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32-bit capable DACs, Split from: "WASAPI plug-in version 3.0 beta" (94907)
saratoga
post Jun 1 2012, 02:14
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QUOTE (SoNic67 @ May 31 2012, 20:50) *
QUOTE (Arnold B. Krueger @ May 30 2012, 09:29) *
There is no such thing in general. You need to study up on how modern converters work.

Now you upset me...
Want examples? AK4683, page 1 "Zero Detect Function".


From your link:

QUOTE
Zero Detect
When the PCM1794 detects that the audio input data in the L-channel and the R-channel is continuously zero for
1024 LRCKs in the PCM mode or that the audio input data is continuously zero for 1024 WDCKs in the external filter
mode, the PCM1794 sets ZERO (pin 13) to HIGH.


So if the source turns off, it alerts the controller so it can turn off the amp. I don't really understand what you're trying to get at, but I don't think this is it. Its common to turn off circuits when they're not being used for power saving reasons. Modern efficiency standards more or less require it.
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Arnold B. Kruege...
post Jun 1 2012, 12:13
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QUOTE (saratoga @ May 31 2012, 21:14) *
QUOTE (SoNic67 @ May 31 2012, 20:50) *
QUOTE (Arnold B. Krueger @ May 30 2012, 09:29) *
There is no such thing in general. You need to study up on how modern converters work.

Now you upset me...
Want examples? AK4683, page 1 "Zero Detect Function".


From your link:

QUOTE
Zero Detect
When the PCM1794 detects that the audio input data in the L-channel and the R-channel is continuously zero for
1024 LRCKs in the PCM mode or that the audio input data is continuously zero for 1024 WDCKs in the external filter
mode, the PCM1794 sets ZERO (pin 13) to HIGH.


So if the source turns off, it alerts the controller so it can turn off the amp. I don't really understand what you're trying to get at, but I don't think this is it. Its common to turn off circuits when they're not being used for power saving reasons. Modern efficiency standards more or less require it.


Agreed.

This is a high-function DAC with a built-in level control. An ideal digital level control that allows level changes at arbitrary times will cause what is called "The zipper effect" which is essentially a series of clicks caused by manually operating an ideal digital level control. This effect can be avoided or minimized by allowing level changes to only occur when the signal naturally passes through zero.

Similar logic is used in light dimmers to minimize EMI generation.

As an aside in the early 1970s I devised a FM muting circuit that seemingly acted instantly and fully but without loud clicks, by waiting for a natural zero-crossing to change state.

This post has been edited by Arnold B. Krueger: Jun 1 2012, 12:15
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knutinh
post Jun 1 2012, 13:36
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QUOTE (Arnold B. Krueger @ May 30 2012, 15:23) *
There are several other DACs that come close Probably very little but money keeps you from connecting a bunch of them in parallel in such away that the noise drops -3 dB every time you double the number of DACs connected in parallel.

I was under the impression that this kind of statement makes assumptions about the DAC errors that is not totally true. I.e. you cannot achieve any (large) increase in "signal to error level" by doubling the number of DAC chips a (large) number of times. They will have some signal-dependant non-linearity, clock pulse bleed-through or whatever that will add constructively, and at some point these errors will become the significant source of error (that does not improve by increasing the amount of DACs).

I don't know about using totally different designs (how many fundamentally different high-quality DAC chips are available?), or liquid Nitrogen cooling?

Of course, going from e.g. 128 to 256 audio ADC/DAC chips is mostly an academic exercise anyways. If the true gain is 1dB less error rather than the expected 3dB, very few of us will ever notice (due to the low levels in the first place, and the low probability that anyone finds it practical to use such a setup)

-k

This post has been edited by knutinh: Jun 1 2012, 13:39
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Arnold B. Kruege...
post Jun 1 2012, 13:52
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QUOTE (knutinh @ Jun 1 2012, 08:36) *
QUOTE (Arnold B. Krueger @ May 30 2012, 15:23) *
There are several other DACs that come close Probably very little but money keeps you from connecting a bunch of them in parallel in such away that the noise drops -3 dB every time you double the number of DACs connected in parallel.

