Are purely-analog audio devices immune to aliasing?
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Green Xenon [Radium]
Is it true that purely-analog audio devices [such as analog cassette, AM
radio, and the pre-digital telephone systems*] are immune to aliasing?
*By pre-digital telephone systems, I am referring to how these systems
operated prior to using digital technology. Nowadays, many analog phone
systems do use DSP somewhere along the line. Similar applies to the
analog AM radio, it used to be just analog but now it utilizes some
amount of DSP indirectly.
Thanks,
Radium
geoff
Um, just because something is digital (ie PCM telephone transmission) does
NOT imply that there is any DSP involved.
Are you troubled by aliasing on your telephone ?
geoff
Richard Crowley
> Are you troubled by aliasing on your telephone ?
He seems to be troubled by aliasing with his own name.
He can't even decide which alias to use, so he uses both.
Ron Capik
No, they are not "immune" (as if that were
some disease) to aliasing.
It would seem you still have much to
learn about communications theory
and the associated math.
Later...
Ron Capik
--
Randy Yates
> Hi:
>
> Is it true that purely-analog audio devices [such as analog cassette,
> AM radio, and the pre-digital telephone systems*] are immune to
> aliasing?
Yes.
There is a way to see this very clearly, but it involves some
calculus. If you're prepared for that, go to a library and find the book
"Signals and Systems" and read the section on sampling.
If you're not prepared for that, and you really want to understand this
topic, go take a couple (or three) semesters of calculus at a junior
college or university and then tackle it.
--Randy
PS: Radium, you remind me of a man who says he's thirsty, and yet is
unwilling to walk to a well 1/2 mile away to get a drink. Many of the
questions you ask are fundamental if you'd be willing to expend a little
effort and get some education.
@BOOK{signalsandsystems,
title = "{Signals and Systems}",
author = "{Alan~V.~Oppenheim, Alan~S.~Willsky, with Ian~T.~Young}",
publisher = "Prentice Hall",
year = "1983"}
--
% Randy Yates % "My Shangri-la has gone away, fading like
%% Fuquay-Varina, NC % the Beatles on 'Hey Jude'"
%%% 919-577-9882 %
%%%% <ya...@ieee.org> % 'Shangri-La', *A New World Record*, ELO
http://www.digitalsignallabs.com
Earl Kiosterud
"Green Xenon [Radium]" <gluc...@excite.com> wrote in message
news:4828deb3$0$5698$4c36...@roadrunner.com...
The pre-digital telephone system used frequency-division multiplexing, each voice channel in
a 4 KHz slot. Nyquist was 4 KHz. Without pre-filtering, they'd have the very aliasing
that digital systems can have. Without post-filtering, you'd hear sidebands around 8 KHz.
No difference, except the sampling frequency.
--
Earl
Chronic Philharmonic
"Randy Yates" <ya...@ieee.org> wrote in message
news:m3k5hzj...@ieee.org...
> "Green Xenon [Radium]" <gluc...@excite.com> writes:
>
>> Hi:
>>
>> Is it true that purely-analog audio devices [such as analog cassette,
>> AM radio, and the pre-digital telephone systems*] are immune to
>> aliasing?
>
> Yes.
Well... AM radio is not immune to aliasing, although we rarely encounter it
in practice (although FM stereo encoders require anti-aliasing filters*).
Pre-digital phones are probably immune to aliasing unless they were doing
analog multiplexing on long distance trunk lines. Analog cassettes might be
immune to aliasing per se, but if the recording frequency gets high enough
(ultrasonic), it will produce audible beats with the bias frequency.
*mathematically, FM stereo is double sideband suppressed carrier, which is
mathematically equivalent to alternately sampling the left and right
channels at 38 KHz. So although it is not quantized (digital), it is
sampled. Many analog systems fall into this category.
Randy Yates
> "Randy Yates" <ya...@ieee.org> wrote in message
> news:m3k5hzj...@ieee.org...
>> "Green Xenon [Radium]" <gluc...@excite.com> writes:
>>
>>> Hi:
>>>
>>> Is it true that purely-analog audio devices [such as analog cassette,
>>> AM radio, and the pre-digital telephone systems*] are immune to
>>> aliasing?
