Since he referred to a time delay, I presumed there was probably insufficient open loop bandwidth?
There are a lot of little details to observe depending on your hardware.
A fun test is to play a Dolby AC3 encoded wav through SPDIF out. Since wav is actually just a container format you can pretty much put whatever you want in it.
If your software chain is bit perfect, a decoder will light up. If not you'll get awful noise.
A fun test is to play a Dolby AC3 encoded wav through SPDIF out. Since wav is actually just a container format you can pretty much put whatever you want in it.
If your software chain is bit perfect, a decoder will light up. If not you'll get awful noise.
Another way to look at the signal related phase shift is the classic video differential phase measurement. It's not difficult to do today. Just measure the change in phase with DC offset from zero. Good video amps are in the .1 degree for that test at 3 MHz. I suspect that's also true for in band signals in decent audio amps. DSL modulation would be hopeless if there was any phase shift since the information is coded in phase relationships.
I see lots of arm waving stuff about "dispersion" in AD-DA links. Another phase issue?. Again I have seen no published measurements showing what this is about and I seem to get very good waveform fidelity in audio AD-DA systems.
However phase coherence as a prime goal is really problematic for speakers. Inverted tweeters to make crossovers work are intrinsically non-coherent. Many tweeters have their acoustic center move with frequency which would be another issue. And the change off axis would further mess with coherency. Ed Simon can probably shed light on the impact of distance and humidity on frequency and phase in air paths. I have read many moons ago that human hearing is not phase sensitive except perhaps phase shifts that create amplitude changes or phase changing on a steady state signal (but that creates sidebands so its really a modulated signal).
I see lots of arm waving stuff about "dispersion" in AD-DA links. Another phase issue?. Again I have seen no published measurements showing what this is about and I seem to get very good waveform fidelity in audio AD-DA systems.
However phase coherence as a prime goal is really problematic for speakers. Inverted tweeters to make crossovers work are intrinsically non-coherent. Many tweeters have their acoustic center move with frequency which would be another issue. And the change off axis would further mess with coherency. Ed Simon can probably shed light on the impact of distance and humidity on frequency and phase in air paths. I have read many moons ago that human hearing is not phase sensitive except perhaps phase shifts that create amplitude changes or phase changing on a steady state signal (but that creates sidebands so its really a modulated signal).
A musical signal isn’t coherent in any of it’s attributes on a 2-3 seconds time scale, else it wouldn’t represent music (some music genres excluded).
I am not capable to comment if phase modulation has any place in the workings of an audio amplifier (except those due to the phase change at the edges of frequency shaping filters) but the time derivative of the phase modulation is frequency modulation, so there maybe a way for the effects to become measurable. How these can be segregated from the HD and IMD products is another problem.
https://en.wikipedia.org/wiki/Phase_modulation#/media/File:Phase-modulation.gif
George
Re time delay, how about speakers "all pass" linear distortion (time domain response), anyone dare to speak about time delays in properly designed electronics? You may test your ears on audibility of "all pass distortion".
Audibility of allpass phase distortion (test)
Audibility of allpass phase distortion (test)
Wasn't the idea here that "phase in-coherence" would create modulated noise? Thats how I understood it - and if so, I think it is something to pursue because I think that music correlated distorsion and noise are real party poopers.
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What setting is that? And how do you know if the user has used it - again, no controls & I never see anybody querying the setup used when a null result is reported from Foobar ABX?
Well if an internal soundcard is used & DS is resampling then it's not Foobar that doing 'something' to the files, is it?
Sure, why not - let's test for small impairments based on any system - no need to evaluate the system is capable - it's all good as long as a null result is returned - we'll only ask questions if a positive result is returned
This was directed at your comment from a technical perspective (and John's blind cheerleading of anything that is in opposition, no matter the merit, good or bad, of certain folks positions).
In short, it doesn't add up as something major (if at all), and would show up as grossly muddying transient events that are sitting close to the noise floor far beyond a stationary signal at the same level. Or show up as a sidelobe in a nulling/bridge experiment, where incoherence lights up as a imperfect cancellation. E.g gain up a signal, divide it back down and feed the gained and straight signal into two sides of a bridge. Does anything show up in burst tones that looks like a modulated noise effect?