I was under the impression that this kind of statement makes assumptions about the DAC errors that is not totally true. I.e. you cannot achieve any (large) increase in "signal to error level" by doubling the number of DAC chips a (large) number of times. They will have some signal-dependant non-linearity, clock pulse bleed-through or whatever that will add constructively, and at some point these errors will become the significant source of error (that does not improve by increasing the amount of DACs).


Agreed. Hence my careful choice of words: "noise drops". The Sabre DACs may be specifically designed to be free of most of the other non-noise artifacts as they recommend paralleling up to 4 DACs on the same chip. They give SNR specs for this configuration, and they do specify that the resulting nonlinear distortion is well above the noise. Ya got to be a spec sheet lawyer! ;-)

QUOTE
I don't know about using totally different designs (how many fundamentally different high-quality DAC chips are available?), or liquid Nitrogen cooling?

Of course, going from e.g. 128 to 256 audio ADC/DAC chips is mostly an academic exercise anyways. If the true gain is 1dB less error rather than the expected 3dB, very few of us will ever notice (due to the low levels in the first place, and the low probability that anyone finds it practical to use such a setup)


I don't' know how far people have actually gone with this game. It gets real expensive if you try to go very far. And as you say for what? Even one DAC on the 4 section chip is vast overkill.


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SoNic67
post Jun 3 2012, 19:20
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Paralelling is done especially on passive I/V conversion. For normal OpAmp conversion, the limit is the OpAmp noise.
After a few paralleled DAC's, the noise still gets lower (more current allows for lower passive I/V resistance), but the distortion doesn't, it "settles" to the architecture minimum value. Noise is statistically uncorrelated, distortion is correlated.
For Sabre32 9018, the SNR can go as low as -135dB using paralleling of the 8 internal DAC's (129dB 8ch, 133dB stereo, 135dB mono), but THD remains in all cases at -120dB (20 bit quality). This is their data. At some point there is no quality return for paralleling DAC chips.

This post has been edited by SoNic67: Jun 3 2012, 19:28
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IgorC
post Jun 3 2012, 19:50
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Those are nice and probably real specifications.
The thing is that there are no headphone or speaker which will benefit from such high precision. The best headphones have THD -80 dB (best case). The sum of -80 dB + DAC's THD -100 dB is still the same - 80dB. DAC's THD better than -95 ... -100 dB doesn't improve anything in real audio experience but the specifications itself.

It makes only sense if phones have low impedance as saratoga has already mention (but still You won't get any better than -100dB)

This post has been edited by IgorC: Jun 3 2012, 19:58
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saratoga
post Jun 3 2012, 22:11
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QUOTE (IgorC @ Jun 3 2012, 14:50) *
The thing is that there are no headphone or speaker which will benefit from such high precision. The best headphones have THD -80 dB (best case). The sum of -80 dB + DAC's THD -100 dB is still the same - 80dB.


THD is relative to the signal intensity though. For dynamic music, you might have loud and quiet passages, in which case having a lower noise floor then the THD implies is still worthwhile so that you don't hit the noise floor during softer portions of a track.

I agree that having a noise floor less then 90-100dB below your typical listening level is entirely pointless.

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Arnold B. Kruege...
post Jun 4 2012, 13:21
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QUOTE (saratoga @ Jun 3 2012, 17:11) *
QUOTE (IgorC @ Jun 3 2012, 14:50) *
The thing is that there are no headphone or speaker which will benefit from such high precision. The best headphones have THD -80 dB (best case). The sum of -80 dB + DAC's THD -100 dB is still the same - 80dB.


