HydrogenAudio

Hydrogenaudio Forum => Scientific Discussion => Topic started by: stranhoROX on 2011-10-16 18:03:03

Title: What's the point of higher sampling rates in audio?
Post by: stranhoROX on 2011-10-16 18:03:03
Could someone explain to me why there are some recordings with sample rates as high as 192kHz? If most of us hear up to 20kHz, wouldn't 44.1kHz or even 48kHz be enough? Or is there other practical aspects besides boosting maximum frequency in higher sampling rates I am not aware of?
Title: What's the point of higher sampling rates in audio?
Post by: C.R.Helmrich on 2011-10-16 18:18:32
Good question! My answer: People record with 96 or 192 kHz simply because they can and because they believe it sounds better. Yes, 48 kHz is enough, and no solid scientific evidence on the perceptual superiority of higher sampling rates has been presented to date. There were some experiments by Dr. Kunchur, for example, but these involved synthetic signals and actually provided more questions than answers. You can find plenty of discussions in older threads here on HA.

Chris
Title: What's the point of higher sampling rates in audio?
Post by: Brand on 2011-10-16 18:29:44
Yeah, this has been discussed a lot on HA.

But in short, I think that if (!) we assume that people can't hear frequencies over 20 kHz, then 44.1 kHz should be enough when listening to the final product.

However, using higher sample rates can indeed have some advantages during production. Off top of my head: pitch shifting of samples is more precise at higher sample rates, you get lower latency and some synths and effects might sound better (the good ones should sound good always and oversampling can be used).
Title: What's the point of higher sampling rates in audio?
Post by: benski on 2011-10-16 20:38:18
There are some advantages to recording, mixing and producing in higher samplerates, similar to using higher bit depths.  In particular, non-linear digital audio manipulation will produce overtones according to the order of the equation. 

For example, tape saturation distortion has an effect roughly equivalent to output=log(input).  If we use the fourth order taylor series expansion of log(x)
(http://upload.wikimedia.org/wikipedia/en/math/5/9/d/59d343fa4438505cc72e6e1abb14358d.png)

We will produce distortion overtones at 4x the frequency of the existing frequencies in the input.  Using a higher sample rate can help prevent or reduce aliasing due to this effect.  Sure, you can upsample before the effect, and then downsample after, but it might be easier (less CPU) to just have the whole processing chain be 192kHz or 384kHz, and use a lowpass filter before the effect to lop off any input spectra that would lead to aliasing.

Another advantage is that resonant IIR filters (the kind used in synthesizer filters and guitar effects) typically have non-linear responses between theoretical Fc and actual Fc (due to frequency warping and the non-linear effects of resonance, especially if you use a sub-sample delay in the feedback path to compensate for phase differences, see section 5.3 of http://dafx04.na.infn.it/WebProc/Proc/P_061.pdf (http://dafx04.na.infn.it/WebProc/Proc/P_061.pdf)).  It might be mostly linear, however, to Fs/8, so using a higher sample rate allows the usable audio band to have a linear (and predictable) response to the filter controls.
Title: What's the point of higher sampling rates in audio?
Post by: saratoga on 2011-10-16 21:32:57
However, using higher sample rates can indeed have some advantages during production. Off top of my head: pitch shifting of samples is more precise at higher sample rates, you get lower latency and some synths and effects might sound better (the good ones should sound good always and oversampling can be used).


True, but if you can always record at 48k, upsample to 192k for processing, and then downsample again for distribution if you think your software has problems (really bugs...) at lower sample rates.  So thats not really a reason to record at 192k, only a reason you might want to use it to work around problems with software.
Title: What's the point of higher sampling rates in audio?
Post by: ExUser on 2011-10-16 21:50:10
Trivial case where higher-sampling rates can become audible: Slowing the recording down. For playback at 1/N times the original speed, you need N times the sampling rate to capture all audible frequencies.
Title: What's the point of higher sampling rates in audio?
Post by: Notat on 2011-10-16 22:48:32
+1 to saratoga

Canar's scenario is applicable to recordings of bats. I don't think you generally want formerly ultrasonic material to become audible on a slowed recording.

There have been demonstrations where listeners can reliably distinguish 48 and 96 kHz recordings. There are generally more variables in play (e.g. ADC behavior) than just sample rate in these trials.

Production engineers like to think of their process as an information funnel. They capture enormous amounts of information and, as they work, whittle it down to what they want people to hear in the final product. In this context, many are comforted by recording equipment and procedures that capture much more than necessary.
Title: What's the point of higher sampling rates in audio?
Post by: krabapple on 2011-10-17 03:57:42
There have been demonstrations where listeners can reliably distinguish 48 and 96 kHz recordings. There are generally more variables in play (e.g. ADC behavior) than just sample rate in these trials.


Links?
Title: What's the point of higher sampling rates in audio?
Post by: Juha on 2011-10-17 04:14:30
Still there are plenty of audiointerfaces which won't give flat enough frequency response when running @44.1kHz (nor @48kHz) mode but, does it when set to 96kHz or higher.

(IIRC, many plug-ins (VST) works @192kHz (or maybe even at higher samplerate) internally.)

Juha
Title: What's the point of higher sampling rates in audio?
Post by: 2Bdecided on 2011-10-17 10:07:57
@benski,

...but if you're simply relying on the high sample rate to keep processing aliases out of the audible band, you need to go into the MHz for some operations. Or do them in a smarter way. I agree that keeping the base Nyquist limit a little higher when you're resampling through each of many different DSPs can help in an objective and measurable sense: it can mitigate some of the effects of careless processing and/or resampling in each DSP module. This can be audible is the processing is bad enough - e.g. really trashy resampling. But is the improvement enough to turn trash into perfection? No.

C.R.Helmrich's response is right IMO - people do it because they can, and they believe it sounds better. Despite an almost total lack of any corroborating evidence. Plus a few people will pay more for a recording at a higher sample rate, for the same reason.

EDIT Examples:
http://www.soundkeeperrecordings.com/purchase.htm (http://www.soundkeeperrecordings.com/purchase.htm)
http://www.paulmccartney.com/bandontherun/gbp.html (http://www.paulmccartney.com/bandontherun/gbp.html)

I guess the question becomes: why wouldn't you do it? If it costs nothing, and sells one more copy, or lets you charge some people more, or gets you bragging rights in some circles, then it's "worth it" - even if the technical benefit is zero.

Cheers,
David.
Title: What's the point of higher sampling rates in audio?
Post by: benski on 2011-10-17 11:47:11
@benski,

...but if you're simply relying on the high sample rate to keep processing aliases out of the audible band, you need to go into the MHz for some operations. Or do them in a smarter way. I agree that keeping the base Nyquist limit a little higher when you're resampling through each of many different DSPs can help in an objective and measurable sense: it can mitigate some of the effects of careless processing and/or resampling in each DSP module. This can be audible is the processing is bad enough - e.g. really trashy resampling. But is the improvement enough to turn trash into perfection? No.
David.


Sure, but for polynomial equations (or polynomial approximations of transcendental functions), the aliasing is predictable.  Because multiplying two signals M and N will produce a sideband at M+N, each polynomial order will require an equivalent increase in sampling frequency. Certainly there are other aliasing-reducing techniques such as using an all-pass for fractional delays and minBLEP for waveform generation, but for resonant IIR filters, especially, the higher sample rate can really make a difference (and admittedly most plugins probably upsample/downsample internally)

Note that I'm not at all trying to imply that high sample rate in final, delivered, consumer audio is justified.  I'm just pointing out that doing the mixing and production at high sample rate is worthwhile.  And if you already have a 192kHz master, why not try to sell it for a few bucks more?
Title: What's the point of higher sampling rates in audio?
Post by: Northpack on 2011-10-17 16:34:06
And if you already have a 192kHz master, why not try to sell it for a few bucks more?

Because you don't want to make your money by fooling people?
Title: What's the point of higher sampling rates in audio?
Post by: dhromed on 2011-10-17 17:05:08
And if you already have a 192kHz master, why not try to sell it for a few bucks more?


Great, so why not sell the 44.1 version for a few bucks less?
Title: What's the point of higher sampling rates in audio?
Post by: knutinh on 2011-10-17 18:34:17
I'm just pointing out that doing the mixing and production at high sample rate is worthwhile.

Sounds to me like a sound engineer wasting bandwidth, storage and money assuming that he is better at analyzing dsp algorithms than the guy designing the dsp algorithm was. If a given algorithms works better at 192kHz than 44.1kHz, why didnt the algorithm designer just do the resampling internally? Or redesign the algorithm so that it worked satisfactory at the lower rate?


Not saying that there could never be a situation where what you are saying makes sense for the well-informed technician, but I get the feeling that we could have better sounding CDs and radio transmissions if sound engineers stopped thinking of sample rates and focused on how to _use_ their EQs and dynamic processors instead.

-k
Title: What's the point of higher sampling rates in audio?
Post by: Wombat on 2011-10-17 18:45:30
And if you already have a 192kHz master, why not try to sell it for a few bucks more?


Great, so why not sell the 44.1 version for a few bucks less?

They do sell both versions sometimes at least. Reading the reviews of such sales and how much better the 96kHz version sounds and with 192kHz even the micro-macro details come to shine you have to wonder.

Enough said about PCM. PCM is so yesterday. Real audiophiles need DSD these days. I see the market for this slowly coming.
All ways of playing PCM and forcing DSD to be PCM on PCM hardware is just bad!! You need to buy DSD DACs and DSD recordings, this is the real deal today
----Sarcasm mode off----

Sorry if some think this is OT but these days DSD is part of the Hires delusion.

I lately stumbled across an eye opening thread here on Hydrogen that should be read carefully. Much knowledge in these few pages!
http://www.hydrogenaudio.org/forums/index....=37717&st=0 (http://www.hydrogenaudio.org/forums/index.php?showtopic=37717&st=0)
I even have to quote this cool sentense of David here:
"...ultrasonic information is somehow audio - this is plain and simple 'all the ultrasonic junk changes the way imperfect audio equipment works'"
Title: What's the point of higher sampling rates in audio?
Post by: HTS on 2011-10-18 03:45:10
Do microphones have to be good enough to take advantage of the high sample rates? Otherwise why not just record at 44.1 and up sample to 192khz for release?

http://www.lavryengineering.com/documents/...ling_Theory.pdf (http://www.lavryengineering.com/documents/Sampling_Theory.pdf)

This guy says that microphones don't "respond" to stuff at and over 96khz, and musical instruments don't produce overtones that go that high.
Title: What's the point of higher sampling rates in audio?
Post by: saratoga on 2011-10-18 03:47:28
Otherwise why not just record at 44.1 and up sample to 192khz for release?


