Frequency accuracy in older RXs
 

"rickman" wrote in message Yes, it is good to start
with a signal that has less harmonic content.
Can your function generator put out a sine wave? I bet that has pretty
low harmonic content... ;)


Yes my function generator can put out square, triangle and sine waves. I
was just using the square waves to refresh my memory on what hapens when
they go through a low pass filter. I have not played with that in many
years.



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Frequency accuracy in older RXs
 

rickman wrote in :

What circuit clips a
tone into a square wave just so it could be run through a low pass filter?


Sound synthesis pitch following tone generator. Ramp generator based on
natural pitch input (to get triangle wave). Preconditioner for faster capture
in 4046 PLL to use the input that would otherwise be too sensitive to errors
caused by irregular pulse width ratios... I suspect there are several cases
for this.

Frequency accuracy in older RXs
 

rickman wrote in :

On 10/7/2014 2:07 PM, gareth wrote:
"Ralph Mowery" wrote in message
...
I thought the triangle wave had even harmonics in it, but found out it
actually has odd harmonics but they decrease with the square of the order
instead of just a simple 1/N.

It is the sawtooth (equal slopes at rise and decay) that is made up of
even harmonics.

You seem to have them backwards...

http://en.wikipedia.org/wiki/Triangle_wave

http://en.wikipedia.org/wiki/Sawtooth_wave


(I cannot speak with any authority in the triangle wave)

Or much authority in the sawtooth wave as it turns out... ;)


Well, he's not far wrong as it happens... I'm new to RF and radio but have
spent a lot more time with audio synthesisers. A triangle is a special case
of a sawtooth (properly, ramp) wave. Like the square, it has no even
harmonics, but the moment you change the speeds of the two parts of the ramp
in one cycle, even harmonics occur with increasing strength the more it gets
skewed. The really awkward bit with sawtooth waves is that they are a VERY
general case, and include ramps but also nonlinear slopes. This is something
I don't get into with maths, but it has strong implications for timbres
especially when emulating natural instruments like brass or strings or even
old synths that used relaxation oscillators. In radio techniques I imagine
the clipping and filtering (or other means) to precondition a rough signal
for the XOR phase comparator input of a PLL, taking advantage of its high
noise immunity even with the raw clipped signal (it still requires a 50%
width ratio), though a bit of filtering after clipping can help there.

Frequency accuracy in older RXs
 

rickman wrote in :

You say the filter removes "all" of the harmonics... that is obviously
not correct. The filter may reduce them, but it does not and can not
completely remove them. The nearest tones (which are also the largest
amplitude tones) will only be reduced a small amount really. Or maybe
you are planning to use a brick wall filter?


One thing I tried briefly, before getting into computers are doing things
with code still a bit shy of getting into digital filtering just yet, was a
switched capacitor filter, the attenuation there is excellent, and you can
easily filter out the clock signal if it's 100 times the pass frequency.

My question intended to ask *WHY* would anyone design a circuit to
produce a square wave and then spend the time and trouble to filter it?


I had a go at answerign that in my other posts just now.. Basically,
conditioning a rough wave by Schmitt trigger into somethign much easier to
process later, often for PLL input. Controlling hysteresis (using positive
feedback) can be especially useful to elimnate some problems with complex
waves that cross the centre more than twice per cycle.

Frequency accuracy in older RXs
 

rickman wrote in :

You posed a problem; I gave you an answer. BTW there cannot be an
easier item to lash to a microcontroller than a DDS

Thank you for your suggestion.


Assuming that DDS is Direct Digital Synthesis, I'm not sure anything needs to
be lashed to anything. :) Just use a phase accumulator in a single DWORD or
whatever native data size gives adequate resolution. Then just feed a DAC,
assuming the processor has one. I think very many of them do.

Frequency accuracy in older RXs
 

Michael Black wrote in
news:alpine.LNX.2.02.1410071314370.14980@darkstar. example.org:

If the input to the mixer isnt' well filtered, then I think you're going
to get all kinds of weird responses, since something can mix with a
harmonic of the oscillator and be converted down to the IF.

The best wheeze I came up with (never built it though), is a tracking filter:
a Schmitt trigger with a bit of positive feedback hysteresis to condition the
signal for a PLL and divide-by-100 based on two decade counters, to create a
stable clock signal for a 100X switched capacitor filter that takes a copy of
the original, direct signal. The output would be damn close to a sine wave no
matter what the input was like, but there is a clock signal that may need
removing, but possibly a simple one-pole filter will do that well enough.

