patrick.melet@dmradiocom.fr wrote:
> hi
> 
> in your article : Efficient Multitone detection
> 
> you have a low pass filter wk(n) = wk(n-1) + uk(n)/16 - wk(n-1)/16
> which have a 10 Hz cutoff at 8 kHz sampling

No. It has 10Hz cutoff at Fs = 1kHz .

> how to compute the new filter for a 300 kHz sampling frequency and 30
> kHz sampling ?

You are the Ph.D., aren't you?
Here is the Ph.D. stuff:

y += (x-y)*A

A = 1 - exp(-2*Pi*Fs/Fsa);



Vladimir Vassilevsky

DSP and Mixed Signal Design Consultant

http://www.abvolt.com

hi

in your article : Efficient Multitone detection

you have a low pass filter wk(n) = wk(n-1) + uk(n)/16 - wk(n-1)/16
which have a 10 Hz cutoff at 8 kHz sampling

how to compute the new filter for a 300 kHz sampling frequency and 30
kHz sampling ?

Tim Wescott wrote:

>>> I made a digital PLL, with multiplier based phase detector, digital
>>> loop filter C1 + C2/(1-z^-1) and a digital NCO with 16 bits
>>> accumulator
>>>
>>> My problem is when I don't have the 22170 Hz signal, my output filter
>>> sweep very slowly

[...]

>>
>> 1. You should use the state machine frequency/phase detector, not a 
>> multiplier. This will guarantee the reliable quick PLL pull-in.
>>
> Unless you're trying to lock to a signal that's embedded in noise, in 
> which case a state machine detector will be worse than useless.
> 
> If you sample the data first, then implement your PLL, a pure-logic 
> state machine frequency/phase detector will have an uncertainty around 
> the edges of +/- 1/2 sample, which quite broad for many purposes.  I 
> rarely encounter an application that can suffer this much noise being 
> injected by the algorithm itself.
> Assuming that the tone is strong enough you can get the same effect by 
> estimating the incoming signal's phase and counting rotations -- this 
> works very well in practice, with much noisier signals than you can hope 
> to use a frequency/phase detector on, and doesn't have the edge 
> uncertainty.  It does have the drawback that it twists your brain in 
> loops, but once you get it figured out you'll be smarter than you were 
> going in.

Agreed completely. The state machine does not have to be like a dual 
flip-flop thing from the textbook. The point is to make a frequency 
sensitive phase detector, and that can be accomplished by different 
ways. Counting the phase rollovers cost me a couple of days to make it 
work, but that was also the phase modulated signal :)

BTW, implementing a PLL as PID rather then PI also makes it frequency 
sensitive.


> 
>> 2. The pull-in time depends on the loop filter parameters. There is a 
>> tradeoff between the PLL noise bandwidth and the pull-in time.
>>
> Yea verily.  If you can detect lock you can switch (or slide) to a 
> narrower bandwidth once the phase & frequency are acquired.

There is also some shamanistic theories for using the nonlinear filters 
for that purpose.

>> 3. The RMS variation of the NCO frequency can be used as the measure 
>> of PLL lock.
> 
> 
> Assuming a pure enough tone -- here again, in a noisy environment it can 
> be rough to distinguish noise-induced RMS variation of the error signal 
> from an unlocked loop.  If you're doing something extreme like digging a 
> tone that's buried several dB deep into the noise then you can have a 
> perfectly adequate lock yet have a very difficult time pulling it out -- 
> in this case you'd have to low-pass filter the NCO frequency variations, 
> and look at the RMS variation of the resulting signal.  It could be 
> done, but you'd have to approach the problem of determining when you 
> have a good lock very carefully lest you simply build a false-positive 
> generator.

The detecting of a PLL lock requires some sort of measurement of the RMS 
noise in the loop. I don't know if there can be the other way for doing 
that.



Vladimir Vassilevsky

DSP and Mixed Signal Design Consultant

http://www.abvolt.com

patrick.melet@dmradiocom.fr wrote:

> Hi all,
> 
> I made a digital PLL, with multiplier based phase detector, digital
> loop filter C1 + C2/(1-z^-1) and a digital NCO with 16 bits
> accumulator
> 
Sounds fine so far, assuming your accumulator will give you the 
frequency precision you need.