Indeed.  Why precisely processing needs to be coupled to the recording and distribution is beyond me.  Its very easy to change the sampling rate, and all kinds of DSP processing do just that in practice. 
Title: What's the point of higher sampling rates in audio?
Post by: FreaqyFrequency on 2011-10-18 04:12:10
There are, in fact, at least two omni capsules out there now which extend well beyond 20 kHz, both made by Microtech Gefell: MK222, and MK221.  The 222 is flat within +/- 1 dB from 0.5 Hz to 80 kHz, and the 221 is the same spec from 10 Hz to 80 kHz.  So there exist microphones do now have the ability to capture those sounds, but it will remain true that human beings cannot hear them.

The only usage I could see for high sample rates is for recording and analysis of high frequency sounds in nature.  As Lavry explains (quite well) in that white paper, there is just no justification for such high rates when working with a human audience.  All filtering can be dealt with benignly at the antialiasing stage using DSP, and after that there should be no issue getting proper, accurate results with 44.1 as a final delivery format.
Title: What's the point of higher sampling rates in audio?
Post by: Dirk95100 on 2011-10-18 05:06:45
I'm just pointing out that doing the mixing and production at high sample rate is worthwhile.

Sounds to me like a sound engineer wasting bandwidth, storage and money assuming that he is better at analyzing dsp algorithms than the guy designing the dsp algorithm was. If a given algorithms works better at 192kHz than 44.1kHz, why didnt the algorithm designer just do the resampling internally? Or redesign the algorithm so that it worked satisfactory at the lower rate?


This.



Mics that have a high frequency responce, also produce more noise. So total resolution does not increase with just adding frequency responce. Its a thing explained by information theory. You can not have the cake and eat it.
Title: What's the point of higher sampling rates in audio?
Post by: hellokeith on 2011-10-18 07:14:57
Seems like a worthwhile experiment would be to run a set of samples - all from the same source but at different sampling rates - through a series of mixing, eq, fx, etc and then examine if the highest sampling rate sample has fewer issues from the series of processes.
Title: What's the point of higher sampling rates in audio?
Post by: unekdoud on 2011-10-18 08:01:09
Could someone explain to me why there are some recordings with sample rates as high as 192kHz? If most of us hear up to 20kHz, wouldn't 44.1kHz or even 48kHz be enough? Or is there other practical aspects besides boosting maximum frequency in higher sampling rates I am not aware of?


I doubt my answer is accurate, but the first thing that came to mind is that the same 192kHz signal can be downsampled to both 44.1 and 48kHz without creating artifacts. If the audio was recorded at 44.1kHz and resampled to 48kHz the result would be different from just recording at 48kHz.
Title: What's the point of higher sampling rates in audio?
Post by: 2Bdecided on 2011-10-18 11:26:46
And if you already have a 192kHz master, why not try to sell it for a few bucks more?

Because you don't want to make your money by fooling people?
It's retailing. It's based on fooling people, and people want to be fooled. They want to believe expensive food tastes better, for example. They want to read about all the reasons it tastes wonderful. The sunshine in the fields. The beautiful maidens who picked each crop by hand. Reading about all those reasons will make the food taste better to them - even though none of those reasons changes the actual taste of the food at all.

The biggest problem IMO is when there's only fooling, and real progress disappears where it might otherwise have been possible and beneficial. Also, where outright lies are told.

I don't mind a free market where several quality levels are offered; I can try them, and pay for the one I find acceptable.

I know certain people will claim that they hear differences I cannot. However, with all these quality levels available, I can set up rigorous ABX testing


Anyway, back to reality: if you are making high quality recordings, and some of the people purchasing your high quality recordings want to pay you $10 extra for a 192kHz version, why on earth wouldn't you make one available? As long as it doesn't make the quality worse, and doesn't cost you more than the financial return, it's really not a problem if people want to pay more for no tangible benefit.

I think it should be quite clear to anyone here that it's of no audible benefit what-so-ever, but it may create an excuse (that the accountants will accept) to create better (re-)masters. Which will then be used for the 44.1kHz version that can now be bought for a bargain price.  Everyone's a winner.

Does this explain why parts of the industry are heading down this route? And those who should speak out, don't? Granted, it could be The Emperor's New Clothes all over again, but I suspect many people know exactly what they're doing.

The downside is that we don't get proper surround. Though some people are still quietly working on that too.

Cheers,
David.
Title: What's the point of higher sampling rates in audio?
Post by: Dirk95100 on 2011-10-18 12:33:12
Anyway, back to reality: if you are making high quality recordings, and some of the people purchasing your high quality recordings want to pay you $10 extra for a 192kHz version, why on earth wouldn't you make one available? As long as it doesn't make the quality worse, and doesn't cost you more than the financial return, it's really not a problem if people want to pay more for no tangible benefit.

I think it should be quite clear to anyone here that it's of no audible benefit what-so-ever, but it may create an excuse (that the accountants will accept) to create better (re-)masters. Which will then be used for the 44.1kHz version that can now be bought for a bargain price.  Everyone's a winner.


Not necessaraly so....

A general rule of measurements is that accuracy and measurement time are related. Low measuring times means low accuracy and high accuracy means long measuring times.
So there comes a time that while you think you increase the amount of information (increase sampling rates), you are actualy decreacing the amount of information. Again information theory explains this all.
Title: What's the point of higher sampling rates in audio?
Post by: Cavaille on 2011-10-18 13:34:50
Guys, you all disappoint me. The point of higher sampling rates is capturing at best quality possible, regardless if the quality improvement to 16/44.1 is only theoretical. The point is: you can always dowsample later. Really, all your shiny, biting and ironic arguments would have made sense ten years ago but in present times it´s completely superfluous: nowadays we have plenty of HDD space, more than capable hardware, fast processing (even 32/384.000 can be edited and processed very fast if I may be allowed to indulge in that exaggeration fantasy) and extremely fast internet connections. It just doesn´t make sense to use 16/44.1 or lossy compression nowadays and even less sense to promote it as the only format worthwile.

Did it occur to you that for example the Grammy foundation and the Deutsche Grammophon are doing backups of their precious analogue tapes with 24/192 before the tapes degrade and are lost? They are doing this because for example the IASA recommends it. They also don´t care if the quality is better or not, they just want to capture everything. They apparently do it according to the motto 'Better safe than sorry'. Are you calling them stupid or deluded? They obviously know the times they are in and behave accordingly, they exploit the technical possibilities they have and don´t stand still. Which is what has been missing here since a few years now. You guys are still doing comparisons of lossy formats at 96 kBit/s! Why? The world won´t need them for much longer. Furthermore, you make fun of people who don´t share your opinion. Granted, some of them deserve it. But for a few years now everyone who promotes something like higher samplerates, 24 bit, hell, anything else that deviates from established norms is treated like a little dumb child. You have become close minded people, living in their own secluded world that is becoming smaller and smaller every day.

This forum clearly has lost its edge, I really feel that the world has turned - but without you.

I will no longer participate here (you´ll applaud it, I´m sure) and I know that sentences will be put out of context. If there would be a delete-button for my profile I´d use it.
Title: What's the point of higher sampling rates in audio?
Post by: 2Bdecided on 2011-10-18 14:46:18
Anyway, back to reality: if you are making high quality recordings, and some of the people purchasing your high quality recordings want to pay you $10 extra for a 192kHz version, why on earth wouldn't you make one available? As long as it doesn't make the quality worse, and doesn't cost you more than the financial return, it's really not a problem if people want to pay more for no tangible benefit.

I think it should be quite clear to anyone here that it's of no audible benefit what-so-ever, but it may create an excuse (that the accountants will accept) to create better (re-)masters. Which will then be used for the 44.1kHz version that can now be bought for a bargain price.  Everyone's a winner.


Not necessaraly so....

A general rule of measurements is that accuracy and measurement time are related. Low measuring times means low accuracy and high accuracy means long measuring times.
So there comes a time that while you think you increase the amount of information (increase sampling rates), you are actualy decreacing the amount of information. Again information theory explains this all.
There are two ways to realise this is misleading...

1) It's the RMS noise that typically increases as sampling frequency increases. The dB/Hz noise doesn't. So the noise level within the audio band remains roughly constant as sample rate is increased. When we didn't have enough bits to out-do the real world, the in-band noise fell as the sample rate increased.

2) No ADCs or DACs run at 44.1kHz natively. They run at a higher rate internally. The 96kHz output is no less accurate than the 48kHz output - both are usually derived from the same higher rate version internally. The 96kHz version can't be worse than the 48kHz version.

Cheers,
David.
Title: What's the point of higher sampling rates in audio?
Post by: 2Bdecided on 2011-10-18 14:55:20
Guys, you all disappoint me. The point of higher sampling rates is capturing at best quality possible, regardless if the quality improvement to 16/44.1 is only theoretical. The point is: you can always dowsample later. Really, all your shiny, biting and ironic arguments would have made sense ten years ago but in present times it´s completely superfluous: nowadays we have plenty of HDD space, more than capable hardware, fast processing (even 32/384.000 can be edited and processed very fast if I may be allowed to indulge in that exaggeration fantasy) and extremely fast internet connections. It just doesn´t make sense to use 16/44.1 or lossy compression nowadays and even less sense to promote it as the only format worthwile.

Did it occur to you that for example the Grammy foundation and the Deutsche Grammophon are doing backups of their precious analogue tapes with 24/192 before the tapes degrade and are lost? They are doing this because for example the IASA recommends it. They also don´t care if the quality is better or not, they just want to capture everything. They apparently do it according to the motto 'Better safe than sorry'. Are you calling them stupid or deluded? They obviously know the times they are in and behave accordingly, they exploit the technical possibilities they have and don´t stand still. Which is what has been missing here since a few years now. You guys are still doing comparisons of lossy formats at 96 kBit/s! Why? The world won´t need them for much longer. Furthermore, you make fun of people who don´t share your opinion. Granted, some of them deserve it. But for a few years now everyone who promotes something like higher samplerates, 24 bit, hell, anything else that deviates from established norms is treated like a little dumb child. You have become close minded people, living in their own secluded world that is becoming smaller and smaller every day.

This forum clearly has lost its edge, I really feel that the world has turned - but without you.

I will no longer participate here (you´ll applaud it, I´m sure) and I know that sentences will be put out of context. If there would be a delete-button for my profile I´d use it.
The audio world disappoints me, solving non-problems by adopting the increases in sample rate and bit depth that Moor's Law brings almost for free, while ignoring the possibilities of decent surround sound, the high cost and/or poor performance of transducers, and the lousy sound in most consumers' homes.