>>
>> Yes.
>
> Well... AM radio is not immune to aliasing, although we rarely encounter it
> in practice
How would AM ever alias?
--
% Randy Yates % "Midnight, on the water...
%% Fuquay-Varina, NC % I saw... the ocean's daughter."
%%% 919-577-9882 % 'Can't Get It Out Of My Head'
%%%% <ya...@ieee.org> % *El Dorado*, Electric Light Orchestra
http://www.digitalsignallabs.com
Green Xenon [Radium]
> Without pre-filtering, they'd have the very aliasing
> that digital systems can have.
But won't an analog device just smoothly cut-off a frequency that is too
high -- i.e. at a certain point the cut-off gradually beings and the
higher an incoming frequency is, the more it will be attenuated --
without any aliasing?
geoff
> "Chronic Philharmonic" <karl.u...@verizon.net> writes:
>
>> "Randy Yates" <ya...@ieee.org> wrote in message
>> news:m3k5hzj...@ieee.org...
>>> "Green Xenon [Radium]" <gluc...@excite.com> writes:
>>>
>>>> Hi:
>>>>
>>>> Is it true that purely-analog audio devices [such as analog
>>>> cassette, AM radio, and the pre-digital telephone systems*] are
>>>> immune to aliasing?
>>>
>>> Yes.
>>
>> Well... AM radio is not immune to aliasing, although we rarely
>> encounter it in practice
>
> How would AM ever alias?
USB / LSB ?
geoff
Don Pearce
No that isn't aliasing, it is adjacent channel interference - a totally
different thing. Aliasing is a product of sampling, an ambiguation of
the signal caused by the fact that it is sampled at discrete points and
whatever is in between those points must be filled in with assumptions
about its nature. In audio, the assumption is generally that the signal
between those points contains the lowest possible frequency solution
below the Nyquist frequency. But the other solutions, involving higher
frequencies are equally valid from a mathematical point of view - they
are the alias solutions.
d
Green Xenon [Radium]
> the cut-off gradually beings
Sorry that should read "the cut-off gradually *begins*"
Don Pearce
d
Earl Kiosterud
> Green Xenon [Radium] wrote:
>
>
>> the cut-off gradually beings
>
>
> Sorry that should read "the cut-off gradually *begins*"
Radium,
An alias is a modulation product that we don't want -- one that appears in our passband.
Any modulation system produces various modulation products -- we just have to structure it
so that they don't wind up where we don't want them. The fact that we're doing
discrete-amplitude sampling (16-bit or whatever), has nothing to do with it -- it's the fact
that we're doing it in discrete time intervals (a form of amplitude modulation, based on a
carrier, like 44.1 KHz.).
The following is to illustrate the potential for aliases in an analog system, not to depict
how carrier systems were actually implemented.
Imagine it's the 60s, and you're building a telephone carrier system where you want to
translate a voice channel, 0-4KHz, to the 4-8KHz channel. So you modulate a 4 Khz sine-wave
carrier with your baseband audio. This produces sum and difference freqencies. The sum
frequencies fall right in the channel you want, 4-8KHz. For example, a 1 Khz baseband
(original audio) component produces 5 Khz in your channel, exactly as you want. 2 KHz
produces 6 KHz, etc. The difference frequencies fall in the 4-zero KHz range, looking like
upside down audio. For example, 1 Khz of baseband produces 3 Khz. That's OK, because
you're going to filter the after-modulation signal (post-filter) for only 4-8KHz. If you
don't, you'll have adjacent-channel interference, as Don points out in his reply.
Similarly, the sum-frequency products of baseband signals above 4 KHz would wind up above 8
KHz, messing up other channels (i.e.: 5 KHz would wind up at 9 KHz, messing up the 8-12KHz
channel). More adjacent-channel interference.
