Query about the "jitterstream"

[pkane2001]

7 hours ago, CG said:

Truer words have seldom been written!

 

In communications systems, it's generally more valuable to measure the spectrum of the phase noise, as John describes.  Not easy or cheap, by a long shot.

 

For audio reproduction, you'd think that there'd be a way to measure the phase noise sidebands of a synthesized tone at the output of a DAC (DAC as in a complete digital audio converter, not just a DAC chip).  It wouldn't be easy or cheap, but possible.  That might give a clue as to the effect of phase noise on the audio output.  Of course, it would also be combined with other effects, but those affect the sound, too.

 

One gotcha:  Almost all measurements in the audio world are made using various averaging or smoothing techniques.  This is done with the idea that noise is random, while the desired tones you are trying to measure are not.  So, if you take a bunch of repeat measurements, the noise will average to a much lower level because the noise is assumed to be completely random, while the desired tones will remain constant.  This lets you detect lower level signals within the noise.

 

That is true, but probably misses a key point.  

 

Many "phase noise events" are transitory by nature.  So, you'd only see them once in a while.  Once in a while, as in every 1000 sample sweeps.  There you might witness an awful spectrum due to a bunch of factors.  The next sample sweep could be fine.  In a communications system, that symbol is lost and is a loss of actual bits.  But, one lost symbol out of a thousand gives the misleading impression that everything is ok when you apply averaging.  Not so if you count bit errors.  (If you don't believe me, try measuring MER versus BER some time...)

 

But, is one bad audio sample out of a thousand actually audible?  I'd guess so, but I don't know of any research that concludes anything either way.  (If anybody knows of any, please post a link!)

 

The point is, you probably really need to use a different approach to measuring the effects of phase noise in an audio system as I described.  (You could argue this is necessary for plain old distortion, too, but that's another topic for another forum.)  Perhaps using a peak or max hold function would be more valuable.  You'd just retain the highest value in every FFT bin for several thousands sweeps and display that.  This isn't that difficult to do in software.  (Here's one simple example - http://www.w7ay.net/site/Applications/Amici/index.html)

 

In addition, phase noise on a clock effectively modulates each and every converted tone in the audio spectrum with that noise content.  If the frequency content of the phase noise is low enough, that's pretty much the same thing as rocking your loudspeakers back and forth at whatever rate the phase noise modulation is.  Maybe that's bad; maybe that's even good!  Dunno.

 

I apologize to the OP for going off into the esoteric here, but the topic probably is worth investigating.

 

Here's a way to try to go about determining audibility of various types of jitter: https://distortaudio.org/

 

DISTORT is still in development (I call it a continuous beta ), but already has a few useful features.

 

One, you can apply any number, frequency and level of sine-modulated clock jitter to an existing recording and then listen. You can add random jitter. Or correlated jitter. Or 1/f noise-modulated (close-in) jitter. Or any combination of the above. The recording can be as simple as a sine wave, or as complex as a symphony -- your choice. Apply jitter, save it, then play it back on your 'perfect' system and listen to see if you can hear the effect. Or even listen directly on your computer with DISTORT playing the audio file. You can then switch nearly instantaneously between the original and the distorted tracks using the Bypass button.

 

Here's some correlated, 1/f and sine-modulated jitter applied to a simple 1kHz sine wave:

 

Oh,  and best of all -- no FFTs are used to generate jitter or harmonic distortion, everything is done in the time domain.

 

 

[pkane2001]

9 minutes ago, CG said:

BTW, how did you determine the out-of-band linearity performance of the amplifiers in question?  Is it extrapolation from the published in-band performance?  I missed that.

 

http://www.audiodesignguide.com/Ibridone/Sen_Semi_Diode_Apps-quik108.pdf

 

The out of band analysis you saw is caused by the non-linearity of the system causing inter-modulation distortion (IMD) and aliasing from the energy in frequencies above the audio range to reflect back into the audible range. While aliasing of out-of-band frequencies is unlikely to happen in a DAC (much more likely in an ADC), IMD can happen in both.