DG should use more than 192kHz - people have already made great use of the bias signal recovered from tapes to stabilise wow and flutter. Better make sure they're capturing it faithfully to allow this processing in the future.

Of course, if there is ever any objective evidence that 192kHz brings audible benefits for home listening, HA would be extremely interested.

Cheers,
David.

P.S. I've found hobbyists who insist on transferring vinyl at 192kHz while using a $200 turntable. I wonder where they got the idea that high sampling rates were so important? Not from HA.
Title: What's the point of higher sampling rates in audio?
Post by: saratoga on 2011-10-18 17:41:08
Guys, you all disappoint me. The point of higher sampling rates is capturing at best quality possible, regardless if the quality improvement to 16/44.1 is only theoretical. The point is: you can always dowsample later.


And you can always up-sample later, so who cares?
Title: What's the point of higher sampling rates in audio?
Post by: Wombat on 2011-10-18 19:48:27
I love the steps forward high resolution movies come to me now on Blu-Ray and HDTV. I did some step ups on my TV hardware over the time and always enjoyed it getting better and better.
I love to watch HD-TV, especially sport events, it simply makes fun.
At night me old fart puts on his 3D glasses and slaughters enemies in 3D on a Playstation 3. Looking good on a recent 47" LED TV only wasting 60Watts of power.

But i still can´t hear, as end user the stated benefits of Hires music against its same version properly downsampled. Believe me my audio system is not to shabby and allows native Hires playback.
I don´t question to archive the raw capture of a recording somewhere in the best possible, recent technology. I am just fed up that on some forums it is a given all these Hires formats sound always better.
If someoene wants to tell me i do not because i missed the world turning i have to wonder...
Title: What's the point of higher sampling rates in audio?
Post by: punkrockdude on 2011-10-18 21:13:40
People record with 96 or 192 kHz simply because they can and because they believe it sounds better.

Try a few VST compressors at 44100 Hz and then on a project using 192000 Hz and listen to the difference in attack and release. this does not mean that the percieved quality of the recorded audio is better but some plugins sound a lot better at higher sample rates. Regards.
Title: What's the point of higher sampling rates in audio?
Post by: [JAZ] on 2011-10-18 22:09:35
And are you 100% sure that the plugin is samplerate aware???

I've been part of this past year fixing most of the plugins of my program* so that they don't perform differently at different sample rates. There's no plausible reason why a compressor would act that much different at different sampling rates. The slope should increase accordingly.
(Of course, by the nature of a compressor, a higher samplerate allows to control better the envelope, but we're talking about a couple of samples better, not something that could clearly be heard).



* http://sourceforge.net/projects/psycle (http://sourceforge.net/projects/psycle) . I'm doing the last tests and adjustments for a new official release. (Should have already released it this weekend).
Title: What's the point of higher sampling rates in audio?
Post by: C.R.Helmrich on 2011-10-18 22:42:13
It just doesn´t make sense to use 16/44.1 or lossy compression nowadays and even less sense to promote it as the only format worthwile.
... You guys are still doing comparisons of lossy formats at 96 kBit/s! Why? The world won´t need them for much longer. Furthermore, you make fun of people who don´t share your opinion.

I completely agree with you on the fun-making part. Such people don't contribute anything but rather deteriorate a forum's reputation. But I strongly disagree with you on the rest.

Digital archiving of historical recordings is a completely separate issue, where data rate doesn't matter. So yes, if the analog tape creates distortion up to 48 kHz, digitize at 96 kHz, why not? But why at 24 or 32 bits? You won't capture any more information than with 16 (or 14) bits per sample. And, luckily, at HA many know a thing or two about information theory which most of the world's population doesn't.

Sorry to disappoint you, but already nowadays you hear more compressed than lossless audio in your daily life. Digital TV, radio, phone, basically every animation or stream on the Internet use it. Give me an application other than physical media (CD, Blu-Ray, etc.) and private FLAC/ALAC collections where lossless audio is being distributed/used by more than a tiny minority. That's why we test things like 96 kbps: to see how low you can go with lossy compression and still achieve excellent (but of course not transparent) quality, and to make other people aware of our findings. It's a hard truth, but I think we are reaching a point where lossless audio is becoming unimportant. The industry is e.g. pushing digital radio to channel bit-rates far below 100 kbps stereo! Which is ridiculous if you ask me, but that's how it is. So there are much more important things to worry about than whether to use 48 kHz or higher sampling rates.

Edit: Thought about it some more... Of course nowadays we would also want to archive e.g. some contemporary amateur cell-phone videos with "historical value". Isn't it more important to get the crappy cell-phone video and sound right in the next phone generation than to be able to preserve it in 24/96?

My 0.02 Euro.

Chris
Title: What's the point of higher sampling rates in audio?
Post by: krabapple on 2011-10-18 23:10:04
The downside is that we don't get proper surround. Though some people are still quietly working on that too.

Cheers,
David.


I'm intrigued by this part...how does industry support for silly sample rates keep us from getting proper surround?

Title: What's the point of higher sampling rates in audio?
Post by: krabapple on 2011-10-18 23:26:24
Guys, you all disappoint me. The point of higher sampling rates is capturing at best quality possible, regardless if the quality improvement to 16/44.1 is only theoretical. The point is: you can always dowsample later. Really, all your shiny, biting and ironic arguments would have made sense ten years ago but in present times it´s completely superfluous: nowadays we have plenty of HDD space, more than capable hardware, fast processing (even 32/384.000 can be edited and processed very fast if I may be allowed to indulge in that exaggeration fantasy) and extremely fast internet connections. It just doesn´t make sense to use 16/44.1 or lossy compression nowadays and even less sense to promote it as the only format worthwile.

Did it occur to you that for example the Grammy foundation and the Deutsche Grammophon are doing backups of their precious analogue tapes with 24/192 before the tapes degrade and are lost? They are doing this because for example the IASA recommends it. They also don´t care if the quality is better or not, they just want to capture everything. They apparently do it according to the motto 'Better safe than sorry'. Are you calling them stupid or deluded? They obviously know the times they are in and behave accordingly, they exploit the technical possibilities they have and don´t stand still. Which is what has been missing here since a few years now. You guys are still doing comparisons of lossy formats at 96 kBit/s! Why? The world won´t need them for much longer. Furthermore, you make fun of people who don´t share your opinion. Granted, some of them deserve it. But for a few years now everyone who promotes something like higher samplerates, 24 bit, hell, anything else that deviates from established norms is treated like a little dumb child. You have become close minded people, living in their own secluded world that is becoming smaller and smaller every day.

This forum clearly has lost its edge, I really feel that the world has turned - but without you.

I will no longer participate here (you´ll applaud it, I´m sure) and I know that sentences will be put out of context. If there would be a delete-button for my profile I´d use it.



Jeez, such dramatics! 

High SR and high bit depth have their place at the recording and production end -- the utility of 24 or 32bit processing is a given here, and 2bdecided even noted one of the most compelling (and little-cited) reasons to use super-high SR for archiving master tapes: it allows later use of software that eliminates flutter (Plangent processing). 

However, they're largely bling at the delivery format end....i.e. the media that the consumer buys for the mere purpose of *listening*. 
 

Title: What's the point of higher sampling rates in audio?
Post by: Juha on 2011-10-19 04:33:22
I bet most of these 16/44.1 fanatics rips/ripped their best tapes/vinyls using least 24/96.


Juha
Title: What's the point of higher sampling rates in audio?
Post by: hellokeith on 2011-10-19 05:30:51
why at 24 or 32 bits?

Chris,

I was under the impression that 24 bits offers headroom (for lack of a better word) for destructive editing operations?
Title: What's the point of higher sampling rates in audio?
Post by: Dirk95100 on 2011-10-19 05:35:11
Anyway, back to reality: if you are making high quality recordings, and some of the people purchasing your high quality recordings want to pay you $10 extra for a 192kHz version, why on earth wouldn't you make one available? As long as it doesn't make the quality worse, and doesn't cost you more than the financial return, it's really not a problem if people want to pay more for no tangible benefit.

I think it should be quite clear to anyone here that it's of no audible benefit what-so-ever, but it may create an excuse (that the accountants will accept) to create better (re-)masters. Which will then be used for the 44.1kHz version that can now be bought for a bargain price.  Everyone's a winner.


Not necessaraly so....

A general rule of measurements is that accuracy and measurement time are related. Low measuring times means low accuracy and high accuracy means long measuring times.
So there comes a time that while you think you increase the amount of information (increase sampling rates), you are actualy decreacing the amount of information. Again information theory explains this all.
There are two ways to realise this is misleading...

1) It's the RMS noise that typically increases as sampling frequency increases. The dB/Hz noise doesn't. So the noise level within the audio band remains roughly constant as sample rate is increased. When we didn't have enough bits to out-do the real world, the in-band noise fell as the sample rate increased.

2) No ADCs or DACs run at 44.1kHz natively. They run at a higher rate internally. The 96kHz output is no less accurate than the 48kHz output - both are usually derived from the same higher rate version internally. The 96kHz version can't be worse than the 48kHz version.

Cheers,
David.


http://www.lavryengineering.com/documents/...ling_Theory.pdf (http://www.lavryengineering.com/documents/Sampling_Theory.pdf)
Title: What's the point of higher sampling rates in audio?
Post by: saratoga on 2011-10-19 05:51:17
http://www.lavryengineering.com/documents/...ling_Theory.pdf (http://www.lavryengineering.com/documents/Sampling_Theory.pdf)


Yeah, no ones going to read that.

Is there any particular point you'd like to make?  Or do you agree with 2Bdecided's (essentially correct) response.
Title: What's the point of higher sampling rates in audio?
Post by: Woodinville on 2011-10-19 07:45:57
1) It's the RMS noise that typically increases as sampling frequency increases. The dB/Hz noise doesn't. So the noise level within the audio band remains roughly constant as sample rate is increased. When we didn't have enough bits to out-do the real world, the in-band noise fell as the sample rate increased.



?huh? In what sense do you mean that????
Title: What's the point of higher sampling rates in audio?
Post by: Dirk95100 on 2011-10-19 08:42:19
http://www.lavryengineering.com/documents/...ling_Theory.pdf (http://www.lavryengineering.com/documents/Sampling_Theory.pdf)


Yeah, no ones going to read that.

Is there any particular point you'd like to make?  Or do you agree with 2Bdecided's (essentially correct) response.

