BC
Bob Camp
Sun, Feb 7, 2010 1:52 AM
Hi
Any approach that includes building a low noise synthesizer is opening up a whole new set of issues. I would much prefer to do my building at audio. Audio parts are cheap, and performance is usually a lot easier to check than at RF.
Bob
On Feb 6, 2010, at 8:30 PM, Bruce Griffiths wrote:
Which just leaves the minor problem of the offset oscillator.
One option is to use a phase truncation spur free output frequency from a DDS.
If one is using the Costas receiver approach the beat frequency need not be a nice round number like 1.0000KHz.
Another method is to use a crystal whose frequency is offset a few kHz from 10MHz.
Yet another is the classical method of dividing 10MHz by 100 and subtracting (using an LSB mixer) the resultant 100KHz from 10MHz to produce 9.9MHz, then divide the 9.9MHz signal by 100 and add (using a USB mixer) the resultant 99kHz signal to the 9.99Mhz signal to produce a 9.999MHz output.
Bruce
John Miles wrote:
A sound-card back end has always seemed like a pretty reasonable approach to
me, if you're inclined to go the DMTD route. I wouldn't send a 'baseband'
signal to the sound card, though -- I'd upconvert it to a few kHz to get
away from the numerous bad things that sound cards do near DC.
-- john, KE5FX
Hi
My main concern with the low frequency pole in the sound card is
the quality of the R/C used. You can certainly model what ever
you have. If they used an aluminum electrolytic for the "C" it
may not be the same next time you check it ....
On a 10 Hz system, a 1 Hz pole is probably not an issue. It might
get in the way with a 1 Hz beat note.
Another thing I have only seen in passing: "Sigma Delta's have
poor low frequency noise characteristics". I haven't dug into it
to see if that's really true or not. If you buy your own ADC's,
you certainly would not be restricted to a Sigma Delta.
Even with a cheap pre-built FPGA board, you could look into
higher sample rates than a conventional sound card. You would
drop back to 16 bits, but it might be worth it.
Bob
On Feb 6, 2010, at 6:46 PM, Bruce Griffiths wrote:
Even better is to toss out the mixers and sample the RF signals
However suitable ADCs cost $US100 or more each.
To which one has to add an FPGA and an interface to a PC with
sufficient throughput to handle the down converted I + Q samples.
Hi
You probably could put a couple of cheap DAC's
(ADCs are preferable as it avoids having to implement the
conversion logic plus comparator required when using a DAC.)
on a board with a FPGA and reduce the data on the fly. I'd
guess that would be be in the same $100 range as a half way
decent sound card. Clock the DAC's off of a 10 MHz reference and
eliminate the cal issue.
If you are down around 10 Hz or worse yet 1 Hz, the AC
coupling of the sound card will get in the way, even with a
bandpass approach. You really don't know what they may have in
there at the low end. Build it yourself and that stuff's not an issue.
My sound card has a 1Hz cutoff RC high pass input filter plus
an internal high pass digital filter.
Its not too difficult to measure the sound card frequency
response using a white noise source for example.
On Feb 6, 2010, at 6:12 PM, Bruce Griffiths wrote:
If one has a high end sound card then it could be used to
implement the bandpass filter and replace the zero crossing detector.
It may be necessary to insert a pilot tone to calibrate the
sound card sampling clock frequency.
A noise floor of about 1E-13/Tau should be achievable.
This simplifies the DMTD system by replacing the zero
crossing detector with a low gain linear preamp.
If one analyses the resultant data off line then one can also
try out different techniques such as a Costas receiver rather
than a simple bandpass filter plus zero crossing detector.
However 1000 seconds of data for 2 channels of 24 bit samples
at 192KSPS will result in a file with a size of at least 1.15GB.
Bruce
Bruce Griffiths wrote:
If one were to use a bandpass filter with a Q of 10 to
filter the beat frequency output of the mixer, then if the input
frequency is 10MHz and the filter component tempco is 100ppm/C
then the resultant phase shift tempco is about 16ps/C referred to
the mixer input frequency.
This phase shift tempco is certainly low enough not to have
significant impact when measuring the frequency stability of a
typical 10811A if the temperature fluctuations are kept small
enough during the run.
The effect of using a bandpass filter with too narrow a
bandwidth is to artificially reduce ADEV for small Tau, so it may
be prudent to use a higher beat frequency that 1Hz or even 10Hz
and not calculate ADEV for Tau less than say 10(??) times the
beat frequency period. A trade off between this and the effect of
aliasing is required.
Hi
With most 10811 range oscillators the impact of a simple
bandpass filter is low enough to not be a major issue. That's for
normal lab temperatures with the circuitry in a conventional die
cast box. No guarantee if you open the window and let the fresh
air blow in during the run.
That's true with a heterodyne. I can see no obvious reason
it would not be true on DMTD.
Bob
On Feb 6, 2010, at 5:12 PM, Bruce Griffiths wrote:
The only major issue with DMTD systems is that they
undersample the phase fluctuations and hence are subject to
aliasing effects.
The low pass filter has to have a bandwidth of the same
order as the beat frequency or the beat frequency signal will be
significantly attenuated.
Since the phase is only sampled once per beat frequency
period the phase fluctuations are undersampled.
Various attempts to use both zero crossings have not been
In principle if one can overcome the increased phase shift
tempco associated with a bandpass filter, using a bandpass filter
can in principle ensure that the phase fluctuations are oversampled.
Hi
A straight heterodyne system will get you to the floor of
most 10811's with a very simple (2 stage) limiter. As with the
DMTD, the counter requirements aren't really all that severe.
Bob
On Feb 6, 2010, at 4:24 PM, WarrenS wrote:
"It's possible / likely for injection lock ... to be a
Something I certainly worried about and tested for.
What I found (for MY case) is that injection lock is NOT
The reason being is that unlike most other ways, where
the two OSC have to be completely independent,
The tight loop approach forces the Two Osc to "Lock with
something like 60 + db gain,
so a little stray -80db injection lock coupling that
would very much limit other systems has
no measurable effect at e-13. Just one of the neat
little side effects that make the tight loop approach so simple.
"then a part in 10^14 is going to be at the 100 of
For that example, just need to put a simple discrete 100
in-between the control voltage and the EFC and now you
have a nice workable 10uv.
BUT the bigger point is, probable not needed, cause you
are NOT going to do any better than the stability of the OSC with
a grounded shorted EFC input.
as you said and I agree is so true:
"There is no perfect way to do any of this, only a lot
of compromises ... you need to watch out for".
But you did not offer any easier way to do it, which is
what the original request was for and my answer addressed.
This is the cheapest easiest way BY FAR to get high
performance, at low tau, ADEV numbers that I've seen.
ws
----- Original Message ----- From: "Bob Camp"lists@cq.nu
To: "Discussion of precise time and frequency
Sent: Saturday, February 06, 2010 12:09 PM
Subject: Re: [time-nuts] ADEV vs MDEV
Hi
It's possible / likely to injection lock with the tight
loop approach and get data that's much better than reality. A lot
depends on the specific oscillators under test and the buffers
(if any) between the oscillators and mixer.
If your OCVCXO has a tuning slope of 0.1 ppm / volt
then a part in 10^14 is going to be at the 100 of nanovolts
level. Certainly not impossible, but it does present it's own set
of issues. Lab gear to do it is available, but not all that
common. DC offsets and their temperature coefficients along with
thermocouple effects could make things exciting.
There is no perfect way to do any of this, only a lot
of compromises here or there. Each approach has stuff you need to
watch out for.
Bob
From: "WarrenS"warrensjmail-one@yahoo.com
Sent: Saturday, February 06, 2010 2:19 PM
To: "Discussion of precise time and frequency
Subject: Re: [time-nuts] ADEV vs MDEV
I would appreciate any comments or observations on
the topic of apparatus with demonstrated stability measurements.
My motivation is to discover the SIMPLEST scheme for
making stability measurements at the 1E-13 in 1s performance level.
If you accept that the measurement is going to limited
for Low COST and SIMPLE, with the ability to measure
Can't beat a simple analog version of NIST's "Tight
Phase-Lock Loop Method of measuring Freq stability".
counter& Printer with a Radio shack type PC data logging DVM,
It can be up and running from scratch in under an Hr,
with no high end test equipment needed.
If you want performance that exceeds the best of most
DMTD at low Tau it takes a little more work
and a higher speed oversampling ADC data logger and a
I must add this is not a popular solution (Or a
IF you know analog and have a GOOD osc with EFC to
as far as I've been able to determine it is the BEST
SIMPLE answer that allows High performance.
Limited by My HP10811 Ref OSC, I'm getting better than
1e-12 in 0.1 sec (at 30 Hz Bandwidth)
Basic modified NIST Block Diag attached:
The NIST paper sums it up quite nicely:
'It is not difficult to achieve a sensitivity of a
part in e14 per Hz resolution
so one has excellent precision capabilities with this system.'
This does not address your other question of ADEV vs MDEV,
What I've described is just a simple way to get the
What you then do with that Data is a different subject.
You can run the raw data thru one of the many ADEV
programs out there, 'Plotter' being my choice.
Have fun
ws
[time-nuts] ADEV vs MDEV
Pete Rawson peterawson at earthlink.net
Sat Feb 6 03:59:18 UTC 2010
Efforts are underway to develop a low cost DMTD apparatus with
demonstrated stability measurements of 1E-13 in 1s. It
existing TI counters can reach this goal in 10s.
or 100+s. using ADEV estimate). The question is; does
provide an appropriate measure of stability in this
the ADEV estimate a more correct answer?
The TI performance I'm referring to is the 20-25 ps,
typical for theHP5370A/B, the SR620 or the CNT81/91. I
from my CNT81showing MDEV< 1E-13 in 10s. and I believe the
other counters behave similarly.
I would appreciate any comments or observations on this topic.
My motivation is to discover the simplest scheme for making
stability measurements at this performance level; this is NOT
even close to the state-of-the-art, but can still be useful.
Pete Rawson
Hi
Any approach that includes building a low noise synthesizer is opening up a whole new set of issues. I would much prefer to do my building at audio. Audio parts are cheap, and performance is usually a lot easier to check than at RF.
Bob
On Feb 6, 2010, at 8:30 PM, Bruce Griffiths wrote:
> Which just leaves the minor problem of the offset oscillator.
>
> One option is to use a phase truncation spur free output frequency from a DDS.
> If one is using the Costas receiver approach the beat frequency need not be a nice round number like 1.0000KHz.
>
> Another method is to use a crystal whose frequency is offset a few kHz from 10MHz.
>
> Yet another is the classical method of dividing 10MHz by 100 and subtracting (using an LSB mixer) the resultant 100KHz from 10MHz to produce 9.9MHz, then divide the 9.9MHz signal by 100 and add (using a USB mixer) the resultant 99kHz signal to the 9.99Mhz signal to produce a 9.999MHz output.
>
> Bruce
>
> John Miles wrote:
>> A sound-card back end has always seemed like a pretty reasonable approach to
>> me, if you're inclined to go the DMTD route. I wouldn't send a 'baseband'
>> signal to the sound card, though -- I'd upconvert it to a few kHz to get
>> away from the numerous bad things that sound cards do near DC.
>>
>> -- john, KE5FX
>>
>>
>>
>>> Hi
>>>
>>> My main concern with the low frequency pole in the sound card is
>>> the quality of the R/C used. You can certainly model what ever
>>> you have. If they used an aluminum electrolytic for the "C" it
>>> may not be the same next time you check it ....
>>>
>>> On a 10 Hz system, a 1 Hz pole is probably not an issue. It might
>>> get in the way with a 1 Hz beat note.
>>>
>>> Another thing I have only seen in passing: "Sigma Delta's have
>>> poor low frequency noise characteristics". I haven't dug into it
>>> to see if that's really true or not. If you buy your own ADC's,
>>> you certainly would not be restricted to a Sigma Delta.
>>>
>>> Even with a cheap pre-built FPGA board, you could look into
>>> higher sample rates than a conventional sound card. You would
>>> drop back to 16 bits, but it might be worth it.
>>>
>>> Bob
>>>
>>>
>>> On Feb 6, 2010, at 6:46 PM, Bruce Griffiths wrote:
>>>
>>>
>>>> Even better is to toss out the mixers and sample the RF signals
>>>>
>>> directly.
>>>
>>>> However suitable ADCs cost $US100 or more each.
>>>> To which one has to add an FPGA and an interface to a PC with
>>>>
>>> sufficient throughput to handle the down converted I + Q samples.
>>>
>>>> Bob Camp wrote:
>>>>
>>>>> Hi
>>>>>
>>>>> You probably could put a couple of cheap DAC's
>>>>>
>>>> (ADCs are preferable as it avoids having to implement the
>>>>
>>> conversion logic plus comparator required when using a DAC.)
>>>
>>>>
>>>>> on a board with a FPGA and reduce the data on the fly. I'd
>>>>>
>>> guess that would be be in the same $100 range as a half way
>>> decent sound card. Clock the DAC's off of a 10 MHz reference and
>>> eliminate the cal issue.
>>>
>>>>> If you are down around 10 Hz or worse yet 1 Hz, the AC
>>>>>
>>> coupling of the sound card will get in the way, even with a
>>> bandpass approach. You really don't know what they may have in
>>> there at the low end. Build it yourself and that stuff's not an issue.
>>>
>>>>> Bob
>>>>>
>>>>>
>>>>>
>>>> My sound card has a 1Hz cutoff RC high pass input filter plus
>>>>
>>> an internal high pass digital filter.
>>>
>>>> Its not too difficult to measure the sound card frequency
>>>>
>>> response using a white noise source for example.
>>>
>>>> Bruce
>>>>
>>>>> On Feb 6, 2010, at 6:12 PM, Bruce Griffiths wrote:
>>>>>
>>>>>
>>>>>
>>>>>> If one has a high end sound card then it could be used to
>>>>>>
>>> implement the bandpass filter and replace the zero crossing detector.
>>>
>>>>>> It may be necessary to insert a pilot tone to calibrate the
>>>>>>
>>> sound card sampling clock frequency.
>>>
>>>>>> A noise floor of about 1E-13/Tau should be achievable.
>>>>>> This simplifies the DMTD system by replacing the zero
>>>>>>
>>> crossing detector with a low gain linear preamp.
>>>
>>>>>> If one analyses the resultant data off line then one can also
>>>>>>
>>> try out different techniques such as a Costas receiver rather
>>> than a simple bandpass filter plus zero crossing detector.
>>>
>>>>>> However 1000 seconds of data for 2 channels of 24 bit samples
>>>>>>
>>> at 192KSPS will result in a file with a size of at least 1.15GB.
>>>
>>>>>> Bruce
>>>>>>
>>>>>>
>>>>>> Bruce Griffiths wrote:
>>>>>>
>>>>>>
>>>>>>> If one were to use a bandpass filter with a Q of 10 to
>>>>>>>
>>> filter the beat frequency output of the mixer, then if the input
>>> frequency is 10MHz and the filter component tempco is 100ppm/C
>>> then the resultant phase shift tempco is about 16ps/C referred to
>>> the mixer input frequency.
>>>
>>>>>>> This phase shift tempco is certainly low enough not to have
>>>>>>>
>>> significant impact when measuring the frequency stability of a
>>> typical 10811A if the temperature fluctuations are kept small
>>> enough during the run.
>>>
>>>>>>> The effect of using a bandpass filter with too narrow a
>>>>>>>
>>> bandwidth is to artificially reduce ADEV for small Tau, so it may
>>> be prudent to use a higher beat frequency that 1Hz or even 10Hz
>>> and not calculate ADEV for Tau less than say 10(??) times the
>>> beat frequency period. A trade off between this and the effect of
>>> aliasing is required.