Now imagine a 9KHz baseband component. It would wind up at 5 KHz -- smack in the middle of
your channel, but at a different frequency. These things don't sound pretty. Your
post-filtering would not remove it, because it's in your channel. That's an alias. So you
must filter your baseband (before modulation, or pre-filtering) below 8 KHz to prevent this
alias from being generated. In reality, you'd probably have pre-filtered your baseband from
0-4KHz anyway.
My point is that there was no sampling, just pure simple sine-carrier analog amplitude
modulation, and we still have the potential for aliases. The theory is the same, whether
it's digital (discrete) or plain old garden-variety analog amplitude modulation.
Also, you can see that there's no inherent cutoff in such analog systems, as you asked
about. You have to provide it to prevent aliases and modulation artifacts from winding up
in your audio.
Now you didn't ask about this, but since I'm this far, I'll take this to the digital world
of CD audio. If the baseband were not pre-filtered below Nyquist (22.05 KHz) before
sampling, then e.g.: a 24 KHz audio component would appear as 19.9 Khz. (44.1 KHz - 24 Khz).
That'd also be an alias -- a simple difference product. The higher the baseband is allowed
to creep above Nyquist, the more alias junk we hear creeping downwards into the audio band,
mirrored around the Nyquist frequency. We must filter the baseband to below Nyquist to
prevent aliases, and similarly post filter the sampled signal to prevent those sum and
difference signals around 44.1 KHz from appearing with our recovered audio. We wouldn't
hear it, but it'd give our amplifiers and speakers stuff to deal with unnecessarily.
--
Earl
Earl Kiosterud
"Randy Yates" <ya...@ieee.org> wrote in message news:m31w46s...@ieee.org...
An AM radio station operating at 1 MHz would have to have the baseband limited to 500 KHz,
or aliases would appear. For example a 600 KHz baseband component would appear at 400 KHz.
This obviously ain't gonna happen in any real world situation, but imagine an AM station
operating at 10 KHz. It would have to keep the audio limited to below it's Nyquist
frequency of 5 Khz.
I'm using the term Nyquist frequency somewhat incorrectly, as it technically applies to
sampled systems. But it still works, and we know what we mean! :)
--
Earl
Ron Capik
> "Chronic Philharmonic" <karl.u...@verizon.net> writes:
>
> > "Randy Yates" <ya...@ieee.org> wrote in message
> > news:m3k5hzj...@ieee.org...
> >> "Green Xenon [Radium]" <gluc...@excite.com> writes:
> >>
> >>> Hi:
> >>>
> >>> Is it true that purely-analog audio devices [such as analog cassette,
> >>> AM radio, and the pre-digital telephone systems*] are immune to
> >>> aliasing?
> >>
> >> Yes.
> >
> > Well... AM radio is not immune to aliasing, although we rarely encounter it
> > in practice
>
> How would AM ever alias?
> --
> % Randy Yates % "Midnight, on the water...
AM is a sampling technique. If the signal bandwidth exceeds the Nyquist rate
there will be aliasing.
Later...
Ron Capik
--
Don Pearce
AM isn't sampling. It is a continuous process. And of course there is no
aliasing because the output of an amplitude modulator contains only the
carrier and its unique sidebands - there is no baseband, and no train of
other alias frequencies.
d
Earl Kiosterud
"Earl Kiosterud" <som...@nowhere.com> wrote in message news:ipiWj.342$lj.202@trnddc01...
It occurs to me that I'm going to catch some flak for using the term "alias" in non-sampled
systems. I think the more correct terms are "image" and "mirror." But "alias" is used
interchangeably, it seems. Bottom line: you have the same problems. It doesn't sound as
ignorant as "RMS Power," but the idea is similar. It ain't right, but we know what we
mean.
--
Earl
Randy Yates
Hi Earl,
I see what you mean now. I'd say that comes as close to an "analog
alias" as you'll get. It is mathematically almost the same thing. (I say
almost since the analog system involves only two Dirac delta functions
while a digital one involves an infinite number.)
But I really don't think this is what was in Radium's mind when he
asked.
--
% Randy Yates % "...the answer lies within your soul
%% Fuquay-Varina, NC % 'cause no one knows which side
%%% 919-577-9882 % the coin will fall."