Sure, namely a part of the 4th paragraph in the article:
Quote
There is also a tradeoff between speed and accuracy. Conversion at
100MHz yield around 8 bits, conversion at 1MHz may yield near 16 bits and as we approach
50-60Hz we get near 24 bits. Speed related inaccuracies are due to real circuit considerations,
such as charging capacitors, amplifier settling and more. Slowing down improves accuracy.


And in general it´s the shorter time you take to measure, the higher the tolerances.
Iaw: You can´t have the cake and eat it.
Title: What's the point of higher sampling rates in audio?
Post by: SebastianG on 2011-10-19 09:38:58
And in general it´s the shorter time you take to measure, the higher the tolerances.

Don't forget the fact that at a higher sampling rate you have a higher bandwidth the noise can spread into. You may get more noise but how much of it will go into the band-of-interest? The interesting question is: How does the power spectral density of the measurement noise change in the band-of-interest with a different sampling rate?

Edit: Oh, I just saw that 2B basically touched the same question. Power spectral density is usually measured in dB/Hz if I recall correctly.
Title: What's the point of higher sampling rates in audio?
Post by: 2Bdecided on 2011-10-19 10:50:29
The downside is that we don't get proper surround. Though some people are still quietly working on that too.

I'm intrigued by this part...how does industry support for silly sample rates keep us from getting proper surround?

There's only so much effort, marketing, messaging etc - and the parts of those aimed at "better sound" are mostly going the wrong way IMO. Resources, and audiophile's attention, are not infinite.

Or to put it another way, they've got to sell something. They've chosen to sell imagined improvements, rather than real ones, a) because it's easier, and b) because plenty of people who should know better haven't called "foul".
Title: What's the point of higher sampling rates in audio?
Post by: 2Bdecided on 2011-10-19 11:01:17
http://www.lavryengineering.com/documents/...ling_Theory.pdf (http://www.lavryengineering.com/documents/Sampling_Theory.pdf)
As others have pointed out, that's contradictory - it makes the point that virtually all sampling now uses hundreds of kHz or MHz...
Quote
For example, most front ends of modern AD
(the modulator section) work at rates between 64 and 512 faster than a basic 44.1 or 48KHz
system. This is 16 to 128 times faster than 192KHz. Such speedy operation yields only a few
bits. Following such high speed low bits intermediary outcome is a process called decimation,
slowing down the speed for more bits. There is a tradeoff between speed and accuracy. The
localized converter circuit (few bits at MHz speeds) is followed by a decimation circuit, yielding
the required bits at the final sample rate.

...but then claims...
Quote
Sampling audio signals at 192KHz is about 3 times faster than the optimal rate.
It compromises the accuracy which ends up as audio distortions.


So there's a point in the ADC where the input is a many kHz few bits signal. This can be converted to 48kHz, 96kHz, or 192kHz. In each case, the quality in the audio band is the same. Downsampling to 192kHz rather than 48kHz cannot introduce more audio distortions.


There is no technical point to using 192kHz - I agree with the paper in this respect - but this specific point "It compromises the accuracy which ends up as audio distortions" - or the idea that there's more noise in the audio band when sampling at 192kHz vs 48kHz, is wrong.

Cheers,
David.

P.S.
Quote
There is also a tradeoff between speed and accuracy. Conversion at
100MHz yield around 8 bits, conversion at 1MHz may yield near 16 bits and as we approach
50-60Hz we get near 24 bits. Speed related inaccuracies are due to real circuit considerations,
such as charging capacitors, amplifier settling and more. Slowing down improves accuracy.
This is true, but
a) it doesn't tell you anything about the noise in the audio band, and
b) in the context of 48kHz vs 192kHz, we're talking about the exact same analogue electronics (capacitors etc) - it's only the digital (mathemataical!) downsampling that's adjusted.

I am not denying information theory - I'm saying it's being misapplied.
Title: What's the point of higher sampling rates in audio?
Post by: Dirk95100 on 2011-10-19 12:50:47
http://www.lavryengineering.com/documents/...ling_Theory.pdf (http://www.lavryengineering.com/documents/Sampling_Theory.pdf)
As others have pointed out, that's contradictory - it makes the point that virtually all sampling now uses hundreds of kHz or MHz...
Quote
For example, most front ends of modern AD
(the modulator section) work at rates between 64 and 512 faster than a basic 44.1 or 48KHz
system. This is 16 to 128 times faster than 192KHz. Such speedy operation yields only a few
bits. Following such high speed low bits intermediary outcome is a process called decimation,
slowing down the speed for more bits. There is a tradeoff between speed and accuracy. The
localized converter circuit (few bits at MHz speeds) is followed by a decimation circuit, yielding
the required bits at the final sample rate.

...but then claims...
Quote
Sampling audio signals at 192KHz is about 3 times faster than the optimal rate.
It compromises the accuracy which ends up as audio distortions.


So there's a point in the ADC where the input is a many kHz few bits signal. This can be converted to 48kHz, 96kHz, or 192kHz. In each case, the quality in the audio band is the same. Downsampling to 192kHz rather than 48kHz cannot introduce more audio distortions.


There is no technical point to using 192kHz - I agree with the paper in this respect - but this specific point "It compromises the accuracy which ends up as audio distortions" - or the idea that there's more noise in the audio band when sampling at 192kHz vs 48kHz, is wrong.

Cheers,
David.

P.S.
Quote
There is also a tradeoff between speed and accuracy. Conversion at
100MHz yield around 8 bits, conversion at 1MHz may yield near 16 bits and as we approach
50-60Hz we get near 24 bits. Speed related inaccuracies are due to real circuit considerations,
such as charging capacitors, amplifier settling and more. Slowing down improves accuracy.
This is true, but
a) it doesn't tell you anything about the noise in the audio band, and
b) in the context of 48kHz vs 192kHz, we're talking about the exact same analogue electronics (capacitors etc) - it's only the digital (mathemataical!) downsampling that's adjusted.

I am not denying information theory - I'm saying it's being misapplied.


What matters is how fast the actual information is gathered, at 192kHz its faster than at 44.1kHz. Therefore the tolerances at 192kHz will be higher than at 44.1kHz.
Title: What's the point of higher sampling rates in audio?
Post by: SebastianG on 2011-10-19 13:52:48
What matters is how fast the actual information is gathered, at 192kHz its faster than at 44.1kHz. Therefore the tolerances at 192kHz will be higher than at 44.1kHz.

So?

What do you think is preferable?
(a) use a sampler/quantizer at 44.1 kHz which adds Gaussian white noise with a variance (http://en.wikipedia.org/wiki/Variance) of v
(b) use a sample/quantizer at 192 kHz with adds Gaussian white noise with a variance of 3v (three times as high)?

I'll tell you: (b) is preferable because the noise's power spectral density is 1.6 dB/Hz lower compared to (a). It means: less noise in the audible band.
Title: What's the point of higher sampling rates in audio?
Post by: pdq on 2011-10-19 14:00:12
What matters is how fast the actual information is gathered, at 192kHz its faster than at 44.1kHz. Therefore the tolerances at 192kHz will be higher than at 44.1kHz.

This is total BS. It has been pointed out several times already that the information is "gathered" at the same rate (hundreds of KHz or MHz) regardless of how it is resampled to its final sample rate.
Title: What's the point of higher sampling rates in audio?
Post by: C.R.Helmrich on 2011-10-19 15:50:30
why at 24 or 32 bits?

Chris,

I was under the impression that 24 bits offers headroom (for lack of a better word) for destructive editing operations?

All of what I say here exclusively focuses on recording. Of course for editing/processing, it makes sense to work with 24 or more bits, just like you would work with a 24- or 32-bit image before creating an 8-bit GIF or PNG of it. Also note in response to punkrockdude (http://www.hydrogenaudio.org/forums/index.php?s=&showtopic=91302&view=findpost&p=772593) that (in my wishful thinking) I assumed that nowadays, plug-in developers know what they're doing when it comes to audible aliasing and other distortions. From what people tell me here, it seems that this is not always the case...

Chris
Title: What's the point of higher sampling rates in audio?
Post by: greynol on 2011-10-19 16:34:21
It has little to do with editing.  People record at higher bit depths so that they don't have to worry about running out of headroom during the recording process from unexpected transients.  You can easily leave 20dB of headroom without sacrificing SNR, at least from a mathematical perspective.

IOW, you don't need to be as careful about your levels with 24-bit as you do recording in 16-bit.  If the levels were optimized at 16-bit upon recording vinyl then recording at 24-bit would be completely unnecessary, since 16-bit is more than adequate in presenting material sourced from vinyl.
Title: What's the point of higher sampling rates in audio?
Post by: Arnold B. Krueger on 2011-11-01 14:08:32
It has little to do with editing.  People record at higher bit depths so that they don't have to worry about running out of headroom during the recording process from unexpected transients.  You can easily leave 20dB of headroom without sacrificing SNR, at least from a mathematical perspective.


If you goal is 20 dB headroom, 16 bits can almost always provide it.

A really quiet room with musicans in it rarely provides a maximum signal that is more than 66 dB above the acoustic noise floor, and you can observe this given reasonbly quiet mics.  That's about 30 dB below the 93-96 dB of nominal dynamic range that is available with 16 bits. If you noise-shape the quantization, then there is even more dynamic range than 96 dB with just 16 bits.

Title: What's the point of higher sampling rates in audio?
Post by: greynol on 2011-11-01 15:48:47
That's great, Arny, though I don't know that I would necessarily center the discussion around acoustic recording.  Lots of things can go direct these days.
Title: What's the point of higher sampling rates in audio?
Post by: punkrockdude on 2011-11-01 15:55:41

And are you 100% sure that the plugin is samplerate aware???

I've been part of this past year fixing most of the plugins of my program* so that they don't perform differently at different sample rates. There's no plausible reason why a compressor would act that much different at different sampling rates. The slope should increase accordingly.
(Of course, by the nature of a compressor, a higher samplerate allows to control better the envelope, but we're talking about a couple of samples better, not something that could clearly be heard).



* http://sourceforge.net/projects/psycle (http://sourceforge.net/projects/psycle) . I'm doing the last tests and adjustments for a new official release. (Should have already released it this weekend).

I used Reaper. Regards.
Title: What's the point of higher sampling rates in audio?
Post by: Arnold B. Krueger on 2011-11-01 20:27:51
That's great, Arny, though I don't know that I would necessarily center the discussion around acoustic recording.  Lots of things can go direct these days.


Please tell me how to do  vocals without going acoustic. ;-)

Also, if you actually see how most top bands work, their drums and percussion are also acoustic.

Very often even the nominally electronic instruments such as synths, guitars and bass are recorded from mics on their instrument amps.

