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Old 15th May 2014, 02:22 PM   #61
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Emm... no.

They call it 90 degrees, but this never happens in the real world, kinda. It is sought for, to be able to tune the PLL_F close enough to DUT_F to get diff signal within ADC BW.
If it tracks 100% 90 degrees, then mixer outputs zero, therefore there is no PN.

As far as i understand
Look at these PLLs used in measurement equipment. They are quite broadband. We don't need wide tuning range for audio clock frequencies.

I found this doc,
http://dspace.cc.tut.fi/dpub/bitstre...pdf?sequence=3
Page 41 shows actual FFT of beat frequency ,centered at 55kHz.
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Old 15th May 2014, 02:24 PM   #62
1audio is offline 1audio  United States
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There are techniques for measuring with the difference a low audio frequency. I think its called something like DMTD. You need a huge dynamic range to use it however, if the noise floor is in the -150 dB plus level. Its usually used to look at Allen Variance stability.

Here is the standard text on measuring with pretty complete instructions: Techniques for Measuring Phase Noise You can substitute a low noise opamp (AD797, LME49990) for the jFET amp and not lose much performance. Make sure the DBM is properly terminated and low pass filtered.
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Old 15th May 2014, 02:52 PM   #63
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Quote:
Originally Posted by s3tup View Post
They call it 90 degrees, but this never happens in the real world, kinda. It is sought for, to be able to tune the PLL_F close enough to DUT_F to get diff signal within ADC BW. If it tracks 100% 90 degrees, then mixer outputs zero, therefore there is no PN.
Of course the PLL bandwidth is usually very low, below the lowest offset frequency of interest (although there are techniques to compensate the low-frequency effect of the PLL).

With basic phase detectors (e.g. double balanced mixer), the sensitivity to phase noise is highest, and to amplitude noise lowest, if the signals are in quadrature (and vice versa for the in-phase condition). For a free-running condition, the test setup invariably changes from measuring phase noise to amplitude noise. The DMTD technique Demian mentiones probably compensates the resulting effect mathematically (perhaps with the assumption that amplitude noise is much lower than phase noise (which is valid for most oscillators, in particular those with CMOS output)), but for a DIY setup the PLL method seems much easier and safer.

Some more reading:

Phase Noise Measurement: Phase Detectors
http://www.home.agilent.com/agilent/...fr=-35185.0.00

Samuel
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Old 16th May 2014, 06:23 AM   #64
1audio is offline 1audio  United States
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I found the DMTD info. Here is a detailed intro: http://cdn.intechopen.com/pdfs-wm/24305.pdf It more for measuring frequency with very high resolution athan looking at phase noise but phase is preserved so it may allow phase noise measurements with very high resolution. Its effectively a variation on the cross correlation technique. http://www.holzworth.com/Aux_docs/Ho...at_Feb2011.pdf HPw added the cross correlation to his analyzer software HpW-Works so with two reference oscillators you can measure below the noise floor of either. The setup to do this right will get more complex. The minicircuits phase detectors are optimized for this service Phase Detectors - Mini Circuits There are others that may be better but way more money.

At one point I started to lay out a PCB for this but the many different packages for oscillators and the less than clear impact of phase noise once its below the noise of the DAC/ADC suggest more was to be found in the audio output of the device. I may still make one for sorting oscillators. A test socket for the surface mount oscillators is a PITA to resolve. They do not have common footprints.
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Old 27th May 2014, 12:11 PM   #65
lemon is offline lemon  Greece
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I have a couple of days this nice digital generator.

As a digital generator is very well. Here are some captures from an ES9023 based dac (THD, Jitter). They present two captures from either spdif or i2s input, the analogue to digital capture was done by EMU0404 (usb) with a 10K ladder attenuator.
I have attached another Jitter measurement from the same dac but the role of digital generator had a CMedia 6631A usb->i2s card.
As you can see the Altor JKGEN 384_12 has cleared jitter presentation.