%%%% <ya...@ieee.org> % 'Big Wheels', *Out of the Blue*, ELO
http://www.digitalsignallabs.com
dpierce.ca...@gmail.com
> But I really don't think this is what was in Radium's
> mind when he asked.
Other than the internet equivalent of an annoying,
bratty, kicking, screaming child trying to grab the
steering wheel from his car seat, it's impossible
to know what is in his, uhm 'mind.'
And, to add, it's just not important, either.
Earl Kiosterud
"Randy Yates" <ya...@ieee.org> wrote in message news:m3hcd2p...@ieee.org...
Hi Randy,
Oops. I spoke too soon. It occurred to me while I was working in the yard. A 10KHz AM
station could modulate to 10 KHz. As Don pointed out in another thread, there's no baseband
in the modulator output, so there's no concern about the difference component (sideband)
overlapping with it, as there is with sampling. But if you try to modulate above 10K, the
difference product ends up negative, and folds into the band. 12KHz modulation would
produce a component at 2 KHz. I don't know of a demodulator that could properly sort that
out. I've always considered that an alias, but it's not the same as in a sampling system,
such as when the difference component creeps downward past Nyquist as the baseband creeps
upwards past it.
I realize I've probably been misusing the term "alias." When in Rome. :)
--
Earl
Green Xenon [Radium]
Do analog, non-electronic musical instruments ever alias?
Green Xenon [Radium]
Green Xenon [Radium]
Well, if analog entities are not immune to aliasing...
... then if some extraordinary ultrasonic transducer attempts to
generate a 50 MHz pure sine-wave tone in Earth's troposphere ...
... will the air molecules generate aliased waveforms because those
molecules cannot vibrate faster than 30 MHz??
Note: 30 MHz is the fastest the air molecules of Earth's troposphere can
oscillate.
geoff
> Note: 30 MHz is the fastest the air molecules of Earth's troposphere
> can oscillate.
Really ?!! Well I won't even bother with a super-tweeter then ;-(
geoff
Green Xenon [Radium]
Please answer my question:
Earl Kiosterud
Suppose we set up an amplitude-modulation system using a multiplier, except that we reverse
the carrier (44.1 KHz) and the baseband audio. We offset the 44.1 KHz signal (so it is
always positive) but not the baseband -- it's allowed to operate in two quadrants of our
multiplier. You could say that the carrier is modulating the baseband, instead of the usual
other way around. The resulting signal would include the baseband components, and a set of
AM sidebands around 44.1KHz, images of the audio components, just as with CD audio (except
there wouldn't be sidebands around 88.2 KHz, 132.3 KHz., etc). If we allow our audio to go
past 22.05 Khz, we'll have spurious stuff in our baseband frequency range. For example, an
audio component at 30 KHz would produce a component at 14.1 Khz.
Isn't that the same spurious component (same frequency) we'd get with aliasing in the case
of CD audio? This system is continuous (the audio isn't sampled -- it's allowed to change
continuously), yet we have a sort of a Nyquist frequency under which our baseband must stay
in order to not get signal components in our output that are ambiguous.
--
Earl
"Don Pearce" <nos...@nospam.com> wrote in message
news:9K2dnS2KKdBNgLTVnZ2dnUVZ8rCdnZ2d@plusnet...
Don Pearce
<som...@nowhere.com> wrote:
>Hi Don,
>
>Suppose we set up an amplitude-modulation system using a multiplier, except that we reverse
>the carrier (44.1 KHz) and the baseband audio. We offset the 44.1 KHz signal (so it is
>always positive) but not the baseband -- it's allowed to operate in two quadrants of our
>multiplier. You could say that the carrier is modulating the baseband, instead of the usual
>other way around. The resulting signal would include the baseband components, and a set of
>AM sidebands around 44.1KHz, images of the audio components, just as with CD audio (except
>there wouldn't be sidebands around 88.2 KHz, 132.3 KHz., etc). If we allow our audio to go
>past 22.05 Khz, we'll have spurious stuff in our baseband frequency range. For example, an
>audio component at 30 KHz would produce a component at 14.1 Khz.
