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Debate of DAC design regarding DSD vs PCM among 5 VIPs


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Agreed about importance of DAC latch.

 

"flicker" is a technical term described extensively by Rubiola who also describes the process of "parametric up conversion" where voltage/current noise results in phase noise. This is extensively discussed in the reference -- the very basis of why a good power supply for the oscillator is important. When a signal leaks that is, in fact, noise.

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27 minutes ago, jabbr said:

"flicker" is a technical term described extensively by Rubiola who also describes the process of "parametric up conversion" where voltage/current noise results in phase noise. This is extensively discussed in the reference -- the very basis of why a good power supply for the oscillator is important. When a signal leaks that is, in fact, noise.

 

In audio, correlated jitter typically means all the non-random processes.

 

Random sources such as thermal current/voltage noise causes uncorrelated jitter. There is no systematic relationship between audio signal and the disturbance. Since for example DPLL always has some low-pass frequency slope and corner frequency this causes the frequency lobe to have widening lobe, even if the input noise is white. This doesn't mean that the noise itself would be correlated in any way.

 

Same way as dither decorrelates the systematic rounding error with random process.

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43 minutes ago, Miska said:

 

In audio, correlated jitter typically means all the non-random processes.

 

Random sources such as thermal current/voltage noise causes uncorrelated jitter. There is no systematic relationship between audio signal and the disturbance. Since for example DPLL always has some low-pass frequency slope and corner frequency this causes the frequency lobe to have widening lobe, even if the input noise is white. This doesn't mean that the noise itself would be correlated in any way.

Flicker or 1/f noise is definitely correlated and since it's within the PLL corner, not easy to improve even with the "new" femtosecond chips which typically quote at 1khz or greater offset because that's where the number look the best.

 

This article is publicly available: http://www.fhnw.ch/technik/ime/publikationen/2008/circuits-at-the-nanoscale-communications-imaging-and-sensing

 

you might note the similarity between the "chopper" amplifier and "noise shaping" where upsampling pushes the noise into a higher frequency range -- above the PLL corner frequency in the former but I think the analogy is apropos

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8 hours ago, jabbr said:

Flicker or 1/f noise is definitely correlated and since it's within the PLL corner, not easy to improve even with the "new" femtosecond chips which typically quote at 1khz or greater offset because that's where the number look the best.

 

Correlated with the audio or correlated with the clock? Remember these are different things.

 

We regularly have a problem when we talk about audio clocking, is that people look at the clocks in separation, and not from the audio point of view. One big reason is that books about clocks usually look at the clocks alone, and not in relation to data conversion circuits and how the conversion circuit architecture affects the behavior. For example the DSC1 type of DAC reduces effect of that kind of jitter, while R2R type emphasizes it. Same actually goes for the correlated jitters too.

 

What I think Andreas was talking about is the jitter embedded into data from ADC and how to whiten that at the DAC side. And sure, for example DSD upsampling, especially to a different rate family, is one effective way to reduce it, but it can be further whitened with some clock tricks depending on type of the DAC. But that requires DSP processing of the clock signal.

 

People seem to worry a lot about DAC clocking, but not so much about the clocking used at ADC side for recording... ;)

 

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4 hours ago, Miska said:

 

There is no "good jitter". But random (uncorrelated) jitter is usually less audible than non-random (correlated) jitter. I picked up some examples...

 

This could be considered to be less bad "good" jitter:

goodjitter.thumb.png.8b1632aba6c8bad1d5d36169792b8085.png

 

And this could be considered to be more bad "bad" jitter:

badjitter.thumb.png.4af8f15e10a2da4e0e152887c4c099ae.png

 

You can see the first one has widening base of the main lobe, while the second one has narrower main lobe, but strong distinct side lobes.

 

Upper bad good jitter graph looks as if a flutter noise is added to the pure sinusoidal signal :)

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On 4/28/2017 at 5:03 AM, Miska said:

 

There is no "good jitter". But random (uncorrelated) jitter is usually less audible than non-random (correlated) jitter. I picked up some examples...