As a rule, to sell recordings modern groups tour, and as a rule touring groups are still very reliant on microphones.
Title: What's the point of higher sampling rates in audio?
Post by: Arnold B. Krueger on 2011-11-01 20:44:32

And are you 100% sure that the plugin is samplerate aware???

I've been part of this past year fixing most of the plugins of my program* so that they don't perform differently at different sample rates.


If nonlinear processing is done in the digital domain, certain high harmonics can reflect back down into the audible range. Any signal that is generated in the digital domain above the Nyquist frequency  aliases around the Nyquist frequency.  For example, an 8 KHz tone in a 44.1 KHz sampled sytem is distorted by a fourth order digitally-implemented nonlinearity and would be expected to produce a fourth harmonic at 32 KHz.  Since 32 KHz is 10 Khz higher than the 44.1 KHz Nyquist frequency of 22 KHz, the  4th harmonic is aliased down to 12 KHz where it is far more audible than it would be in a digital system with a much higher sample rate or an analog system. In a 96 KHz system, the same nonlinearity would produce a foruth harmonic at 32 KHz where it would not be audible.

Aliasing can also impact certain dynamic range compression or expansion algorithms that are based on nonlinarities implemented in the digital domain.

Most kinds of processing that are used in music synthesis and mixing are linear, so aliasing is not a common problem.
Title: What's the point of higher sampling rates in audio?
Post by: greynol on 2011-11-01 21:16:17
Also, if you actually see how most top bands work, their drums and percussion are also acoustic.
Argumentum ad populum; yawn.

Please tell me how to do  vocals without going acoustic. ;-)
Not all music has vocals.

As a rule, to sell recordings modern groups tour, and as a rule touring groups are still very reliant on microphones.
Non sequitur.

Seriously, this overgeneralizing on your part isn't getting us very far.

If I had said 30dB would this have kept you from nit picking?
Title: What's the point of higher sampling rates in audio?
Post by: 2Bdecided on 2011-11-02 11:32:08
If nonlinear processing is done in the digital domain, certain high harmonics can reflect back down into the audible range. Any signal that is generated in the digital domain above the Nyquist frequency  aliases around the Nyquist frequency.  For example, an 8 KHz tone in a 44.1 KHz sampled sytem is distorted by a fourth order digitally-implemented nonlinearity and would be expected to produce a fourth harmonic at 32 KHz.  Since 32 KHz is 10 Khz higher than the 44.1 KHz Nyquist frequency of 22 KHz, the  4th harmonic is aliased down to 12 KHz where it is far more audible than it would be in a digital system with a much higher sample rate or an analog system. In a 96 KHz system, the same nonlinearity would produce a foruth harmonic at 32 KHz where it would not be audible.
A simple non-linear transfer function (i.e. literally mapping sample values to new sample values using a look-up table which, if plotted input vs output on a graph, would show a curve) - that can produce harmonics above the noise into the MHz range. You don't prevent audible aliasing just be using a "slightly" higher sample rate.

It helps a bit, but "just" 96kHz isn't really a solution.

Cheers,
David.
Title: What's the point of higher sampling rates in audio?
Post by: benski on 2011-11-02 14:40:03
A simple non-linear transfer function (i.e. literally mapping sample values to new sample values using a look-up table which, if plotted input vs output on a graph, would show a curve) - that can produce harmonics above the noise into the MHz range. You don't prevent audible aliasing just be using a "slightly" higher sample rate.

It helps a bit, but "just" 96kHz isn't really a solution.

Cheers,
David.


This isn't completely true, David.  If you have an arbitrary non-linear function, yes you will have sky-high THD generation.  However, if you have a function with a known polynomial order (such as a Taylor series approximation of a transcendental function), then you will also have a predictable limit on the number of harmonics produced.  This is because because a polynomial function is the equivalent of amplitude modulation - x^2 is the signal x modulating the amplitude of signal x - and AM has predictable sidebands (M-N and M+N, in the x^2 case, that would be 0 and 2x)
Title: What's the point of higher sampling rates in audio?
Post by: 2Bdecided on 2011-11-02 15:18:48
Yes, I agree. Sorry, that's what I was implying (thought I'd said it earlier in the thread - but there have been a lot of threads like this) - the way you avoid aliasing is to design the processing properly/carefully, so you know what it's doing. You may then need some oversampling, but you'll know how much.

Just using oversampling to fix aliasing without understanding is what I was criticising (and maybe not what Arny was talking about).

I feel sure some of the commercial DSP/plug-ins that allegedly "work better" at higher sample rates haven't done their calculations properly. It could be they apply temporal parameters per sample rather than per second, but sometimes I think the issue is that any attempts to design out aliasing are inadequate. In this case, even jumping to 96kHz doesn't guarantee the DSP works properly - just less badly. If this is the case, we're not safe until we hit 10s of MHz.

Cheers,
David.
Title: What's the point of higher sampling rates in audio?
Post by: astroidmist on 2011-11-07 06:00:19
I don't know if this has been posted or addressed yet, but it's a very good read related to this stuff.  It's worth reading entirely, because each part of it relates to the other parts. 

http://www.lavryengineering.com/documents/...ling_Theory.pdf (http://www.lavryengineering.com/documents/Sampling_Theory.pdf)
Title: What's the point of higher sampling rates in audio?
Post by: Roseval on 2011-11-07 23:01:21
Nyquist in reverse

One should sample at the double of the highest frequency.
The reverse holds too, if you sample at 44, there shouldn't be any signal above 22kHz in the source.
Practical consequence, the input must be band limited.
If you want to cover everything up to 20 kHz, the only thing you can do is using a pretty steep low pass filter (brick wall).
In general this type of filter produces artifacts like pre-ringing.

If you sample at 88, your problems remain the same, no frequency above Nyquist please.
This time our Nyquist is 44.
We might decide to use a brick wall again but this time it is much farther out of our audible range so probably has less impact.

As the late Julian Dunn phrased it
A direct effect of the higher sampling rate is that for an identical filter design the time
displacements will scale inversely with sample rate. Hence an improvement can be
made just from raising the sample rate - even for those who cannot hear above
20kHz.


There isn't  much musical energy at this level.
We also might decide to use a smoother low pass filter e.g. starting at 30 kHz.

There might indeed be a (small) benefit using higher sample rates.

BTW: James Boyk measured a couple of instruments demonstrating that not only cymbals produces sound above 20 kHz
http://www.cco.caltech.edu/~boyk/spectra/spectra.htm (http://www.cco.caltech.edu/~boyk/spectra/spectra.htm)
Title: What's the point of higher sampling rates in audio?
Post by: Slipstreem on 2011-11-08 01:01:38
This forum clearly has lost its edge, I really feel that the world has turned - but without you.

To save everyone having to read the same meaningless dross again, I'm selectively quoting just this line to make it clear that it's Cavaille's post I'm replying to, and this post has effectively dragged me out of a self-imposed retirement from HA due to being repeated hectored by a seemingly psychotic member who shall remain nameless, so you can guess how much this post has got my back up! 

Are you suggesting that the generation of today are in some way responsible for preserving audible content above ~22kHz for the sake of future genetically superior human beings who populate the planet long after we're gone? 99.9% of the world's human population don't stand a bat's chance in hell of hearing anything above 22kHz ever, so it's totally pointless sampling accurately at anything above 44.1kHz for the final delivery format. 96 or even 192kHz has its place during the editing process for the blindingly obvious reasons covered by previous posters, but anything beyond 44.1kHz for the final delivery format makes absolutely no sense whatsoever no matter which angle you approach the argument from for the vast majority of the human population.

HA's standards have certainly slipped since I was a regualr contributor if posts like Cavaille's are allowed to remain unedited, so I'm glad (if slightly saddened in some ways) to no longer be a regular HA contributor. 
Title: What's the point of higher sampling rates in audio?
Post by: Wombat on 2011-11-08 01:14:17
One should sample at the double of the highest frequency.
The reverse holds too, if you sample at 44, there shouldn't be any signal above 22kHz in the source.
Practical consequence, the input must be band limited.
If you want to cover everything up to 20 kHz, the only thing you can do is using a pretty steep low pass filter (brick wall).
In general this type of filter produces artifacts like pre-ringing.

Whats the problem with the "Brickwall" filter? If done correctly this evil pre-ringing is happening above 21kHz and shouldn´t matter. You may also decide to allow some aliasing above 20kHz and filter from there on. Not much pre-ringing left at all. Both attempts should be absolutely transparent. Good luck on abxing that against!
Title: What's the point of higher sampling rates in audio?
Post by: pappaapa on 2011-11-08 14:38:42
Question:

Shouldn't the rest of the worlds acoustics be considered as well?
I mean, non audible frequencies pass through materials and become audible. So they need to be there if you want a natural sound reproduction.
It's not just our eardrums in the room, is it?

To make a comparison with light:
Your day at the office would be pretty dark if you were to take the non visible part of the spectra out of the equation.
As the flourecent lights fully depends on ultraviolet light, a higher frequency light just outside the visible spectra.

NOTE: I'm a musician and an audiophile, not a scientist. So I could be horribly wrong.
And maybe this has been said a million times before.
But to me it feels pretty obvious.

You should start to slow the soundwaves down right at the speaker covers.

/Levi
Title: What's the point of higher sampling rates in audio?
Post by: Wombat on 2011-11-08 14:55:15
Question:

Shouldn't the rest of the worlds acoustics be considered as well?
I mean, non audible frequencies pass through materials and become audible. So they need to be there if you want a natural sound reproduction.
It's not just our eardrums in the room, is it?

To make a comparison with light:
Your day at the office would be pretty dark if you were to take the non visible part of the spectra out of the equation.
As the flourecent lights fully depends on ultraviolet light, a higher frequency light just outside the visible spectra.

NOTE: I'm a musician and an audiophile, not a scientist. So I could be horribly wrong.
And maybe this has been said a million times before.
But to me it feels pretty obvious.

You should start to slow the soundwaves down right at the speaker covers.

/Levi

To answer your question is: NO!
This sounds like audiophile gibberish.
The part of your speaker that is pruducing the high frequencies above 20kHz do move some mm² of light material with pretty low energy. This can´t activate any other materials in your room.

P.S.: These kind of discussions are becoming a real pita.
Title: What's the point of higher sampling rates in audio?
Post by: 2Bdecided on 2011-11-08 17:00:02
To make a comparison with light:
Your day at the office would be pretty dark if you were to take the non visible part of the spectra out of the equation.
It wouldn't make any visible difference, because they're non visible.

Quote
As the flourecent lights fully depends on ultraviolet light, a higher frequency light just outside the visible spectra.
But speakers don't depend on ultrasonic sound to create audible sound (well, there's a device that does, but it's not a normal speaker).