>>>
>>>>>>> Bruce
>>>>>>>
>>>>>>> Bob Camp wrote:
>>>>>>>
>>>>>>>
>>>>>>>> Hi
>>>>>>>>
>>>>>>>> With most 10811 range oscillators the impact of a simple
>>>>>>>>
>>> bandpass filter is low enough to not be a major issue. That's for
>>> normal lab temperatures with the circuitry in a conventional die
>>> cast box. No guarantee if you open the window and let the fresh
>>> air blow in during the run.
>>>
>>>>>>>> That's true with a heterodyne. I can see no obvious reason
>>>>>>>>
>>> it would not be true on DMTD.
>>>
>>>>>>>> Bob
>>>>>>>>
>>>>>>>>
>>>>>>>> On Feb 6, 2010, at 5:12 PM, Bruce Griffiths wrote:
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>> The only major issue with DMTD systems is that they
>>>>>>>>>
>>> undersample the phase fluctuations and hence are subject to
>>> aliasing effects.
>>>
>>>>>>>>> The low pass filter has to have a bandwidth of the same
>>>>>>>>>
>>> order as the beat frequency or the beat frequency signal will be
>>> significantly attenuated.
>>>
>>>>>>>>> Since the phase is only sampled once per beat frequency
>>>>>>>>>
>>> period the phase fluctuations are undersampled.
>>>
>>>>>>>>> Various attempts to use both zero crossings have not been
>>>>>>>>>
>>> successful.
>>>
>>>>>>>>> In principle if one can overcome the increased phase shift
>>>>>>>>>
>>> tempco associated with a bandpass filter, using a bandpass filter
>>> can in principle ensure that the phase fluctuations are oversampled.
>>>
>>>>>>>>>
>>>>>>>>> Bruce
>>>>>>>>>
>>>>>>>>> Bob Camp wrote:
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>> Hi
>>>>>>>>>>
>>>>>>>>>> A straight heterodyne system will get you to the floor of
>>>>>>>>>>
>>> most 10811's with a very simple (2 stage) limiter. As with the
>>> DMTD, the counter requirements aren't really all that severe.
>>>
>>>>>>>>>> Bob
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> On Feb 6, 2010, at 4:24 PM, WarrenS wrote:
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>> "It's possible / likely for injection lock ... to be a
>>>>>>>>>>>>
>>> problem ..."
>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>> Something I certainly worried about and tested for.
>>>>>>>>>>> What I found (for MY case) is that injection lock is NOT
>>>>>>>>>>>
>>> a problem.
>>>
>>>>>>>>>>> The reason being is that unlike most other ways, where
>>>>>>>>>>>
>>> the two OSC have to be completely independent,
>>>
>>>>>>>>>>> The tight loop approach forces the Two Osc to "Lock with
>>>>>>>>>>>
>>> something like 60 + db gain,
>>>
>>>>>>>>>>> so a little stray -80db injection lock coupling that
>>>>>>>>>>>
>>> would very much limit other systems has
>>>
>>>>>>>>>>> no measurable effect at e-13. Just one of the neat
>>>>>>>>>>>
>>> little side effects that make the tight loop approach so simple.
>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>> "then a part in 10^14 is going to be at the 100 of
>>>>>>>>>>>>
>>> nanovolts level."
>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>> For that example, just need to put a simple discrete 100
>>>>>>>>>>>
>>> to 1 resistor divider
>>>
>>>>>>>>>>> in-between the control voltage and the EFC and now you
>>>>>>>>>>>
>>> have a nice workable 10uv.
>>>
>>>>>>>>>>> BUT the bigger point is, probable not needed, cause you
>>>>>>>>>>>
>>> are NOT going to do any better than the stability of the OSC with
>>> a grounded shorted EFC input.
>>>
>>>>>>>>>>> as you said and I agree is so true:
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>> "There is no perfect way to do any of this, only a lot
>>>>>>>>>>>>
>>> of compromises ... you need to watch out for".
>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>> But you did not offer any easier way to do it, which is
>>>>>>>>>>>
>>> what the original request was for and my answer addressed.
>>>
>>>>>>>>>>> This is the cheapest easiest way BY FAR to get high
>>>>>>>>>>>
>>> performance, at low tau, ADEV numbers that I've seen.
>>>
>>>>>>>>>>> ws
>>>>>>>>>>> ***************
>>>>>>>>>>>
>>>>>>>>>>> ----- Original Message ----- From: "Bob Camp"<lists@cq.nu>
>>>>>>>>>>> To: "Discussion of precise time and frequency
>>>>>>>>>>>
>>> measurement"<time-nuts@febo.com>
>>>
>>>>>>>>>>> Sent: Saturday, February 06, 2010 12:09 PM
>>>>>>>>>>> Subject: Re: [time-nuts] ADEV vs MDEV
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>> Hi
>>>>>>>>>>>>
>>>>>>>>>>>> It's possible / likely to injection lock with the tight
>>>>>>>>>>>>
>>> loop approach and get data that's much better than reality. A lot
>>> depends on the specific oscillators under test and the buffers
>>> (if any) between the oscillators and mixer.
>>>
>>>>>>>>>>>> If your OCVCXO has a tuning slope of 0.1 ppm / volt
>>>>>>>>>>>>
>>> then a part in 10^14 is going to be at the 100 of nanovolts
>>> level. Certainly not impossible, but it does present it's own set
>>> of issues. Lab gear to do it is available, but not all that
>>> common. DC offsets and their temperature coefficients along with
>>> thermocouple effects could make things exciting.
>>>
>>>>>>>>>>>> There is no perfect way to do any of this, only a lot
>>>>>>>>>>>>
>>> of compromises here or there. Each approach has stuff you need to
>>> watch out for.
>>>
>>>>>>>>>>>> Bob
>>>>>>>>>>>>
>>>>>>>>>>>> --------------------------------------------------
>>>>>>>>>>>> From: "WarrenS"<warrensjmail-one@yahoo.com>
>>>>>>>>>>>> Sent: Saturday, February 06, 2010 2:19 PM
>>>>>>>>>>>> To: "Discussion of precise time and frequency
>>>>>>>>>>>>
>>> measurement"<time-nuts@febo.com>
>>>
>>>>>>>>>>>> Subject: Re: [time-nuts] ADEV vs MDEV
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>> Peat said:
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>> I would appreciate any comments or observations on
>>>>>>>>>>>>>>
>>> the topic of apparatus with demonstrated stability measurements.
>>>
>>>>>>>>>>>>>> My motivation is to discover the SIMPLEST scheme for
>>>>>>>>>>>>>>
>>> making stability measurements at the 1E-13 in 1s performance level.
>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>> If you accept that the measurement is going to limited
>>>>>>>>>>>>>
>>> by the Reference Osc,
>>>
>>>>>>>>>>>>> for Low COST and SIMPLE, with the ability to measure
>>>>>>>>>>>>>
>>> ADEVs at that level,
>>>
>>>>>>>>>>>>> Can't beat a simple analog version of NIST's "Tight
>>>>>>>>>>>>>
>>> Phase-Lock Loop Method of measuring Freq stability".
>>>
>>>>>>>>>>>>> http://tf.nist.gov/phase/Properties/one.htm#oneone Fig 1.7
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> By replacing the "Voltage to freq converter, Freq
>>>>>>>>>>>>>
>>> counter& Printer with a Radio shack type PC data logging DVM,
>>>
>>>>>>>>>>>>> It can be up and running from scratch in under an Hr,
>>>>>>>>>>>>>
>>> with no high end test equipment needed.
>>>
>>>>>>>>>>>>> If you want performance that exceeds the best of most
>>>>>>>>>>>>>
>>> DMTD at low Tau it takes a little more work
>>>
>>>>>>>>>>>>> and a higher speed oversampling ADC data logger and a
>>>>>>>>>>>>>
>>> good offset voltage.
>>>
>>>>>>>>>>>>> I must add this is not a popular solution (Or a
>>>>>>>>>>>>>
>>> general Purpose one) but
>>>
>>>>>>>>>>>>> IF you know analog and have a GOOD osc with EFC to
>>>>>>>>>>>>>
>>> use for the reference,
>>>
>>>>>>>>>>>>> as far as I've been able to determine it is the BEST
>>>>>>>>>>>>>
>>> SIMPLE answer that allows High performance.
>>>
>>>>>>>>>>>>> Limited by My HP10811 Ref OSC, I'm getting better than
>>>>>>>>>>>>>
>>> 1e-12 in 0.1 sec (at 30 Hz Bandwidth)
>>>
>>>>>>>>>>>>> Basic modified NIST Block Diag attached:
>>>>>>>>>>>>> The NIST paper sums it up quite nicely:
>>>>>>>>>>>>> 'It is not difficult to achieve a sensitivity of a
>>>>>>>>>>>>>
>>> part in e14 per Hz resolution
>>>
>>>>>>>>>>>>> so one has excellent precision capabilities with this system.'
>>>>>>>>>>>>>
>>>>>>>>>>>>> This does not address your other question of ADEV vs MDEV,
>>>>>>>>>>>>> What I've described is just a simple way to get the
>>>>>>>>>>>>>
>>> Low cost, GOOD Raw data.
>>>
>>>>>>>>>>>>> What you then do with that Data is a different subject.
>>>>>>>>>>>>>
>>>>>>>>>>>>> You can run the raw data thru one of the many ADEV
>>>>>>>>>>>>>
>>> programs out there, 'Plotter' being my choice.
>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> Have fun
>>>>>>>>>>>>> ws
>>>>>>>>>>>>>
>>>>>>>>>>>>> *************
>>>>>>>>>>>>>
>>>>>>>>>>>>> [time-nuts] ADEV vs MDEV
>>>>>>>>>>>>> Pete Rawson peterawson at earthlink.net
>>>>>>>>>>>>> Sat Feb 6 03:59:18 UTC 2010
>>>>>>>>>>>>>
>>>>>>>>>>>>> Efforts are underway to develop a low cost DMTD apparatus with
>>>>>>>>>>>>> demonstrated stability measurements of 1E-13 in 1s. It
>>>>>>>>>>>>>
>>> seems that
>>>
>>>>>>>>>>>>> existing TI counters can reach this goal in 10s.
>>>>>>>>>>>>>
>>> (using MDEV estimate
>>>
>>>>>>>>>>>>> or 100+s. using ADEV estimate). The question is; does
>>>>>>>>>>>>>
>>> the MDEV tool
>>>
>>>>>>>>>>>>> provide an appropriate measure of stability in this
>>>>>>>>>>>>>
>>> time range, or is
>>>
>>>>>>>>>>>>> the ADEV estimate a more correct answer?
>>>>>>>>>>>>>
>>>>>>>>>>>>> The TI performance I'm referring to is the 20-25 ps,
>>>>>>>>>>>>>
>>> single shot TI,
>>>
>>>>>>>>>>>>> typical for theHP5370A/B, the SR620 or the CNT81/91. I
>>>>>>>>>>>>>
>>> have data
>>>
>>>>>>>>>>>>> from my CNT81showing MDEV< 1E-13 in 10s. and I believe the
>>>>>>>>>>>>> other counters behave similarly.
>>>>>>>>>>>>>
>>>>>>>>>>>>> I would appreciate any comments or observations on this topic.
>>>>>>>>>>>>> My motivation is to discover the simplest scheme for making
>>>>>>>>>>>>> stability measurements at this performance level; this is NOT
>>>>>>>>>>>>> even close to the state-of-the-art, but can still be useful.
>>>>>>>>>>>>>
>>>>>>>>>>>>> Pete Rawson
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>
>
>
> _______________________________________________
> time-nuts mailing list -- time-nuts@febo.com
> To unsubscribe, go to https://www.febo.com/cgi-bin/mailman/listinfo/time-nuts
> and follow the instructions there.
>
BG
Bruce Griffiths
Sun, Feb 7, 2010 2:02 AM
JPL resorted to using a commercial synthesiser set for an offset of
123Hz (to minimise spurs and other artifacts) in their 100MHz N channel
mixer system.
Occasionally one comes across 5.000055MHz OCXOs that use 10.000110MHz
crystals internally.
The resultant 55Hz (with 5MHz source) or 110Hz (with 10MHz source) beat
frequencies are lie between the hamonincs of either 50Hz or 60Hz line
frequencies.
Bruce
Bob Camp wrote:
Hi
Any approach that includes building a low noise synthesizer is opening up a whole new set of issues. I would much prefer to do my building at audio. Audio parts are cheap, and performance is usually a lot easier to check than at RF.
Bob
On Feb 6, 2010, at 8:30 PM, Bruce Griffiths wrote:
Which just leaves the minor problem of the offset oscillator.
One option is to use a phase truncation spur free output frequency from a DDS.
If one is using the Costas receiver approach the beat frequency need not be a nice round number like 1.0000KHz.
Another method is to use a crystal whose frequency is offset a few kHz from 10MHz.
Yet another is the classical method of dividing 10MHz by 100 and subtracting (using an LSB mixer) the resultant 100KHz from 10MHz to produce 9.9MHz, then divide the 9.9MHz signal by 100 and add (using a USB mixer) the resultant 99kHz signal to the 9.99Mhz signal to produce a 9.999MHz output.
Bruce
John Miles wrote:
A sound-card back end has always seemed like a pretty reasonable approach to
me, if you're inclined to go the DMTD route. I wouldn't send a 'baseband'
signal to the sound card, though -- I'd upconvert it to a few kHz to get
away from the numerous bad things that sound cards do near DC.
-- john, KE5FX
Hi
My main concern with the low frequency pole in the sound card is
the quality of the R/C used. You can certainly model what ever
you have. If they used an aluminum electrolytic for the "C" it
may not be the same next time you check it ....
On a 10 Hz system, a 1 Hz pole is probably not an issue. It might
get in the way with a 1 Hz beat note.
Another thing I have only seen in passing: "Sigma Delta's have
poor low frequency noise characteristics". I haven't dug into it
to see if that's really true or not. If you buy your own ADC's,
you certainly would not be restricted to a Sigma Delta.
Even with a cheap pre-built FPGA board, you could look into
higher sample rates than a conventional sound card. You would
drop back to 16 bits, but it might be worth it.
Bob
JPL resorted to using a commercial synthesiser set for an offset of
123Hz (to minimise spurs and other artifacts) in their 100MHz N channel
mixer system.
Occasionally one comes across 5.000055MHz OCXOs that use 10.000110MHz
crystals internally.
The resultant 55Hz (with 5MHz source) or 110Hz (with 10MHz source) beat
frequencies are lie between the hamonincs of either 50Hz or 60Hz line
frequencies.
Bruce
Bob Camp wrote:
> Hi
>
> Any approach that includes building a low noise synthesizer is opening up a whole new set of issues. I would much prefer to do my building at audio. Audio parts are cheap, and performance is usually a lot easier to check than at RF.
>
> Bob
>
>
> On Feb 6, 2010, at 8:30 PM, Bruce Griffiths wrote:
>
>
>> Which just leaves the minor problem of the offset oscillator.
>>
>> One option is to use a phase truncation spur free output frequency from a DDS.
>> If one is using the Costas receiver approach the beat frequency need not be a nice round number like 1.0000KHz.
>>
>> Another method is to use a crystal whose frequency is offset a few kHz from 10MHz.
>>
>> Yet another is the classical method of dividing 10MHz by 100 and subtracting (using an LSB mixer) the resultant 100KHz from 10MHz to produce 9.9MHz, then divide the 9.9MHz signal by 100 and add (using a USB mixer) the resultant 99kHz signal to the 9.99Mhz signal to produce a 9.999MHz output.
