Very interesting. May I ask how that was measured? I'd imagine applying a stimulus voltage via a series resistor to the output of the regulator and connecting a spectrum analyser across the regulator output?
Did anybody try that with stimulus other than a sine? Multi tone? Music?
Did anybody try that with stimulus other than a sine? Multi tone? Music?
Jack didn't provide detailed info about the test setup. It was something he did as sort of as an afterthought subsequent to publishing his regulator comparison article in Linear Audio. Linear Audio | your tech audio resource
Here it is a comparison chart, by Jack, from Linear Audio, maybe this is what was asked for?
And don't forget, draw a flat line at 100uOhm (10^-4), ~up to 100kHz. That is the LDO performance on that chart.
Ciao, George
http://linearaudio.net/sites/linearaudio.net/files/V4 JW F8.pdf
And don't forget, draw a flat line at 100uOhm (10^-4), ~up to 100kHz. That is the LDO performance on that chart.
Ciao, George
http://linearaudio.net/sites/linearaudio.net/files/V4 JW F8.pdf
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Possibly some more of interest: iFi iDSD Micro DSD512 / PCM768 DAC and Headphone Amp. Impressions, Reviews and Comments. | Page 475 | Headphone Reviews and Discussion - Head-Fi.org
An opinion was expressed that is out of what mainstream EEs normally tend to believe about audio design and measurements. Often out-of-mainstream opinions are scoffed at and sometimes ridiculed by mainstream believers.
Skill at careful listening takes time and effort to pay off, but it can be valuable if pursued systematically. The easiest thing to hear is probably IMD on something like choral vocals or even a vocal duet. Do you hear distinct, individual voices or one sound that is a multiple voices muddled and intermodulated together? To what extent is the mixing-together? Try to memorize that sound long enough to compare with something else. Then try focusing on just that one factor when listening to different dacs and amplifiers, and you will start to understand a bit of what is possible.
An opinion was expressed that is out of what mainstream EEs normally tend to believe about audio design and measurements. Often out-of-mainstream opinions are scoffed at and sometimes ridiculed by mainstream believers.
Skill at careful listening takes time and effort to pay off, but it can be valuable if pursued systematically. The easiest thing to hear is probably IMD on something like choral vocals or even a vocal duet. Do you hear distinct, individual voices or one sound that is a multiple voices muddled and intermodulated together? To what extent is the mixing-together? Try to memorize that sound long enough to compare with something else. Then try focusing on just that one factor when listening to different dacs and amplifiers, and you will start to understand a bit of what is possible.
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What listening set up should be used so that placebo and other external bias don't taint the result? Or did you even compensate for such factors when you did your listening comparisons?
JohnW,
Given your stated views on HQ Player, it might be interesting to take a look at the following regarding the very expensive Chord dacs and HQ Player: HQPlayer -vs- Chord WTA1 - HQ Player - Roon Labs Community
Given your stated views on HQ Player, it might be interesting to take a look at the following regarding the very expensive Chord dacs and HQ Player: HQPlayer -vs- Chord WTA1 - HQ Player - Roon Labs Community
I should probably mention that Markw4's audibility claims should be ignored because they are only based on casual subjective anecdotes.
That ifi post would most likely have been by Thorsten Loesch, the head
designer behind their products, also did AMR. He used to knock around these
woods frequently but like many, probably just didn't have time to deal with
the BS. Very knowledgeable and achieved designer. Outspoken, often
controversial but solid diverse experience in HiFi and Pro. Apparently used to
be a pretty wild guy in his youth. I liked him and his contributions here.
TCD
ifi now has some association with John Curl. Could be they will be doing something together at some point. As you may know, he consults with more than one company although Parasound seems to be the main client.
Anyway, ifi's association with John is would probably be another clue as to their approach to sound.
Anyway, ifi's association with John is would probably be another clue as to their approach to sound.
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Well I can only comment on my own personal experience with HQ Player and i'm not impressed - it does make me wonder if the filters / modulators are designed by ear or purely technically - I suspect the later..
March 12th I have a couple of guys popping over from the UK with the Chord Dave / Hugo upscaler / HugoII DAC for a weekend listening session in the lab, so looking forward to this...
Last night was our first listen to the little Discrete DAC design which sounded pretty amazing... over the next couple of weeks I'll tweak the design to see how far it can be pushed...
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Very interesting news. Hope it turns out great! We could certainly use some better dacs than we have now.
Also, will be interested in your impressions of the Chord products after the March listening session. Some people believe they are the best sounding dacs made today in terms of space, ambiance, sound field, and euphorically over all.