>
>Isn't that the same spurious component (same frequency) we'd get with aliasing in the case
>of CD audio? This system is continuous (the audio isn't sampled -- it's allowed to change
>continuously), yet we have a sort of a Nyquist frequency under which our baseband must stay
>in order to not get signal components in our output that are ambiguous.
>--
>Earl
Let me think about this! I'll do the maths later. I think you are
right that you will get stuff all over the place that you don't want,
but I think I would tend to call them images rather than aliases.
I also suspect that the use of a synchronous demodulator might let you
recover the signals (as you can recover modulation over 100% this
way), meaning that the signals aren't truly jumbled - they just appear
that way. Aliasing is truly there for ever once it happens - there is
no way back.
OK, let me change my mind. I've just used Mathcad to look at this. I
made two FFT spectra - one with 44.1k as the carrier and 3k as the
modulation and the other with 3k as the carrier and 44.1k as the
modulation.
For the first I see a carrier at 44.1kHz, and sidebands at 41.1 and
47.1kHz. Exactly as you expect.
For the second I see a carrier at 3kHz and sidebands at 44.1 and
47.1kHz. Which is exactly as you expect once you know what to expect
;-)
Which will do for me. It would even be a nice way to generate
suppressed carrier AM - just filter away the 3kHz when done
modulating.
d
--
Pearce Consulting
http://www.pearce.uk.com
Earl Kiosterud
"Earl Kiosterud" <som...@nowhere.com> wrote in message news:KDmWj.52$vE.31@trnddc03...
Hey Randy,
Oops again. I said that I didn't know of a demodulator that could sort out the 2 KHz
component that was the result of 12K audio. A plain old envelope (diode) detector would be
confounded, and the transmitter would have to be constrained to 10K audio. But my
favorite, the AM synchronous demodulator, could easily do that, since it could be set up to
use only the upper set of sidebands. In that case the transmitter could modulate up to 20K,
above which point the difference-frequency sidebands would creep into our upper set of
sidebands and muck up the works.
I don't know what Radium's intentions are, but he sometimes sparks discussions that get us
thinking and we end up sorting things out.
Regards from Virginia Beach,
--
Earl
Earl Kiosterud
"Don Pearce" <nos...@nospam.com> wrote in message news:482e5bea...@news.plus.net...
(snip)
>
> Let me think about this! I'll do the maths later. I think you are
> right that you will get stuff all over the place that you don't want,
> but I think I would tend to call them images rather than aliases.
>
> I also suspect that the use of a synchronous demodulator might let you
> recover the signals (as you can recover modulation over 100% this
> way), meaning that the signals aren't truly jumbled - they just appear
> that way. Aliasing is truly there for ever once it happens - there is
> no way back.
>
> OK, let me change my mind. I've just used Mathcad to look at this. I
> made two FFT spectra - one with 44.1k as the carrier and 3k as the
> modulation and the other with 3k as the carrier and 44.1k as the
> modulation.
>
> For the first I see a carrier at 44.1kHz, and sidebands at 41.1 and
> 47.1kHz. Exactly as you expect.
>
> For the second I see a carrier at 3kHz and sidebands at 44.1 and
> 47.1kHz. Which is exactly as you expect once you know what to expect
> ;-)
>
> Which will do for me. It would even be a nice way to generate
> suppressed carrier AM - just filter away the 3kHz when done
> modulating.
>
> d
>
>
> --
> Pearce Consulting
> http://www.pearce.uk.com
Hey Don,
Here's how I see this. In your first case, the 41.1K and 47.1 K components are sidebands of
the 44.1K carrier. In the second case those same components are the (folded) sidebands (or
negative-frequency, which is perfectly valid)) of a 3 KHz carrier. The result is the
same -- they're the same sum and difference frequencies -- they don't really care who's
modulating whom! :) A sum frequency is a sum, and a difference is a difference. The only
real difference is if you add 0 Hz (DC offset) to either the audio or to the carrier. In
the case of the former, you get the 44.1K carrier component as well as the sidebands around
it, and in the latter you get baseband audio as well as the same sidebands around 44.1K.