>>
>> Bruce
>>
>> John Miles wrote:
>>
>>> A sound-card back end has always seemed like a pretty reasonable approach to
>>> me, if you're inclined to go the DMTD route. I wouldn't send a 'baseband'
>>> signal to the sound card, though -- I'd upconvert it to a few kHz to get
>>> away from the numerous bad things that sound cards do near DC.
>>>
>>> -- john, KE5FX
>>>
>>>
>>>
>>>
>>>> Hi
>>>>
>>>> My main concern with the low frequency pole in the sound card is
>>>> the quality of the R/C used. You can certainly model what ever
>>>> you have. If they used an aluminum electrolytic for the "C" it
>>>> may not be the same next time you check it ....
>>>>
>>>> On a 10 Hz system, a 1 Hz pole is probably not an issue. It might
>>>> get in the way with a 1 Hz beat note.
>>>>
>>>> Another thing I have only seen in passing: "Sigma Delta's have
>>>> poor low frequency noise characteristics". I haven't dug into it
>>>> to see if that's really true or not. If you buy your own ADC's,
>>>> you certainly would not be restricted to a Sigma Delta.
>>>>
>>>> Even with a cheap pre-built FPGA board, you could look into
>>>> higher sample rates than a conventional sound card. You would
>>>> drop back to 16 bits, but it might be worth it.
>>>>
>>>> Bob
>>>>
BC
Bob Camp
Sun, Feb 7, 2010 2:07 AM
Hi
Occasionally you also come across 5.000055 MHz OCXO's that have 5 MHz crystals in them. Then you discover just how much short term stability can degrade when they move the crystal 55 Hz. Same vendor crystal, same crystal spec., same oscillator circuit, not even close on short term stability....
Bob
On Feb 6, 2010, at 9:02 PM, Bruce Griffiths wrote:
JPL resorted to using a commercial synthesiser set for an offset of 123Hz (to minimise spurs and other artifacts) in their 100MHz N channel mixer system.
Occasionally one comes across 5.000055MHz OCXOs that use 10.000110MHz crystals internally.
The resultant 55Hz (with 5MHz source) or 110Hz (with 10MHz source) beat frequencies are lie between the hamonincs of either 50Hz or 60Hz line frequencies.
Bruce
Bob Camp wrote:
Hi
Any approach that includes building a low noise synthesizer is opening up a whole new set of issues. I would much prefer to do my building at audio. Audio parts are cheap, and performance is usually a lot easier to check than at RF.
Bob
On Feb 6, 2010, at 8:30 PM, Bruce Griffiths wrote:
Which just leaves the minor problem of the offset oscillator.
One option is to use a phase truncation spur free output frequency from a DDS.
If one is using the Costas receiver approach the beat frequency need not be a nice round number like 1.0000KHz.
Another method is to use a crystal whose frequency is offset a few kHz from 10MHz.
Yet another is the classical method of dividing 10MHz by 100 and subtracting (using an LSB mixer) the resultant 100KHz from 10MHz to produce 9.9MHz, then divide the 9.9MHz signal by 100 and add (using a USB mixer) the resultant 99kHz signal to the 9.99Mhz signal to produce a 9.999MHz output.
Bruce
John Miles wrote:
A sound-card back end has always seemed like a pretty reasonable approach to
me, if you're inclined to go the DMTD route. I wouldn't send a 'baseband'
signal to the sound card, though -- I'd upconvert it to a few kHz to get
away from the numerous bad things that sound cards do near DC.
-- john, KE5FX
Hi
My main concern with the low frequency pole in the sound card is
the quality of the R/C used. You can certainly model what ever
you have. If they used an aluminum electrolytic for the "C" it
may not be the same next time you check it ....
On a 10 Hz system, a 1 Hz pole is probably not an issue. It might
get in the way with a 1 Hz beat note.
Another thing I have only seen in passing: "Sigma Delta's have
poor low frequency noise characteristics". I haven't dug into it
to see if that's really true or not. If you buy your own ADC's,
you certainly would not be restricted to a Sigma Delta.
Even with a cheap pre-built FPGA board, you could look into
higher sample rates than a conventional sound card. You would
drop back to 16 bits, but it might be worth it.
Bob
Hi
Occasionally you also come across 5.000055 MHz OCXO's that have 5 MHz crystals in them. Then you discover just how much short term stability can degrade when they move the crystal 55 Hz. Same vendor crystal, same crystal spec., same oscillator circuit, not even close on short term stability....
Bob
On Feb 6, 2010, at 9:02 PM, Bruce Griffiths wrote:
> JPL resorted to using a commercial synthesiser set for an offset of 123Hz (to minimise spurs and other artifacts) in their 100MHz N channel mixer system.
>
> Occasionally one comes across 5.000055MHz OCXOs that use 10.000110MHz crystals internally.
> The resultant 55Hz (with 5MHz source) or 110Hz (with 10MHz source) beat frequencies are lie between the hamonincs of either 50Hz or 60Hz line frequencies.
>
> Bruce
>
> Bob Camp wrote:
>> Hi
>>
>> Any approach that includes building a low noise synthesizer is opening up a whole new set of issues. I would much prefer to do my building at audio. Audio parts are cheap, and performance is usually a lot easier to check than at RF.
>>
>> Bob
>>
>>
>> On Feb 6, 2010, at 8:30 PM, Bruce Griffiths wrote:
>>
>>
>>> Which just leaves the minor problem of the offset oscillator.
>>>
>>> One option is to use a phase truncation spur free output frequency from a DDS.
>>> If one is using the Costas receiver approach the beat frequency need not be a nice round number like 1.0000KHz.
>>>
>>> Another method is to use a crystal whose frequency is offset a few kHz from 10MHz.
>>>
>>> Yet another is the classical method of dividing 10MHz by 100 and subtracting (using an LSB mixer) the resultant 100KHz from 10MHz to produce 9.9MHz, then divide the 9.9MHz signal by 100 and add (using a USB mixer) the resultant 99kHz signal to the 9.99Mhz signal to produce a 9.999MHz output.
>>>
>>> Bruce
>>>
>>> John Miles wrote:
>>>
>>>> A sound-card back end has always seemed like a pretty reasonable approach to
>>>> me, if you're inclined to go the DMTD route. I wouldn't send a 'baseband'
>>>> signal to the sound card, though -- I'd upconvert it to a few kHz to get
>>>> away from the numerous bad things that sound cards do near DC.
>>>>
>>>> -- john, KE5FX
>>>>
>>>>
>>>>
>>>>
>>>>> Hi
>>>>>
>>>>> My main concern with the low frequency pole in the sound card is
>>>>> the quality of the R/C used. You can certainly model what ever
>>>>> you have. If they used an aluminum electrolytic for the "C" it
>>>>> may not be the same next time you check it ....
>>>>>
>>>>> On a 10 Hz system, a 1 Hz pole is probably not an issue. It might
>>>>> get in the way with a 1 Hz beat note.
>>>>>
>>>>> Another thing I have only seen in passing: "Sigma Delta's have
>>>>> poor low frequency noise characteristics". I haven't dug into it
>>>>> to see if that's really true or not. If you buy your own ADC's,
>>>>> you certainly would not be restricted to a Sigma Delta.
>>>>>
>>>>> Even with a cheap pre-built FPGA board, you could look into
>>>>> higher sample rates than a conventional sound card. You would
>>>>> drop back to 16 bits, but it might be worth it.
>>>>>
>>>>> Bob
>>>>>
>
>
> _______________________________________________
> time-nuts mailing list -- time-nuts@febo.com
> To unsubscribe, go to https://www.febo.com/cgi-bin/mailman/listinfo/time-nuts
> and follow the instructions there.
>
BG
Bruce Griffiths
Sun, Feb 7, 2010 2:40 AM
As a matter of interest just how bad were those OCXOs?
e.g. what was the ballpark ADEV for 1s, 10s etc.?
Bruce
Bob Camp wrote:
Hi
Occasionally you also come across 5.000055 MHz OCXO's that have 5 MHz crystals in them. Then you discover just how much short term stability can degrade when they move the crystal 55 Hz. Same vendor crystal, same crystal spec., same oscillator circuit, not even close on short term stability....
Bob
On Feb 6, 2010, at 9:02 PM, Bruce Griffiths wrote:
JPL resorted to using a commercial synthesiser set for an offset of 123Hz (to minimise spurs and other artifacts) in their 100MHz N channel mixer system.
Occasionally one comes across 5.000055MHz OCXOs that use 10.000110MHz crystals internally.
The resultant 55Hz (with 5MHz source) or 110Hz (with 10MHz source) beat frequencies are lie between the hamonincs of either 50Hz or 60Hz line frequencies.
Bruce
Bob Camp wrote:
Hi
Any approach that includes building a low noise synthesizer is opening up a whole new set of issues. I would much prefer to do my building at audio. Audio parts are cheap, and performance is usually a lot easier to check than at RF.
Bob
On Feb 6, 2010, at 8:30 PM, Bruce Griffiths wrote:
Which just leaves the minor problem of the offset oscillator.
One option is to use a phase truncation spur free output frequency from a DDS.
If one is using the Costas receiver approach the beat frequency need not be a nice round number like 1.0000KHz.
Another method is to use a crystal whose frequency is offset a few kHz from 10MHz.
Yet another is the classical method of dividing 10MHz by 100 and subtracting (using an LSB mixer) the resultant 100KHz from 10MHz to produce 9.9MHz, then divide the 9.9MHz signal by 100 and add (using a USB mixer) the resultant 99kHz signal to the 9.99Mhz signal to produce a 9.999MHz output.
Bruce
John Miles wrote:
A sound-card back end has always seemed like a pretty reasonable approach to
me, if you're inclined to go the DMTD route. I wouldn't send a 'baseband'
signal to the sound card, though -- I'd upconvert it to a few kHz to get
away from the numerous bad things that sound cards do near DC.
-- john, KE5FX
Hi
My main concern with the low frequency pole in the sound card is
the quality of the R/C used. You can certainly model what ever
you have. If they used an aluminum electrolytic for the "C" it
may not be the same next time you check it ....
On a 10 Hz system, a 1 Hz pole is probably not an issue. It might
get in the way with a 1 Hz beat note.
Another thing I have only seen in passing: "Sigma Delta's have
poor low frequency noise characteristics". I haven't dug into it
to see if that's really true or not. If you buy your own ADC's,
you certainly would not be restricted to a Sigma Delta.
Even with a cheap pre-built FPGA board, you could look into
higher sample rates than a conventional sound card. You would
drop back to 16 bits, but it might be worth it.
Bob
As a matter of interest just how bad were those OCXOs?
e.g. what was the ballpark ADEV for 1s, 10s etc.?
Bruce
Bob Camp wrote:
> Hi
>
> Occasionally you also come across 5.000055 MHz OCXO's that have 5 MHz crystals in them. Then you discover just how much short term stability can degrade when they move the crystal 55 Hz. Same vendor crystal, same crystal spec., same oscillator circuit, not even close on short term stability....
>
> Bob
>
>
> On Feb 6, 2010, at 9:02 PM, Bruce Griffiths wrote:
>
>
>> JPL resorted to using a commercial synthesiser set for an offset of 123Hz (to minimise spurs and other artifacts) in their 100MHz N channel mixer system.
>>
>> Occasionally one comes across 5.000055MHz OCXOs that use 10.000110MHz crystals internally.
>> The resultant 55Hz (with 5MHz source) or 110Hz (with 10MHz source) beat frequencies are lie between the hamonincs of either 50Hz or 60Hz line frequencies.
>>
>> Bruce
>>
>> Bob Camp wrote:
>>
>>> Hi
>>>
>>> Any approach that includes building a low noise synthesizer is opening up a whole new set of issues. I would much prefer to do my building at audio. Audio parts are cheap, and performance is usually a lot easier to check than at RF.
>>>
>>> Bob
>>>
>>>
>>> On Feb 6, 2010, at 8:30 PM, Bruce Griffiths wrote:
>>>
>>>
>>>
>>>> Which just leaves the minor problem of the offset oscillator.
>>>>
>>>> One option is to use a phase truncation spur free output frequency from a DDS.
>>>> If one is using the Costas receiver approach the beat frequency need not be a nice round number like 1.0000KHz.
>>>>
>>>> Another method is to use a crystal whose frequency is offset a few kHz from 10MHz.
>>>>
>>>> Yet another is the classical method of dividing 10MHz by 100 and subtracting (using an LSB mixer) the resultant 100KHz from 10MHz to produce 9.9MHz, then divide the 9.9MHz signal by 100 and add (using a USB mixer) the resultant 99kHz signal to the 9.99Mhz signal to produce a 9.999MHz output.
>>>>
>>>> Bruce
>>>>
>>>> John Miles wrote:
>>>>
>>>>
>>>>> A sound-card back end has always seemed like a pretty reasonable approach to
>>>>> me, if you're inclined to go the DMTD route. I wouldn't send a 'baseband'
>>>>> signal to the sound card, though -- I'd upconvert it to a few kHz to get
>>>>> away from the numerous bad things that sound cards do near DC.
>>>>>
>>>>> -- john, KE5FX
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>> Hi
>>>>>>
>>>>>> My main concern with the low frequency pole in the sound card is
>>>>>> the quality of the R/C used. You can certainly model what ever
>>>>>> you have. If they used an aluminum electrolytic for the "C" it
>>>>>> may not be the same next time you check it ....
>>>>>>
>>>>>> On a 10 Hz system, a 1 Hz pole is probably not an issue. It might
>>>>>> get in the way with a 1 Hz beat note.
>>>>>>
>>>>>> Another thing I have only seen in passing: "Sigma Delta's have
>>>>>> poor low frequency noise characteristics". I haven't dug into it
>>>>>> to see if that's really true or not. If you buy your own ADC's,
>>>>>> you certainly would not be restricted to a Sigma Delta.
>>>>>>
>>>>>> Even with a cheap pre-built FPGA board, you could look into
>>>>>> higher sample rates than a conventional sound card. You would
>>>>>> drop back to 16 bits, but it might be worth it.
>>>>>>
>>>>>> Bob
>>>>>>
>>>>>>
>>
>>
JM
John Miles
Sun, Feb 7, 2010 3:28 AM
Any noise or drift in the "2nd LO", so to speak, would be common-mode
between the two channels. It shouldn't be all that critical.
-- john, KE5FX
-----Original Message-----
From: time-nuts-bounces@febo.com [mailto:time-nuts-bounces@febo.com]On
Behalf Of Bob Camp
Sent: Saturday, February 06, 2010 5:52 PM
To: Discussion of precise time and frequency measurement
Subject: Re: [time-nuts] ADEV vs MDEV - using sound card
Hi
Any approach that includes building a low noise synthesizer is
opening up a whole new set of issues. I would much prefer to do
my building at audio. Audio parts are cheap, and performance is
usually a lot easier to check than at RF.
Bob
On Feb 6, 2010, at 8:30 PM, Bruce Griffiths wrote:
Which just leaves the minor problem of the offset oscillator.
One option is to use a phase truncation spur free output
If one is using the Costas receiver approach the beat frequency
need not be a nice round number like 1.0000KHz.
Another method is to use a crystal whose frequency is offset a
Yet another is the classical method of dividing 10MHz by 100
and subtracting (using an LSB mixer) the resultant 100KHz from
10MHz to produce 9.9MHz, then divide the 9.9MHz signal by 100 and
add (using a USB mixer) the resultant 99kHz signal to the 9.99Mhz
signal to produce a 9.999MHz output.