Any hypothetical intermodulation of ultrasonics from the original instruments, mixing in air to create audible components, will be strongest near the instruments themselves - and will be captured, in the audible range, by the microphones at the original event. So you don't need to create them in the listening room - they're already in the recording if they exist.

Far more real (i.e. measurable and sometimes audible) is unwanted intermodulation in the speakers themselves. That never exited in the original performance, and can be measurably reduced by removing all ultrasonics cleanly.

Cheers,
David.
Title: What's the point of higher sampling rates in audio?
Post by: krabapple on 2011-11-08 17:04:44
Question:

Shouldn't the rest of the worlds acoustics be considered as well?
I mean, non audible frequencies pass through materials and become audible.



No, that's distortion.
Title: What's the point of higher sampling rates in audio?
Post by: Wombat on 2011-11-08 18:31:46
Far more real (i.e. measurable and sometimes audible) is unwanted intermodulation in the speakers themselves. That never exited in the original performance, and can be measurably reduced by removing all ultrasonics cleanly.


Once again a sentense from you i should keep and remember well!
I lately looked over some frequency response plots of the latest B&W speakers. They introduce a resonance and peak at 40kHz of +10dB! I can imagine while you feed them with music that has content at 40kHz the tweeter may introduce unwanted behaviour in audible frequencies. I doubt this is what anyone should want.
Title: What's the point of higher sampling rates in audio?
Post by: pappaapa on 2011-11-08 20:20:58
First: Thanks for your replies!

And just to tell you. Personally, I'm only curious. I have no preference (but vinyl).
I've never participated in this kind of discussion before and I haven't done much more reading than the physics in school (15 years ago).

Though I did do some reading before writing this and all I can say is: This is waaay too complicated for me.

Turns out there are actually several ultrasonic speakers. Where the air it passes through slows down the soundwaves, so that it will be audible at a certain point.
You can find them at disneyworld and in your local mall. Aimed at you.

But really, I can't understand how anyone without a degree in physics can say either YES or NO to my question.
You really need to know your way around Nonlinear Acoustics.
Here's a place to start: Nonlinear Acoustics Wiki (http://en.wikipedia.org/wiki/Nonlinear_acoustics)

So unless anybody can explain this in a simple (and informed) way, I'm just gonna have to leave this alone.
Really sorry for wasting your time.

/Levi
Title: What's the point of higher sampling rates in audio?
Post by: drewfx on 2011-11-08 21:42:51
First: Thanks for your replies!

And just to tell you. Personally, I'm only curious. I have no preference (but vinyl).
I've never participated in this kind of discussion before and I haven't done much more reading than the physics in school (15 years ago).

Though I did do some reading before writing this and all I can say is: This is waaay too complicated for me.

Turns out there are actually several ultrasonic speakers. Where the air it passes through slows down the soundwaves, so that it will be audible at a certain point.
You can find them at disneyworld and in your local mall. Aimed at you.

But really, I can't understand how anyone without a degree in physics can say either YES or NO to my question.
You really need to know your way around Nonlinear Acoustics.
Here's a place to start: Nonlinear Acoustics Wiki (http://en.wikipedia.org/wiki/Nonlinear_acoustics)

So unless anybody can explain this in a simple (and informed) way, I'm just gonna have to leave this alone.
Really sorry for wasting your time.

/Levi


My understanding is ultrasonic speakers work similar to AM radio - you have an (inaudible) ultrasonic wave modulated by the audio you want to produce. You never hear the ultrasonic signal itself; it's just a carrier for the audible stuff.
Title: What's the point of higher sampling rates in audio?
Post by: Soap on 2011-11-08 22:24:01
So unless anybody can explain this in a simple (and informed) way, I'm just gonna have to leave this alone.


You already were offered an explanation in a simple and informed way:


Any hypothetical intermodulation of ultrasonics from the original instruments, mixing in air to create audible components, will be strongest near the instruments themselves - and will be captured, in the audible range, by the microphones at the original event. So you don't need to create them in the listening room - they're already in the recording if they exist.

(emphasis mine).

horse.
water.
drink.
Title: What's the point of higher sampling rates in audio?
Post by: pappaapa on 2011-11-08 22:50:56
My understanding is ultrasonic speakers work similar to AM radio - you have an (inaudible) ultrasonic wave modulated by the audio you want to produce. You never hear the ultrasonic signal itself; it's just a carrier for the audible stuff.


Well that's one kind. Where you have a reciever remodulating the ultrasonic sound.

The kind I'm refering to needs no reciever. It's just air slowing the soundwaves down and at a desired distance the sound becomes audible. As I understand it they shape the ultrasonic sound so that when the air has slowed it down to certain wavelengths interferance amongst the short soundwaves produces longer wavelengths and thus a high fidelity audible sound.
The mechanism is called Parametric Array.

Here's some reading: Parametric array - Wiki (http://en.wikipedia.org/wiki/Parametric_array)
Here's an interesting link of an implementation in an art installation: Link (http://blog.svconline.com/briefingroom/2008/11/07/sennheisers-audio-beam-creates-a-sound-ceiling/)

The question here is how often and how much does this occur at normal listening volumes in the nonaudible (but reproducable) frequencies. And to me it's just too complicated.

I tried to find some clever guy with some info on the web. But I really have to go to sleep now.

Good night from Sweden!
/Levi
Title: What's the point of higher sampling rates in audio?
Post by: pappaapa on 2011-11-08 23:10:34
Any hypothetical intermodulation of ultrasonics from the original instruments, mixing in air to create audible components, will be strongest near the instruments themselves - and will be captured, in the audible range, by the microphones at the original event. So you don't need to create them in the listening room - they're already in the recording if they exist.


@ David & Soap
Sorry guys I didn't fully understand this explanation when I first read it. Learning as I go along.
Very true, this!

But still, it leaves out all the sounds that hasn't been recorded through a mic. Synths, or a lined bass.
And more important (i'm guessing) it leaves out all the sounds together, in a mix.

One more try with the sleep thing.
Levi
Title: What's the point of higher sampling rates in audio?
Post by: Soap on 2011-11-08 23:48:26
But still, it leaves out all the sounds that hasn't been recorded through a mic. Synths, or a lined bass.
And more important (i'm guessing) it leaves out all the sounds together, in a mix.


What you are now describing is audio harmonics the artist themselves would not have heard. 



That is (unless we're talking calculated harmonic interference ala someone like The Hafler Trio or Aphex Twin) by definition outside the scope of artistic intent and thus undesirable noise - distortion.


If we ARE talking calculated harmonics only present on playback then clearly the artist would publish in a medium capable of creating said harmonics.

Title: What's the point of higher sampling rates in audio?
Post by: drewfx on 2011-11-09 02:58:00
My understanding is ultrasonic speakers work similar to AM radio - you have an (inaudible) ultrasonic wave modulated by the audio you want to produce. You never hear the ultrasonic signal itself; it's just a carrier for the audible stuff.


Well that's one kind. Where you have a reciever remodulating the ultrasonic sound.

The kind I'm refering to needs no reciever. It's just air slowing the soundwaves down and at a desired distance the sound becomes audible. As I understand it they shape the ultrasonic sound so that when the air has slowed it down to certain wavelengths interferance amongst the short soundwaves produces longer wavelengths and thus a high fidelity audible sound.
The mechanism is called Parametric Array.

Here's some reading: Parametric array - Wiki (http://en.wikipedia.org/wiki/Parametric_array)
Here's an interesting link of an implementation in an art installation: Link (http://blog.svconline.com/briefingroom/2008/11/07/sennheisers-audio-beam-creates-a-sound-ceiling/)


In both of these examples, the ultrasonic wave is used as a carrier wave. The fact that it is demodulated "naturally" without a receiver is irrelevant - the ultrasonic part is still just a carrier wave, and the modulation signal contains the audible frequency information you hear after demodulation.

From sennheiser audio beam (http://www.sennheiserusa.com/tech-spec-database_downloads_spec-sheets_EN_105793) (the speaker used in the art installation):

"The AudioBeam directional loudspeaker works with ultra-sound, modulating the audible sound onto an ultrasonic carrier frequency, much like a radio station does, and then emitting this signal via 150 special piezoelectric pressure transducers.
Audible sound is only generated at a distance from the AudioBeam, when the signal is demodulated because of the non-linearity of air."


[EDIT - Also note that these systems are explicitly taking advantage of the fact the ultrasonic carrier wave is inaudible - if the ultrasonic carrier wave was audible on it's own, you wouldn't want to be anywhere near these types of speakers.]
Title: What's the point of higher sampling rates in audio?
Post by: krabapple on 2011-11-15 16:29:44
I bet most of these 16/44.1 fanatics rips/ripped their best tapes/vinyls using least 24/96.


Juha



My vinyl and cassette rips are at 24/44.1 (the capture card is an M-Audio 24/196) -- the higher bit depth at transfer merely saves me the minor trouble of having Audition convert it upwards.  The technical reason for the higher depth is twofold -- 1) more headroom during transfer (I'm lazy and don't always want to seek out the absolute highest peak beforehand) and 2) keeps rounding errors inaudible when you're passing the audio through a digital processing/production workflow.

If the source audio is an SACD, I'll use an 88 kHz SR...just because I can.    But that's for archiving.  For listening I'm quite happy to use downsampled versions if need be. 


Title: What's the point of higher sampling rates in audio?
Post by: mjb2006 on 2011-11-15 21:12:32
[24-bit] keeps rounding errors inaudible when you're passing the audio through a digital processing/production workflow.

I'm betting they were inaudible anyway. Post ABX results and details of your workflow if you disagree
Title: What's the point of higher sampling rates in audio?
Post by: krabapple on 2011-11-15 21:45:08
[24-bit] keeps rounding errors inaudible when you're passing the audio through a digital processing/production workflow.

I'm betting they were inaudible anyway. Post ABX results and details of your workflow if you disagree



I don't.  But I bet I could make a pathological case where they weren't. I hope you agree that there are legitimate grounds for high bitdepth digital audio workflows, even if not always necessary.
Title: What's the point of higher sampling rates in audio?
Post by: Roseval on 2011-11-15 21:49:18
Having digitized some vinyl I know for sure I won’t go through that again.
A very time consuming affair.
If I had to do it over again, I would go for as much overkill I can afford.
Better save then sorry

Title: What's the point of higher sampling rates in audio?
Post by: 2Bdecided on 2011-11-16 12:38:57
But still, it leaves out all the sounds that hasn't been recorded through a mic. Synths, or a lined bass.
And more important (i'm guessing) it leaves out all the sounds together, in a mix.
It's a good job none of this really matters, because the measurable distortion in speakers is several orders of magnitude larger than the measurable distortion in air. If this mattered, these recordings with high levels of ultrasonics would be unlistenably bad.

