is wonderful. I just found it on youtube. Thank you very much.
Thanks a lot for the useful lecture!
Thanks for such a simple to understand video, its helped me understand a new subject.
Very impressive. & easy to digest. Really a wonderful job.
Good explanation to the basics. i would like to add that you also need another band pass filter at the RF stage to counteract the signals at image frequencies
Wow. Fantastic. Thanks!
Excellent presentation!! Thank you!
This video is really helpful for me. Thank you very much
Clear explanation of presentation, thanks
Excellent explanation thank you so much
Very good explanations.
Well done!
Thank you
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.
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.
omg. Finally I got my answers
At 7:40 I believe that’s only true when you don’t care about the negative frequencies.
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
Perfect and Simple and Humble and to this and to tan and con and the back! Perfect Okay!
is there anyone who can help me how to design a GNSS/GPS receiver using Simulink or GNU Radio
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 ?
The term bandwidth is used profusely in this presentation but is not defined mathematically or graphically in this presentation as far as I can see.
So the reason you multiply/mix the input by an analog RF tuner, is only to bring the signal to the area of the band-pass hardware filter? Would Direct-Sampling remove this Intermediate-Frequency requirement, and if so, why isn't it the preferred approach? However, it still feels like a free lunch that you can sample and get a *single* ADC value, but then digitally mix (DDC) to get 2 baseband samples (the cosine (Q) and sine data (I))! After-all, couldn't more operations be done on the same sample to get even more bandwidth?
Because of the low sampling rate that original waveform is being sampled with large gaps, so the phase shift allows you to sample that missing part.
you can but you need expansive ADC , the circuit would be antenna -->LNA->BPF->ADC
At 4:45 you say sin of f2 is real and cosin of f2 is imaginary, but in the previous slide this was reversed; sin was imaginary and cosin was real. Can you explain this discrepancy?
It's wrong. cos is real (i), sin is imag (q). The slide is incorrect.
@@m1geo Thanks, I was totally confused until I read your comment
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.
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.
Thank you