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GStreamer: Past, present, and future

GStreamer: Past, present, and future

Posted Oct 29, 2010 16:04 UTC (Fri) by paulj (subscriber, #341)
In reply to: GStreamer: Past, present, and future by nix
Parent article: GStreamer: Past, present, and future

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.


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GStreamer: Past, present, and future

Posted Oct 29, 2010 18:34 UTC (Fri) by alankila (guest, #47141) [Link] (10 responses)

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 (guest, #313) [Link] (9 responses)

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] (1 responses)

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 (guest, #47141) [Link] (4 responses)

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] (3 responses)

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 (guest, #47141) [Link] (2 responses)

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 (subscriber, #51683) [Link] (1 responses)

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 (guest, #47141) [Link]

True, true.

GStreamer: Past, present, and future

Posted Nov 3, 2010 2:42 UTC (Wed) by cmccabe (guest, #60281) [Link] (1 responses)

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.


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