Alex, I found the encoder very sensitive (J26), many times it jumps 2-3 click until one. What type is and I will appreciate if there is equivalent more precisely encoder in the market.
Attached Images
File Type: png thd.png (21.9 KB, 206 views)
File Type: png Jittter.png (20.6 KB, 207 views)
File Type: png Jitter.png (23.9 KB, 205 views)

Last edited by lemon; 27th May 2014 at 12:25 PM.
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Old 4th June 2014, 02:08 PM   #66
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4 June 2014: a new firmware released: Downloads - ALTOR AUDIO

Ver 3.4.07:
  • Impulse Test Added
  • CrossSample Test Added
  • Sine Squared Signal Added
  • Frequency Sweep improved
  • Digital Attenuation improved (0..-20dB 1dB step, -20..-90dB 10dB step)
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Old 4th June 2014, 02:21 PM   #67
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if its a regular encoder, you can debounce with an RC filter. better to combine hw debouncing and sw debouncing, but if the code is not open, then hw debounce is all you can do. try a .1 or .01 cap and try various resistors (or even none). search controller forums (like arduino) for examples since this problem comes up frequently.
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Old 5th June 2014, 04:13 AM   #68
1audio is offline 1audio  United States
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Default New waveforms tests

Here are some captures of the new waveforms using different filter options (AK4414 for the filter details short delay vs. normal and fast vs slow rolloff). The impulse tests are with a single sample pulse at full amplitude all at 44.1 KHz and captured at 192 KHz to not lose much detail.

First the impulse test, it will "violate" Nyquist in that it will have frequency content not possible to encode with an appropriate bandlimit. Essentially a recording should never have that encoded. The next waveform, sine squared (or raised cosine http://www.tsc.uc3m.es/~mlazaro/Doce..._cosine-2p.pdf) is a valid waveform with the out of band info reduced (by 60 dB at least) so it can be recorded (which I did to show what the waveform is). Ringing on an "impulse" is not really valid but ringing on the sine squared would be an artifact worth understanding. Finally is the "cross sample" wave which is a burst with a 180 degree phase shift in the middle. This also is not valid with an infinitely fast change but it shows how some different filter solutions handle issues.

Some DACs actually clip on some of threse tests, it because the internal filter does not take into account the peaking of the low pass filter.
Attached Images
File Type: png Impulse SD+ fast 44.1.PNG (16.5 KB, 164 views)
File Type: png Impulse SD+Slow 44.1.PNG (11.4 KB, 161 views)
File Type: png Impulse fast 44.1.PNG (14.9 KB, 32 views)
File Type: png Impulse Slow 44.1.PNG (10.2 KB, 32 views)
File Type: png Sine Squared SD+ fast 44.1.PNG (12.1 KB, 28 views)
File Type: png Sine Squared SD+Slow 44.1.PNG (12.6 KB, 31 views)
File Type: png Cross Sample.PNG (43.9 KB, 34 views)
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Old 5th June 2014, 04:24 AM   #69
RNMarsh is offline RNMarsh  United States
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Quote:
Originally Posted by 1audio View Post
Here are some captures of the new waveforms using different filter options (AK4414 for the filter details short delay vs. normal and fast vs slow rolloff). The impulse tests are with a single sample pulse at full amplitude all at 44.1 KHz and captured at 192 KHz to not lose much detail.

First the impulse test, it will "violate" Nyquist in that it will have frequency content not possible to encode with an appropriate bandlimit. Essentially a recording should never have that encoded. The next waveform, sine squared (or raised cosine http://www.tsc.uc3m.es/~mlazaro/Doce..._cosine-2p.pdf) is a valid waveform with the out of band info reduced (by 60 dB at least) so it can be recorded (which I did to show what the waveform is). Ringing on an "impulse" is not really valid but ringing on the sine squared would be an artifact worth understanding. Finally is the "cross sample" wave which is a burst with a 180 degree phase shift in the middle. This also is not valid with an infinitely fast change but it shows how some different filter solutions handle issues.

Some DACs actually clip on some of threse tests, it because the internal filter does not take into account the peaking of the low pass filter.
Thank you, Demian. Good demo/info.



THX-RNMarsh
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Old 17th June 2014, 01:54 PM   #70
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Hello friends,

I put an "errata sheet" to the website.
Direct link: https://drive.google.com/file/d/0B1T...it?usp=sharing

Actually to all devices, with Main PCB revision 1.3 and 1.3a.

(Before making these modifications, please check - may be your board are already modified!)
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