> I want to lock on a 22170 Hz sine signal in 10 ms lock time
> 
> My problem is when I don't have the 22170 Hz signal, my output filter
> sweep very slowly (1 second) beetween min and max of my input NCO and
> don't lock when my 22170 Hz signal is on

The fact that it sweeps in one direction indicates that you have a 
roundoff problem in your multiplier.  If you're doing the multiplication 
then letting the processor naturally truncate the data it'll be biased 
-- this will cause the drift.
> 
> So I put a window at the ouput of my filter to only sweep in a 2600 Hz
> window
> And when this ouput is out of this window I reset the filter
> accumulator to its maximum value, now the ouput of the filter sweep
> this window very slowly ( about 2 seconds)
> 
> So with this when I put my 22170 Hz off and on my loop locks into it,
> but I have a very long lock time ( 200 ms) and not 10 ms...
> 
> Is in acquisition mode I have to "manually" sweep my input NCO in the
> 2600 Hz window and look at the ouput of the loop filter to say if I'm
> lock or not ?
> 
How well do you know your tone frequency ahead of time?  Is the tone 
guaranteed to be there all the time?  What's the signal to noise ratio 
of your tone?  Is the noise colored?

If you know your tone frequency quite well ahead of time, and you know 
for sure that the tone will be there when you start to acquire, you 
should start up with the integrator value set to your incoming tone 
frequency.

If the tone is guaranteed to be there all the time and and it's SNR is 
high, then you can schedule the gains of your PLL to get a fast, 
reliable acquire followed by a nice low-bandwidth track phase. 
Theoretically you'll do this by going through all the math to generate 
an extended Kalman filter; in the real world you can often get by with 
tuning each phase experimentally to do well enough while optimizing the 
filter for design time.

If the noise is white then you can worry less.  If it may contain 
interfering tones close to the target tone then you can worry more.

If the tone frequency is uncertain, if the tone goes on and off, if it's 
buried in noise, and if there are interfering tones -- you have a long 
row to hoe, and you may want to hit the books or hire a consultant.

Assuming that you have the easy case of a known-frequency, strong, 
continuous tone then you shouldn't be having this trouble.  I'd suspect 
that you have some implementation problem, rather than an algorithmic one.

-- 

Tim Wescott
Wescott Design Services
http://www.wescottdesign.com

Posting from Google?  See http://cfaj.freeshell.org/google/

"Applied Control Theory for Embedded Systems" came out in April.
See details at http://www.wescottdesign.com/actfes/actfes.html

Vladimir Vassilevsky wrote:

> 
> 
> patrick.melet@dmradiocom.fr wrote:
> 
> 
>> I made a digital PLL, with multiplier based phase detector, digital
>> loop filter C1 + C2/(1-z^-1) and a digital NCO with 16 bits
>> accumulator
>>
>> I want to lock on a 22170 Hz sine signal in 10 ms lock time
>>
>> My problem is when I don't have the 22170 Hz signal, my output filter
>> sweep very slowly (1 second) beetween min and max of my input NCO and
>> don't lock when my 22170 Hz signal is on
>>
>> So I put a window at the ouput of my filter to only sweep in a 2600 Hz
>> window
>> And when this ouput is out of this window I reset the filter
>> accumulator to its maximum value, now the ouput of the filter sweep
>> this window very slowly ( about 2 seconds)
>>
>> So with this when I put my 22170 Hz off and on my loop locks into it,
>> but I have a very long lock time ( 200 ms) and not 10 ms...
>>
>> Is in acquisition mode I have to "manually" sweep my input NCO in the
>> 2600 Hz window and look at the ouput of the loop filter to say if I'm
>> lock or not ?
> 
> 
> 1. You should use the state machine frequency/phase detector, not a 
> multiplier. This will guarantee the reliable quick PLL pull-in.
> 
Unless you're trying to lock to a signal that's embedded in noise, in 
which case a state machine detector will be worse than useless.

If you sample the data first, then implement your PLL, a pure-logic 
state machine frequency/phase detector will have an uncertainty around 
the edges of +/- 1/2 sample, which quite broad for many purposes.  I 
rarely encounter an application that can suffer this much noise being 
injected by the algorithm itself.

Assuming that the tone is strong enough you can get the same effect by 
estimating the incoming signal's phase and counting rotations -- this 
works very well in practice, with much noisier signals than you can hope 
to use a frequency/phase detector on, and doesn't have the edge 
uncertainty.  It does have the drawback that it twists your brain in 
loops, but once you get it figured out you'll be smarter than you were 
going in.