 

This could be considered to be less bad "good" jitter:

goodjitter.thumb.png.8b1632aba6c8bad1d5d36169792b8085.png

 

And this could be considered to be more bad "bad" jitter:

badjitter.thumb.png.4af8f15e10a2da4e0e152887c4c099ae.png

 

You can see the first one has widening base of the main lobe, while the second one has narrower main lobe, but strong distinct side lobes.

 

OK, I see what you are writing. I think the problem would be the definition of the term "jitter" itself (I dislike this term).

 

The top would be what I would consider "phase error" ... the random part is the baseline (e.g thermal noise). The widening of the peak is related to 1/f or "flicker noise". This is correlated to signal in that it is both current/voltage dependent as well as "memory" dependent.

 

The bottom -- the sidebands are nearly the same amplitude as the center, so this is what I would call "distortion" and the levels are quite high. The signal surely would not be a clean sinusoid. Could be clipping, could be duty-cycle, could be significant cross talk. This would seem to be more of a "frequency error" rather than "phase error". Do you have a trace of the signal?

 

But again, I think the term "jitter" itself is problematic, and encompasses processes that are radically different.

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45 minutes ago, jabbr said:

 

OK, I see what you are writing. I think the problem would be the definition of the term "jitter" itself (I dislike this term).

 

The top would be what I would consider "phase error" ... the random part is the baseline (e.g thermal noise). The widening of the peak is related to 1/f or "flicker noise". This is correlated to signal in that it is both current/voltage dependent as well as "memory" dependent.

 

The bottom -- the sidebands are nearly the same amplitude as the center, so this is what I would call "distortion" and the levels are quite high. The signal surely would not be a clean sinusoid. Could be clipping, could be duty-cycle, could be significant cross talk. This would seem to be more of a "frequency error" rather than "phase error". Do you have a trace of the signal?

 

But again, I think the term "jitter" itself is problematic, and encompasses processes that are radically different.

jitter is basically an aggregate timing error measurement. Which doesn't help us when a jitter issue is frequency related.

Regards,

Dave

 

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On 4/28/2017 at 10:03 AM, Miska said:

 

There is no "good jitter". But random (uncorrelated) jitter is usually less audible than non-random (correlated) jitter. I picked up some examples...

 

This could be considered to be less bad "good" jitter:

goodjitter.thumb.png.8b1632aba6c8bad1d5d36169792b8085.png

 

And this could be considered to be more bad "bad" jitter:

badjitter.thumb.png.4af8f15e10a2da4e0e152887c4c099ae.png

 

You can see the first one has widening base of the main lobe, while the second one has narrower main lobe, but strong distinct side lobes.

Something that I hope might be answerable here - how to interpret the top FFT plot. I know it represents close-in phase noise, in other words the 1KHz signal is not always reproduced accurately at 1KHz because of this fluctuation of the clock frequency - its phase noise.

 

Looking at that plot in full size, the skirt spans about 500Hz either side of the 11KHz test signal. What this means is that a small amount of the time the signal is wrong by +/- 500Hz & as we move closer to the 11KHz signal the skirt increases in amplitude i.e. the signal is wrong more often as we get closer to the test signal frequency. So if we read this FFT plot as showing amplitude of the error signal either side of the 11KHz would seem to be mistaken. Do people agree with this?

 

Let's say we zoomed into around this 11KHz signal to a much more resolved FFT plot which showed 1Hz divisions on  the x-axis - we would see that the skirt @1Hz either side of the 11KHz was close in amplitude to the test signal. Which I interpret as the signal is wrong by 1Hz quiet often.

 

The point is that the skirts should not be read as down in amplitude @ -160dB growing to -90dB (test signal amplitude, in this case) - the frequency errors are at -90dB but at varying times - it's a statistical plot of how often the errors occur. But the same would apply if the test signal was -3dB - errors in frequency would have an amplitude of -3dB, not -150dB or whatever

 

If this analysis is correct, it's perceptually far different to have an frequency error of the signal at the same amplitude  rather than a frequency error -150dB down in amplitude.

