Quadrature low pass filter question
in [DSP]
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From: jacobfenton on 11 May 2010 13:00 I and Q are used for FM demodulation, it seems I would not want to have non-linear phase. -JF
From: Jerry Avins on 11 May 2010 13:29 On 5/11/2010 12:46 PM, Randy Yates wrote: > Jerry Avins<jya(a)ieee.org> writes: >> [...] >> It's the same thing. > > So, e.g., using complex sampling doesn't allow you to gather > more bandwidth using today's technology in ADCs than real > sampling does? Numbers will clarify what I mean. Sampling I and Q each at 1 KHz suffices for a (nearly) 1 KHz bandwidth. You should expect no less from the 2000 samples/second that you are collecting. Jerry -- "I view the progress of science as ... the slow erosion of the tendency to dichotomize." --Barbara Smuts, U. Mich. �����������������������������������������������������������������������
From: glen herrmannsfeldt on 11 May 2010 13:30 Randy Yates <yates(a)ieee.org> wrote: > Jerry Avins <jya(a)ieee.org> writes: >> [...] >> It's the same thing. > So, e.g., using complex sampling doesn't allow you to gather > more bandwidth using today's technology in ADCs than real > sampling does? There are many tricks that could be used to sample faster than ADC technology allows. You could, for example, use two ADCs and the appropriate sample and hold such that each one received every other sample (and correct for any differences in the two ADCs later). Though the system I still remember from many years ago (likely still used, but with different numbers): Sample at high speed but at reduced resolution, such as only four bits per sample. Convert the result back to analog with an equivalent speed DAC. Analog subtract from the original (possibly delayed through sample and hold type circuitry). Send the analog difference to another ADC, to get the low bits of the result. This works if the threshold levels of the first ADC are accurate enough, yet allows for a much faster conversion. Otherwise, it reminds me of stories from the early days of CD players, using one ADC for both stereo channels, alternating between the two. The resulting half sample period delay was said to be audible by some people. -- glen
From: Randy Yates on 11 May 2010 15:30 glen herrmannsfeldt <gah(a)ugcs.caltech.edu> writes: > Randy Yates <yates(a)ieee.org> wrote: >> Jerry Avins <jya(a)ieee.org> writes: >>> [...] >>> It's the same thing. > >> So, e.g., using complex sampling doesn't allow you to gather >> more bandwidth using today's technology in ADCs than real >> sampling does? > > There are many tricks that could be used to sample faster > than ADC technology allows. True, but those wouldn't be considered plain old real sampling, would they? I.e., apples to oranges. > [...] > Otherwise, it reminds me of stories from the early days of > CD players, using one ADC for both stereo channels, alternating > between the two. The resulting half sample period delay was > said to be audible by some people. I remember that! I do remember my 2nd-generation player sounding MUCH better than the first, especially in the high-end. I do not know why, but I suspected it was the phase distortion in the brickwall filters of the first player. -- Randy Yates % "And all that I can do Digital Signal Labs % is say I'm sorry, mailto://yates(a)ieee.org % that's the way it goes..." http://www.digitalsignallabs.com % Getting To The Point', *Balance of Power*, ELO
From: robert bristow-johnson on 11 May 2010 15:54
On May 11, 12:34 pm, Jerry Avins <j...(a)ieee.org> wrote: > > For complex signals, there are two samples per > sample period: one real, the other imaginary. That is effectively > sampling twice as fast, so the bandwidth is doubled. > > If one wants to get mathematically hoity-toity, one can say that real > sampling covers the range from -fs/2 to +fs/2, while I/Q sampling covers > the range from 0 to fs. not if Q is the Hilbert of I. if Q = Hilbert{I}, then the range is 0 to fs/2. > Personally, I think that obscures the truth that > there really /are/ twice as many samples. The second set of samples > needn't be Q. d(Real)/dt, for example, would serve as well. the way to compare the two is that for real signals, the sampled spectrum that covers -fs/2 < f < +fs/2 and that conjugate symmetry so that a spectrum 0 < f < +fs/2 has just as much information (it would no longer be purely real, but the imaginary part, which is the Hilbert of the real part, contains no more information than just the real part alone). then for a *general* I/Q signal (where maybe we cannot assume that Q is the Hilbert of I) then it's twice as much information, but i am not sure that it would be 0 < f < +fs. not all complex signals have missing negative frequency components. r b-j |