> 2. The pull-in time depends on the loop filter parameters. There is a 
> tradeoff between the PLL noise bandwidth and the pull-in time.
> 
Yea verily.  If you can detect lock you can switch (or slide) to a 
narrower bandwidth once the phase & frequency are acquired.

> 3. The RMS variation of the NCO frequency can be used as the measure of 
> PLL lock.

Assuming a pure enough tone -- here again, in a noisy environment it can 
be rough to distinguish noise-induced RMS variation of the error signal 
from an unlocked loop.  If you're doing something extreme like digging a 
tone that's buried several dB deep into the noise then you can have a 
perfectly adequate lock yet have a very difficult time pulling it out -- 
in this case you'd have to low-pass filter the NCO frequency variations, 
and look at the RMS variation of the resulting signal.  It could be 
done, but you'd have to approach the problem of determining when you 
have a good lock very carefully lest you simply build a false-positive 
generator.

-- 

Tim Wescott
Wescott Design Services
http://www.wescottdesign.com

Posting from Google?  See http://cfaj.freeshell.org/google/

"Applied Control Theory for Embedded Systems" came out in April.
See details at http://www.wescottdesign.com/actfes/actfes.html

patrick.melet@dmradiocom.fr wrote:

> I have read your article : Efficient Multitone detection...
>=20
> It's a fully good article with lots of tips and tricks...
>=20
> Monday I'm going to test it in our application where we need to detect
> 6 tones from 1kHz to 75 kHz
>=20
> The difference with the DTMF and our application is that we can have
> the 6 tones presents WITH a phase modulation (180=B0) at very low
> frequency 67Hz or 100Hz present in our 6 tones

Since the output is I and Q for every frequency, the addition of the=20
phase demodulators should not be a problem at all.

Still I am wondering what is the application which requires that strange =

form of signaling.

> thanks and good luck for the future, you seems to be young (PhD in
> 1997), I have too a PhD in Microelectronics form France in 2001...

No wonder. Just about anyone in this newsgroup is a geek of the advanced =

degree. Some people are trying to conceal, but one can easily notice the =

specific mental damage made by the higher education.

Vladimir Vassilevsky

DSP and Mixed Signal Design Consultant

http://www.abvolt.com

patrick.melet@dmradiocom.fr wrote:

> I have a loop bandwith of 1500 Hz

Is it a PLL loop bandwidth or just the LPF bandwidth? Make sure you 
don't confuse one for another.


> The problem is my acquisition process...
> 
> there's a long time constant due to my loop filter that accumulates
> the values

Looks like not enough of loop gain. PLL is the same thing as the classic 
PI regulator. Refer to any book on the control theory.

Vladimir Vassilevsky

DSP and Mixed Signal Design Consultant

http://www.abvolt.com

I have a frequency sampling of 300 kHz
Then after de multipler (phase detector) I decimate with a FIR filter
to be 10 kHz sampling to the loop filter
so the loop filter is at 10 kHz sampling
because I have a strange output of this filter when I compute the
coefficients for a sampling frequency of 300 kHz
when i sampling the filter at 300 kHz I have a 44340 Hz at the output
of the mixer (detector) that is not filtered by the loop filter

I think it should work with 300 kHz sampling

I have Kd = 16384, Ko = 1.4e-9
with Fs=10kHz I have C1 = 14335 and C2=3041
and with Fs=300 kHz I have C1=586 and C2=4.14

I think ther's a problem here

On Mar 16, 7:54 am, patrick.me...@dmradiocom.fr wrote:
> I have a loop bandwith of 1500 Hz
>
> The problem is my acquisition process...
>
> there's a long time constant due to my loop filter that accumulates
> the values
>
> If I make a sweep on my input NCO, the output of the loop filter don't
> converge
>
> I don't know how to measure the rms of the variation of the frequency
> of the nco

It sounds like something isn't right in the implementation. If the
signal is dead center and SNR is decent, acquisition should occur much
faster than 200 ms. You should not have to sweep unless the input
frequency is too far off. How did you compute the loop filter gains?

John

I have a loop bandwith of 1500 Hz

The problem is my acquisition process...

there's a long time constant due to my loop filter that accumulates
the values

If I make a sweep on my input NCO, the output of the loop filter don't
converge

I don't know how to measure the rms of the variation of the frequency
of the nco