A sound-card back end has always seemed like a pretty
me, if you're inclined to go the DMTD route. I wouldn't send
signal to the sound card, though -- I'd upconvert it to a few
away from the numerous bad things that sound cards do near DC.
-- john, KE5FX
Hi
My main concern with the low frequency pole in the sound card is
the quality of the R/C used. You can certainly model what ever
you have. If they used an aluminum electrolytic for the "C" it
may not be the same next time you check it ....
On a 10 Hz system, a 1 Hz pole is probably not an issue. It might
get in the way with a 1 Hz beat note.
Another thing I have only seen in passing: "Sigma Delta's have
poor low frequency noise characteristics". I haven't dug into it
to see if that's really true or not. If you buy your own ADC's,
you certainly would not be restricted to a Sigma Delta.
Even with a cheap pre-built FPGA board, you could look into
higher sample rates than a conventional sound card. You would
drop back to 16 bits, but it might be worth it.
Bob
On Feb 6, 2010, at 6:46 PM, Bruce Griffiths wrote:
Even better is to toss out the mixers and sample the RF signals
However suitable ADCs cost $US100 or more each.
To which one has to add an FPGA and an interface to a PC with
sufficient throughput to handle the down converted I + Q samples.
Hi
You probably could put a couple of cheap DAC's
(ADCs are preferable as it avoids having to implement the
conversion logic plus comparator required when using a DAC.)
on a board with a FPGA and reduce the data on the fly. I'd
guess that would be be in the same $100 range as a half way
decent sound card. Clock the DAC's off of a 10 MHz reference and
eliminate the cal issue.
If you are down around 10 Hz or worse yet 1 Hz, the AC
coupling of the sound card will get in the way, even with a
bandpass approach. You really don't know what they may have in
there at the low end. Build it yourself and that stuff's not an issue.
My sound card has a 1Hz cutoff RC high pass input filter plus
an internal high pass digital filter.
Its not too difficult to measure the sound card frequency
response using a white noise source for example.
On Feb 6, 2010, at 6:12 PM, Bruce Griffiths wrote:
If one has a high end sound card then it could be used to
implement the bandpass filter and replace the zero crossing detector.
It may be necessary to insert a pilot tone to calibrate the
sound card sampling clock frequency.
A noise floor of about 1E-13/Tau should be achievable.
This simplifies the DMTD system by replacing the zero
crossing detector with a low gain linear preamp.
If one analyses the resultant data off line then one can also
try out different techniques such as a Costas receiver rather
than a simple bandpass filter plus zero crossing detector.
However 1000 seconds of data for 2 channels of 24 bit samples
at 192KSPS will result in a file with a size of at least 1.15GB.
Bruce
Bruce Griffiths wrote:
If one were to use a bandpass filter with a Q of 10 to
filter the beat frequency output of the mixer, then if the input
frequency is 10MHz and the filter component tempco is 100ppm/C
then the resultant phase shift tempco is about 16ps/C referred to
the mixer input frequency.
This phase shift tempco is certainly low enough not to have
significant impact when measuring the frequency stability of a
typical 10811A if the temperature fluctuations are kept small
enough during the run.
The effect of using a bandpass filter with too narrow a
bandwidth is to artificially reduce ADEV for small Tau, so it may
be prudent to use a higher beat frequency that 1Hz or even 10Hz
and not calculate ADEV for Tau less than say 10(??) times the
beat frequency period. A trade off between this and the effect of
aliasing is required.
Hi
With most 10811 range oscillators the impact of a simple
bandpass filter is low enough to not be a major issue. That's for
normal lab temperatures with the circuitry in a conventional die
cast box. No guarantee if you open the window and let the fresh
air blow in during the run.
That's true with a heterodyne. I can see no obvious reason
it would not be true on DMTD.
Bob
On Feb 6, 2010, at 5:12 PM, Bruce Griffiths wrote:
The only major issue with DMTD systems is that they
undersample the phase fluctuations and hence are subject to
aliasing effects.
The low pass filter has to have a bandwidth of the same
order as the beat frequency or the beat frequency signal will be
significantly attenuated.
Since the phase is only sampled once per beat frequency
period the phase fluctuations are undersampled.
Various attempts to use both zero crossings have not been
In principle if one can overcome the increased phase shift
tempco associated with a bandpass filter, using a bandpass filter
can in principle ensure that the phase fluctuations are oversampled.
Hi
A straight heterodyne system will get you to the floor of
most 10811's with a very simple (2 stage) limiter. As with the
DMTD, the counter requirements aren't really all that severe.
Bob
On Feb 6, 2010, at 4:24 PM, WarrenS wrote:
"It's possible / likely for injection lock ... to be a
Something I certainly worried about and tested for.
What I found (for MY case) is that injection lock is NOT
The reason being is that unlike most other ways, where
the two OSC have to be completely independent,
The tight loop approach forces the Two Osc to "Lock with
something like 60 + db gain,
so a little stray -80db injection lock coupling that
would very much limit other systems has
no measurable effect at e-13. Just one of the neat
little side effects that make the tight loop approach so simple.
"then a part in 10^14 is going to be at the 100 of
For that example, just need to put a simple discrete 100
in-between the control voltage and the EFC and now you
have a nice workable 10uv.
BUT the bigger point is, probable not needed, cause you
are NOT going to do any better than the stability of the OSC with
a grounded shorted EFC input.
as you said and I agree is so true:
"There is no perfect way to do any of this, only a lot
of compromises ... you need to watch out for".
But you did not offer any easier way to do it, which is
what the original request was for and my answer addressed.
This is the cheapest easiest way BY FAR to get high
performance, at low tau, ADEV numbers that I've seen.
ws
----- Original Message ----- From: "Bob Camp"lists@cq.nu
To: "Discussion of precise time and frequency
Sent: Saturday, February 06, 2010 12:09 PM
Subject: Re: [time-nuts] ADEV vs MDEV
Hi
It's possible / likely to injection lock with the tight
loop approach and get data that's much better than reality. A lot
depends on the specific oscillators under test and the buffers
(if any) between the oscillators and mixer.
If your OCVCXO has a tuning slope of 0.1 ppm / volt
then a part in 10^14 is going to be at the 100 of nanovolts
level. Certainly not impossible, but it does present it's own set
of issues. Lab gear to do it is available, but not all that
common. DC offsets and their temperature coefficients along with
thermocouple effects could make things exciting.
There is no perfect way to do any of this, only a lot
of compromises here or there. Each approach has stuff you need to
watch out for.
Bob
From: "WarrenS"warrensjmail-one@yahoo.com
Sent: Saturday, February 06, 2010 2:19 PM
To: "Discussion of precise time and frequency
Subject: Re: [time-nuts] ADEV vs MDEV
I would appreciate any comments or observations on
the topic of apparatus with demonstrated stability measurements.
My motivation is to discover the SIMPLEST scheme for
making stability measurements at the 1E-13 in 1s performance level.
If you accept that the measurement is going to limited
for Low COST and SIMPLE, with the ability to measure
Can't beat a simple analog version of NIST's "Tight
Phase-Lock Loop Method of measuring Freq stability".
By replacing the "Voltage to freq converter, Freq
counter& Printer with a Radio shack type PC data logging DVM,
It can be up and running from scratch in under an Hr,
with no high end test equipment needed.
If you want performance that exceeds the best of most
DMTD at low Tau it takes a little more work
and a higher speed oversampling ADC data logger and a
I must add this is not a popular solution (Or a
IF you know analog and have a GOOD osc with EFC to
as far as I've been able to determine it is the BEST
SIMPLE answer that allows High performance.
Limited by My HP10811 Ref OSC, I'm getting better than
1e-12 in 0.1 sec (at 30 Hz Bandwidth)
Basic modified NIST Block Diag attached:
The NIST paper sums it up quite nicely:
'It is not difficult to achieve a sensitivity of a
part in e14 per Hz resolution
so one has excellent precision capabilities with
This does not address your other question of ADEV vs MDEV,
What I've described is just a simple way to get the
What you then do with that Data is a different subject.
You can run the raw data thru one of the many ADEV
programs out there, 'Plotter' being my choice.
Have fun
ws
[time-nuts] ADEV vs MDEV
Pete Rawson peterawson at earthlink.net
Sat Feb 6 03:59:18 UTC 2010
Efforts are underway to develop a low cost DMTD
demonstrated stability measurements of 1E-13 in 1s. It
existing TI counters can reach this goal in 10s.
or 100+s. using ADEV estimate). The question is; does
provide an appropriate measure of stability in this
the ADEV estimate a more correct answer?
The TI performance I'm referring to is the 20-25 ps,
typical for theHP5370A/B, the SR620 or the CNT81/91. I
from my CNT81showing MDEV< 1E-13 in 10s. and I
other counters behave similarly.
I would appreciate any comments or observations on
My motivation is to discover the simplest scheme for making
stability measurements at this performance level;
even close to the state-of-the-art, but can still be useful.
Pete Rawson
and follow the instructions there.
Any noise or drift in the "2nd LO", so to speak, would be common-mode
between the two channels. It shouldn't be all that critical.
-- john, KE5FX
> -----Original Message-----
> From: time-nuts-bounces@febo.com [mailto:time-nuts-bounces@febo.com]On
> Behalf Of Bob Camp
> Sent: Saturday, February 06, 2010 5:52 PM
> To: Discussion of precise time and frequency measurement
> Subject: Re: [time-nuts] ADEV vs MDEV - using sound card
>
>
> Hi
>
> Any approach that includes building a low noise synthesizer is
> opening up a whole new set of issues. I would much prefer to do
> my building at audio. Audio parts are cheap, and performance is
> usually a lot easier to check than at RF.
>
> Bob
>
>
> On Feb 6, 2010, at 8:30 PM, Bruce Griffiths wrote:
>
> > Which just leaves the minor problem of the offset oscillator.
> >
> > One option is to use a phase truncation spur free output
> frequency from a DDS.
> > If one is using the Costas receiver approach the beat frequency
> need not be a nice round number like 1.0000KHz.
> >
> > Another method is to use a crystal whose frequency is offset a
> few kHz from 10MHz.
> >
> > Yet another is the classical method of dividing 10MHz by 100
> and subtracting (using an LSB mixer) the resultant 100KHz from
> 10MHz to produce 9.9MHz, then divide the 9.9MHz signal by 100 and
> add (using a USB mixer) the resultant 99kHz signal to the 9.99Mhz
> signal to produce a 9.999MHz output.
> >
> > Bruce
> >
> > John Miles wrote:
> >> A sound-card back end has always seemed like a pretty
> reasonable approach to
> >> me, if you're inclined to go the DMTD route. I wouldn't send
> a 'baseband'
> >> signal to the sound card, though -- I'd upconvert it to a few
> kHz to get
> >> away from the numerous bad things that sound cards do near DC.
> >>
> >> -- john, KE5FX
> >>
> >>
> >>
> >>> Hi
> >>>
> >>> My main concern with the low frequency pole in the sound card is
> >>> the quality of the R/C used. You can certainly model what ever
> >>> you have. If they used an aluminum electrolytic for the "C" it
> >>> may not be the same next time you check it ....
> >>>
> >>> On a 10 Hz system, a 1 Hz pole is probably not an issue. It might
> >>> get in the way with a 1 Hz beat note.
> >>>
> >>> Another thing I have only seen in passing: "Sigma Delta's have
> >>> poor low frequency noise characteristics". I haven't dug into it
> >>> to see if that's really true or not. If you buy your own ADC's,
> >>> you certainly would not be restricted to a Sigma Delta.
> >>>
> >>> Even with a cheap pre-built FPGA board, you could look into
> >>> higher sample rates than a conventional sound card. You would
> >>> drop back to 16 bits, but it might be worth it.
> >>>
> >>> Bob
> >>>
> >>>
> >>> On Feb 6, 2010, at 6:46 PM, Bruce Griffiths wrote:
> >>>
> >>>
> >>>> Even better is to toss out the mixers and sample the RF signals
> >>>>
> >>> directly.
> >>>
> >>>> However suitable ADCs cost $US100 or more each.
> >>>> To which one has to add an FPGA and an interface to a PC with
> >>>>
> >>> sufficient throughput to handle the down converted I + Q samples.
> >>>
> >>>> Bob Camp wrote:
> >>>>
> >>>>> Hi
> >>>>>
> >>>>> You probably could put a couple of cheap DAC's
> >>>>>
> >>>> (ADCs are preferable as it avoids having to implement the
> >>>>
> >>> conversion logic plus comparator required when using a DAC.)
> >>>
> >>>>
> >>>>> on a board with a FPGA and reduce the data on the fly. I'd
> >>>>>
> >>> guess that would be be in the same $100 range as a half way
> >>> decent sound card. Clock the DAC's off of a 10 MHz reference and
> >>> eliminate the cal issue.
> >>>
> >>>>> If you are down around 10 Hz or worse yet 1 Hz, the AC
> >>>>>
> >>> coupling of the sound card will get in the way, even with a
> >>> bandpass approach. You really don't know what they may have in
> >>> there at the low end. Build it yourself and that stuff's not an issue.
> >>>
> >>>>> Bob
> >>>>>
> >>>>>
> >>>>>
> >>>> My sound card has a 1Hz cutoff RC high pass input filter plus
> >>>>
> >>> an internal high pass digital filter.
> >>>
> >>>> Its not too difficult to measure the sound card frequency
> >>>>
> >>> response using a white noise source for example.
> >>>
> >>>> Bruce
> >>>>
> >>>>> On Feb 6, 2010, at 6:12 PM, Bruce Griffiths wrote:
> >>>>>
> >>>>>
> >>>>>
> >>>>>> If one has a high end sound card then it could be used to
> >>>>>>
> >>> implement the bandpass filter and replace the zero crossing detector.
> >>>
> >>>>>> It may be necessary to insert a pilot tone to calibrate the
> >>>>>>
> >>> sound card sampling clock frequency.
> >>>
> >>>>>> A noise floor of about 1E-13/Tau should be achievable.
> >>>>>> This simplifies the DMTD system by replacing the zero
> >>>>>>
> >>> crossing detector with a low gain linear preamp.
> >>>
> >>>>>> If one analyses the resultant data off line then one can also
> >>>>>>
> >>> try out different techniques such as a Costas receiver rather
> >>> than a simple bandpass filter plus zero crossing detector.
> >>>
> >>>>>> However 1000 seconds of data for 2 channels of 24 bit samples
> >>>>>>
> >>> at 192KSPS will result in a file with a size of at least 1.15GB.
> >>>
> >>>>>> Bruce
> >>>>>>
> >>>>>>
> >>>>>> Bruce Griffiths wrote:
> >>>>>>
> >>>>>>
> >>>>>>> If one were to use a bandpass filter with a Q of 10 to
> >>>>>>>
> >>> filter the beat frequency output of the mixer, then if the input
> >>> frequency is 10MHz and the filter component tempco is 100ppm/C
> >>> then the resultant phase shift tempco is about 16ps/C referred to
> >>> the mixer input frequency.
> >>>
> >>>>>>> This phase shift tempco is certainly low enough not to have
> >>>>>>>
> >>> significant impact when measuring the frequency stability of a
> >>> typical 10811A if the temperature fluctuations are kept small
> >>> enough during the run.
> >>>
> >>>>>>> The effect of using a bandpass filter with too narrow a
> >>>>>>>
> >>> bandwidth is to artificially reduce ADEV for small Tau, so it may
> >>> be prudent to use a higher beat frequency that 1Hz or even 10Hz
> >>> and not calculate ADEV for Tau less than say 10(??) times the
> >>> beat frequency period. A trade off between this and the effect of
> >>> aliasing is required.