THD is relative to the signal intensity though. For dynamic music, you might have loud and quiet passages, in which case having a lower noise floor then the THD implies is still worthwhile so that you don't hit the noise floor during softer portions of a track.


Do you know of anybody with a logical interest in -120 dB THD and 129 dB SNR than test equipment manufacturers? I guess the 120dB+ SNR might make sense for a preamp-in microphone application as it eliminates the need for a remote volume control.

QUOTE
I agree that having a noise floor less then 90-100dB below your typical listening level is entirely pointless.


Agreed, unless you have someone who wants to hook 100 dB/w speakers to 1 Kw power amps and not hear any noise when they put their ears next to the speakers.
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SoNic67
post Jun 5 2012, 01:09
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I can hear very clear sounds at -60dB (running volume at normal level of audition) - based on spectrum analizers. I would like to have as less distortion as possible at that level. If the DAC has only -80dB THD+N, that leaves only 20dB between sounds and distortion/noise. That is perfectly audible, it will influence the sound... Any PC speakers with a 'normal' THD+N of -85dB will prove easily that fact.
If the DAC has -110dB THD+N, that leaves -50dB between sound and noise/distortion. I cannot hear that and that means that the DAC will not "colour" the sound.

Now... headphones/speakers have listed -80...90dB THD, but that is at nominal levels of power. At lower levels, because of different mechanism of producing those distortions (mechanical histerezis of suspension and limited membrane rigidity), distortion levels drop significantly. Mind also that thermal noise is generated in speakers/headphones is very, very low due to low impedance. A good pair of headphones probably can achieve -100...105dB of THD when are 'run' at -20dB levels - sufficient for a normal hearing.

Anyway, that means only 20 bit of real resolution!!! So yes, 24 bit is more than enough for audio.

This post has been edited by SoNic67: Jun 5 2012, 01:21
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saratoga
post Jun 5 2012, 01:22
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QUOTE (SoNic67 @ Jun 4 2012, 20:09) *
I can hear very clear sounds at -60dB (running volume at normal level of audition) - based on spectrum analizers.


-60dB relative to what?

QUOTE (SoNic67 @ Jun 4 2012, 20:09) *
I would like to have as less distortion as possible at that level. If the DAC has only -80dB THD+N, that leaves only 20dB between sounds and distortion/noise.


I'm not sure I understand what you're trying to say, but I don't think your logic here is correct. At least not the 20dB part.
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IgorC
post Jun 5 2012, 03:52
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QUOTE (SoNic67 @ Jun 4 2012, 21:09) *
Now... headphones/speakers have listed -80...90dB THD ...

I'd be grateful if You could provide a link to specifications for headphones or speakers which have THD better than -80 dB? Or at least -80 dB.
Example, Sennheiser HD800 has THD less than -74 dB. Maybe a bit less, but not that much. It hardly hits -80 dB. The problem is that it's expensive and presents only less than 1% of all market of headphones.

QUOTE (SoNic67 @ Jun 4 2012, 21:09) *
A good pair of headphones probably can achieve -100...105dB of THD when are 'run' at -20dB levels - sufficient for a normal hearing.

THD of headphones doesn't improve with a lower levels. Example http://www.innerfidelity.com/images/SennheiserHD650.pdf

This post has been edited by IgorC: Jun 5 2012, 04:01
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Arnold B. Kruege...
post Jun 5 2012, 13:00
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QUOTE (SoNic67 @ Jun 4 2012, 20:09) *
I can hear very clear sounds at -60dB (running volume at normal level of audition) - based on spectrum analizers.


This situation probably ignores the effects of masking which are huge.


QUOTE
I would like to have as less distortion as possible at that level. If the DAC has only -80dB THD+N, that leaves only 20dB between sounds and distortion/noise. That is perfectly audible, it will influence the sound... Any PC speakers with a 'normal' THD+N of -85dB will prove easily that fact.