Frequency accuracy in older RXs
 

On 09/10/2014 20:21, Lostgallifreyan wrote:
rickman wrote in :

You posed a problem; I gave you an answer. BTW there cannot be an
easier item to lash to a microcontroller than a DDS

Thank you for your suggestion.


Assuming that DDS is Direct Digital Synthesis, I'm not sure anything needs to
be lashed to anything. :) Just use a phase accumulator in a single DWORD or
whatever native data size gives adequate resolution. Then just feed a DAC,
assuming the processor has one. I think very many of them do.


Fine, but first find a microcontroller with such a fast (minimum 60MHz)
DAC. Even if you do you'll spend many times the cost of a simple
microcontroller plus DDS chip system - and you won't rival the cheaper
system's performance until your DAC can work at several times 60MHz.

PA

Frequency accuracy in older RXs
 

Peter Able stuck@home wrote in news:ks2dncC5fJrIAqrJnZ2dnUVZ7s-
:

Fine, but first find a microcontroller with such a fast (minimum 60MHz)
DAC. Even if you do you'll spend many times the cost of a simple
microcontroller plus DDS chip system - and you won't rival the cheaper
system's performance until your DAC can work at several times 60MHz.


Good point. Almost certainly easier to just send step size from CPU to the
accumulator in fast hardware. The step size needs changing far less often
than it needs stepping.

Frequency accuracy in older RXs
 

Lostgallifreyan wrote:
Peter Able stuck@home wrote in news:ks2dncC5fJrIAqrJnZ2dnUVZ7s-
:

Fine, but first find a microcontroller with such a fast (minimum 60MHz)
DAC. Even if you do you'll spend many times the cost of a simple
microcontroller plus DDS chip system - and you won't rival the cheaper
system's performance until your DAC can work at several times 60MHz.


Good point. Almost certainly easier to just send step size from CPU to the
accumulator in fast hardware. The step size needs changing far less often
than it needs stepping.

That is what the external DDS chip is doing.

When you require only FM-modulated signals and no sinewave, existing
microcontrollers can do it using their onboard timers and programmable
clock dividers. For example, the Raspberry Pi has been turned into
FM-broadcast and amateur radio shortwave FSK/ASK transmitter, the RF
signal appears (as a square wave) directly on a GPIO pin. Just filter
and amplify, or when you don't ca just connect a random wire as an
antenna to transmit a couple of mW (and harmonics).

Frequency accuracy in older RXs
 

Rob wrote in
:

Lostgallifreyan wrote:
Peter Able stuck@home wrote in news:ks2dncC5fJrIAqrJnZ2dnUVZ7s-
:

Fine, but first find a microcontroller with such a fast (minimum
60MHz) DAC. Even if you do you'll spend many times the cost of a
simple microcontroller plus DDS chip system - and you won't rival the
cheaper system's performance until your DAC can work at several times
60MHz.


Good point. Almost certainly easier to just send step size from CPU to
the accumulator in fast hardware. The step size needs changing far less
often than it needs stepping.

That is what the external DDS chip is doing.

When you require only FM-modulated signals and no sinewave, existing
microcontrollers can do it using their onboard timers and programmable
clock dividers. For example, the Raspberry Pi has been turned into
FM-broadcast and amateur radio shortwave FSK/ASK transmitter, the RF
signal appears (as a square wave) directly on a GPIO pin. Just filter
and amplify, or when you don't ca just connect a random wire as an
antenna to transmit a couple of mW (and harmonics).


An off-topic question, but very interesting at least to me... Do DDS chips
exist with 128 or even 256 phase accumulators onboard with the step size
adjustment being capable of matching the speed of the stepping itself
(though taking external control of local phase modulations between
accumulators), and allowing mixing of all outputs, perhaps in user-selected
groups based on binary fractions of the total accumulator count?

I ask because if they do it might be possible for me to convert my phase mod
synth code to dedicated hardware without resorting to very fast CPU's...
While the rates are audio only, the huge parallel array gets demanding of CPU
time as it is.

I suspect the answer to all that may be 'no' without custom VLSI chips
because of the relatively complex paths between accumulators needed for a
phase mod synth of N operators per algorithm, but maybe DDS chips come in
enough varieties to surprise me. :)