> >>>
> >>>>>>> Bruce
> >>>>>>>
> >>>>>>> Bob Camp wrote:
> >>>>>>>
> >>>>>>>
> >>>>>>>> Hi
> >>>>>>>>
> >>>>>>>> With most 10811 range oscillators the impact of a simple
> >>>>>>>>
> >>> bandpass filter is low enough to not be a major issue. That's for
> >>> normal lab temperatures with the circuitry in a conventional die
> >>> cast box. No guarantee if you open the window and let the fresh
> >>> air blow in during the run.
> >>>
> >>>>>>>> That's true with a heterodyne. I can see no obvious reason
> >>>>>>>>
> >>> it would not be true on DMTD.
> >>>
> >>>>>>>> Bob
> >>>>>>>>
> >>>>>>>>
> >>>>>>>> On Feb 6, 2010, at 5:12 PM, Bruce Griffiths wrote:
> >>>>>>>>
> >>>>>>>>
> >>>>>>>>
> >>>>>>>>> The only major issue with DMTD systems is that they
> >>>>>>>>>
> >>> undersample the phase fluctuations and hence are subject to
> >>> aliasing effects.
> >>>
> >>>>>>>>> The low pass filter has to have a bandwidth of the same
> >>>>>>>>>
> >>> order as the beat frequency or the beat frequency signal will be
> >>> significantly attenuated.
> >>>
> >>>>>>>>> Since the phase is only sampled once per beat frequency
> >>>>>>>>>
> >>> period the phase fluctuations are undersampled.
> >>>
> >>>>>>>>> Various attempts to use both zero crossings have not been
> >>>>>>>>>
> >>> successful.
> >>>
> >>>>>>>>> In principle if one can overcome the increased phase shift
> >>>>>>>>>
> >>> tempco associated with a bandpass filter, using a bandpass filter
> >>> can in principle ensure that the phase fluctuations are oversampled.
> >>>
> >>>>>>>>>
> >>>>>>>>> Bruce
> >>>>>>>>>
> >>>>>>>>> Bob Camp wrote:
> >>>>>>>>>
> >>>>>>>>>
> >>>>>>>>>> Hi
> >>>>>>>>>>
> >>>>>>>>>> A straight heterodyne system will get you to the floor of
> >>>>>>>>>>
> >>> most 10811's with a very simple (2 stage) limiter. As with the
> >>> DMTD, the counter requirements aren't really all that severe.
> >>>
> >>>>>>>>>> Bob
> >>>>>>>>>>
> >>>>>>>>>>
> >>>>>>>>>> On Feb 6, 2010, at 4:24 PM, WarrenS wrote:
> >>>>>>>>>>
> >>>>>>>>>>
> >>>>>>>>>>
> >>>>>>>>>>
> >>>>>>>>>>>
> >>>>>>>>>>>> "It's possible / likely for injection lock ... to be a
> >>>>>>>>>>>>
> >>> problem ..."
> >>>
> >>>>>>>>>>>>
> >>>>>>>>>>>>
> >>>>>>>>>>> Something I certainly worried about and tested for.
> >>>>>>>>>>> What I found (for MY case) is that injection lock is NOT
> >>>>>>>>>>>
> >>> a problem.
> >>>
> >>>>>>>>>>> The reason being is that unlike most other ways, where
> >>>>>>>>>>>
> >>> the two OSC have to be completely independent,
> >>>
> >>>>>>>>>>> The tight loop approach forces the Two Osc to "Lock with
> >>>>>>>>>>>
> >>> something like 60 + db gain,
> >>>
> >>>>>>>>>>> so a little stray -80db injection lock coupling that
> >>>>>>>>>>>
> >>> would very much limit other systems has
> >>>
> >>>>>>>>>>> no measurable effect at e-13. Just one of the neat
> >>>>>>>>>>>
> >>> little side effects that make the tight loop approach so simple.
> >>>
> >>>>>>>>>>>
> >>>>>>>>>>>
> >>>>>>>>>>>
> >>>>>>>>>>>> "then a part in 10^14 is going to be at the 100 of
> >>>>>>>>>>>>
> >>> nanovolts level."
> >>>
> >>>>>>>>>>>>
> >>>>>>>>>>>>
> >>>>>>>>>>> For that example, just need to put a simple discrete 100
> >>>>>>>>>>>
> >>> to 1 resistor divider
> >>>
> >>>>>>>>>>> in-between the control voltage and the EFC and now you
> >>>>>>>>>>>
> >>> have a nice workable 10uv.
> >>>
> >>>>>>>>>>> BUT the bigger point is, probable not needed, cause you
> >>>>>>>>>>>
> >>> are NOT going to do any better than the stability of the OSC with
> >>> a grounded shorted EFC input.
> >>>
> >>>>>>>>>>> as you said and I agree is so true:
> >>>>>>>>>>>
> >>>>>>>>>>>
> >>>>>>>>>>>
> >>>>>>>>>>>> "There is no perfect way to do any of this, only a lot
> >>>>>>>>>>>>
> >>> of compromises ... you need to watch out for".
> >>>
> >>>>>>>>>>>>
> >>>>>>>>>>>>
> >>>>>>>>>>> But you did not offer any easier way to do it, which is
> >>>>>>>>>>>
> >>> what the original request was for and my answer addressed.
> >>>
> >>>>>>>>>>> This is the cheapest easiest way BY FAR to get high
> >>>>>>>>>>>
> >>> performance, at low tau, ADEV numbers that I've seen.
> >>>
> >>>>>>>>>>> ws
> >>>>>>>>>>> ***************
> >>>>>>>>>>>
> >>>>>>>>>>> ----- Original Message ----- From: "Bob Camp"<lists@cq.nu>
> >>>>>>>>>>> To: "Discussion of precise time and frequency
> >>>>>>>>>>>
> >>> measurement"<time-nuts@febo.com>
> >>>
> >>>>>>>>>>> Sent: Saturday, February 06, 2010 12:09 PM
> >>>>>>>>>>> Subject: Re: [time-nuts] ADEV vs MDEV
> >>>>>>>>>>>
> >>>>>>>>>>>
> >>>>>>>>>>>
> >>>>>>>>>>>
> >>>>>>>>>>>
> >>>>>>>>>>>> Hi
> >>>>>>>>>>>>
> >>>>>>>>>>>> It's possible / likely to injection lock with the tight
> >>>>>>>>>>>>
> >>> loop approach and get data that's much better than reality. A lot
> >>> depends on the specific oscillators under test and the buffers
> >>> (if any) between the oscillators and mixer.
> >>>
> >>>>>>>>>>>> If your OCVCXO has a tuning slope of 0.1 ppm / volt
> >>>>>>>>>>>>
> >>> then a part in 10^14 is going to be at the 100 of nanovolts
> >>> level. Certainly not impossible, but it does present it's own set
> >>> of issues. Lab gear to do it is available, but not all that
> >>> common. DC offsets and their temperature coefficients along with
> >>> thermocouple effects could make things exciting.
> >>>
> >>>>>>>>>>>> There is no perfect way to do any of this, only a lot
> >>>>>>>>>>>>
> >>> of compromises here or there. Each approach has stuff you need to
> >>> watch out for.
> >>>
> >>>>>>>>>>>> Bob
> >>>>>>>>>>>>
> >>>>>>>>>>>> --------------------------------------------------
> >>>>>>>>>>>> From: "WarrenS"<warrensjmail-one@yahoo.com>
> >>>>>>>>>>>> Sent: Saturday, February 06, 2010 2:19 PM
> >>>>>>>>>>>> To: "Discussion of precise time and frequency
> >>>>>>>>>>>>
> >>> measurement"<time-nuts@febo.com>
> >>>
> >>>>>>>>>>>> Subject: Re: [time-nuts] ADEV vs MDEV
> >>>>>>>>>>>>
> >>>>>>>>>>>>
> >>>>>>>>>>>>
> >>>>>>>>>>>>
> >>>>>>>>>>>>> Peat said:
> >>>>>>>>>>>>>
> >>>>>>>>>>>>>
> >>>>>>>>>>>>>
> >>>>>>>>>>>>>> I would appreciate any comments or observations on
> >>>>>>>>>>>>>>
> >>> the topic of apparatus with demonstrated stability measurements.
> >>>
> >>>>>>>>>>>>>> My motivation is to discover the SIMPLEST scheme for
> >>>>>>>>>>>>>>
> >>> making stability measurements at the 1E-13 in 1s performance level.
> >>>
> >>>>>>>>>>>>>>
> >>>>>>>>>>>>>>
> >>>>>>>>>>>>> If you accept that the measurement is going to limited
> >>>>>>>>>>>>>
> >>> by the Reference Osc,
> >>>
> >>>>>>>>>>>>> for Low COST and SIMPLE, with the ability to measure
> >>>>>>>>>>>>>
> >>> ADEVs at that level,
> >>>
> >>>>>>>>>>>>> Can't beat a simple analog version of NIST's "Tight
> >>>>>>>>>>>>>
> >>> Phase-Lock Loop Method of measuring Freq stability".
> >>>
> >>>>>>>>>>>>> http://tf.nist.gov/phase/Properties/one.htm#oneone
> Fig 1.7
> >>>>>>>>>>>>>
> >>>>>>>>>>>>>
> >>>>>>>>>>>>> By replacing the "Voltage to freq converter, Freq
> >>>>>>>>>>>>>
> >>> counter& Printer with a Radio shack type PC data logging DVM,
> >>>
> >>>>>>>>>>>>> It can be up and running from scratch in under an Hr,
> >>>>>>>>>>>>>
> >>> with no high end test equipment needed.
> >>>
> >>>>>>>>>>>>> If you want performance that exceeds the best of most
> >>>>>>>>>>>>>
> >>> DMTD at low Tau it takes a little more work
> >>>
> >>>>>>>>>>>>> and a higher speed oversampling ADC data logger and a
> >>>>>>>>>>>>>
> >>> good offset voltage.
> >>>
> >>>>>>>>>>>>> I must add this is not a popular solution (Or a
> >>>>>>>>>>>>>
> >>> general Purpose one) but
> >>>
> >>>>>>>>>>>>> IF you know analog and have a GOOD osc with EFC to
> >>>>>>>>>>>>>
> >>> use for the reference,
> >>>
> >>>>>>>>>>>>> as far as I've been able to determine it is the BEST
> >>>>>>>>>>>>>
> >>> SIMPLE answer that allows High performance.
> >>>
> >>>>>>>>>>>>> Limited by My HP10811 Ref OSC, I'm getting better than
> >>>>>>>>>>>>>
> >>> 1e-12 in 0.1 sec (at 30 Hz Bandwidth)
> >>>
> >>>>>>>>>>>>> Basic modified NIST Block Diag attached:
> >>>>>>>>>>>>> The NIST paper sums it up quite nicely:
> >>>>>>>>>>>>> 'It is not difficult to achieve a sensitivity of a
> >>>>>>>>>>>>>
> >>> part in e14 per Hz resolution
> >>>
> >>>>>>>>>>>>> so one has excellent precision capabilities with
> this system.'
> >>>>>>>>>>>>>
> >>>>>>>>>>>>> This does not address your other question of ADEV vs MDEV,
> >>>>>>>>>>>>> What I've described is just a simple way to get the
> >>>>>>>>>>>>>
> >>> Low cost, GOOD Raw data.
> >>>
> >>>>>>>>>>>>> What you then do with that Data is a different subject.
> >>>>>>>>>>>>>
> >>>>>>>>>>>>> You can run the raw data thru one of the many ADEV
> >>>>>>>>>>>>>
> >>> programs out there, 'Plotter' being my choice.
> >>>
> >>>>>>>>>>>>>
> >>>>>>>>>>>>> Have fun
> >>>>>>>>>>>>> ws
> >>>>>>>>>>>>>
> >>>>>>>>>>>>> *************
> >>>>>>>>>>>>>
> >>>>>>>>>>>>> [time-nuts] ADEV vs MDEV
> >>>>>>>>>>>>> Pete Rawson peterawson at earthlink.net
> >>>>>>>>>>>>> Sat Feb 6 03:59:18 UTC 2010
> >>>>>>>>>>>>>
> >>>>>>>>>>>>> Efforts are underway to develop a low cost DMTD
> apparatus with
> >>>>>>>>>>>>> demonstrated stability measurements of 1E-13 in 1s. It
> >>>>>>>>>>>>>
> >>> seems that
> >>>
> >>>>>>>>>>>>> existing TI counters can reach this goal in 10s.
> >>>>>>>>>>>>>
> >>> (using MDEV estimate
> >>>
> >>>>>>>>>>>>> or 100+s. using ADEV estimate). The question is; does
> >>>>>>>>>>>>>
> >>> the MDEV tool
> >>>
> >>>>>>>>>>>>> provide an appropriate measure of stability in this
> >>>>>>>>>>>>>
> >>> time range, or is
> >>>
> >>>>>>>>>>>>> the ADEV estimate a more correct answer?
> >>>>>>>>>>>>>
> >>>>>>>>>>>>> The TI performance I'm referring to is the 20-25 ps,
> >>>>>>>>>>>>>
> >>> single shot TI,
> >>>
> >>>>>>>>>>>>> typical for theHP5370A/B, the SR620 or the CNT81/91. I
> >>>>>>>>>>>>>
> >>> have data
> >>>
> >>>>>>>>>>>>> from my CNT81showing MDEV< 1E-13 in 10s. and I
> believe the
> >>>>>>>>>>>>> other counters behave similarly.
> >>>>>>>>>>>>>
> >>>>>>>>>>>>> I would appreciate any comments or observations on
> this topic.
> >>>>>>>>>>>>> My motivation is to discover the simplest scheme for making
> >>>>>>>>>>>>> stability measurements at this performance level;
> this is NOT
> >>>>>>>>>>>>> even close to the state-of-the-art, but can still be useful.
> >>>>>>>>>>>>>
> >>>>>>>>>>>>> Pete Rawson
> >>>>>>>>>>>>>
> >>>>>>>>>>>>>
> >>>>>>>>>>>>>
> >
> >
> >
> > _______________________________________________
> > time-nuts mailing list -- time-nuts@febo.com
> > To unsubscribe, go to
https://www.febo.com/cgi-bin/mailman/listinfo/time-nuts
> and follow the instructions there.
>
_______________________________________________
time-nuts mailing list -- time-nuts@febo.com
To unsubscribe, go to
https://www.febo.com/cgi-bin/mailman/listinfo/time-nuts
and follow the instructions there.
BC
Bob Camp
Sun, Feb 7, 2010 4:24 AM
Hi
The 5 MHz stuff was down at or below 1.5x10^-12 at one second by our measure. Others measured them a bit lower than that. We didn't do 100% testing at 10 sec, so I don't have a lot of data there. The ones 55 Hz higher often came at or above 4x10^-12.
Bob
On Feb 6, 2010, at 9:40 PM, Bruce Griffiths wrote:
As a matter of interest just how bad were those OCXOs?
e.g. what was the ballpark ADEV for 1s, 10s etc.?
Bruce
Bob Camp wrote:
Hi
Occasionally you also come across 5.000055 MHz OCXO's that have 5 MHz crystals in them. Then you discover just how much short term stability can degrade when they move the crystal 55 Hz. Same vendor crystal, same crystal spec., same oscillator circuit, not even close on short term stability....
Bob
On Feb 6, 2010, at 9:02 PM, Bruce Griffiths wrote:
JPL resorted to using a commercial synthesiser set for an offset of 123Hz (to minimise spurs and other artifacts) in their 100MHz N channel mixer system.