A DAC or headphone amp with only -80 dB (0.01%) spurious responses is by modern standards, a serious POS among SOTA devices. 100 dB shouldn't be rocket science. But, its overkill.

QUOTE
If the DAC has -110dB THD+N, that leaves -50dB between sound and noise/distortion. I cannot hear that and that means that the DAC will not "colour" the sound.


Just as people can drown in 3 inches of water and dead is dead no matter how shallow the water is, 10 dB below audibility is for all practical purposes as good as 50 dB or 100 dB.

QUOTE
Now... headphones/speakers have listed -80...90dB THD, but that is at nominal levels of power. At lower levels, because of different mechanism of producing those distortions (mechanical histerezis of suspension and limited membrane rigidity), distortion levels drop significantly. Mind also that thermal noise is generated in speakers/headphones is very, very low due to low impedance. A good pair of headphones probably can achieve -100...105dB of THD when are 'run' at -20dB levels - sufficient for a normal hearing.


You badly need to find out what humans can actually hear in the way of nonlinear distortion. I feel like I'm listening to a runner who thinks he can run 10 second miles!

QUOTE
Anyway, that means only 20 bit of real resolution!!! So yes, 24 bit is more than enough for audio.


So is 16 bits in distribution formats.
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Arnold B. Kruege...
post Jun 5 2012, 13:06
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QUOTE (IgorC @ Jun 4 2012, 22:52) *
THD of headphones doesn't improve with a lower levels. Example http://www.innerfidelity.com/images/SennheiserHD650.pdf


That's probably only because you are looking at THD+N, not the results of a proper spectrum analysis. At lower levels, the noise may make the results look higher than the nonlinear distoriton actually is.

Typically, nonlinear distortion goes down with level, unless it is due to a discontinuity, such as crossover distortion. Crossover distortion is essentially impossible in a sigma-delta DAC (that covers just about every modern DAC for audio), and exceedingly rare or absent in properly designed amplifiers. Transducers can't have sharp discontinuities in their transfer curves unless there is something like sticking.
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SoNic67
post Jun 6 2012, 01:11
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QUOTE (IgorC @ Jun 4 2012, 22:52) *
QUOTE (SoNic67 @ Jun 4 2012, 21:09) *
Now... headphones/speakers have listed -80...90dB THD ...

I'd be grateful if You could provide a link to specifications for headphones or speakers which have THD better than -80 dB? Or at least -80 dB.

You didn't read my whole statement... at nominal levels (that can damage hearing) THD+N is at -80dB...-90dB. Lower that at norma listening and we are talking a different story. As I said, that's because of hysterezis of friction (suspension), membrane finite rigidity (for bass)...
http://www.headphone.com/headphones/grado-gs-1000i.php#tabs

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saratoga
post Jun 6 2012, 02:39
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QUOTE (SoNic67 @ Jun 5 2012, 20:11) *
QUOTE (IgorC @ Jun 4 2012, 22:52) *
QUOTE (SoNic67 @ Jun 4 2012, 21:09) *
Now... headphones/speakers have listed -80...90dB THD ...

I'd be grateful if You could provide a link to specifications for headphones or speakers which have THD better than -80 dB? Or at least -80 dB.

You didn't read my whole statement... at nominal levels (that can damage hearing) THD+N is at -80dB...-90dB. Lower that at norma listening and we are talking a different story. As I said, that's because of hysterezis of friction (suspension), membrane finite rigidity (for bass)...
http://www.headphone.com/headphones/grado-gs-1000i.php#tabs


That link says -60dB . . .
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Arnold B. Kruege...
post Jun 6 2012, 14:10
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QUOTE (saratoga @ Jun 5 2012, 21:39) *
QUOTE (SoNic67 @ Jun 5 2012, 20:11) *
QUOTE (IgorC @ Jun 4 2012, 22:52) *
QUOTE (SoNic67 @ Jun 4 2012, 21:09) *
Now... headphones/speakers have listed -80...90dB THD ...