And that's the result of the sum and difference frequencies too. Ain't no getting away from
it!
You mentioned generating suppressed-carrier generation, by filtering out the 3K component.
I think you can just multiply the signal with the baseband, adding 0 Hz (DC) to neither, and
get that. You have to use a four-quadrant multiplier (handles both negative and positive
signals). You should get no carrier and no baseband in the output -- just the sidebands.
ATSC TV adds a little DC to get a little bit of carrier (if I understand it correctly, and
I'm not at all sure I do) for recovery. At least that's one way of implementing it.
Forgive my excess of clarification comments in all this -- I don't mean to be pedantic, but
I try to be as clear as possible, and I think they might be useful to some of those who
might be following this thread.
The terms "mirror" and "image," I think, at least as used in superhet radio, refer to such
things as a modulation product that is the sum frequency, where only the difference was
wanted. It's a mirror around the local oscillator frequency. But in a more general sense,
any modulation product that winds up in your frequency band of interest (audio, in our case)
is an ambiguous component, in that we can't distinguish it from a real signal component.
I've always thought of any as an alias.
Here's my main thrust: I think the aliases we get in sampled audio where the audio goes
above Nyquist are simply the difference modulation products, which creep into our baseband
if the audio goes above Nyquist. As the audio gets higher in frequency, the sidebands get
lower, from 44.1K. Nyquist is simply the midpoint, where they meet. Sampling is a case of
general modulation theory.
Hoping to hear your comments.
--
Regards from Virginia Beach,
Earl
Don Pearce
Yes - that is how I see it.
>You mentioned generating suppressed-carrier generation, by filtering out the 3K component.
>I think you can just multiply the signal with the baseband, adding 0 Hz (DC) to neither, and
>get that. You have to use a four-quadrant multiplier (handles both negative and positive
>signals). You should get no carrier and no baseband in the output -- just the sidebands.
>ATSC TV adds a little DC to get a little bit of carrier (if I understand it correctly, and
>I'm not at all sure I do) for recovery. At least that's one way of implementing it.
>
Right - I've done this with a double balanced mixer and a bit of
adjustable DC bias to null the carrier completely (more or less).
>Forgive my excess of clarification comments in all this -- I don't mean to be pedantic, but
>I try to be as clear as possible, and I think they might be useful to some of those who
>might be following this thread.
>
No, that's fine. This can all get very complicated unless you have
some simple way of visualising it.
>The terms "mirror" and "image," I think, at least as used in superhet radio, refer to such
>things as a modulation product that is the sum frequency, where only the difference was
>wanted. It's a mirror around the local oscillator frequency. But in a more general sense,
>any modulation product that winds up in your frequency band of interest (audio, in our case)
>is an ambiguous component, in that we can't distinguish it from a real signal component.
>I've always thought of any as an alias.
>
An alias is a specific kind of image with a specific set of
properties. It can only occur as a result of discrete-time sampling.
All the stuff that comes of a normal modulator is an image - or an
intermod product if we have any non-linearity.
>Here's my main thrust: I think the aliases we get in sampled audio where the audio goes
>above Nyquist are simply the difference modulation products, which creep into our baseband
>if the audio goes above Nyquist. As the audio gets higher in frequency, the sidebands get
>lower, from 44.1K. Nyquist is simply the midpoint, where they meet. Sampling is a case of
>general modulation theory.
>
>Hoping to hear your comments.
Well, since sampling is a form of modulation, I have to agree. But as
I say it is a form of modulation product that can only exist in a
sampled system, so I think it is fair to treat it as a special case.
Also it is occurring in a situation we would not usually think of as
modulation (even though it is). Anywhere two signals are multiplied
(an ADC multiplies the audio by the sampling pulse) there will be
modulation products.
geoff
> Let me think about this! I'll do the maths later. I think you are
> right that you will get stuff all over the place that you don't want,
> but I think I would tend to call them images rather than aliases.
That was the analogy I was drawing with my "LSB/USB" post. That the sum and
difference frequencies are in som way analogous to the frequency products
from aliasing.
geoff
Don Pearce
Yes they are analogous in that a sampling device will also produce them.