Occasionally one comes across 5.000055MHz OCXOs that use 10.000110MHz crystals internally.
The resultant 55Hz (with 5MHz source) or 110Hz (with 10MHz source) beat frequencies are lie between the hamonincs of either 50Hz or 60Hz line frequencies.
Bruce
Bob Camp wrote:
Hi
Any approach that includes building a low noise synthesizer is opening up a whole new set of issues. I would much prefer to do my building at audio. Audio parts are cheap, and performance is usually a lot easier to check than at RF.
Bob
On Feb 6, 2010, at 8:30 PM, Bruce Griffiths wrote:
Which just leaves the minor problem of the offset oscillator.
One option is to use a phase truncation spur free output frequency from a DDS.
If one is using the Costas receiver approach the beat frequency need not be a nice round number like 1.0000KHz.
Another method is to use a crystal whose frequency is offset a few kHz from 10MHz.
Yet another is the classical method of dividing 10MHz by 100 and subtracting (using an LSB mixer) the resultant 100KHz from 10MHz to produce 9.9MHz, then divide the 9.9MHz signal by 100 and add (using a USB mixer) the resultant 99kHz signal to the 9.99Mhz signal to produce a 9.999MHz output.
Bruce
John Miles wrote:
A sound-card back end has always seemed like a pretty reasonable approach to
me, if you're inclined to go the DMTD route. I wouldn't send a 'baseband'
signal to the sound card, though -- I'd upconvert it to a few kHz to get
away from the numerous bad things that sound cards do near DC.
-- john, KE5FX
Hi
My main concern with the low frequency pole in the sound card is
the quality of the R/C used. You can certainly model what ever
you have. If they used an aluminum electrolytic for the "C" it
may not be the same next time you check it ....
On a 10 Hz system, a 1 Hz pole is probably not an issue. It might
get in the way with a 1 Hz beat note.
Another thing I have only seen in passing: "Sigma Delta's have
poor low frequency noise characteristics". I haven't dug into it
to see if that's really true or not. If you buy your own ADC's,
you certainly would not be restricted to a Sigma Delta.
Even with a cheap pre-built FPGA board, you could look into
higher sample rates than a conventional sound card. You would
drop back to 16 bits, but it might be worth it.
Bob
Hi
The 5 MHz stuff was down at or below 1.5x10^-12 at one second by our measure. Others measured them a bit lower than that. We didn't do 100% testing at 10 sec, so I don't have a lot of data there. The ones 55 Hz higher often came at or above 4x10^-12.
Bob
On Feb 6, 2010, at 9:40 PM, Bruce Griffiths wrote:
> As a matter of interest just how bad were those OCXOs?
>
> e.g. what was the ballpark ADEV for 1s, 10s etc.?
>
> Bruce
>
> Bob Camp wrote:
>> Hi
>>
>> Occasionally you also come across 5.000055 MHz OCXO's that have 5 MHz crystals in them. Then you discover just how much short term stability can degrade when they move the crystal 55 Hz. Same vendor crystal, same crystal spec., same oscillator circuit, not even close on short term stability....
>>
>> Bob
>>
>>
>> On Feb 6, 2010, at 9:02 PM, Bruce Griffiths wrote:
>>
>>
>>> JPL resorted to using a commercial synthesiser set for an offset of 123Hz (to minimise spurs and other artifacts) in their 100MHz N channel mixer system.
>>>
>>> Occasionally one comes across 5.000055MHz OCXOs that use 10.000110MHz crystals internally.
>>> The resultant 55Hz (with 5MHz source) or 110Hz (with 10MHz source) beat frequencies are lie between the hamonincs of either 50Hz or 60Hz line frequencies.
>>>
>>> Bruce
>>>
>>> Bob Camp wrote:
>>>
>>>> Hi
>>>>
>>>> Any approach that includes building a low noise synthesizer is opening up a whole new set of issues. I would much prefer to do my building at audio. Audio parts are cheap, and performance is usually a lot easier to check than at RF.
>>>>
>>>> Bob
>>>>
>>>>
>>>> On Feb 6, 2010, at 8:30 PM, Bruce Griffiths wrote:
>>>>
>>>>
>>>>
>>>>> Which just leaves the minor problem of the offset oscillator.
>>>>>
>>>>> One option is to use a phase truncation spur free output frequency from a DDS.
>>>>> If one is using the Costas receiver approach the beat frequency need not be a nice round number like 1.0000KHz.
>>>>>
>>>>> Another method is to use a crystal whose frequency is offset a few kHz from 10MHz.
>>>>>
>>>>> Yet another is the classical method of dividing 10MHz by 100 and subtracting (using an LSB mixer) the resultant 100KHz from 10MHz to produce 9.9MHz, then divide the 9.9MHz signal by 100 and add (using a USB mixer) the resultant 99kHz signal to the 9.99Mhz signal to produce a 9.999MHz output.
>>>>>
>>>>> Bruce
>>>>>
>>>>> John Miles wrote:
>>>>>
>>>>>
>>>>>> A sound-card back end has always seemed like a pretty reasonable approach to
>>>>>> me, if you're inclined to go the DMTD route. I wouldn't send a 'baseband'
>>>>>> signal to the sound card, though -- I'd upconvert it to a few kHz to get
>>>>>> away from the numerous bad things that sound cards do near DC.
>>>>>>
>>>>>> -- john, KE5FX
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>> Hi
>>>>>>>
>>>>>>> My main concern with the low frequency pole in the sound card is
>>>>>>> the quality of the R/C used. You can certainly model what ever
>>>>>>> you have. If they used an aluminum electrolytic for the "C" it
>>>>>>> may not be the same next time you check it ....
>>>>>>>
>>>>>>> On a 10 Hz system, a 1 Hz pole is probably not an issue. It might
>>>>>>> get in the way with a 1 Hz beat note.
>>>>>>>
>>>>>>> Another thing I have only seen in passing: "Sigma Delta's have
>>>>>>> poor low frequency noise characteristics". I haven't dug into it
>>>>>>> to see if that's really true or not. If you buy your own ADC's,
>>>>>>> you certainly would not be restricted to a Sigma Delta.
>>>>>>>
>>>>>>> Even with a cheap pre-built FPGA board, you could look into
>>>>>>> higher sample rates than a conventional sound card. You would
>>>>>>> drop back to 16 bits, but it might be worth it.
>>>>>>>
>>>>>>> Bob
>>>>>>>
>>>>>>>
>>>
>>>
>
>
>
> _______________________________________________
> time-nuts mailing list -- time-nuts@febo.com
> To unsubscribe, go to https://www.febo.com/cgi-bin/mailman/listinfo/time-nuts
> and follow the instructions there.
>
BC
Bob Camp
Sun, Feb 7, 2010 4:31 AM
Hi
If it's a DMTD the offset oscillator is less of an issue than either of the oscillators being tested.
If it's a heterodyne beat note system, then there are only two oscillators. They both contribute equally.
A DMTD with the reference synthesized off of one of the "DUT" inputs looks a lot like a heterodyne if you do everything right.
The constraints on the LO used to drive the sound card or counter are somewhat relaxed since they are on the other side of the down conversion. That's true weather you are talking about the heterodyne or DMTD.
Bob
On Feb 6, 2010, at 10:28 PM, John Miles wrote:
Any noise or drift in the "2nd LO", so to speak, would be common-mode
between the two channels. It shouldn't be all that critical.
-- john, KE5FX
-----Original Message-----
From: time-nuts-bounces@febo.com [mailto:time-nuts-bounces@febo.com]On
Behalf Of Bob Camp
Sent: Saturday, February 06, 2010 5:52 PM
To: Discussion of precise time and frequency measurement
Subject: Re: [time-nuts] ADEV vs MDEV - using sound card
Hi
Any approach that includes building a low noise synthesizer is
opening up a whole new set of issues. I would much prefer to do
my building at audio. Audio parts are cheap, and performance is
usually a lot easier to check than at RF.
Bob
On Feb 6, 2010, at 8:30 PM, Bruce Griffiths wrote:
Which just leaves the minor problem of the offset oscillator.
One option is to use a phase truncation spur free output
If one is using the Costas receiver approach the beat frequency
need not be a nice round number like 1.0000KHz.
Another method is to use a crystal whose frequency is offset a
Yet another is the classical method of dividing 10MHz by 100
and subtracting (using an LSB mixer) the resultant 100KHz from
10MHz to produce 9.9MHz, then divide the 9.9MHz signal by 100 and
add (using a USB mixer) the resultant 99kHz signal to the 9.99Mhz
signal to produce a 9.999MHz output.
A sound-card back end has always seemed like a pretty
me, if you're inclined to go the DMTD route. I wouldn't send
signal to the sound card, though -- I'd upconvert it to a few
away from the numerous bad things that sound cards do near DC.
-- john, KE5FX
Hi
My main concern with the low frequency pole in the sound card is
the quality of the R/C used. You can certainly model what ever
you have. If they used an aluminum electrolytic for the "C" it
may not be the same next time you check it ....
On a 10 Hz system, a 1 Hz pole is probably not an issue. It might
get in the way with a 1 Hz beat note.
Another thing I have only seen in passing: "Sigma Delta's have
poor low frequency noise characteristics". I haven't dug into it
to see if that's really true or not. If you buy your own ADC's,
you certainly would not be restricted to a Sigma Delta.
Even with a cheap pre-built FPGA board, you could look into
higher sample rates than a conventional sound card. You would
drop back to 16 bits, but it might be worth it.
Bob
On Feb 6, 2010, at 6:46 PM, Bruce Griffiths wrote:
Even better is to toss out the mixers and sample the RF signals
However suitable ADCs cost $US100 or more each.
To which one has to add an FPGA and an interface to a PC with
sufficient throughput to handle the down converted I + Q samples.
Hi
You probably could put a couple of cheap DAC's
(ADCs are preferable as it avoids having to implement the
conversion logic plus comparator required when using a DAC.)
on a board with a FPGA and reduce the data on the fly. I'd
guess that would be be in the same $100 range as a half way
decent sound card. Clock the DAC's off of a 10 MHz reference and
eliminate the cal issue.
If you are down around 10 Hz or worse yet 1 Hz, the AC
coupling of the sound card will get in the way, even with a
bandpass approach. You really don't know what they may have in
there at the low end. Build it yourself and that stuff's not an issue.
My sound card has a 1Hz cutoff RC high pass input filter plus
an internal high pass digital filter.
Its not too difficult to measure the sound card frequency
response using a white noise source for example.
On Feb 6, 2010, at 6:12 PM, Bruce Griffiths wrote:
If one has a high end sound card then it could be used to
implement the bandpass filter and replace the zero crossing detector.
It may be necessary to insert a pilot tone to calibrate the
sound card sampling clock frequency.
A noise floor of about 1E-13/Tau should be achievable.
This simplifies the DMTD system by replacing the zero
crossing detector with a low gain linear preamp.
If one analyses the resultant data off line then one can also
try out different techniques such as a Costas receiver rather
than a simple bandpass filter plus zero crossing detector.
However 1000 seconds of data for 2 channels of 24 bit samples
at 192KSPS will result in a file with a size of at least 1.15GB.
Bruce
Bruce Griffiths wrote:
If one were to use a bandpass filter with a Q of 10 to
filter the beat frequency output of the mixer, then if the input
frequency is 10MHz and the filter component tempco is 100ppm/C
then the resultant phase shift tempco is about 16ps/C referred to
the mixer input frequency.
This phase shift tempco is certainly low enough not to have
significant impact when measuring the frequency stability of a
typical 10811A if the temperature fluctuations are kept small
enough during the run.
The effect of using a bandpass filter with too narrow a
bandwidth is to artificially reduce ADEV for small Tau, so it may
be prudent to use a higher beat frequency that 1Hz or even 10Hz
and not calculate ADEV for Tau less than say 10(??) times the
beat frequency period. A trade off between this and the effect of
aliasing is required.
Hi
With most 10811 range oscillators the impact of a simple
bandpass filter is low enough to not be a major issue. That's for
normal lab temperatures with the circuitry in a conventional die
cast box. No guarantee if you open the window and let the fresh
air blow in during the run.
That's true with a heterodyne. I can see no obvious reason
it would not be true on DMTD.
Bob
On Feb 6, 2010, at 5:12 PM, Bruce Griffiths wrote:
The only major issue with DMTD systems is that they
undersample the phase fluctuations and hence are subject to
aliasing effects.
The low pass filter has to have a bandwidth of the same
order as the beat frequency or the beat frequency signal will be
significantly attenuated.
Since the phase is only sampled once per beat frequency
period the phase fluctuations are undersampled.
Various attempts to use both zero crossings have not been
In principle if one can overcome the increased phase shift
tempco associated with a bandpass filter, using a bandpass filter
can in principle ensure that the phase fluctuations are oversampled.
Hi
A straight heterodyne system will get you to the floor of
most 10811's with a very simple (2 stage) limiter. As with the
DMTD, the counter requirements aren't really all that severe.
Bob
On Feb 6, 2010, at 4:24 PM, WarrenS wrote:
"It's possible / likely for injection lock ... to be a
Something I certainly worried about and tested for.
What I found (for MY case) is that injection lock is NOT
The reason being is that unlike most other ways, where
the two OSC have to be completely independent,
The tight loop approach forces the Two Osc to "Lock with
something like 60 + db gain,
so a little stray -80db injection lock coupling that
would very much limit other systems has
no measurable effect at e-13. Just one of the neat
little side effects that make the tight loop approach so simple.
"then a part in 10^14 is going to be at the 100 of
For that example, just need to put a simple discrete 100
in-between the control voltage and the EFC and now you
have a nice workable 10uv.
BUT the bigger point is, probable not needed, cause you
are NOT going to do any better than the stability of the OSC with
a grounded shorted EFC input.
as you said and I agree is so true:
"There is no perfect way to do any of this, only a lot
of compromises ... you need to watch out for".
But you did not offer any easier way to do it, which is
what the original request was for and my answer addressed.
This is the cheapest easiest way BY FAR to get high
performance, at low tau, ADEV numbers that I've seen.
ws
----- Original Message ----- From: "Bob Camp"lists@cq.nu
To: "Discussion of precise time and frequency
Sent: Saturday, February 06, 2010 12:09 PM
Subject: Re: [time-nuts] ADEV vs MDEV
Hi
It's possible / likely to injection lock with the tight
loop approach and get data that's much better than reality. A lot
depends on the specific oscillators under test and the buffers
(if any) between the oscillators and mixer.
If your OCVCXO has a tuning slope of 0.1 ppm / volt
then a part in 10^14 is going to be at the 100 of nanovolts
level. Certainly not impossible, but it does present it's own set
of issues. Lab gear to do it is available, but not all that
common. DC offsets and their temperature coefficients along with
thermocouple effects could make things exciting.
There is no perfect way to do any of this, only a lot
of compromises here or there. Each approach has stuff you need to
watch out for.
Bob
From: "WarrenS"warrensjmail-one@yahoo.com
Sent: Saturday, February 06, 2010 2:19 PM
To: "Discussion of precise time and frequency
Subject: Re: [time-nuts] ADEV vs MDEV
I would appreciate any comments or observations on
the topic of apparatus with demonstrated stability measurements.
My motivation is to discover the SIMPLEST scheme for
making stability measurements at the 1E-13 in 1s performance level.
If you accept that the measurement is going to limited
for Low COST and SIMPLE, with the ability to measure
Can't beat a simple analog version of NIST's "Tight
Phase-Lock Loop Method of measuring Freq stability".
