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GStreamer: Past, present, and futureGStreamer: Past, present, and futurePosted Oct 29, 2010 14:28 UTC (Fri) by nix (subscriber, #2304)In reply to: GStreamer: Past, present, and future by Spudd86 Parent article: GStreamer: Past, present, and future
A foolish question, perhaps, but what are all these far-past-48kHz audio samples targetted at? Not even dogs can hear that high. Bats, perhaps?
What hardware would you play them back on?
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GStreamer: Past, present, and future Posted Oct 29, 2010 16:04 UTC (Fri) by paulj (subscriber, #341) [Link]
I've wondered that too. Then recently I saw Monty from Xiph explain it in http://xiph.org/video/vid1.shtml - from about 11min in. Basically, it's cause you don't want any freqs > nyquist frequency for your sample rate to remain in the signal, or it'll cause aliasing. The ultra-high sample rates basically give you more margin for your low-pass filter, making them easier/cheaper to build.
GStreamer: Past, present, and future Posted Oct 29, 2010 18:34 UTC (Fri) by alankila (subscriber, #47141) [Link]
While that is important to folks that do sampling in the analog domain, once you have actually captured the signal, digital techniques can easily do the rest and represent artifact-free 44.1 kHz audio with frequency response cut at 20 kHz. There are other reasons to prefer high sampling rates during processing, such as the reduction of artifacts due to bilinear transform and having free spectrum available for spectral expansion before aliasing occurs due to nonlinear effects. Not all applications need those things, though.
However, the idea of consumer-level 96 kHz audio (as opposed to 44.1 kHz audio) is pointless. It may sell some specialized, expensive equipment at high markup for people who are into that sort of thing, but there appear to be no practical improvements in the actual sound quality.
GStreamer: Past, present, and future Posted Oct 29, 2010 23:32 UTC (Fri) by dlang (✭ supporter ✭, #313) [Link]
I really question the 'common knowledge' and 'studies show' statements that say that people can't tell the difference between a 20KHz signal playing with 44KHz samples, and played at 96KHz samples.
I remember when the same statements were being made about video, how anything over 24Hz refresh rate was a waste of time because we had decades of studies that showed that people couldn't tell the difference.
Well, they found out that they were wrong there, at 24Hz people stopped seeing things as separate pictures and saw things as motion instead, but there are still benefits to higher refresh rates.
I think the same thing is in play on the audio side.
not everyone will be able to tell the difference, and it may even be that the mythical 'average man' cannot, but that doesn't mean that it's not worthwhile for some people. It also doesn't mean that people who don't report a difference in a test won't see a difference over a longer timeframe of useage (for example, going from 30Hz refresh rates to 80Hz refresh rates appears to decrease eye strain and headaches for people over long time periods, even for people who can't tell the difference between the two when they sit down in front of the two side by side.
GStreamer: Past, present, and future Posted Oct 30, 2010 0:12 UTC (Sat) by jspaleta (subscriber, #50639) [Link]
I think the existence of inaudible dog whistles is serious blow against your hypothesis. We've had a much longer experience with audio frequencies near the edge of human perception than you would perhaps realize at first blush. Much of that history pre-dates any attempt at digital sampling. If 99.9% of people can't perceive dog whistles at 22 Khz, they aren't going to hear it played on their Alpine speakers in their car either.
Video framing on the other hand is relatively quite new...unless you count thumb powered flipbooks pen and paper animations.
-jef
GStreamer: Past, present, and future Posted Oct 30, 2010 15:01 UTC (Sat) by corbet (editor, #1) [Link] For all of our experience with audio, there was a small subset of us who were driven absolutely nuts by the weird high-pitched chirper things that the Japanese seem to like to put into doorways for whatever reason. Everybody else wondered what we were griping about. Some people hear higher than others.The other thing that nobody has pointed out: if you're sampling at 44KHz, you need a pretty severe low-pass filter if you want to let a 20KHz signal through. That will cause significant audio distortion at the upper end of the frequency range, there's no way to avoid it. A higher sampling rate lets you move the poles up much higher where you don't mess with stuff in the audio range. That said, I'm not such an audiophile that I'm not entirely happy with CD-quality audio.
GStreamer: Past, present, and future Posted Oct 30, 2010 14:42 UTC (Sat) by alankila (subscriber, #47141) [Link]
Let's just say that I remain skeptical.