Luckily, they're mostly OK. And any distortion in air, 10s of dBs lower than the distortion in the speakers, is irrelevant.

Unless, I suppose, you're running a PA rig at home - but then it's the distortion (and damage) in your ears that should concern you.

Cheers,
David.
Title: What's the point of higher sampling rates in audio?
Post by: HTS on 2012-03-20 02:35:58

And are you 100% sure that the plugin is samplerate aware???

I've been part of this past year fixing most of the plugins of my program* so that they don't perform differently at different sample rates.


If nonlinear processing is done in the digital domain, certain high harmonics can reflect back down into the audible range. Any signal that is generated in the digital domain above the Nyquist frequency  aliases around the Nyquist frequency.  For example, an 8 KHz tone in a 44.1 KHz sampled sytem is distorted by a fourth order digitally-implemented nonlinearity and would be expected to produce a fourth harmonic at 32 KHz.  Since 32 KHz is 10 Khz higher than the 44.1 KHz Nyquist frequency of 22 KHz, the  4th harmonic is aliased down to 12 KHz where it is far more audible than it would be in a digital system with a much higher sample rate or an analog system. In a 96 KHz system, the same nonlinearity would produce a foruth harmonic at 32 KHz where it would not be audible.

Aliasing can also impact certain dynamic range compression or expansion algorithms that are based on nonlinarities implemented in the digital domain.

Most kinds of processing that are used in music synthesis and mixing are linear, so aliasing is not a common problem.


Aren't algorithmic reverbs based on non-linear transformations? Especially these days they offer many types of adjustments to the parameters for effects.
Title: What's the point of higher sampling rates in audio?
Post by: Woodinville on 2012-04-07 05:41:38
Aren't algorithmic reverbs based on non-linear transformations? Especially these days they offer many types of adjustments to the parameters for effects.


How so? huge IIR filters are not nonlinear.
Title: What's the point of higher sampling rates in audio?
Post by: knutinh on 2012-04-09 12:20:52
How so? huge IIR filters are not nonlinear.

I was under the impression that standard reverb algorithms introduced time-variant "stuff" in the filter topology to reduce audible artifacts, making them non-LTI?

Nevertheless, I assume that the kind of nonlinearities "warned" about by Arnold in terms of aliasing was of a very different kind: Signal-dependent gain with non-smooth envelope.

-k
Title: Re: What's the point of higher sampling rates in audio?
Post by: shusse82 on 2016-11-20 09:38:41
The highest frequency at 44.1 kHz 16 bit rate is 22.05 kHz according to Nyquist. Right?

In one second, the 44,100 samples would just pick up the crest and trough of the 22,050 waves. Do our ears just need the alternation between top and bottom of every wave to hear it? Or do we fill in the blanks?

Wouldn't higher sample rates like 192 kHz help when lots of instruments are playing at once? Take the 16 vs 24 bits out of the equation since it is only used for volume. Let's assume you only need the crest and the trough, which would expressed as a 1. Anything in between, and also no sound, would be expressed as a 0. If there are multiple instruments, then you would get a 1 for that sample, if there are any instruments at all it would be 1, but if there are no instruments hitting a frequency crest or trough, then you would get a 0.

Playback would be do the opposite. The player would look for rates of repeating ons and offs to produce a pitch. The problem I see here is that a 1 kHz note would perfectly overlay a 100 Hz note. And you would just hear the 1 kHz note. Of course real notes rarely line up like that.

This still leaves you with having to split up the 44.1 thousand samples amongst several instruments in a full band. Rhythm guitar has 6 notes with half a dozen harmonics each (36), plus the one bass note (6), plus the four at most notes of a drummer times maybe 6 harmonics (24), plus the one to three notes at once of the lead guitar times 6 harmonics (18) equals 84 pitches that need to be sampled. This is actually more like a 100 pitches estimated since notes move up and down a little within even a second.

If the computer can only do 41,100 samples per second total, then each harmonic only gets 411 samples per second. This would be fast enough for low notes, but you can clearly hear high notes on full band CDs.

I've read somewhere that the samples are constant, so maybe the computer is predicting sounds for playback based on whether the switches are on of off for every one of the 41 thousand samples per second. If the computer sees a crest or trough at sample 1, 101 and 201, etc, but also sees something at 1001, 2001, etc. It probably drops the bass note.

I've noticed that when I play a bass note perfectly in time and pitch, I can't hear it, but if the note is way off, it's very noticeable. Problem is you need notes to be heard. Which leads me to conclude that the higher rate of 192 kHz would be better, so the computer can hear instruments that are almost overlapping and catch more stuff as long as the lower notes are not perfectly perfect.

You might not run into overlapping frequencies at higher pitches because they are spread apart like 980 Hz then 1030 Hz etc, but very low notes around 20 Hz are so close together they would overlap with higher frequencies more easily. ie. If a low B is 25 Hz, Low C is 27 Hz, and a Low C# is 30 Hz, then one of those frequencies is bound to be an exact fraction of a higher note so that the computer just doesn't see it.

I have a solution. The computer would record in pitch groups. First would be a section between 4 Hz and 7 Hz. It would run 7  samples per second just for this, then test every second to see if there is movement in this range. Next, the computer would record 15 samples from 8 Hz to 15 Hz, then test for this band, and so on. Because this has a doubling pattern, the code would simply be 2 to the x through 2 to the x + 1 minus 1.

The test would be harder, and would only work with real musicians/vocalists. The computer would look for the naturally slight pitch shifts and know that if a note disappears, but then comes back, when there are higher overlapping notes, then the notes still exists and should be produced at let's say 30 Hz.

The computer would use three samples for this check. First, is there imperfect notes before, second, are there imperfect notes after, third, in the empty space is there possibly overlapping notes that are half, third etc a higher frequency. The lowest bands would check less often, and the higher notes would check a lot, but this would be fine if you have 192 thousand samples.

The only problem is a bass player often isn't playing for half a second when the rest of the band is playing. At 240 BPM, you get 4 beats per second, and 8 eighth notes per second in 4/4. A measure at 240 is 1 second. If the recorder is sampling right in line with the beginning of the measure then there would be 2.8 thousand samples at 44.1 between the 1st and 2nd eighth note, but if the bass lines up perfectly as a fraction of a higher pitch, then the computer needs to see the sudden drop outs in relation to the beat. The bass player would drop out at the end of his note of course, but the computer should watch for when he drops out, then comes back suddenly anytime before the 2.8 thousandth sample. If he comes in anytime after 2 thousand or so and continues, then it is part of a new note being played.

This solution would prevent bassists from going any faster than 8th notes at 240 BPM or 16th notes at 120 BPM. Personally, I don't play any faster than this because it just turns into mush even still in analogue.

To wrap it all up, a higher sample rate would catch even the smallest differences so that what overlaps at 44.1 kHz doesn't overlap at 192 kHz. Then you don't need some complex work around to try and figure out where the bass notes are.
Title: Re: What's the point of higher sampling rates in audio?
Post by: dhromed on 2016-11-20 12:23:43
Quote
This still leaves you with having to split up the 44.1 thousand samples amongst several instruments in a full band.
How is it possible to know absolutely nothing about sampling and still have enough self-confidence to write FIFTEEN paragraphs of nonsense?
Title: Re: What's the point of higher sampling rates in audio?
Post by: [JAZ] on 2016-11-20 12:25:19
@shusse82 : Please, don't take this as an attack to you, but your post is extremely crazy.

Nyquist says that you need 2*N samples to record frequencies up to and not including N. This means every frequency, even non-integer frequencies.
The bit depth can, in fact, help adding precision to non-integer frequencies, so bit depth is not only a volume issue (volume understood as max value where noise is under the threshold of hearing. Obviously volume is not limited itself by bit depth)

It needs 2*N since it samples a wave, and a wave oscillates between a peak value and a valley value. This representation is what becomes the frequency in itself when an amplifier and a speaker reproduces it. To understand how the ear interprets this variation in pressure that the oscillating wave produces is relatively easy, but you need to study about the cochlea and the inner ear ( see https://en.wikipedia.org/wiki/Ear and specifically the inner ear link). Suffice to say it is not about detecting a peak.


From here on, your post is where it starts to be crazy. I can't even try to guess where do you want to go. What I first think is that you imply that 44Khz is not enough, that you could use 192Khz, and then you seem to try to describe a way to encode audio differently, very limited to what it can store (because you focus only on instruments, so I may ask... will it be able to store vocals?) and even you describe difficulties in how you could store what you try to store.

That's why I won't reply about that other part, more so when you end saying that maybe with 192Khz you would have enough.
Title: Re: What's the point of higher sampling rates in audio?
Post by: Wombat on 2016-11-20 15:49:52
Reading that reasoning above on the same page with J.D. posting must be some kind of comedy  :)
Title: Re: What's the point of higher sampling rates in audio?
Post by: Apesbrain on 2016-11-20 16:08:56
How is it possible to know absolutely nothing...and still have enough self-confidence...(to invent your own "facts".)
Welcome to the new reality. Why bother to understand how things really are when it's so much easier to just make it up?
Title: Re: What's the point of higher sampling rates in audio?
Post by: greynol on 2016-11-20 17:22:07
Funny how Meridian could contrive a test that could only manage to get a dataset that showed a 56% success rate.  With this latest wrinkle it's a wonder how digital audio could even work at all!
Title: Re: What's the point of higher sampling rates in audio?
Post by: jumpingjackflash5 on 2016-11-20 17:53:33
I wonder why those topics are such often repeated, even if I in the past also contributed to their length.

I think it is clear given todays knowledge.

48 kHz is the optimal sample rate for human audio and rates above that we can use, but it does not bring any practical (audible) nor theoretical (impacting spectrum of human hearing) advantages. We can do it but only for technical/futureproof reasons. 44.1 kHz is the CD standard and since majority of recordings is available in that format it is very important for DACs and other audio equipment to support it well.

16 bit audio is completely recommended for practice - audibly transparent playback. When we record in 24 bit we can keep it in that format (if space is not a concern) even for playback, then we do not have to dither/noise shape and we have theoretically better source for subsequent conversions, if required.

Until somebody proves - by ABX or other reproducible method of testing - something else, then these discussions do not bring new knowledge.
Title: Re: What's the point of higher sampling rates in audio?
Post by: greynol on 2016-11-20 18:38:58
1974, WTF???
Title: Re: What's the point of higher sampling rates in audio?
Post by: xnor on 2016-11-20 18:49:39
Doing a very simple Bayesian analysis of  90 successes out of 160 attempts in the style of my "Bayes Factors for ABX tests" post, the result is actually evidence for the null hypothesis.