By replacing the "Voltage to freq converter, Freq
counter& Printer with a Radio shack type PC data logging DVM,
It can be up and running from scratch in under an Hr,
with no high end test equipment needed.
If you want performance that exceeds the best of most
DMTD at low Tau it takes a little more work
and a higher speed oversampling ADC data logger and a
I must add this is not a popular solution (Or a
IF you know analog and have a GOOD osc with EFC to
as far as I've been able to determine it is the BEST
SIMPLE answer that allows High performance.
Limited by My HP10811 Ref OSC, I'm getting better than
1e-12 in 0.1 sec (at 30 Hz Bandwidth)
Basic modified NIST Block Diag attached:
The NIST paper sums it up quite nicely:
'It is not difficult to achieve a sensitivity of a
part in e14 per Hz resolution
so one has excellent precision capabilities with
This does not address your other question of ADEV vs MDEV,
What I've described is just a simple way to get the
What you then do with that Data is a different subject.
You can run the raw data thru one of the many ADEV
programs out there, 'Plotter' being my choice.
Have fun
ws
[time-nuts] ADEV vs MDEV
Pete Rawson peterawson at earthlink.net
Sat Feb 6 03:59:18 UTC 2010
Efforts are underway to develop a low cost DMTD
demonstrated stability measurements of 1E-13 in 1s. It
existing TI counters can reach this goal in 10s.
or 100+s. using ADEV estimate). The question is; does
provide an appropriate measure of stability in this
the ADEV estimate a more correct answer?
The TI performance I'm referring to is the 20-25 ps,
typical for theHP5370A/B, the SR620 or the CNT81/91. I
from my CNT81showing MDEV< 1E-13 in 10s. and I
other counters behave similarly.
I would appreciate any comments or observations on
My motivation is to discover the simplest scheme for making
stability measurements at this performance level;
even close to the state-of-the-art, but can still be useful.
Pete Rawson
and follow the instructions there.
Hi
If it's a DMTD the offset oscillator is less of an issue than either of the oscillators being tested.
If it's a heterodyne beat note system, then there are only two oscillators. They both contribute equally.
A DMTD with the reference synthesized off of one of the "DUT" inputs looks a lot like a heterodyne if you do everything right.
The constraints on the LO used to drive the sound card or counter are somewhat relaxed since they are on the other side of the down conversion. That's true weather you are talking about the heterodyne or DMTD.
Bob
On Feb 6, 2010, at 10:28 PM, John Miles wrote:
> Any noise or drift in the "2nd LO", so to speak, would be common-mode
> between the two channels. It shouldn't be all that critical.
>
> -- john, KE5FX
>
>> -----Original Message-----
>> From: time-nuts-bounces@febo.com [mailto:time-nuts-bounces@febo.com]On
>> Behalf Of Bob Camp
>> Sent: Saturday, February 06, 2010 5:52 PM
>> To: Discussion of precise time and frequency measurement
>> Subject: Re: [time-nuts] ADEV vs MDEV - using sound card
>>
>>
>> Hi
>>
>> Any approach that includes building a low noise synthesizer is
>> opening up a whole new set of issues. I would much prefer to do
>> my building at audio. Audio parts are cheap, and performance is
>> usually a lot easier to check than at RF.
>>
>> Bob
>>
>>
>> On Feb 6, 2010, at 8:30 PM, Bruce Griffiths wrote:
>>
>>> Which just leaves the minor problem of the offset oscillator.
>>>
>>> One option is to use a phase truncation spur free output
>> frequency from a DDS.
>>> If one is using the Costas receiver approach the beat frequency
>> need not be a nice round number like 1.0000KHz.
>>>
>>> Another method is to use a crystal whose frequency is offset a
>> few kHz from 10MHz.
>>>
>>> Yet another is the classical method of dividing 10MHz by 100
>> and subtracting (using an LSB mixer) the resultant 100KHz from
>> 10MHz to produce 9.9MHz, then divide the 9.9MHz signal by 100 and
>> add (using a USB mixer) the resultant 99kHz signal to the 9.99Mhz
>> signal to produce a 9.999MHz output.
>>>
>>> Bruce
>>>
>>> John Miles wrote:
>>>> A sound-card back end has always seemed like a pretty
>> reasonable approach to
>>>> me, if you're inclined to go the DMTD route. I wouldn't send
>> a 'baseband'
>>>> signal to the sound card, though -- I'd upconvert it to a few
>> kHz to get
>>>> away from the numerous bad things that sound cards do near DC.
>>>>
>>>> -- john, KE5FX
>>>>
>>>>
>>>>
>>>>> Hi
>>>>>
>>>>> My main concern with the low frequency pole in the sound card is
>>>>> the quality of the R/C used. You can certainly model what ever
>>>>> you have. If they used an aluminum electrolytic for the "C" it
>>>>> may not be the same next time you check it ....
>>>>>
>>>>> On a 10 Hz system, a 1 Hz pole is probably not an issue. It might
>>>>> get in the way with a 1 Hz beat note.
>>>>>
>>>>> Another thing I have only seen in passing: "Sigma Delta's have
>>>>> poor low frequency noise characteristics". I haven't dug into it
>>>>> to see if that's really true or not. If you buy your own ADC's,
>>>>> you certainly would not be restricted to a Sigma Delta.
>>>>>
>>>>> Even with a cheap pre-built FPGA board, you could look into
>>>>> higher sample rates than a conventional sound card. You would
>>>>> drop back to 16 bits, but it might be worth it.
>>>>>
>>>>> Bob
>>>>>
>>>>>
>>>>> On Feb 6, 2010, at 6:46 PM, Bruce Griffiths wrote:
>>>>>
>>>>>
>>>>>> Even better is to toss out the mixers and sample the RF signals
>>>>>>
>>>>> directly.
>>>>>
>>>>>> However suitable ADCs cost $US100 or more each.
>>>>>> To which one has to add an FPGA and an interface to a PC with
>>>>>>
>>>>> sufficient throughput to handle the down converted I + Q samples.
>>>>>
>>>>>> Bob Camp wrote:
>>>>>>
>>>>>>> Hi
>>>>>>>
>>>>>>> You probably could put a couple of cheap DAC's
>>>>>>>
>>>>>> (ADCs are preferable as it avoids having to implement the
>>>>>>
>>>>> conversion logic plus comparator required when using a DAC.)
>>>>>
>>>>>>
>>>>>>> on a board with a FPGA and reduce the data on the fly. I'd
>>>>>>>
>>>>> guess that would be be in the same $100 range as a half way
>>>>> decent sound card. Clock the DAC's off of a 10 MHz reference and
>>>>> eliminate the cal issue.
>>>>>
>>>>>>> If you are down around 10 Hz or worse yet 1 Hz, the AC
>>>>>>>
>>>>> coupling of the sound card will get in the way, even with a
>>>>> bandpass approach. You really don't know what they may have in
>>>>> there at the low end. Build it yourself and that stuff's not an issue.
>>>>>
>>>>>>> Bob
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>> My sound card has a 1Hz cutoff RC high pass input filter plus
>>>>>>
>>>>> an internal high pass digital filter.
>>>>>
>>>>>> Its not too difficult to measure the sound card frequency
>>>>>>
>>>>> response using a white noise source for example.
>>>>>
>>>>>> Bruce
>>>>>>
>>>>>>> On Feb 6, 2010, at 6:12 PM, Bruce Griffiths wrote:
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>> If one has a high end sound card then it could be used to
>>>>>>>>
>>>>> implement the bandpass filter and replace the zero crossing detector.
>>>>>
>>>>>>>> It may be necessary to insert a pilot tone to calibrate the
>>>>>>>>
>>>>> sound card sampling clock frequency.
>>>>>
>>>>>>>> A noise floor of about 1E-13/Tau should be achievable.
>>>>>>>> This simplifies the DMTD system by replacing the zero
>>>>>>>>
>>>>> crossing detector with a low gain linear preamp.
>>>>>
>>>>>>>> If one analyses the resultant data off line then one can also
>>>>>>>>
>>>>> try out different techniques such as a Costas receiver rather
>>>>> than a simple bandpass filter plus zero crossing detector.
>>>>>
>>>>>>>> However 1000 seconds of data for 2 channels of 24 bit samples
>>>>>>>>
>>>>> at 192KSPS will result in a file with a size of at least 1.15GB.
>>>>>
>>>>>>>> Bruce
>>>>>>>>
>>>>>>>>
>>>>>>>> Bruce Griffiths wrote:
>>>>>>>>
>>>>>>>>
>>>>>>>>> If one were to use a bandpass filter with a Q of 10 to
>>>>>>>>>
>>>>> filter the beat frequency output of the mixer, then if the input
>>>>> frequency is 10MHz and the filter component tempco is 100ppm/C
>>>>> then the resultant phase shift tempco is about 16ps/C referred to
>>>>> the mixer input frequency.
>>>>>
>>>>>>>>> This phase shift tempco is certainly low enough not to have
>>>>>>>>>
>>>>> significant impact when measuring the frequency stability of a
>>>>> typical 10811A if the temperature fluctuations are kept small
>>>>> enough during the run.
>>>>>
>>>>>>>>> The effect of using a bandpass filter with too narrow a
>>>>>>>>>
>>>>> bandwidth is to artificially reduce ADEV for small Tau, so it may
>>>>> be prudent to use a higher beat frequency that 1Hz or even 10Hz
>>>>> and not calculate ADEV for Tau less than say 10(??) times the
>>>>> beat frequency period. A trade off between this and the effect of
>>>>> aliasing is required.
>>>>>
>>>>>>>>> Bruce
>>>>>>>>>
>>>>>>>>> Bob Camp wrote:
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>> Hi
>>>>>>>>>>
>>>>>>>>>> With most 10811 range oscillators the impact of a simple
>>>>>>>>>>
>>>>> bandpass filter is low enough to not be a major issue. That's for
>>>>> normal lab temperatures with the circuitry in a conventional die
>>>>> cast box. No guarantee if you open the window and let the fresh
>>>>> air blow in during the run.
>>>>>
>>>>>>>>>> That's true with a heterodyne. I can see no obvious reason
>>>>>>>>>>
>>>>> it would not be true on DMTD.
>>>>>
>>>>>>>>>> Bob
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> On Feb 6, 2010, at 5:12 PM, Bruce Griffiths wrote:
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>> The only major issue with DMTD systems is that they
>>>>>>>>>>>
>>>>> undersample the phase fluctuations and hence are subject to
>>>>> aliasing effects.
>>>>>
>>>>>>>>>>> The low pass filter has to have a bandwidth of the same
>>>>>>>>>>>
>>>>> order as the beat frequency or the beat frequency signal will be
>>>>> significantly attenuated.
>>>>>
>>>>>>>>>>> Since the phase is only sampled once per beat frequency
>>>>>>>>>>>
>>>>> period the phase fluctuations are undersampled.
>>>>>
>>>>>>>>>>> Various attempts to use both zero crossings have not been
>>>>>>>>>>>
>>>>> successful.
>>>>>
>>>>>>>>>>> In principle if one can overcome the increased phase shift
>>>>>>>>>>>
>>>>> tempco associated with a bandpass filter, using a bandpass filter
>>>>> can in principle ensure that the phase fluctuations are oversampled.
>>>>>
>>>>>>>>>>>
>>>>>>>>>>> Bruce
>>>>>>>>>>>
>>>>>>>>>>> Bob Camp wrote:
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>> Hi
>>>>>>>>>>>>
>>>>>>>>>>>> A straight heterodyne system will get you to the floor of
>>>>>>>>>>>>
>>>>> most 10811's with a very simple (2 stage) limiter. As with the
>>>>> DMTD, the counter requirements aren't really all that severe.
>>>>>
>>>>>>>>>>>> Bob
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> On Feb 6, 2010, at 4:24 PM, WarrenS wrote:
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>> "It's possible / likely for injection lock ... to be a
>>>>>>>>>>>>>>
>>>>> problem ..."
>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>> Something I certainly worried about and tested for.
>>>>>>>>>>>>> What I found (for MY case) is that injection lock is NOT
>>>>>>>>>>>>>
>>>>> a problem.
>>>>>
>>>>>>>>>>>>> The reason being is that unlike most other ways, where
>>>>>>>>>>>>>
>>>>> the two OSC have to be completely independent,
>>>>>
>>>>>>>>>>>>> The tight loop approach forces the Two Osc to "Lock with
>>>>>>>>>>>>>
>>>>> something like 60 + db gain,
>>>>>
>>>>>>>>>>>>> so a little stray -80db injection lock coupling that
>>>>>>>>>>>>>
>>>>> would very much limit other systems has
>>>>>
>>>>>>>>>>>>> no measurable effect at e-13. Just one of the neat
>>>>>>>>>>>>>
>>>>> little side effects that make the tight loop approach so simple.
>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>> "then a part in 10^14 is going to be at the 100 of
>>>>>>>>>>>>>>
>>>>> nanovolts level."
>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>> For that example, just need to put a simple discrete 100
>>>>>>>>>>>>>
>>>>> to 1 resistor divider
>>>>>
>>>>>>>>>>>>> in-between the control voltage and the EFC and now you
>>>>>>>>>>>>>
>>>>> have a nice workable 10uv.
>>>>>
>>>>>>>>>>>>> BUT the bigger point is, probable not needed, cause you
>>>>>>>>>>>>>
>>>>> are NOT going to do any better than the stability of the OSC with
>>>>> a grounded shorted EFC input.
>>>>>
>>>>>>>>>>>>> as you said and I agree is so true:
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>> "There is no perfect way to do any of this, only a lot
>>>>>>>>>>>>>>
>>>>> of compromises ... you need to watch out for".
>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>> But you did not offer any easier way to do it, which is
>>>>>>>>>>>>>
>>>>> what the original request was for and my answer addressed.
>>>>>
>>>>>>>>>>>>> This is the cheapest easiest way BY FAR to get high
>>>>>>>>>>>>>
>>>>> performance, at low tau, ADEV numbers that I've seen.
>>>>>
>>>>>>>>>>>>> ws
>>>>>>>>>>>>> ***************
>>>>>>>>>>>>>
>>>>>>>>>>>>> ----- Original Message ----- From: "Bob Camp"<lists@cq.nu>
>>>>>>>>>>>>> To: "Discussion of precise time and frequency
>>>>>>>>>>>>>
>>>>> measurement"<time-nuts@febo.com>
>>>>>
>>>>>>>>>>>>> Sent: Saturday, February 06, 2010 12:09 PM
>>>>>>>>>>>>> Subject: Re: [time-nuts] ADEV vs MDEV
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>> Hi
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> It's possible / likely to injection lock with the tight
>>>>>>>>>>>>>>
>>>>> loop approach and get data that's much better than reality. A lot
>>>>> depends on the specific oscillators under test and the buffers
>>>>> (if any) between the oscillators and mixer.
>>>>>
>>>>>>>>>>>>>> If your OCVCXO has a tuning slope of 0.1 ppm / volt
>>>>>>>>>>>>>>
>>>>> then a part in 10^14 is going to be at the 100 of nanovolts
>>>>> level. Certainly not impossible, but it does present it's own set
>>>>> of issues. Lab gear to do it is available, but not all that
>>>>> common. DC offsets and their temperature coefficients along with
>>>>> thermocouple effects could make things exciting.
>>>>>
>>>>>>>>>>>>>> There is no perfect way to do any of this, only a lot
>>>>>>>>>>>>>>
>>>>> of compromises here or there. Each approach has stuff you need to
>>>>> watch out for.
