Just because I'm paranoid doesn't mean no one is following me.Conspiracy theorists of the world unite, someone out there is out to get you.
Just because I'm paranoid doesn't mean no one is following me.Conspiracy theorists of the world unite, someone out there is out to get you.
May 2014 - including papers on 1-bit audio and MLP among others. IIRC there were discussions going on around that time on the audibility of watermarking that most publishers were including on high resolution formats. Bob Stuart agreed with someone's argument that watermarks had to be audible in order to meet their goal of remaining identifiable when used with a notionally perfect lossy codec as anything inaudible would be discarded. I wonder if the original forum discussion still exists.The last issue on high resolution audio was in 2004, May's was an update.
Lots of wheel spinning here. While what say is true enough, but it never gets anywhere. None of this indicates timing is inadequate the way things are done. The timing is like 100 times better than needed, imagine the improvement if it were 10,000 times better. Well I'm not hearing it.Perfect!
So, my point was, that expectations based on the theory of Linear Time Invariant (LTI) systems, which are traditionally analyzed with the help of Fourier transform, are breaking down for Mammal Hearing System (MHS), which is neither linear nor time invariant.
In LTI, we care about durations, frequencies, sampling rates, and amplitudes in the time and frequency domains. In MHS, we also have to care about onset times, recuperation periods, levels of perceived loudness, inter-frequencies masking etc. "Four sounds a little louder than One" is not what LTI predicts, yet it makes perfect sense in the MHS framework.
The experiment illustrates at least two things:
(1) In MHS, perceived loudness depends not only on amplitude, but also on duration. This is a robust effect, linked to the hearing system's "slow" integrator, operating over tens of milliseconds. There also exists a less robust effect, not demonstrated by this experiment, due to "fast" integrator, operating over tens of microseconds, which makes a perceived onset time depend on the amplitude.
(2) Some of you will be able to differentiate between One and Two, some not. Or between Two and Three. Virtually everyone will be able to differentiate between One and Four. And this is for the "slow" integrator, considered rather consistent! Individual differences in functioning of the fast integrator are more difficult to elicit experimentally, yet they do exist.
Qualitatively, the number of dimension LTI operates in is smaller than the number of MHS dimensions. If we hold constant the value(s) in one or more of MHS dimension(s), we take the dimension(s) out of play, and then MHS behavior follows the LTI-predicted behavior more closely.
That's the general reason why "simple" music, mostly consisting of a small number of sinusoids slowly changing their amplitudes and frequencies over time, is more readily amenable to LTI analysis. The effects of the perceptual integrators fade away. Onset times matter less.
The "complex" music, with large number of sinusoids exhibiting fast and frequent onsets and fadeouts, chirps, and transients, is not as amenable to LTI analysis. The integrators play an important role in this case. We better preserve the information about the onset times more accurately.
Thanks. But that misses the last part of my sentence. I like to see @Sergei run his listening tests and post his observation and files. Then we can get somewhere as opposed to a theoretical discussion, or dismissal of the results after the fact because the test files were not this way or that way.See @miero thread on signal generation with sox which starts with synthesis of 1kHz tone files. The sox website has downloads for Windows and MacOS so you don't need to be using linux. I think these should produce the signals asked for, although you may want to change the sample rate and depth, and the attenuation from full scale:
Start with the irrelevant, end up with the repeated, ummm, misunderstanding. The perfect circle.
So it basically says you can use shorter reconstruction filters to get a result that is presumably at least equivalent to using sinc.Not sure if it has already come up, but there was an interesting paper on "Modern Sampling" in May's AES journal. It is an open access paper so free to download. Modern Sampling: A Tutorial.
What is your explanation of the four-tone experiment?
Can you link to a paper about the fast integrator?There also exists a less robust effect, not demonstrated by this experiment, due to "fast" integrator, operating over tens of microseconds, which makes a perceived onset time depend on the amplitude.
Fair point. With my devil's advicate hat on I'd argue telling you what you're supposed to hear before you hear it would affect what you hear. Is there a way to do a sealed post with the "here's what you should have heard and why" part that can only be opened some time later, or do we just have to trust people not to open spoiler tags in this sort of situation?Thanks. But that misses the last part of my sentence. I like to see @Sergei run his listening tests and post his observation and files. Then we can get somewhere as opposed to a theoretical discussion, or dismissal of the results after the fact because the test files were not this way or that way.
That we can do better.So what is the argument here?
A perfect square wave is a mathematical construct that doesn't exist in nature, let alone music. Every square-wave-like sound that actually exists, is bandwidth limited. And our perception is also bandwidth limited.... Here's an example: "1sec square wave at 0db"
That took 23 bytes and contains more data than any wav file ever can at any sample rate, since a square wave has unlimited BW.
I agree. 44-16 isn't quite fully transparent to all humans. But evidence suggests it doesn't take much more for the digital encoding & reconstruction to be fully transparent. I am all in favor of a higher standard, say 64-24 or whatever it would take to be fully transparent with a reasonable safety margin.... we can do better. ...
Electronic music can contain perfect square waves since its synthesizedA perfect square wave is a mathematical construct that doesn't exist in nature, let alone music. Every square-wave-like sound that actually exists, is bandwidth limited. And our perception is also bandwidth limited.
Electronic music can contain perfect square waves since its synthesized
Actually, it can't because electronics with infinite bandwidth don't exist. Also, an actual sound is made from changing air pressure. And the mathematical derivative of a perfect square wave is undefined at its transition point. That means a perfect square wave, to propagate as sound in the air, would require an infinite rate of change in air pressure, which is not physically possible.Electronic music can contain perfect square waves since its synthesized
Yes you can. Not in wav though.No it can't, because you can't encode for two different voltages at the same time point.
You can synthesize whatever you want, including a signal with infinite BW, just dont store it as a wav.Actually, it can't because electronics with infinite bandwidth don't exist.
The reason for this is that a square wave moves from one amplitude to another in no time at all. It is an infinitely short period of time between state A and state B. Because sampling works by giving you one sample per time interval, the best you can do is approximate it. It doesn't happen in nature either for obvious reasons. (And no, we are not dealing with quantum effects here haha)No it can't, because you can't encode for two different voltages at the same time point.