Even if we start with a completely dishonest prior that distributes the 95% most credible values (of the parameter that describes the performance of the listener) above 50% with the peak at exactly 90/160%, the result is still only very weak evidence for the alternative hypothesis.


@greynol: Simply remove a few numbers from the quote timestamp and you'll end up with such dates.
Title: Re: What's the point of higher sampling rates in audio?
Post by: dhromed on 2016-11-20 18:51:14
the quoter puts the time in ms in the tag but it's clipping the 3 least significant digits or something.

edit
ha, ninja'd
Title: Re: What's the point of higher sampling rates in audio?
Post by: greynol on 2016-11-20 19:04:24
I wonder why those topics are such often repeated, even if I in the past also contributed to their length.
...and there you were thinking that being mocked and ridiculed with such vehemence was personal.  Sorry bro, neither you nor your placebophile tendencies/misunderstandings were(?) all that unique/special.

48 kHz is the optimal sample rate for human audio
Declaring what is "optimal" will likely get you into trouble; especially when the ability to hear pure tones does not neatly correlate with the ability to hear the same frequencies in actual consumer content, pesky psychoacoustics!

technical/futureproof reasons.
theoretically better source for subsequent conversions
The needle on my speculative//hand-waving bullshit meter just moved.  How fun!
Title: Re: What's the point of higher sampling rates in audio?
Post by: Thad E Ginathom on 2016-11-20 19:58:43
The highest frequency at 44.1 kHz 16 bit rate is 22.05 kHz according to Nyquist. Right?

...
...
...

To wrap it all up, a higher sample rate would catch even the smallest differences so that what overlaps at 44.1 kHz doesn't overlap at 192 kHz. Then you don't need some complex work around to try and figure out where the bass notes are.

Amazing. I've seen heaps of people, first and foremost myself, completely misunderstand digital audio, but to follow a fictional understanding  with a fictional alternative is quite novel. Well, you have an idea. Go for it! See if it works. If you can.

In my experience of systems management and support, people who say things like, "the computer would..." actually don't have a clue how, or whether their request is even feasible. The have no clue how "the computer would..." Sorry, making it up as you go along does not write working software or create usable devices.

I am very much a novice at this stuff. But, even if I have not, and probably never will, master the maths of what happens in the gaps between those samples, at least I now understand that I am not, actually, as some of the audiophiles would have it, listening to something like speeded-up stuttering. I had to read a lot, watch some videos, listen to some actually-well-informed people etc etc to get this far. Now its your turn. If you choose to do so, then welcome to the club.  :)
Title: Re: What's the point of higher sampling rates in audio?
Post by: xnor on 2016-11-20 22:28:14
...
No, just no.

Start with the basics. This is a nice introductory video: Digital Show & Tell (https://xiph.org/video/vid2.shtml)
Title: Re: What's the point of higher sampling rates in audio?
Post by: Apesbrain on 2016-11-20 23:49:46
1974, WTF???
I caused that, sorry. Editing on mobile and I got overly aggressive with the backspace key.
Title: Re: What's the point of higher sampling rates in audio?
Post by: jumpingjackflash5 on 2016-11-21 06:03:30


technical/futureproof reasons.
theoretically better source for subsequent conversions
The needle on my speculative//hand-waving bullshit meter just moved.  How fun!

I know that meter (would not call it bullshit however) is very sensitive :), and it is good to have it working ...

I meant by those reasons digitizing very unique analog content when it is necessary/desired to digitally store  information e.g. up to 30 kHz. Then we can theoretically use 96 kHz sample rate, but no more and even then it is only for archival purposes, we still cannot hear that. For normal/standard audio recordings (including common vinyl records), there is nothing what calls for or forces us to use higher SR that 48 kHz.

I agree that word "optimal" is risky to use, especially in discussions here.

Similar to many audio processing standards like Mastered for ITunes or workflow of good engineers like Bob Ludwig, I prefer to use 24bit source for resampling/editing/creating MP3s if that processing is required and 24 bit source is available. This, however, does not claim that using 16 bit source for that purpose would be practically (audibly) different.






Title: Re: What's the point of higher sampling rates in audio?
Post by: greynol on 2016-11-21 20:01:54
This, however, does not claim that using 16 bit source for that purpose would be practically (audibly) different.
Which is all that matters.

Re:Apple/ Bob Ludwig
I see you're reverting to hollow pleas to authority instead of providing evidence once again, sigh.
https://hydrogenaud.io/index.php/topic,111271.msg918333.html#msg918333
https://hydrogenaud.io/index.php/topic,111271.msg918387.html#msg918387
Title: Re: What's the point of higher sampling rates in audio?
Post by: Chibisteven on 2016-11-21 21:56:42
Just got done recording a bat conversation @ 576 KHz.  The problem is I couldn't hear any of it.  :P
Title: Re: What's the point of higher sampling rates in audio?
Post by: splice on 2016-11-22 04:56:25
Just got done recording a bat conversation @ 576 KHz.  The problem is I couldn't hear any of it.  :P

What sort of bandwidth does a bat chirp cover? As in, what are the typical lowest and highest frequencies present in a chirp?
Title: Re: What's the point of higher sampling rates in audio?
Post by: Cartoon on 2016-11-22 07:03:44
What sort of bandwidth does a bat chirp cover? As in, what are the typical lowest and highest frequencies present in a chirp?

A quick google says 10-100kHz, with some sources claiming >150kHz. That would leave some merit to sampling at 192kHz or even higher if you are into bat music or sound effects, or possibly dolphin and whale music as they can hear similar frequencies according to my quick google  :) Since sound is such an important sensory input for bats, like our vision, would bat music make them disoriented or "drunk"? Maybe a bat could ABX 16 bit/48kHz from 32 bit/192kHz?

But should we care?
Title: Re: What's the point of higher sampling rates in audio?
Post by: Chibisteven on 2016-11-22 08:08:40
I did a quick search on hearing range and read this wikipedia article (https://en.wikipedia.org/wiki/Hearing_range) before posting my funny post making fun of higher sample rates to at least be sure.
Title: Re: What's the point of higher sampling rates in audio?
Post by: Thad E Ginathom on 2016-11-22 09:15:41
Just got done recording a bat conversation @ 576 KHz.  The problem is I couldn't hear any of it.  :P
OK, so you posted this for the humour... but some people must have such needs in real life research. Hearing it? Well, slow the tape down of course!
Title: Re: What's the point of higher sampling rates in audio?
Post by: Woodinville on 2016-11-24 13:16:54
How so? huge IIR filters are not nonlinear.
I was under the impression that standard reverb algorithms introduced time-variant "stuff" in the filter topology to reduce audible artifacts, making them non-LTI?

Nevertheless, I assume that the kind of nonlinearities "warned" about by Arnold in terms of aliasing was of a very different kind: Signal-dependent gain with non-smooth envelope.

-k

That would depend on the reverb, I might say.
Title: Re: What's the point of higher sampling rates in audio?
Post by: smok3 on 2016-11-25 19:44:46
An interesting test would be 32 kHz vs 44.1 kHz (or 48 kHz). I did a limited one some years ago and irc lots of people failed even that.

p.s. Actually DCC (https://en.wikipedia.org/wiki/Digital_Compact_Cassette) used that sampling rate.
Title: Re: What's the point of higher sampling rates in audio?
Post by: greynol on 2016-11-26 04:36:34
Exactly.
Title: Re: What's the point of higher sampling rates in audio?
Post by: Shinsekai on 2016-11-26 06:00:07
Interesting test, indeed.

I've taken a sample from one of my favorite tracks (sample1). Then resampled that to 32 kHz using lvqcl's SoX component (default settings), and again to 44.1 kHz (sample2).

And...
Code: [Select]
foo_abx 2.0.2 report
foobar2000 v1.3.13
2016-11-25 22:27:37

File A: sample1.flac
SHA1: 765511fc2863f842b789ce304cb1a1cafa5e31a5
Gain adjustment: -6.21 dB
File B: sample2(32kHz).flac
SHA1: 8283b03fe88a153c88681b8a98b5ec0acbc9ccd9
Gain adjustment: -6.11 dB

Output:
WASAPI (event) : Speakers (Realtek High Definition Audio), 24-bit
Crossfading: NO

22:27:37 : Test started.
22:28:22 : 00/01
22:29:25 : 01/02
22:29:57 : 01/03
22:30:30 : 02/04
22:30:56 : 02/05
22:31:23 : 03/06
22:31:48 : 04/07
22:32:18 : 04/08
22:32:45 : 05/09
22:33:08 : 05/10
22:33:33 : 06/11
22:34:02 : 06/12
22:34:31 : 06/13
22:34:58 : 06/14
22:35:28 : 06/15
22:35:52 : 06/16
22:35:52 : Test finished.

 ----------
Total: 6/16
Probability that you were guessing: 89.5%

 -- signature --
b750e35d423ce912218069de0333cc83c2ab2c26
...Failed miserably.  :D
Title: Re: What's the point of higher sampling rates in audio?
Post by: greynol on 2016-11-26 06:08:43
Pesky psychoacoustics.
Title: Re: What's the point of higher sampling rates in audio?
Post by: rw11 on 2016-11-26 23:13:34
Just got done recording a bat conversation @ 576 KHz.  The problem is I couldn't hear any of it.  :P
OK, so you posted this for the humour... but some people must have such needs in real life research. Hearing it? Well, slow the tape down of course!

IIRC, bat hearing rolls off a lot above 100 kHz; maybe dolphins... (they can image with ultrasound, as the eye does with EMF)
Title: Re: What's the point of higher sampling rates in audio?
Post by: augustine on 2017-04-25 23:44:24
The only possible advantages I can see for 192khz sample rates are lower latency for live instruments into DAWs and digital equipment, and recording wildlife that uses very high frequency sounds .
Title: Re: What's the point of higher sampling rates in audio?
Post by: augustine on 2017-04-25 23:52:32
Just got done recording a bat conversation @ 576 KHz.  The problem is I couldn't hear any of it.  :P
OK, so you posted this for the humour... but some people must have such needs in real life research. Hearing it? Well, slow the tape down of course!

I am one of those people. I have been conducting studies in the Amazon of very high frequency animals (insects etc)  slowed down to audible ranges.  There are some very strange sound sbeen made in the very high frequency bands. My challenge is nyquist of course and I hope to head back out there with equipment capable of recording higher frequencies (ie 576 khz sample rate would "only "record sounds up to 288 khz)