DSPRelated.com
Forums

How to get envelope from AM signal without phase shift

Started by WWalker March 7, 2010
WWalker wrote:
> Hi Hardy, > > Unfortunately, the LPF will phase shift the modulation. So this technique > will not work for me. Do you know of any other way to extract the > modulation without using a filter?
Why do you believe that? Think about what is being filtered. Jerry -- Discovery consists of seeing what everybody has seen, and thinking what nobody has thought. .. Albert Szent-Gyorgi �����������������������������������������������������������������������
In the system I am investigating, the phase speed and group speed are not
the same and are not constant and change with distance. Because of this,
the phase of the carrier is not the same as the phase of the modulation in
the signal. 

As I mentioned one way to get the modulation: without a phase shift,
without modulation distortion, and in between oscillations is to simply
divide the signal by the carrier which can be obtained by using a PLL.
Unfortunatly the technique is very sensitive to noise. But it does show
that it is in principle possible. The resultant modulation using the divide
technique is plagued with large random spikes. Do you know of any signal
processing methods to remove the spikes without distorting the signal or
phase shifting the modulation? I have tried using a running average, and
mean average but I always get a phase shift. Pehaps a Median filter could
be used but my guess is that it will distort the signal and phase shift
it.

Lastly, I should mention I have come up with another interesting method
which is to transmit a modulation signal through the dispersive medium with
a sinusoidal carrier (Q) and another modulation signal with a cosinusoidal
carrier (I). The instantaneous envelope can then be obtained by squaring
each, adding, and taking the square root. This method requires careful
alignment of the signals, but it does work and is a lot less sensitive to
noise.
 
William

>Tim Wescott wrote: >> WWalker wrote: >>> Hi, >>> >>> Does any one know how to extract the envelope of an amplitude
modulated
>>> signal without a phase shift, distortions, and able to determine the >>> envelope in between the signal cycles. One way that almost works is to >>> simply devide the signal by the carrier but, this technique is too >>> sensitive to phase noise. I have also tried using the Hilbert
transform
>>> but, I get some leakage distortions. >> >> Multiplying by the carrier is an accepted and worthwhile practice. There
>> are numerous useful extensions of this, many of which are to deal with >> the phase noise issue, and with selective fading that includes the >> carrier -- search on "exalted carrier" and "synchronous AM" to see the >> variations. > >I think W wants to explore the effects of a dispersive channel with >constant group delay in the band of interest. I don't think any kind of >demodulation is useful for that. > >Jerry >-- >Physics is like sex: sure, it may give some practical results, but >that's not why we do it. -- Richard P. Feynman (Nobel Prize, Physics) >������������������������������������������������������������������������ >
It looks like you don't know what you doing.
Matlab is not a substitute for knowledge.
 From here, you can either seek for professional help or study an ABC=20
book on DSP from cover to cover.

VLV


WWalker wrote:

> In the system I am investigating, the phase speed and group speed are n=
ot
> the same and are not constant and change with distance. Because of this=
,
> the phase of the carrier is not the same as the phase of the modulation=
in
> the signal.=20 >=20 > As I mentioned one way to get the modulation: without a phase shift, > without modulation distortion, and in between oscillations is to simply=
> divide the signal by the carrier which can be obtained by using a PLL. > Unfortunatly the technique is very sensitive to noise. But it does show=
> that it is in principle possible. The resultant modulation using the di=
vide
> technique is plagued with large random spikes. Do you know of any signa=
l
> processing methods to remove the spikes without distorting the signal o=
r
> phase shifting the modulation? I have tried using a running average, an=
d
> mean average but I always get a phase shift. Pehaps a Median filter cou=
ld
> be used but my guess is that it will distort the signal and phase shift=
> it. >=20 > Lastly, I should mention I have come up with another interesting method=
> which is to transmit a modulation signal through the dispersive medium =
with
> a sinusoidal carrier (Q) and another modulation signal with a cosinusoi=
dal
> carrier (I). The instantaneous envelope can then be obtained by squarin=
g
> each, adding, and taking the square root. This method requires careful > alignment of the signals, but it does work and is a lot less sensitive =
to
> noise. > =20 > William >=20 >=20 >>Tim Wescott wrote: >> >>>WWalker wrote: >>> >>>>Hi, >>>> >>>>Does any one know how to extract the envelope of an amplitude >=20 > modulated >=20 >>>>signal without a phase shift, distortions, and able to determine the >>>>envelope in between the signal cycles. One way that almost works is t=
o
>>>>simply devide the signal by the carrier but, this technique is too >>>>sensitive to phase noise. I have also tried using the Hilbert >=20 > transform >=20 >>>>but, I get some leakage distortions. >>> >>>Multiplying by the carrier is an accepted and worthwhile practice. The=
re
>=20 >=20 >>>are numerous useful extensions of this, many of which are to deal with=
=20
>>>the phase noise issue, and with selective fading that includes the=20 >>>carrier -- search on "exalted carrier" and "synchronous AM" to see the=
=20
>>>variations. >> >>I think W wants to explore the effects of a dispersive channel with=20 >>constant group delay in the band of interest. I don't think any kind of=
=20
>>demodulation is useful for that. >> >>Jerry >>--=20 >>Physics is like sex: sure, it may give some practical results, but >>that's not why we do it. -- Richard P. Feynman (Nobel Prize, Physics)=
>>=FF=CF=CD=FF=CF=CD=FF=CF=CD=FF=CF=CD=FF=CF=CD=FF=CF=CD=FF=CF=CD=FF=CF=CD=
=FF=CF=CD=FF=CF=CD=FF=CF=CD=FF=CF=CD=FF=CF=CD=FF=CF=CD=FF=CF=CD=FF=CF=CD=FF= =CF=CD=FF=CF=CD=FF=CF=CD=FF=CF=CD=FF=CF=CD=FF=CF=CD=FF=CF=CD=FF=CF=CD=FF=CF= =CD=FF=CF=CD=FF=CF=CD=FF=CF=CD=FF=CF=CD=FF=CF=CD=FF=CF=CD=FF=CF=CD=FF=CF=CD= =FF=CF=CD=FF=CF=CD=FF=CF=CD=FF=CF=CD=FF=CF=CD=FF=CF=CD=FF=CF=CD=FF=CF=CD=FF= =CF=CD=FF=CF=CD=FF=CF=CD=FF=CF=CD=FF=CF=CD=FF=CF=CD=FF=CF=CD=FF=CF=CD=FF=CF= =CD=FF=CF=CD=FF=CF=CD=FF=CF=CD=FF=CF=CD=FF=CF=CD=FF=CF=CD=FF=CF=CD=FF=CF=CD= =FF=CF=CD=FF=CF=CD=FF=CF=CD=FF=CF=CD=FF=CF=CD=FF=CF=CD=FF=CF=CD=FF=CF=CD=FF= =CF=CD=FF=CF=CD=FF=CF=CD=FF=CF=CD=FF=CF=CD=FF=CF=CD
>>
Hi Jerry, 

The low pass filter is used to filter out the higher harmonic terms
generated by the mixer. But unfortunately, the filter phase shifts the
wanted modulation. In my experiment I am transmitting a 50MHz Modulation
signl with a 500MHz Carrier. If I use a simple 3rd order filter
(1/(j(f/fc)+1)^3), with a 100MHz cutoff, the resultant modulation is phase
shifted about 90 degrees. But, the effect I am trying to measure is a 3
degree change in modulation.  

William



> >Use a PLL to get the carrier frequency and multiply and then low-pass >filter. Synchronous demodulation. >For supressed carrier you need to square the signal first then lock >onto 2f then divide by two and multiple - filter. >For low carrier to noise ratios you may need a different method. > >Hardy >
>WWalker wrote: >> Hi Hardy, >> >> Unfortunately, the LPF will phase shift the modulation. So this
technique
>> will not work for me. Do you know of any other way to extract the >> modulation without using a filter? > >Why do you believe that? Think about what is being filtered. > >Jerry >-- >Discovery consists of seeing what everybody has seen, and thinking what >nobody has thought. .. Albert Szent-Gyorgi >����������������������������������������������������������������������� >
"WWalker" <william.walker@n_o_s_p_a_m.imtek.de> writes:

> Hi Hardy, > > Unfortunately, the LPF will phase shift the modulation.
Not if you do it with a digital linear-phase filter. -- Randy Yates % "So now it's getting late, Digital Signal Labs % and those who hesitate mailto://yates@ieee.org % got no one..." http://www.digitalsignallabs.com % 'Waterfall', *Face The Music*, ELO
I am a professional. Say something intelligent and perhaps we can talk
about it. But being rude does not help.