I'd be grateful if You could provide a link to specifications for headphones or speakers which have THD better than -80 dB? Or at least -80 dB.

You didn't read my whole statement... at nominal levels (that can damage hearing) THD+N is at -80dB...-90dB. Lower that at norma listening and we are talking a different story. As I said, that's because of hysterezis of friction (suspension), membrane finite rigidity (for bass)...
http://www.headphone.com/headphones/grado-gs-1000i.php#tabs


That link says -60dB . . .


Yes, and the frequency scale seems to be misleading. The first spurious response appears to be second harmonic.
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knutinh
post Jun 6 2012, 21:18
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QUOTE (Arnold B. Krueger @ Jun 1 2012, 14:52) *
I don't' know how far people have actually gone with this game. It gets real expensive if you try to go very far. And as you say for what? Even one DAC on the 4 section chip is vast overkill.

I guess it might make sense for some measurement/lab applications. I certainly think it is interesting to see how far things can be taken.

Distortion that is signal-correlated (predictable?) might also be reduced through a feed-forward or feedback-loop?

I am also depressingly aware of how some people would misguide and be misguided into believing that their records would sound audibly better for using one.

-k
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IgorC
post Jun 6 2012, 21:38
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QUOTE (SoNic67 @ Jun 5 2012, 21:11) *
You didn't read my whole statement... at nominal levels (that can damage hearing) THD+N is at -80dB...-90dB. Lower that at norma listening and we are talking a different story. As I said, that's because of hysterezis of friction (suspension), membrane finite rigidity (for bass)...
http://www.headphone.com/headphones/grado-gs-1000i.php#tabs

You're too far to get such good THD(+N) with these headphones on listenable levels.
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SoNic67
post Jun 7 2012, 00:20
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Sooo... In conclusion, we need 32 bit audio?
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stephan_g
post Jun 7 2012, 00:46
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QUOTE (Arnold B. Krueger @ Jun 5 2012, 13:06) *
Typically, nonlinear distortion goes down with level, unless it is due to a discontinuity, such as crossover distortion. Crossover distortion is essentially impossible in a sigma-delta DAC (that covers just about every modern DAC for audio), and exceedingly rare or absent in properly designed amplifiers. Transducers can't have sharp discontinuities in their transfer curves unless there is something like sticking.

I wouldn't say crossover distortion is absent... people do tend to go to considerable pains to keep it in check in low-quiescent-current applications where it easily dominates high-frequency distortion at higher levels.
Borrowing a graph with horribly messed-up labeling from NwAvGuy:

(real labels should be, from bottom to top at 9 kHz: NJM2068 2.5x, NJM2068 7x, OPA2277 2.5x, OPA2277 7x)

The OPA2277 already was the best-performing out of several low-power parts he tried. And yes, of course the decay in distortion above ~10 kHz is only an artifact due to a 20 kHz LPF that filters out the harmonics.

The different characteristics and according distortion spectra of electronics vs. transducers mean that comparing distortion specs has to be carried out with care. Transducers usually have distortion dominated by low-order (2nd/3rd) products which still show considerable masking, while with electronics it is easy to generate a near-flat "zoo" of harmonics that decays very slowly, resulting in much tighter (numerical) requirements since now high-order harmonics have to be kept below the threshold of audibility (usually <=-80 dBr) at all times. Discontinuities + feedback at work there.

Incidentally, since headphone drivers also tend to be numerically optimized these days, I'd expect a rising discrepancy between objective and subjective evaluation of level handling. Same as when introducing feedback, basically - less low-order, more higher-order products. I'd rather not be trading 0.2% less 2nd harmonic for 0.03% more 7th or 8th (since it is quite possible to hear the latter, while -54 dB of 2nd should be a little below the audible minimum found in studies).