Your specific example "20 kHz signal playing with 44 kHz samples, and played at 96 kHz samples" is a particularly poorly example. I assume you meant a pure tone signal? Such a tone can be represented by any sampling with a sampling rate > 40 kHz. So, 44 kHz and 96 kHz are equally good with respect to representing that signal. If there is any difference at all favoring the 96 kHz system, it arises from relatively worse engineering involved with the 44 kHz system -- poorer quality of handling of frequencies around 20 kHz, perhaps -- and not from any intrinsic difference between the representations of the two signals themselves.
Many people seem to think---and I am not implying you are one---that the way digital signals are converted to analog output waveforms occurs as if linear interpolation between sample points were used. From this reasoning, it looks as if higher sampling rates were better, because the linearly interpolated version of 96 kHz signal would look considerably closer to the "original analog waveform" than its 44 kHz sampling interpolated the same way. But that's not how it works. Digital systems are not interpolated by fitting line segments, but by fitting sin waveforms through the sample points. So in both cases, the original 20 kHz sin() could be equally well reconstructed.
GStreamer: Past, present, and future Posted Oct 30, 2010 15:04 UTC (Sat) by corbet (editor, #1) [Link] Sinc waveforms, actually (sin(θ)/θ) :)I knew all those signal processing classes would come in useful eventually...
GStreamer: Past, present, and future Posted Oct 31, 2010 11:27 UTC (Sun) by alankila (subscriber, #47141) [Link]
It is true that the resampling is typically done with convolving the signal with sinc, but the effect of this convolving is as if the interpolation had occurred with sin waveforms fit through the sampled data points.
GStreamer: Past, present, and future Posted Nov 2, 2010 4:02 UTC (Tue) by Spudd86 (guest, #51683) [Link]
Err, generally not sinc, it's usually windowed so as to have better PSNR
GStreamer: Past, present, and future Posted Nov 6, 2010 10:55 UTC (Sat) by alankila (subscriber, #47141) [Link]
True, true.
GStreamer: Past, present, and future Posted Nov 3, 2010 2:42 UTC (Wed) by cmccabe (guest, #60281) [Link]
I built an RC oscillator, chained it with an op-amp, and used it to drive a speaker. Then I cranked it up to the 20 kHz range. So I can tell you that I can hear above 22 kHz. We did "double-blind tests" where someone else was turning the sound on and off. I could always tell.
Some people can hear it, some people can't. Unfortunately, the "can't" people designed the Red Book audio format, apparently. I forget the exact frequency at which it became inaudible.
P.S. A lot of people have hearing damage because they listen to music at a volume which is too loud. You need earplugs at most concerts to avoid this.
GStreamer: Past, present, and future Posted Nov 3, 2010 21:03 UTC (Wed) by paulj (subscriber, #341) [Link]
Gah, yeah.. And even at the cinema - least I've suffered through uncomfortably loud movies at Cineworld in the UK a few times, and block my ears with fingers and/or shoulder.
GStreamer: Past, present, and future Posted Oct 29, 2010 19:20 UTC (Fri) by nicooo (guest, #69134) [Link]
A bottlenose dolphin would need 300 kHz samples
GStreamer: Past, present, and future Posted Oct 31, 2010 13:10 UTC (Sun) by nix (subscriber, #2304) [Link]
Ah! So the dolphins have been manipulating our video format work!
Mice might need it too, for their supersonic squeaks of delight.
Perhaps... Douglas Adams was right?
GStreamer: Past, present, and future Posted Oct 29, 2010 23:12 UTC (Fri) by dlang (✭ supporter ✭, #313) [Link]
it's not that the audio frequencies are > 48KHz (the audio frequencies are almost certainly below 20KHz)
It's that using more samples to represent the data makes the resulting audio cleaner.
remember that you aren't recording the frequency, you are recording the amplitude at specific periods. the more samples you have, the cleaner the result.
GStreamer: Past, present, and future Posted Oct 29, 2010 23:13 UTC (Fri) by dlang (✭ supporter ✭, #313) [Link]
by the way, the nyquist limit isn't the limit for where things sound good, it's the limit beyond which there is no hope of getting anything resembling the correct result.
