Geddes on distortion measurements

Status
This old topic is closed. If you want to reopen this topic, contact a moderator using the "Report Post" button.
PMA said:
My measurement limit is like below. I use sine generator and -100dB/dec LP filter to remove generator harmonics. Generator without filter has 0.003% THD, but some harmonics up to 9th.

0dB here is 1Vrms. The card has fine distortion from -10dBVrms lower, it increases between -10dB and 0dB to some 0.0025%. For about -15dB, the distortion is about 0.0009%, that is my limit.

Dynamic range is up to some 135dB (for 64K record, this is important), but then the card distortion increases.

For the amplifier measurement, there was a voltage divider of some 26dB attenuation in front of the measuring card. The card input accepts no more than some 1Vrms and has low distortion up to some 300mVrms.


Pavel,

Can you show us a CCIF 19+20 kHz measurement of this?

Thanks,
Bob
 
mfc said:


Into 4 ohms that is 63.2 millivolts (for .001watts out).
-60 db below that is 63.2 microvolts.

63.2 microvlolts is about the level you might expect to
pickup on your interconnects from a local AM radio station.
Not to mention hum or power supply noise.

So at these levels most stuff would just be junk picked up
by the amp.

Are you saying the ear can percieve correlated junk
(amp distortion products) at these levels and isn't sensitive to
the rest of the uncorrelated junk at these low levels?

Not necessarily a big problem if you employ a spectrum analyzer for viewing the distortion data.

Bob
 
gedlee said:


This is incorrect basically because it doesn't matter. Sound cards operate independently of windows and don't rely on the PC clock.

But if a very large number of time sychronous samples are used, then it is important that the clocks for the input signal and the sampled data signal be close to one another or the averaging will begin to reduce the signal along with the noise. Note that I said "close" to one another - they don't even have to be exact, the amount of difference will set the SN ratio.
Since the soudncards do not interface with the PC and Windows in a syncronous manner, there is not assurance of syncronized data. Be best you can do is aligned data which statistically would be around half a sample off. I do agree that if you have the data aligned, you could average the number of periods sampled to filter out the noise. This is technique is used much when using MLS signals to measure speakers. So now I understand.
 
Bob Cordell said:

Not necessarily a big problem if you employ a spectrum analyzer for viewing the distortion data.
Bob

Hi,

I should have made my point more clearly. No problem
with obtaining distortion spectra at low levels...
what I was more getting at is the ear's ability.

If the ear can discern stuff -60 db below .001 watt out,
can it also discern stuff -60 db bellow .0001 watt out,
and -60 db below .00001 watt, etc.? I would
think the ears range would become compressed at low signal
levels, but gedlee seems to be saying it doesn't.

Maybe the ear's range is even more than -60 db at
these low levels?

Mike
 
mfc said:


I would
think the ears range would become compressed at low signal
levels, but gedlee seems to be saying it doesn't.


It is absolutely true that the masking decreases with signal, not increases. This means that the ear can resolve more harmonic content from distortion at lower levels than at higher ones.

It does not seem to be well understood that the harmonics of a signal are simply a modification of that signals waveform - we call it timbre in other contexts. So most definatley the ear can perceive these waveform modifications more readily than it perceives incorent noise even if that noise is at a level greater than the harmonics. PLEASE stop thinking that the ear acts like an FFT. It doesn't and so these discussions which use FFT analysis are misleading at best and completly wrong at worst.
 
mige0 said:

Earl, do you think I can get measurement below the limits above with your method ?


Theoretically yes. Synchronous time averaging has no theoretical limit of noise floor, only a practical limit due to true sychronization of the waveform. It seems to me that the limiting factor is how closely the input signal matchs the sampling frequency of the output signal.
 
Bob Cordell said:


Pavel,

Can you show us a CCIF 19+20 kHz measurement of this?

Thanks,
Bob

Bob,

with pleasure (13+14kHz), it is about -2dB under limitation, into 8ohm:

/no averaging/
 

Attachments

  • im13+14.gif
    im13+14.gif
    53.8 KB · Views: 666
The previous image was for 8ohm load. But for 4ohm, we can see transition out of class A at some -12Vp and 12Vp. For the usual class AB low iddle current design, this would happen 1-2V from zero. Now we have inflexion points of transfer curve, source of higher harmonics. But still amp mostly works in linear portion of the transfer curve and does not annoy listeners as the low iddle current designs do.
 

Attachments

  • transfer2.gif
    transfer2.gif
    15.6 KB · Views: 627
gedlee said:


Theoretically yes. Synchronous time averaging has no theoretical limit of noise floor, only a practical limit due to true sychronization of the waveform. It seems to me that the limiting factor is how closely the input signal matchs the sampling frequency of the output signal.

I'm late to this thread, but I've done my share of these measurements. With an external A/D one could phase lock a good analog test signal (I've done ~-140dB at 1kHz) to the A/D clock forcing sychronicity. I mean here multiplying up the analog sine wave frequency to an appropriate clock frequency. If you're doing serious research there is no need to limit one's self to using a sound card only.

Also, even though they are maligned for no good reason to me, bridge null techniques remain a powerful tool. You can get a good -120dB null and amplify the residue. Put this into a nice low frequency spectrum analyser and you can get down to the thermal noise limit easily.
 
Hi



scott wurcer said:


Also, even though they are maligned for no good reason to me, bridge null techniques remain a powerful tool. You can get a good -120dB null and amplify the residue. Put this into a nice low frequency spectrum analyser and you can get down to the thermal noise limit easily.


ScottWurcer, what is " bridge null techniques " in this context ? Could you please explain any further ?
What kind of multiplier would you use to keep additional jitter by this device low ?

Greetings
Michael
 
AX tech editor
Joined 2002
Paid Member
scott wurcer said:
[snip]Also, even though they are maligned for no good reason to me, bridge null techniques remain a powerful tool. You can get a good -120dB null and amplify the residue. Put this into a nice low frequency spectrum analyser and you can get down to the thermal noise limit easily.

I have done similar things with a good old AP S1. Use one channel of the analog analyzer to extract the HD signal, then input that in the other channel's A/D and spectrum analyzer, doing multiple samples averaging. You see the individual harmonics clearly.

Jan Didden
 
gedlee said:


I guess that I'd say - If a sound card and some software works fine what's the point of all that external equipment and expense.

I do serious research. I only use a sound card.

Never meant to say you weren't serious. I guess I missed the point, is it enhancing the resolution of sound card measurements or measuring the true THD at reduced output powers?

BTW one of those amplifiers appears to also have some serious power supply problems (120Hz sidebands on the harmonics).
 
scott wurcer said:

I guess I missed the point, is it enhancing the resolution of sound card measurements or measuring the true THD at reduced output powers?

BTW one of those amplifiers appears to also have some serious power supply problems (120Hz sidebands on the harmonics).

The point that I most want to make is that we must look at
the true THD - mostly the higher harmonics - at reduced output levels. This is never published and seldom done. I use a soundcard, so its natural to look for techniques that work with them.

One amp was made in Asian for 220 volts and may not be working correctly on 110.
 
Status
This old topic is closed. If you want to reopen this topic, contact a moderator using the "Report Post" button.