 

Again, if my analysis is correct, the top FFT plot would certainly not represent "good" jitter or even "less bad" jitter - it could be perceptually very significant, blurring sound somewhat I would guess.   

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39 minutes ago, mmerrill99 said:

Again, if my analysis is correct, the top FFT plot would certainly not represent "good" jitter or even "less bad" jitter - it could be perceptually very significant, blurring sound somewhat I would guess.   

 

As you note, these are frequency rather than phase plots, and with linear, rather than log, scale along frequency(?). I'd say that phase error is better than outright distortion -- so clearly "less bad" ;) ...but ... hard to predict what the actual @1Hz offset error would be ... but let's say @10Hz is -60 db/Hz ... not the best achievable, so perhaps that's why this plot is selected, to demonstrate the error. As @Miska alludes to, the clock phase error is the best, but once it is distributed into the circuit itself, the numbers get worse. These numbers are never published and who knows how often they are actually measured.

 

The errors seen on the bottom plot might be due to more basic electronics issues such as crosstalk, setup timing errors, duty cycle asymmetries in computed clocks etc etc. You know a violin that is a bit blurred is better than a chainsaw ;) 

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24 minutes ago, jabbr said:

 

As you note, these are frequency rather than phase plots, and with linear, rather than log, scale along frequency(?). I'd say that phase error is better than outright distortion -- so clearly "less bad" ;) ...but ... hard to predict what the actual @1Hz offset error would be ... but let's say @10Hz is -60 db/Hz ... not the best achievable, so perhaps that's why this plot is selected, to demonstrate the error. As @Miska alludes to, the clock phase error is the best, but once it is distributed into the circuit itself, the numbers get worse. These numbers are never published and who knows how often they are actually measured.

 

The errors seen on the bottom plot might be due to more basic electronics issues such as crosstalk, setup timing errors, duty cycle asymmetries in computed clocks etc etc. You know a violin that is a bit blurred is better than a chainsaw ;) 

But remember as regards to perception, we are hearing modulating phase shifts, not fixed phase shifts & also remember that non-linear systems (all our playback systems) will produce intermodulation distortions as a result of the phase differences & these will also be modulating.

 

I have perceived a more stable, solid soundstage as a result of using a lower jitter clock

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Interview with Andreas Koch from playback

 

2.6 If you claim that you solved the jitter completely, does it mean that the signal will not be affected anymore by any digital cables and digital interfaces? 

 

Yes, indeed. Once the problem of jitter is eliminated, the impact of various digital cables and interfaces will no longer exist.  However, there are still many problems found in USB interface. Hence, we put lots of effort on optimizing the USB interface and DSD signal transmission so as to make sure that the noise from PC won’t have negative influence on the signal.

 

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1 hour ago, jabbr said:

 

As you note, these are frequency rather than phase plots, and with linear, rather than log, scale along frequency(?). I'd say that phase error is better than outright distortion -- so clearly "less bad" ;) ...but ... hard to predict what the actual @1Hz offset error would be ... but let's say @10Hz is -60 db/Hz ... not the best achievable, so perhaps that's why this plot is selected, to demonstrate the error. As @Miska alludes to, the clock phase error is the best, but once it is distributed into the circuit itself, the numbers get worse. These numbers are never published and who knows how often they are actually measured.

 

The errors seen on the bottom plot might be due to more basic electronics issues such as crosstalk, setup timing errors, duty cycle asymmetries in computed clocks etc etc. You know a violin that is a bit blurred is better than a chainsaw ;) 

What is meant by -60dB/Hz?

I'm suggesting that the amplitude of the error due to a clock drifting around it's fundamental frequency (phase noise) will be the same signal amplitude but reproduced at a frequency slightly different to the correct frequency.