The difference is that the sampler also produces a comb of them up to
infinity. You could say that the analogue image products are the limit
case of aliasing when the dead space approaches zero.
d
Earl Kiosterud
"Don Pearce" <nos...@nospam.com> wrote in message news:482caf60...@news.plus.net...
(snip)
Don
The reason I posed my little AM modulator, using a multiplier with some 0 Hz added to the
44.1K carrier was to show that it's exactly the same in this continuous (non-sampled) system
as the aliasing in CD audio. The result is exactly the same in either system, and my point
is that it's for the same reasons. It's all about modulation products. I don't understand
why you say it can happen only in a sampling system. Could you elaborate?
Do you agree that the output of the DAC will contain the baseband components, and a set of
sidebands around 44.1K patterned like the baseband? (Also around 88.2K, etc.)
--
Regards from soon-to-be-raining -- again -- Virginia Beach,
Earl
dpierce.ca...@gmail.com
dpierce.ca...@gmail.com
The reason is very simple. IN the case of AM modulation,
the carrier is a sine wave, which has but a single component,
the fundamental of the carrier.
When "sampling," in the context you're using it, the
"carrier is actually a train of narrow pulses. Look at
it's specturm: it's a series of components evenly
space out to infinity (assuming 0 rise and fall time),
so you have the images spread out to infinity as
well.
Remember that what we are talking about "sampling"
is often also referred to as "pulse code modulation,"
or PCM.
It's exactly the same (save that the generated "sidebands
are reallu "images," not "aliases"
The reason you have aliases with "sampling (or modulating
a pulse train) is because of all those other components:
something far out of band will also generate an infinite
number of images, which can fall back or "alias" into
your basebnand and thus be indistiguishable from a
true base-band signal.
Can it happen in an all-anolg system? Sure, just come up
with a carrier that's a sufficiently comples waveform.
Condiser, for example, your carrier having TWO components:
a sine at 44.1 kHz and one at 88.2 kHz. Now, modulate that
with a signal at. oh 88.2+-43 kHz, Where do the sidebands
end up?!
Chronic Philharmonic
"Randy Yates" <ya...@ieee.org> wrote in message
news:m31w46s...@ieee.org...
> "Chronic Philharmonic" <karl.u...@verizon.net> writes:
>
>> "Randy Yates" <ya...@ieee.org> wrote in message
>> news:m3k5hzj...@ieee.org...
>>> "Green Xenon [Radium]" <gluc...@excite.com> writes:
>>>
>>>> Hi:
>>>>
>>>> Is it true that purely-analog audio devices [such as analog cassette,
>>>> AM radio, and the pre-digital telephone systems*] are immune to
>>>> aliasing?
>>>
>>> Yes.
>>
>> Well... AM radio is not immune to aliasing, although we rarely encounter
>> it
>> in practice
>
> How would AM ever alias?
Sorry, I was thinking of multiplexing using AM, where the sidebands would
overlap adjacent channels if the modulating frequency is not band limited to
1/2 the carrier spacing (as it happens on the AM Broadcast band at night). A
single isolated AM transmitter would not alias into its own baseband.
Chronic Philharmonic
"Green Xenon [Radium]" <gluc...@excite.com> wrote in message
news:482a0c74$0$31741$4c36...@roadrunner.com...
> Note: 30 MHz is the fastest the air molecules of Earth's troposphere can
> oscillate.
They go to 31.53 MHz at my house.
Don Pearce
Yes, I can see this is a matter of how you view things.
An ADC is like an AM modulator (OK it is an AM modulator) and it
produces image sidebands in exactly the way any of them does. The big
difference with most modulation systems is that in an ADC, the carrier
(44.1kHz) isn't symmetric - it is entirely one sided. For that reason
baseband comes through it and in this case it becomes our wanted signal.
Of course this will happen whether the modulator is sampled, as in an
ADC or continuous. The rest of the alias productsaround 88.2kHz etc can
only happen in a sampling system, and are true aliases.
d
Don Pearce
That's another way of looking at it.
d