If you don't mind, there is something else I would like to ask you about: You said before you have a very elaborate setup for measuring jitter. Also, I presume you are familiar with Jtest and its origin. The question is: How effective would say Jtest is for evaluating all of jitter that is audible?
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Interesting question Mark! I generated such a signal with a small python script the other day myself.
a) to be sure it's correct vs. some random file I downloaded from the net and
b) to understand what the j-signal actually looks like.
What I didn't understand is why this j-signal should show effects of DAC clock jitter in the measured output FFT.
a) to be sure it's correct vs. some random file I downloaded from the net and
b) to understand what the j-signal actually looks like.
What I didn't understand is why this j-signal should show effects of DAC clock jitter in the measured output FFT.
Julian Dunn jitter test signal had been developed exclusively for SPDIF transmission protocol...
True.
Its the jitter test that is widely available (and cheap), so its what people use. If they don't anything they declare jitter a non-issue. Another case of Kahneman's WYSIATI one might ask? A machine for jumping to conclusions
Its the jitter test that is widely available (and cheap), so its what people use. If they don't anything they declare jitter a non-issue. Another case of Kahneman's WYSIATI one might ask? A machine for jumping to conclusions
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So I take it, as I was asking "dumb" questions I'm excluded from the "Kahneman's WYSIATI" crowd? 😉
Seriously. If the DPLL works shouldn't jitter on the incoming signal be a non issue, as long as it fits in the DPLL bandwidth? The ESS does ASRC anyways, DPLL just adjusts resampling factor according to <incoming bitrate> vs <outgoing bitrate>, right? If ASRC works correctly, the translation factor shouldn't matter, correct?
So why would we see any artifacts regarding jitter on the clock input pins in the output signal which is dependent on the incoming signal the DAC has to convert? After all the j-signal is just the smallest 250Hz signal (not even 1 LSB, its 1/2 LSB, not +/- 1LSB, but 0/+1 LSB) superimposed on a 12kHz sine wave signal with -6dB amplitude. Or at least the digital PCM representation of such a 12kHz analogue signal. Talking about a 48kHz sampled signal that is.
I just rephrased my doubts about such a signal because I still don't understand it, and yes, I read the paper MarkW4 cited. But this speaks of a digital input signal who's timing information is directly translated into the timing of the output signal. Being a software engineer I cannot see how this jitter on the input signal could influence the output signal if there's a correctly working DPLL. Nor can I see the direct relation between the input signal, the output signal and the clocking signal given that there's a rather complicated modulation process going on and input signal clock jitter is non existent after ASRC/DPLL.
I'd suspect that the modulator rather produces higher frequency "noise" due to clock input jitter.
After all, this is largely formulated as questions rather than "I know what's going on" type of statement. Sadly I don't have the math background to prove all this in any type of formulae. Any DSP gurus here to back up my statements or able to falsify me? I'd be happy either way, I just want to learn.
Seriously. If the DPLL works shouldn't jitter on the incoming signal be a non issue, as long as it fits in the DPLL bandwidth? The ESS does ASRC anyways, DPLL just adjusts resampling factor according to <incoming bitrate> vs <outgoing bitrate>, right? If ASRC works correctly, the translation factor shouldn't matter, correct?
So why would we see any artifacts regarding jitter on the clock input pins in the output signal which is dependent on the incoming signal the DAC has to convert? After all the j-signal is just the smallest 250Hz signal (not even 1 LSB, its 1/2 LSB, not +/- 1LSB, but 0/+1 LSB) superimposed on a 12kHz sine wave signal with -6dB amplitude. Or at least the digital PCM representation of such a 12kHz analogue signal. Talking about a 48kHz sampled signal that is.
I just rephrased my doubts about such a signal because I still don't understand it, and yes, I read the paper MarkW4 cited. But this speaks of a digital input signal who's timing information is directly translated into the timing of the output signal. Being a software engineer I cannot see how this jitter on the input signal could influence the output signal if there's a correctly working DPLL. Nor can I see the direct relation between the input signal, the output signal and the clocking signal given that there's a rather complicated modulation process going on and input signal clock jitter is non existent after ASRC/DPLL.
I'd suspect that the modulator rather produces higher frequency "noise" due to clock input jitter.
After all, this is largely formulated as questions rather than "I know what's going on" type of statement. Sadly I don't have the math background to prove all this in any type of formulae. Any DSP gurus here to back up my statements or able to falsify me? I'd be happy either way, I just want to learn.