Re: why it is useful to have DR > THD+N, one only needs to consider that the required instantaneous dynamic range in any given playback or recording situation is usually smaller than the total dynamic range for all of them. For example, you may want your 80-odd dB of instantaneous DR regardless of whether you're using speakers with a sensitivity of 85 dB SPL / 2.83 V / m or others with 105 dB. (Or headphones with 95 dB / V vs. others with 130 dB / V.) It's mostly a matter of convenience and not having to worry about stuff.
QUOTE (knutinh @ Jun 6 2012, 21:18) *
Distortion that is signal-correlated (predictable?) might also be reduced through a feed-forward or feedback-loop?

The classic predistortion? That would seem worth a shot. Each batch of chips may need a different set of coefficients (internal EEPROM?), but it should be doable. Obviously that only applies if the analog part is the dominant source of distortion.

Premium DACs are already using balanced output so they're largely getting rid of even-order distortion, but what did one have to do to get rid of odd order again? Think it involved a Hilbert xform for a 90 phase shift? Sounds kinda impractical but you never know.

As far as 32-bit DACs are concerned... seems like you're losing about 4 bits with undithered output, so you're only getting 20 bits from a 24 bit converter. At 192 kHz we're already having 4X oversampling though, for an effective 22 bits in the audio band. So if anything, it would be of concern for 44/48 kHz and more or less "synthetic" material. A very, VERY small fraction of cases.
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nevermind
post Jun 7 2012, 01:56
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QUOTE (stephan_g @ Jun 7 2012, 01:46) *
I wouldn't say crossover distortion is absent... people do tend to go to considerable pains to keep it in check in low-quiescent-current applications where it easily dominates high-frequency distortion at higher levels. ... The OPA2277 already was the best-performing out of several low-power parts he tried.


You do realise that the increase in high frequency distortion is due to the reduction in open loop gain as frequency increases, it has nothing to do with crossover distortion. The opa2277 is the best of all because it has 20db extra loop gain, sort of cheating, really. Oh, and both legends are wrong.

Edit:oops I was thinking of the opa2227.




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Arnold B. Kruege...
post Jun 7 2012, 12:39
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QUOTE (stephan_g @ Jun 6 2012, 19:46) *
QUOTE (Arnold B. Krueger @ Jun 5 2012, 13:06) *
Typically, nonlinear distortion goes down with level, unless it is due to a discontinuity, such as crossover distortion. Crossover distortion is essentially impossible in a sigma-delta DAC (that covers just about every modern DAC for audio), and exceedingly rare or absent in properly designed amplifiers. Transducers can't have sharp discontinuities in their transfer curves unless there is something like sticking.

I wouldn't say crossover distortion is absent... people do tend to go to considerable pains to keep it in check in low-quiescent-current applications where it easily dominates high-frequency distortion at higher levels.
Borrowing a graph with horribly messed-up labeling from NwAvGuy:


(real labels should be, from bottom to top at 9 kHz: NJM2068 2.5x, NJM2068 7x, OPA2277 2.5x, OPA2277 7x)

The OPA2277 already was the best-performing out of several low-power parts he tried.


Key words: "He tried". Just two of how many 100s of similar parts.

You seem to be oblivious to the fact that the above plot is totally inappropriate for the purpose of showing the existence or absence of crossover distortion.

The OPA 2277 is not designed or recommended as a headphone amplfier by its manufacturer: OPA 2277 Manufacturer's spec sheet and application information Neither is the NJM 2068. NJM 2068 Manufacturer's spec sheet

Letsee: Irrelevant data about irrelevant parts.

<Shaking head>

This post has been edited by Arnold B. Krueger: Jun 7 2012, 12:41
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Arnold B. Kruege...
post Jun 7 2012, 12:48
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QUOTE (SoNic67 @ Jun 6 2012, 19:20) *
Sooo... In conclusion, we need 32 bit audio?