Can you otherwise propose an experiment that would show this specific effect and be able to manipulate it to be greater and smaller with a modified differential pair?
I guess I'm wondering why, when I make experiments with events barely poking out of the noise floor, why I don't see the effect you're describing? If it's real, lesson learned and I'll need to be more careful about what parts I pick. If it's real and sits so far into the noise floor, cool effect I don't need to sweat, and if it not real, then that's fine as well. I'm just incredulous, it's my natural state of being.
I think you are misunderstanding the concept of what CG posted - noise floor modulation would not show up on an FFT as the noise floor itself is not specifically shown on an FFT (unless this is specifically being tested for) due to the way FFT works.
A similar fun test for bit perfection can be done with HD flag detection in suitable decoders
I can envisage two possible mechanisms for phase coherence drift being an audible issue:
- All sounds have a sound envelope which maps their various stage Attack, decay, sustain, release (ADSR) & this envelope spans a certain time. If this shape is disturbed from what it should be, because of the phase coherence changes in upper harmonics, it usually has an effect on timbre perception
- If phase coherence drift has a secondary effect in certain electronics that causes noise floor modulation then I can see an audible effect because auditory perception divides the sound into foreground & background (noise, room ambiance etc) - it generally doesn't pay much attention to the background, being more focused on the foreground sound - unless attention is drawn to this background - And this could be caused by the background noise changing, particularly if it is modulating with a pattern to it (signal correlated)
Yeah, how does one assure identical sounding files....my files sounding different experiment still holds with multiple witnesses since then.
Just this afternoon I showed two guys at work how my two short OTG USB cables cause USB headphones to sound different.
One of the guys is trialing my guitar cable tonight and should report back tomorrow including directional differences.
Dan.
That almost sounds like a threat
It's obvious that there is no kind of bias or suggestion going on there, nor with the USB cables for that matter It could be, if all were done on the same effectc, as rasing the sample size means rasing the power which means risk to commit a beta-error (i.e. failing to reject the null hypothesis although it is false) is seriously lowered.
In our case there are hundreds of tests of different EUTs (some of these might actually be undistinguishable while other might be not) are merged to state the "undistinguishability" of each and every effect, regardless of the specific conditions.
It should be imo obvious that a reasonable analysis couldn´t be done that way.
The interesting point (or surprising at least for me) was that the ABX protocol does not lower the risk of false positives, as the guard against that is the SL (or alpha error; leaving aside for simplicity the philosphical differences between concepts for the moment) nor raises the efficiency - it´s still the same dichotomous results as in case of an A/B - but actually lowers the rate of _correct_ positives, if using for analysis the test statistic of accumulated correct trials and the exact binomial test.
I don´t think that it was intentionally done, and unfortunately missed the opportunity to ask Arny about the process/discussions that led to the usage of this specific protocal back in the 1970/1980s.
Leventhal´s analysis would still hold true in case of using A/B protocols, as it was just based on Cohen´s power analysis (and recommendations) so would only reflect the specific problems presented by the ABX protocol by in the assumptions about lowering the percentage of correct answers.
I would advise to try HQPlayer.
You can do your Poo-ging
https://www.signalyst.com/consumer.html
It takes control of OS and shuts everything that doesn't need to be running.
It is pretty much the best quality audio you can get out of a computer ATM - the rest is hardware dependent.
It has far superior digital filters, modulators and OP's whatever format you desire.
DSD up to 1024 x and PCM up to 32 bit / 1.5MHz.
I know people who have achieved really high quality sound with a fairly cheap DAC, for example, by using HQP
and converting to DSD256, as such most of the DAC's filters etc are bypassed. There are many possibilities.
T
That looks like a lot of useful fun. Thank you ----
-Richard
Yes Mark,
It has convolution algorithms for DRC / can also be used with other
programs (Acourate) to implement active crossovers.
It has > 20 digital filters, 4 dithers, 4 noise shapers, 8 modulators (for PCM
-> DSD)
I think it should qualify it as a good program for digital Pooging
T
No worries Richard.
The designer is quite active on Computer Audiophile forum under the name of
Miska. He is super switched on. You can find a lot of useful information
there.
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