 

So if the original signal is @-6dB then these errors in frequency will also be @-6dB. Why would they be reduced in amplitude? 

 

FFTs are not intuitive (i.e. misleading) when plotting at anything other than narrow band signals

 

Just another point - IMO, this is also correlated jitter as Jabbr has said - I believe it tracks the signal i.e only 'blurring' the frequencies in the signal, not producing noise which is unrelated to the signal! 

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BTW, as regards audibility of phase - it seems to be related to cochlea non-linearities - if two frequencies are in different critical bands then intermodulation products will not occur, however if they are in the same critical bands the cochlea will generate intermodulation products.

 

Edit: The critical band is an old concept that defines the frequency bands that the cochlea auditory filter splits the incoming signal into. The more modern & slightly different frequency ranges is called the ERBs

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1 hour ago, mmerrill99 said:

I'm suggesting that the amplitude of the error due to a clock drifting around it's fundamental frequency (phase noise) will be the same signal amplitude but reproduced at a frequency slightly different to the correct frequency.

 

So if the original signal is @-6dB then these errors in frequency will also be @-6dB. Why would they be reduced in amplitude?

 

Makes me think: "The Frequency shifts, but the Amplitude remains the same !" (to the tune of an old advertising slogan)  :P

 

Sorry -- resume discussion...

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1 hour ago, mmerrill99 said:

What is meant by -60dB/Hz?

I'm suggesting that the amplitude of the error due to a clock drifting around it's fundamental frequency (phase noise) will be the same signal amplitude but reproduced at a frequency slightly different to the correct frequency.

 

So if the original signal is @-6dB then these errors in frequency will also be @-6dB. Why would they be reduced in amplitude? 

 

FFTs are not intuitive (i.e. misleading) when plotting at anything other than narrow band signals

 

Just another point - IMO, this is also correlated jitter as Jabbr has said - I believe it tracks the signal i.e only 'blurring' the frequencies in the signal, not producing noise which is unrelated to the signal! 

 

Typically phase error plots do not show frequency per se, rather frequency offset from the signal, and might be log, so 10^-3,10^-2,10^-1,1,10,10^2 on x-axis plotted against dB on y-axis.

 

so -60dB @1Hz means that the phase error component 1 Hz to either side of the carrier is diminished by 60dB

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The "correlation" of a phase error that rises as it gets closer to the carrier is correlated to the carrier -- this describes "slope". An uncorrelated error is flat regardless of distance from carrier i.e. slope is 0.

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23 minutes ago, jabbr said:

 

Typically phase error plots do not show frequency per se, rather frequency offset from the signal, and might be log, so 10^-3,10^-2,10^-1,1,10,10^2 on x-axis plotted against dB on y-axis.

 

so -60dB @1Hz means that the phase error component 1 Hz to either side of the carrier is diminished by 60dB

Ok, let's take it from first principles. We have two clock ticks one is at the exact correct frequency, 12MHz say & the next one is at 1Hz over 12MHz. Let's say this is the audio clock driving a DAC & the DAC is reproducing a pure tone of 11KHz. For simplicity let's not get into the shape of the sine wave & what part of this sine wave is being reproduced but rather that both digital samples should have produced a 11KHz signal at -3dB. The first sample does but the next sample (which is using the clock tick which has slipped by 1Hz) is producing the same amplitude signal but at the wrong time i.e it has shifted the frequency by 1Hz

 

My question is why would the amplitude be diminished? The frequency is wrong but I can't see how the amplitude is affected?

 

I know this skips lots of details & keeps it simplistic, in order to state my issue - maybe the answer is in the details that I'm not seeing?

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29 minutes ago, jabbr said:

The "correlation" of a phase error that rises as it gets closer to the carrier is correlated to the carrier -- this describes "slope". An uncorrelated error is flat regardless of distance from carrier i.e. slope is 0.

But again, this is what I have a problem with - FFTs do not show the correct amplitude of broadband signals - there's a certain process gain needed to calculate the correct amplitude of the broadband signal as it ranges across that 1KHz frequency range that the skirt encompasses.