GStreamer: Past, present, and future Posted Oct 30, 2010 0:09 UTC (Sat) by gmaxwell (subscriber, #30048) [Link] There is lots of misinformation on this subject out there.Given unlimited precision samples a signal which has no energy above the the system nyquist is _perfectly_ re-constructable, not just "good". If the signal does have energy above the nyquist then it's not "no hope": the system is under-determined and there are a number of possible reconstructions. Of course, we don't sample with infinite precision but increasing the sampling rate is a fairly poor way of increasing the SNR for lower frequencies if thats your coal. For example, a 1 bit precision 3MHz process can give as much SNR in the 0-20kHz range as a 20 bit 48khz process but it takes about 3x the bitrate to do so. 24bit converters with >110dB SNR are readily and cheaply available. These systems can represent audio as loud as 'dangerously loud' with the total noise still dwarfed by the thermal noise in your ear and the room around you. It's effectively infinite precision. Heck, given reasonable assumptions (that you don't need enough dynamic range to cover hearing damage to the faintest discernible sounds) well mastered CDDA is nearly so too. There has been extensive study of frequency extension into the ultrasonic, and none of the studies I've seen which weren't obviously flawed could support that hypothesis. If this perception exists it is so weak as to be unmeasurable even in ideal settings (much less your common listening environment which is awash in reflections, distortions, and background noise). There also is no real physiological basis to argue for the existence of significant ultrasonic perception Heck, if you're posting here you're probably old enough that hearing is mostly insignificant even at 18kHz (HF extension falls off dramatically the early twenties for pretty much everyone) much less higher. But hey if you want to _believe_ I've got some dandy homeopathics to sell you.
GStreamer: Past, present, and future Posted Oct 30, 2010 0:36 UTC (Sat) by dlang (✭ supporter ✭, #313) [Link]
Quote: Given unlimited precision samples a signal which has no energy above the the system nyquist is _perfectly_ re-constructable, not just "good".
I disagree with this statement. something can be reproduced, but not neccessarily _perfectly_
also, any time you have more than one frequency involved, they are going to mix in your sensor, and so you are going to have energy above this frequency.
sampling faster may not be the most efficient way to get better SNR, but it's actually much easier to sample faster than to sample with more precision.
using your example, setting something up to sample 1 bit @ 3MHz may be far cheaper than setting up something to sample 20 bits @ 48KHz. In addition, the low-precision bitstream may end up being more amenible to compression than the high precision bitstream. with something as extreme as the 1bit example, simple run-length encoding probably will gain you much more than a 3x compression ratio. That's not to say that a more sophisticated , lossy, compression algorithm couldn't do better with the 20 bit samples, but again, which is simpler?
I am in no way saying that people hear in the ultrasonic directly, However I am saying that some people listening to a 15KHz sine wave vs a 15KHz square wave will be able to hear a difference.
GStreamer: Past, present, and future Posted Oct 30, 2010 14:21 UTC (Sat) by alankila (subscriber, #47141) [Link]
> I disagree with this statement. something can be reproduced, but not neccessarily _perfectly_
This may be confusing two ways to look at it: as mathematical issue, or as engineering problem. Mathematically the discrete representation and the analog waveform are interchangeable: you can get from one to the other. The quality of the conversion between the two can be made as arbitrarily high as you desire -- typically design targets are set beyond assumed limits of human perception.
>also, any time you have more than one frequency involved, they are going to mix in your sensor, and so you are going to have energy above this frequency.
Intermodulation distortion can generate extra tones, and depending on how strong the effect is, they may even matter. Such nonlinearities do not need more than one frequency, though.
This is normally an undesirable artifact, and our ADC/DACs have evolved to a point where they are essentially perfect with respect to this problem. In any case, from viewpoint of a digital system, artifacts that occurred in the analog realm are part of the signal, and are processed perfectly once captured.
> I am in no way saying that people hear in the ultrasonic directly, However I am saying that some people listening to a 15KHz sine wave vs a 15KHz square wave will be able to hear a difference.
The amusing thing is that a 44.1 kHz representation of a 15 kHz square wave will look identical to a 15 kHz sin wave, because none of the pulse's harmonics are within the passband of the system. Do you happen to have a reference where a system such as this was tested with test subjects so that it would be possible to understand how such a test was conducted?
GStreamer: Past, present, and future Posted Oct 30, 2010 16:27 UTC (Sat) by magnus (subscriber, #34778) [Link] Given unlimited precision samples a signal which has no energy above the the system nyquist is _perfectly_ re-constructable, not just "good".Theoretically, you don't only need unlimited precision on each sample, you also need to have an infinite number of samples, from time -∞ to +∞, to perfectly reconstruct the original signal. In practice though, audio signals will have some information (harmonics etc) at higher frequencies and no filters (not even digital ones) can be perfectly brick-wall shaped, so some aliasing will occur plus you will have some attenuation below the Nyqvist frequency. Sampling at 96 kHz might (if well designed) give you a lot more headroom for these effects. I have no experience with 96 kHz audio so I don't know if this is actually audible or just theory+marketing. Since human hearing is non-linear it's also possible that people can pick up harmonics at higher frequencies even if they can't hear beeps at these frequencies. The only way to know is double blind-testing I guess...
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