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"Quadrature signals : Complex, But not complicated" by Richard Lyons

Started by seb August 21, 2003
Hy,

I recently read this good article of Richard Lyons but there is
something i can not understand.


Does someone could help me understand this difficulty ?

It it says :
"Each A/D converter operates at HALF the sampling rate of standard
real-sampling" !!!

How does it work ? It seems that using half the sampling rate of
standard real-sampling, we obtain the same frequency definition as for
a full sampling rate of a "normal" signal ?
seb wrote:
> Hy, > > I recently read this good article of Richard Lyons but there is > something i can not understand. > > > Does someone could help me understand this difficulty ? > > It it says : > "Each A/D converter operates at HALF the sampling rate of standard > real-sampling" !!! > > How does it work ? It seems that using half the sampling rate of > standard real-sampling, we obtain the same frequency definition as for > a full sampling rate of a "normal" signal ?
The Nyquist sample rate for real-valued signals is twice the highest frequency. For complex signals, the Nyquist sample rate is *equivalent* to the highest frequency. OUP
seb wrote:
> > Hy, > > I recently read this good article of Richard Lyons but there is > something i can not understand. > > Does someone could help me understand this difficulty ? > > It it says : > "Each A/D converter operates at HALF the sampling rate of standard > real-sampling" !!! > > How does it work ? It seems that using half the sampling rate of > standard real-sampling, we obtain the same frequency definition as for > a full sampling rate of a "normal" signal ?
What matters is how many samples are collected in a given time. The number of samples collected by a pair of A/Ds acting alternately in a single analog signal (or simultaneously on a quadrature-shifted pair) is the same as the number collected by a single A/D running twice as fast. Jerry -- Engineering is the art of making what you want from things you can get. �����������������������������������������������������������������������
On Fri, 22 Aug 2003 13:52:20 -0400, Jerry Avins <jya@ieee.org> wrote:

>seb wrote: >> >> Hy, >> >> I recently read this good article of Richard Lyons but there is >> something i can not understand. >> >> Does someone could help me understand this difficulty ? >> >> It it says : >> "Each A/D converter operates at HALF the sampling rate of standard >> real-sampling" !!! >> >> How does it work ? It seems that using half the sampling rate of >> standard real-sampling, we obtain the same frequency definition as for >> a full sampling rate of a "normal" signal ? > >What matters is how many samples are collected in a given time. The >number of samples collected by a pair of A/Ds acting alternately in a >single analog signal (or simultaneously on a quadrature-shifted pair) is >the same as the number collected by a single A/D running twice as fast. > >Jerry
Hi Jer, Yep. To give some numbers for seb (original poster); if the original analog bandpass signal had a bandwidth of roughly 10 kHz, we know that normal real (bandpass) sampling would require a sample rate just greater than 20K samples/sec if you used a single A/D. With the quad sampling scheme (also called "complex down-conversion"), two A/D converters would each be running at a rate just greater than 20K/2, or 10K samples/sec. In both case, 20K samples/sec would be generated. In both cases we're satisfying the Nyquist Theorem (named by the great Claude Shannon who died last year with Alzheimer's disease.) [-Rick-]
ricklyon@REMOVE.onemain.com (Rick Lyons) wrote in message news:<3f474377.69195531@news.west.earthlink.net>...
> On Fri, 22 Aug 2003 13:52:20 -0400, Jerry Avins <jya@ieee.org> wrote: > > >seb wrote: > >> > >> Hy, > >> > >> I recently read this good article of Richard Lyons but there is > >> something i can not understand. > >> > >> Does someone could help me understand this difficulty ? > >> > >> It it says : > >> "Each A/D converter operates at HALF the sampling rate of standard > >> real-sampling" !!! > >> > >> How does it work ? It seems that using half the sampling rate of > >> standard real-sampling, we obtain the same frequency definition as for > >> a full sampling rate of a "normal" signal ? > > > >What matters is how many samples are collected in a given time. The > >number of samples collected by a pair of A/Ds acting alternately in a > >single analog signal (or simultaneously on a quadrature-shifted pair) is > >the same as the number collected by a single A/D running twice as fast. > > > >Jerry > > Hi Jer, > > Yep. To give some numbers for seb (original poster); > if the original analog bandpass signal had a bandwidth of > roughly 10 kHz, we know that normal real (bandpass) > sampling would require a sample rate just greater than > 20K samples/sec if you used a single A/D. With the > quad sampling scheme (also called "complex down-conversion"), > two A/D converters would each be running at a rate just > greater than 20K/2, or 10K samples/sec. > > In both case, 20K samples/sec would be generated. > In both cases we're satisfying the Nyquist Theorem > (named by the great Claude Shannon who died last > year with Alzheimer's disease.) > > [-Rick-]
Thinks, It is well understood (excuse my poor english, technical english is easy to read but wrinting is more difficult). There is something else with this paper: There is an example of application. When plug the in-phase and quadrature-phase to the horizontaly and vertical input of a scope, we observe a circle. But the original input is not specify (the modulation is set to 1 Hz). I think that this input signal must be of a particular form. But which ? For example, if i take a single cosinus (with 0 phase : A*cos(2*pi*f*t + 0) with the same frequency as the modulation frequence, the in-phase signal will be A and the quadrature-phase will be 0. So the scope displays just one motionless point (A, 0).
On 24 Aug 2003 07:28:47 -0700, germain1_fr@yahoo.fr (seb) wrote:

  (snipped)
>> >> Hi Jer, >> >> Yep. To give some numbers for seb (original poster); >> if the original analog bandpass signal had a bandwidth of >> roughly 10 kHz, we know that normal real (bandpass) >> sampling would require a sample rate just greater than >> 20K samples/sec if you used a single A/D. With the >> quad sampling scheme (also called "complex down-conversion"), >> two A/D converters would each be running at a rate just >> greater than 20K/2, or 10K samples/sec. >> >> In both case, 20K samples/sec would be generated. >> In both cases we're satisfying the Nyquist Theorem >> (named by the great Claude Shannon who died last >> year with Alzheimer's disease.) >> >> [-Rick-] > >Thinks, > >It is well understood (excuse my poor english, technical english is >easy to read but wrinting is more difficult). > >There is something else with this paper: >There is an example of application. When plug the in-phase and >quadrature-phase to the horizontaly and vertical input of a scope, we >observe a circle. But the original input is not specify (the >modulation is set to 1 Hz). I think that this input signal must be of >a particular form. But which ? > >For example, if i take a single cosinus (with 0 phase : A*cos(2*pi*f*t >+ 0) with the same frequency as the modulation frequence, the in-phase >signal will be A and the quadrature-phase will be 0. So the scope >displays just one motionless point (A, 0).
Hi, That oscillosope example was my effort to give physical meaning to the notion of a complex exponential signal, like e^j*2*pi*Fo*t, and physical meaning to the 'j' operator. Those two real analog signals applied to the oscilloscope were merely a real cosine wave and a real sinewave of the same frequency. That is, both real signals were sinusoids of the same frequency, but they are exactly 90-degrees (pi/2 radians) out of phase with each other. Signal 1: A*cos(2*pi*f*t) = A*sin(2*pi*f*t + pi/2) and Signal 2: A*sin(2*pi*f*t) There were *NO* "modulation" issues associated with that example. Hope that makes some sense. [-Rick-]