We live in a world with -60 dB distortion ears, -80 dB transducers, and -100 dB or more electronics. Obviously, we need better transducers and electronics. ;-)

The problem with the realism of our recordings has nothing to do with the vanishingly low nonlinear distortion in the equipment that we have been discussing.

The frequency response curves of the headphones we've been discussion would be hilarious, if we knew what their frequency response is supposed to be. The way that the FR of these headphones is being measured is controversial to say the least.

Then there is the problem of how to mic a live performance or mix studio recordings for realism. I see no convincing arguments that anybody really knows how to do this.

So, lets continue to obsess about the parts of the chain that are already way into diminishing returns and ignore the real problem, because the real problem must to too hard.

<shaking head>
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stephan_g
post Jun 8 2012, 17:29
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The real elephant in the room is acoustics, no doubt about that. I'm sure there are plenty of people out there with kilobuck amps running broadly radiating bookshelf speakers in rooms with all the wonderful acoustic properties of a gymnasium. (Translation: Ouch!)

I grew up on headphones, which is why I gravitated to a nearfield setup quite naturally. A bit of obsession with amplifiers isn't for nought in this world, since then at least you can guess why your speakers are hissing or buzzing at you (neither being uncommon among less expensive powered speakers, and even integrated amps/receivers still show noticeable hiss at times).
QUOTE (Arnold B. Krueger @ Jun 7 2012, 12:39) *
Key words: "He tried". Just two of how many 100s of similar parts.

Then you try finding a better-performing DIP-8 part for a line-level gain stage (Rf = 1k5, Rg according to gain, load = 10k) with low quiescent current, run on about +/-8.5 or +/- 12 V, respectively. Preferably one that doesn't cost an arm and a leg.
QUOTE (Arnold B. Krueger @ Jun 7 2012, 12:39) *
You seem to be oblivious to the fact that the above plot is totally inappropriate for the purpose of showing the existence or absence of crossover distortion.

It's hard to say for sure without having examined the opamp Samuel Groner style, that's true. Could be ordinary VAS common mode distortion, too. But my educated guess would be crossover distortion, since the output stages on low-power parts are always starved for current (as they tend to contribute a large portion of idle current draw) and output loading isn't exactly negligible in this particular application. Output loading was the main reason why e.g. a TL072 wasn't even in the game.
QUOTE (Arnold B. Krueger @ Jun 7 2012, 12:39) *
The OPA 2277 is not designed or recommended as a headphone amplfier by its manufacturer: OPA 2277 Manufacturer's spec sheet and application information Neither is the NJM 2068. NJM 2068 Manufacturer's spec sheet

Sorry, but you're way off-base on this one. We're talking about the line-level gain stage in the O2 amp, parameters as outlined above. It has to drive nothing but its feedback network (2k5 and 1k75, respectively) and the 10k volume pot. Driving headphones then is performed by a buffer using two NJM4556s in parallel.

NJM2068s aren't uncommon in mass-produced consumer and pro audio equipment. (Yamaha integrated amps and EMU and some Creative soundcards, to name a few.) They're about 5532 class devices, with lower voltage noise and current draw, but less good load driving. Pretty cheap, too. (And I never paid all that much attention to it, but they can also be had in the outdated SIP case, RoHS compliant even. Cool. I've been looking for a less noisy replacement for a 4558L in an old ProLogic receiver's gain stage for ages.)

I guess the most practical approach would be sacrificing a little noise performance and easing up on the feedback network values for reduced output loading. In practice, gain stage noise would commonly be attenuated significantly anyway, besides the vast majority of sources is considerably noisier than the gain stage. The design objective for the O2 was getting about as close to a wire with gain as possible without having to use any fancy boutique parts, and once in a while that involves pushing things a little.
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pdq
post Jun 8 2012, 17:58
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This is getting pretty far off from the original topic, but it is actually easy to turn an op amp into a class A device simply by applying a bias current to the output. You trade off power consumption for linearity, and you eliminate crossover distortion.
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