William

> >It looks like you don't know what you doing. >Matlab is not a substitute for knowledge. > From here, you can either seek for professional help or study an ABC=20 >book on DSP from cover to cover. > >VLV

I am *the* professional.
I need to know the problem statement. I.e. what is the input, what 
should be the output, what is the accuracy and what hardware is 
available. Your problem will probably take 3-4 hours of work. The cost 
is going to be $1000. Is this OK ?

Vladimir Vassilevsky
DSP and Mixed Signal Design Consultant
http://www.abvolt.com



WWalker wrote:

> I am a professional. Say something intelligent and perhaps we can talk > about it. But being rude does not help. > > William > > >>It looks like you don't know what you doing. >>Matlab is not a substitute for knowledge. >>From here, you can either seek for professional help or study an ABC=20 >>book on DSP from cover to cover. >> >>VLV > > >
Hi Hardy,

A (FIR) linear phase filter will phase shift the modulation a small amount
without distorting the signal in the pass band. As I mentioned in a
previous post. I am trying to measure a 3 degree shift of a 50MHz
modulation, 500MHz carrier signal. 

But, I should mention, that the following technique does work. Fourier
Transform the signal. Replace the higher harmonics mixer terms with zeros,
and then inverse Fourier Transform back to the time domain.

William




William

>"WWalker" <william.walker@n_o_s_p_a_m.imtek.de> writes: > >> Hi Hardy, >> >> Unfortunately, the LPF will phase shift the modulation. > >Not if you do it with a digital linear-phase filter. >-- >Randy Yates % "So now it's getting late, >Digital Signal Labs % and those who hesitate >mailto://yates@ieee.org % got no one..." >http://www.digitalsignallabs.com % 'Waterfall', *Face The Music*, ELO >
On 21 Mar, 23:15, "WWalker" <william.walker@n_o_s_p_a_m.imtek.de>
wrote:
> In the system I am investigating, the phase speed and group speed are not > the same and are not constant and change with distance. Because of this, > the phase of the carrier is not the same as the phase of the modulation in > the signal.
If the phase and group velocities are different, the system is dispersive. If you have a dispersive system, you are in far worse trouble than a mere filter or AM demodulator, irrespective of phase responses, can handle. What are you doing? What do you want to achieve? Why do you think *you* are able to handle whatever it is you are up to? Rune
Vladimir, 

Thanks for the offer. I will think about it. In the meantime I would like
to know if there are solutions for this type of problem. From my experience
this is not an easy problem to solve and may require comming up with
something new as I have indicated in my posts. The problem is to measure a
3 degree change in the envelope of an AM Signal (50MHz modulation, 500MHz
Carrier) captured on a 1GHz digital scope. The envelope needs to be
extracted from the signal and compared to the envelope before the signal
propagated. I am trying to measure the group speed. 

William

> > >I am *the* professional. >I need to know the problem statement. I.e. what is the input, what >should be the output, what is the accuracy and what hardware is >available. Your problem will probably take 3-4 hours of work. The cost >is going to be $1000. Is this OK ? > >Vladimir Vassilevsky >DSP and Mixed Signal Design Consultant >http://www.abvolt.com > > > >WWalker wrote: > >> I am a professional. Say something intelligent and perhaps we can talk >> about it. But being rude does not help. >> >> William >> >> >>>It looks like you don't know what you doing. >>>Matlab is not a substitute for knowledge. >>>From here, you can either seek for professional help or study an ABC=20 >>>book on DSP from cover to cover. >>> >>>VLV >> >> >> >