 

I look on that FFT plot as a statistical representation of how many erroneous samples fall into (how much power is analysed)  the bins 1Hz away from the main signal;, 1.1Hz away, 1,2Hz away, etc. The further away from the fundamental signal the erroneous samples, the fewer the samples. The amplitude of each sample is the same BUT the power found in each bin diminishes towards the noise floor the further away from  the fundamental is plotted.

 

Essentially FFTS are like long exposure photography - if an object stays still  during the exposure, it will be reproduced sharply & brightly with fine detail - if an object moves during exposure its brightness will be diminished & it will be a blurry image (the equivalent of jitter). BUT the image that is captured is not a true representation of the brightness of these objects - their apparent brightness (the equivalent of amplitude) is simply due to the sampling happening during exposure  

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1 hour ago, mmerrill99 said:

Ok, let's take it from first principles. We have two clock ticks one is at the exact correct frequency, 12MHz say & the next one is at 1Hz over 12MHz. Let's say this is the audio clock driving a DAC & the DAC is reproducing a pure tone of 11KHz. For simplicity let's not get into the shape of the sine wave & what part of this sine wave is being reproduced but rather that both digital samples should have produced a 11KHz signal at -3dB. The first sample does but the next sample (which is using the clock tick which has slipped by 1Hz) is producing the same amplitude signal but at the wrong time i.e it has shifted the frequency by 1Hz

 

My question is why would the amplitude be diminished? The frequency is wrong but I can't see how the amplitude is affected?

 

I know this skips lots of details & keeps it simplistic, in order to state my issue - maybe the answer is in the details that I'm not seeing?

 

Perhaps this can be considered like "conservation of energy" such that the SPL or energy is preserved, i.e. area under curve, so that when a peak is widened it loses peak amplitude -- holding energy constant ... otherwise a change in energy would be measured i.e. very accurate clocks would draw less power ... don't think that's the case.

 

Does that help? Remember that the FFT usually only shows the "real" or amplitude component, and being laplacian, there is the imaginary component. That's why its vector math, not simple addition/subtraction.

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If we are discussing the effect of phase error in a clock (12 Mhz example) on the reproduction of an 11 kHz tone, consider that the DSD DAC is integrating the pulse widths to product an analog signal and the pulse width variation is integrated ... a PCM DAC will similarly use a series of multibit values that will change with the clock. Using vector math, a change in the voltage representing a signal will have a component change to both the amplitude and phase of the underlying signal (at each frequency component).

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13 minutes ago, jabbr said:

 

Perhaps this can be considered like "conservation of energy" such that the SPL or energy is preserved, i.e. area under curve, so that when a peak is widened it loses peak amplitude -- holding energy constant ... otherwise a change in energy would be measured i.e. very accurate clocks would draw less power ... don't think that's the case.

 

Does that help? Remember that the FFT usually only shows the "real" or amplitude component, and being laplacian, there is the imaginary component. That's why its vector math, not simple addition/subtraction.

Sorry but this doesn't make sense to me - you are talking about how an FFT represents narrow Vs broadband signals - I'm saying that the representation is misleading to an understanding of what is actually happening with close in phase noise of audio clocks & its affect on signals.

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31 minutes ago, jabbr said:

If we are discussing the effect of phase error in a clock (12 Mhz example) on the reproduction of an 11 kHz tone, consider that the DSD DAC is integrating the pulse widths to product an analog signal and the pulse width variation is integrated

I didn't want to get into complexity & different variations of PWM Vs PCM - just wanted to keep it simple

31 minutes ago, jabbr said:

a PCM DAC will similarly use a series of multibit values that will change with the clock. Using vector math, a change in the voltage representing a signal will have a component change to both the amplitude and phase of the underlying signal (at each frequency component).

Sorry but you are losing me - I can't see how this integration can explain a reduction in amplitude to the extent that we see in the FFT plot here - maybe I'm being stupid? 

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