>>>>>
>>>>>>>>>>>>>> Bob
>>>>>>>>>>>>>>
>>>>>>>>>>>>>> --------------------------------------------------
>>>>>>>>>>>>>> From: "WarrenS"<warrensjmail-one@yahoo.com>
>>>>>>>>>>>>>> Sent: Saturday, February 06, 2010 2:19 PM
>>>>>>>>>>>>>> To: "Discussion of precise time and frequency
>>>>>>>>>>>>>>
>>>>> measurement"<time-nuts@febo.com>
>>>>>
>>>>>>>>>>>>>> Subject: Re: [time-nuts] ADEV vs MDEV
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Peat said:
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>> I would appreciate any comments or observations on
>>>>>>>>>>>>>>>>
>>>>> the topic of apparatus with demonstrated stability measurements.
>>>>>
>>>>>>>>>>>>>>>> My motivation is to discover the SIMPLEST scheme for
>>>>>>>>>>>>>>>>
>>>>> making stability measurements at the 1E-13 in 1s performance level.
>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> If you accept that the measurement is going to limited
>>>>>>>>>>>>>>>
>>>>> by the Reference Osc,
>>>>>
>>>>>>>>>>>>>>> for Low COST and SIMPLE, with the ability to measure
>>>>>>>>>>>>>>>
>>>>> ADEVs at that level,
>>>>>
>>>>>>>>>>>>>>> Can't beat a simple analog version of NIST's "Tight
>>>>>>>>>>>>>>>
>>>>> Phase-Lock Loop Method of measuring Freq stability".
>>>>>
>>>>>>>>>>>>>>> http://tf.nist.gov/phase/Properties/one.htm#oneone
>> Fig 1.7
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> By replacing the "Voltage to freq converter, Freq
>>>>>>>>>>>>>>>
>>>>> counter& Printer with a Radio shack type PC data logging DVM,
>>>>>
>>>>>>>>>>>>>>> It can be up and running from scratch in under an Hr,
>>>>>>>>>>>>>>>
>>>>> with no high end test equipment needed.
>>>>>
>>>>>>>>>>>>>>> If you want performance that exceeds the best of most
>>>>>>>>>>>>>>>
>>>>> DMTD at low Tau it takes a little more work
>>>>>
>>>>>>>>>>>>>>> and a higher speed oversampling ADC data logger and a
>>>>>>>>>>>>>>>
>>>>> good offset voltage.
>>>>>
>>>>>>>>>>>>>>> I must add this is not a popular solution (Or a
>>>>>>>>>>>>>>>
>>>>> general Purpose one) but
>>>>>
>>>>>>>>>>>>>>> IF you know analog and have a GOOD osc with EFC to
>>>>>>>>>>>>>>>
>>>>> use for the reference,
>>>>>
>>>>>>>>>>>>>>> as far as I've been able to determine it is the BEST
>>>>>>>>>>>>>>>
>>>>> SIMPLE answer that allows High performance.
>>>>>
>>>>>>>>>>>>>>> Limited by My HP10811 Ref OSC, I'm getting better than
>>>>>>>>>>>>>>>
>>>>> 1e-12 in 0.1 sec (at 30 Hz Bandwidth)
>>>>>
>>>>>>>>>>>>>>> Basic modified NIST Block Diag attached:
>>>>>>>>>>>>>>> The NIST paper sums it up quite nicely:
>>>>>>>>>>>>>>> 'It is not difficult to achieve a sensitivity of a
>>>>>>>>>>>>>>>
>>>>> part in e14 per Hz resolution
>>>>>
>>>>>>>>>>>>>>> so one has excellent precision capabilities with
>> this system.'
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> This does not address your other question of ADEV vs MDEV,
>>>>>>>>>>>>>>> What I've described is just a simple way to get the
>>>>>>>>>>>>>>>
>>>>> Low cost, GOOD Raw data.
>>>>>
>>>>>>>>>>>>>>> What you then do with that Data is a different subject.
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> You can run the raw data thru one of the many ADEV
>>>>>>>>>>>>>>>
>>>>> programs out there, 'Plotter' being my choice.
>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Have fun
>>>>>>>>>>>>>>> ws
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> *************
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> [time-nuts] ADEV vs MDEV
>>>>>>>>>>>>>>> Pete Rawson peterawson at earthlink.net
>>>>>>>>>>>>>>> Sat Feb 6 03:59:18 UTC 2010
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Efforts are underway to develop a low cost DMTD
>> apparatus with
>>>>>>>>>>>>>>> demonstrated stability measurements of 1E-13 in 1s. It
>>>>>>>>>>>>>>>
>>>>> seems that
>>>>>
>>>>>>>>>>>>>>> existing TI counters can reach this goal in 10s.
>>>>>>>>>>>>>>>
>>>>> (using MDEV estimate
>>>>>
>>>>>>>>>>>>>>> or 100+s. using ADEV estimate). The question is; does
>>>>>>>>>>>>>>>
>>>>> the MDEV tool
>>>>>
>>>>>>>>>>>>>>> provide an appropriate measure of stability in this
>>>>>>>>>>>>>>>
>>>>> time range, or is
>>>>>
>>>>>>>>>>>>>>> the ADEV estimate a more correct answer?
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> The TI performance I'm referring to is the 20-25 ps,
>>>>>>>>>>>>>>>
>>>>> single shot TI,
>>>>>
>>>>>>>>>>>>>>> typical for theHP5370A/B, the SR620 or the CNT81/91. I
>>>>>>>>>>>>>>>
>>>>> have data
>>>>>
>>>>>>>>>>>>>>> from my CNT81showing MDEV< 1E-13 in 10s. and I
>> believe the
>>>>>>>>>>>>>>> other counters behave similarly.
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> I would appreciate any comments or observations on
>> this topic.
>>>>>>>>>>>>>>> My motivation is to discover the simplest scheme for making
>>>>>>>>>>>>>>> stability measurements at this performance level;
>> this is NOT
>>>>>>>>>>>>>>> even close to the state-of-the-art, but can still be useful.
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>> Pete Rawson
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>>>>>>>>>>>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> time-nuts mailing list -- time-nuts@febo.com
>>> To unsubscribe, go to
> https://www.febo.com/cgi-bin/mailman/listinfo/time-nuts
>> and follow the instructions there.
>>
>
>
> _______________________________________________
> time-nuts mailing list -- time-nuts@febo.com
> To unsubscribe, go to
> https://www.febo.com/cgi-bin/mailman/listinfo/time-nuts
> and follow the instructions there.
>
>
> _______________________________________________
> time-nuts mailing list -- time-nuts@febo.com
> To unsubscribe, go to https://www.febo.com/cgi-bin/mailman/listinfo/time-nuts
> and follow the instructions there.
>
BG
Bruce Griffiths
Sun, Feb 7, 2010 4:49 AM
The OCXOs with 55Hz offset should still be adequate for a DMTD unless
one is comparing hydrogen masers or even lower noise sources.
For a Costas receiver the rejection of the offset oscillator noise can
be somewhat higher than for a DMTD without needing to keep the
corresponding zero crossings of the 2 beat frequencies approximately
aligned.
Bruce
Bob Camp wrote:
Hi
The 5 MHz stuff was down at or below 1.5x10^-12 at one second by our measure. Others measured them a bit lower than that. We didn't do 100% testing at 10 sec, so I don't have a lot of data there. The ones 55 Hz higher often came at or above 4x10^-12.
Bob
On Feb 6, 2010, at 9:40 PM, Bruce Griffiths wrote:
As a matter of interest just how bad were those OCXOs?
e.g. what was the ballpark ADEV for 1s, 10s etc.?
Bruce
Bob Camp wrote:
Hi
Occasionally you also come across 5.000055 MHz OCXO's that have 5 MHz crystals in them. Then you discover just how much short term stability can degrade when they move the crystal 55 Hz. Same vendor crystal, same crystal spec., same oscillator circuit, not even close on short term stability....
Bob
On Feb 6, 2010, at 9:02 PM, Bruce Griffiths wrote:
JPL resorted to using a commercial synthesiser set for an offset of 123Hz (to minimise spurs and other artifacts) in their 100MHz N channel mixer system.
Occasionally one comes across 5.000055MHz OCXOs that use 10.000110MHz crystals internally.
The resultant 55Hz (with 5MHz source) or 110Hz (with 10MHz source) beat frequencies are lie between the hamonincs of either 50Hz or 60Hz line frequencies.
Bruce
Bob Camp wrote:
Hi
Any approach that includes building a low noise synthesizer is opening up a whole new set of issues. I would much prefer to do my building at audio. Audio parts are cheap, and performance is usually a lot easier to check than at RF.
Bob
On Feb 6, 2010, at 8:30 PM, Bruce Griffiths wrote:
Which just leaves the minor problem of the offset oscillator.
One option is to use a phase truncation spur free output frequency from a DDS.
If one is using the Costas receiver approach the beat frequency need not be a nice round number like 1.0000KHz.
Another method is to use a crystal whose frequency is offset a few kHz from 10MHz.
Yet another is the classical method of dividing 10MHz by 100 and subtracting (using an LSB mixer) the resultant 100KHz from 10MHz to produce 9.9MHz, then divide the 9.9MHz signal by 100 and add (using a USB mixer) the resultant 99kHz signal to the 9.99Mhz signal to produce a 9.999MHz output.
Bruce
The OCXOs with 55Hz offset should still be adequate for a DMTD unless
one is comparing hydrogen masers or even lower noise sources.
For a Costas receiver the rejection of the offset oscillator noise can
be somewhat higher than for a DMTD without needing to keep the
corresponding zero crossings of the 2 beat frequencies approximately
aligned.
Bruce
Bob Camp wrote:
> Hi
>
> The 5 MHz stuff was down at or below 1.5x10^-12 at one second by our measure. Others measured them a bit lower than that. We didn't do 100% testing at 10 sec, so I don't have a lot of data there. The ones 55 Hz higher often came at or above 4x10^-12.
>
> Bob
>
>
> On Feb 6, 2010, at 9:40 PM, Bruce Griffiths wrote:
>
>
>> As a matter of interest just how bad were those OCXOs?
>>
>> e.g. what was the ballpark ADEV for 1s, 10s etc.?
>>
>> Bruce
>>
>> Bob Camp wrote:
>>
>>> Hi
>>>
>>> Occasionally you also come across 5.000055 MHz OCXO's that have 5 MHz crystals in them. Then you discover just how much short term stability can degrade when they move the crystal 55 Hz. Same vendor crystal, same crystal spec., same oscillator circuit, not even close on short term stability....
>>>
>>> Bob
>>>
>>>
>>> On Feb 6, 2010, at 9:02 PM, Bruce Griffiths wrote:
>>>
>>>
>>>
>>>> JPL resorted to using a commercial synthesiser set for an offset of 123Hz (to minimise spurs and other artifacts) in their 100MHz N channel mixer system.
>>>>
>>>> Occasionally one comes across 5.000055MHz OCXOs that use 10.000110MHz crystals internally.
>>>> The resultant 55Hz (with 5MHz source) or 110Hz (with 10MHz source) beat frequencies are lie between the hamonincs of either 50Hz or 60Hz line frequencies.
>>>>
>>>> Bruce
>>>>
>>>> Bob Camp wrote:
>>>>
>>>>
>>>>> Hi
>>>>>
>>>>> Any approach that includes building a low noise synthesizer is opening up a whole new set of issues. I would much prefer to do my building at audio. Audio parts are cheap, and performance is usually a lot easier to check than at RF.
>>>>>
>>>>> Bob
>>>>>
>>>>>
>>>>> On Feb 6, 2010, at 8:30 PM, Bruce Griffiths wrote:
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>> Which just leaves the minor problem of the offset oscillator.
>>>>>>
>>>>>> One option is to use a phase truncation spur free output frequency from a DDS.
>>>>>> If one is using the Costas receiver approach the beat frequency need not be a nice round number like 1.0000KHz.
>>>>>>
>>>>>> Another method is to use a crystal whose frequency is offset a few kHz from 10MHz.
>>>>>>
>>>>>> Yet another is the classical method of dividing 10MHz by 100 and subtracting (using an LSB mixer) the resultant 100KHz from 10MHz to produce 9.9MHz, then divide the 9.9MHz signal by 100 and add (using a USB mixer) the resultant 99kHz signal to the 9.99Mhz signal to produce a 9.999MHz output.
>>>>>>
>>>>>> Bruce
>>>>>>
>>>>>>
LJ
Lux, Jim (337C)
Sun, Feb 7, 2010 5:18 AM
On 2/6/10 6:02 PM, "Bruce Griffiths" bruce.griffiths@xtra.co.nz wrote:
JPL resorted to using a commercial synthesiser set for an offset of
123Hz (to minimise spurs and other artifacts) in their 100MHz N channel
mixer system.
And that was chosen after a lot of experimentation to find the "sweet spot". Obviously, there's no special frequency, so you want to choose the one at which your gear works best.
On 2/6/10 6:02 PM, "Bruce Griffiths" <bruce.griffiths@xtra.co.nz> wrote:
JPL resorted to using a commercial synthesiser set for an offset of
123Hz (to minimise spurs and other artifacts) in their 100MHz N channel
mixer system.
And that was chosen after a lot of experimentation to find the "sweet spot". Obviously, there's no special frequency, so you want to choose the one at which your gear works best.
MD
Magnus Danielson
Sun, Feb 7, 2010 8:56 AM
Hi
My main concern with the low frequency pole in the sound card is the quality of the R/C used. You can certainly model what ever you have. If they used an aluminum electrolytic for the "C" it may not be the same next time you check it ....
Do consider to bypass it. This is routinely done both by audio folks and
various other. The cap is there to remove DC offsets which can be
problematic in audio editing. I am quite sure you feel at home with the
soldering iron to do that upgrade.
On a 10 Hz system, a 1 Hz pole is probably not an issue. It might get in the way with a 1 Hz beat note.
Another thing I have only seen in passing: "Sigma Delta's have poor low frequency noise characteristics". I haven't dug into it to see if that's really true or not. If you buy your own ADC's, you certainly would not be restricted to a Sigma Delta.
Strange, most Sigma Delta's I have seen would have the opposite said
about them. It's the upper end that is problematic.
Even with a cheap pre-built FPGA board, you could look into higher sample rates than a conventional sound card. You would drop back to 16 bits, but it might be worth it.
In fact, one of my FPGA demo-board does 3 MS at 14 bit. Crappy for
audio, but maybe good enough for this application.
Cheers,
Magnus
Bob Camp wrote:
> Hi
>
> My main concern with the low frequency pole in the sound card is the quality of the R/C used. You can certainly model what ever you have. If they used an aluminum electrolytic for the "C" it may not be the same next time you check it ....
Do consider to bypass it. This is routinely done both by audio folks and
various other. The cap is there to remove DC offsets which can be
problematic in audio editing. I am quite sure you feel at home with the
soldering iron to do that upgrade.
> On a 10 Hz system, a 1 Hz pole is probably not an issue. It might get in the way with a 1 Hz beat note.
>
> Another thing I have only seen in passing: "Sigma Delta's have poor low frequency noise characteristics". I haven't dug into it to see if that's really true or not. If you buy your own ADC's, you certainly would not be restricted to a Sigma Delta.
Strange, most Sigma Delta's I have seen would have the opposite said
about them. It's the upper end that is problematic.
> Even with a cheap pre-built FPGA board, you could look into higher sample rates than a conventional sound card. You would drop back to 16 bits, but it might be worth it.
In fact, one of my FPGA demo-board does 3 MS at 14 bit. Crappy for
audio, but maybe good enough for this application.
Cheers,
Magnus