sample rates issue

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Actually that's another misconception that because 44.1 is exactly half of 88.2, the math is simple and it just consists of dividing the number of samples by 2. For proper sample rate reduction, there is still quite a bit of tricky math and averaging going on.

I didn't say it's "simple" math...I said it's "cleaner" math, and there's no misconception about that when you consider the math to go from 96 to 44.1...
...but of course, I agree with you that it's complicated math in either case. :)

And just to put my overall digital perspectives into proper perspective... ;)
If can get my hands on a clean 24-track 2", and maybe also step up one level with my analog console (currently using a TASACM 3500, no complaints, but I would love something higher-grade) ...
...after that, I wouldn't give digital a second thought and all the crazy debates.
Though I have to admit, I enjoy the cut/paste/comp editing power pf digital...but some days I dream of even forgoing that, and just going back to all analog and living with my tracks as-they-fall...no editing at all.
God...I use to be able to knock out tunes in no time.
Who says digital is quicker!?!?

:eek:
The sheer madness of that thought!!!
Going back to all analog!
:D
 
I didn't say it's "simple" math...I said it's "cleaner" math, and there's no misconception about that when you consider the math to go from 96 to 44.1...
...but of course, I agree with you that it's complicated math in either case. :)

And just to put my overall digital perspectives into proper perspective... ;)
If can get my hands on a clean 24-track 2", and maybe also step up one level with my analog console (currently using a TASACM 3500, no complaints, but I would love something higher-grade) ...
...after that, I wouldn't give digital a second thought and all the crazy debates.
Though I have to admit, I enjoy the cut/paste/comp editing power pf digital...but some days I dream of even forgoing that, and just going back to all analog and living with my tracks as-they-fall...no editing at all.
God...I use to be able to knock out tunes in no time.
Who says digital is quicker!?!?

:eek:
The sheer madness of that thought!!!
Going back to all analog!
:D

I completely understand. It's a workflow thing. Because of what I do, the computer is an integral part of it, but believe it or not, when it comes to actually writing notes, I still use pencil and paper :) It might not be as fast as clicking around, but that clicking around gets in the way. Plus all these DAWs are soo damn linear, not to mention it's not easy to write multiple parts together, specially when I work in terms of polyphonic or multitimbral chordal structures, for me, nothing beats a paper score. I don't even like sitting in front of the piano or any type of instrument when developing ideas.

Then, I can sit down and think of sound design, and all that nonsense.

Once that's done, then I can sit down and practice what I wrote so I can actually play the damn thing :D

I have never worked with a quality reel to reel tape, so I don't have that experience to compare to. Although the amount of experimentation that some computer software allows (NI Reaktor, Plogue Bidule, AudioMulch and the like) is integral to my needs. So not knowing tape, I don't miss it, and probably my stuff would be completely different had I started with tape to begin with.

What I miss though is a quality acoustic space, time (it's hard when you work 70 hour weeks and have a 3 month old) and some outboard gear that I'd definitely like to get my hands on eventually (Culture Vulture, some of the Metasonix disasters, a couple of different variety of reverbs to complement my Kurzweil unit and the like) upgrade to a better interface (maybe one of the RME jobs) and I'll be happy.
 
well i was wondering i am suffering from major cpu overload when recording guitar parts in logic... would recording at lower sample rates eleviate this and make it easier for me to record multiple parts without any having the cpu overload which is preventing me from finishing my projects? do they relat relate to each other?
 
well i was wondering i am suffering from major cpu overload when recording guitar parts in logic... would recording at lower sample rates eleviate this and make it easier for me to record multiple parts without any having the cpu overload which is preventing me from finishing my projects? do they relat relate to each other?
Yes, but there are other things you can do as well. For starters, once you go to the mixing stage (presumably this is where you start adding all those plugins, etc) increase the latency of your soundcard. BTW what audio interface are you using?

Buy more memory.

Logic... that means you are on a Mac? Mind if I ask what model?
 
If you allow the debate to rumble away over your head, like a tropical thunderstorm, you can sneak away out the back and take a quick peek at Glen's first response.

You will get satisfactory results using 16/44.1, which was what people had been successfully using for years. If you are worried about CPU load, bog standard 16/44.1 creates smaller files and is less CPU intensive. You should notice an improvement if you increase the bit rate, i.e. go from 16 to 20, 22 or 24, depending on what your interface is capable of.

CPU overload can be attribute to a number of things: the number of tracks you have open, the number of plug-ins you are using, how resource-hungry those plug-ins are. There are also non-audio reasons for CPU overload. For example, having other programs running at the same time (even such things as screen savers). The presence of viruses and malware can severely degrade performance.
 
...but believe it or not, when it comes to actually writing notes, I still use pencil and paper :)

On my console are folders for every project....and inside are notes (handwritten w/pencil) for every tracking session, every instrument...gear used, knob positions, mic placement...even stuff like which guitar & pickup was...etc...etc...etc.

I laso have notes (and digital pics) of all my mixing/mastering...every connection, knob position...etc...etc.

Now I know the ITB guys are laughing at this, 'cuz with ITB you just save your project files and recall as needed.
But...it's still not the same as having hardcopy notes, 'cuz even with ITB, not everything happens ITB.

But yeah...I have handwritten notes from 30 years ago! :D
 
On my console are folders for every project....and inside are notes (handwritten w/pencil) for every tracking session, every instrument...gear used, knob positions, mic placement...even stuff like which guitar & pickup was...etc...etc...etc.

I laso have notes (and digital pics) of all my mixing/mastering...every connection, knob position...etc...etc.

Now I know the ITB guys are laughing at this, 'cuz with ITB you just save your project files and recall as needed.
But...it's still not the same as having hardcopy notes, 'cuz even with ITB, not everything happens ITB.

But yeah...I have handwritten notes from 30 years ago! :D
Sorry, I think I wasn't clear in my post. I referred to musical notes, I mean writing a score on staff paper. For notes of the type you are talking about, I usually use the "notes" feature in Cubase. :)
 
I always wonder about industry pros...the ones with the golden ears...making claims about greater clarity and depth when they switch their expensive converter from 48 to 96 to 192.

I mean....sure there's the math, but what are these guys really hearing?
If they use the same converter and just flip their sample rates...what else is different?

And I don't ask that question to initiate a deep debate about sampling rates...but I'm just pointing out why so many people who are not golden ear pros end up considering that notion that more=better.
They are being TOLD it does...by people they respect/admire/believe...etc,.
First of all, just because someone is an "industry pro" does not necessarily mean they have golden ears. I don't mean that as disrespect, it's just a statement of fact.

The number of pros in this racket that truly have golden ears is probably quite small; amd many of them that *did* have golden ears when they started out in analog 40 years ago, probably don't have the same ears now that they used to, just due to a combination of age and extended exposure to sustained sound levels. (Even listening to music at low levels will wear down the hearing over time when done more often than someone who doesn't do it as often.) I'll bet that if everybody told the truth about it, the number of real pros who work with some level of tinnitus in one or more of their ears would be shocking.

And let's not forget that even the pros are human beings, and there is a tendency for audio engineers to be gear sluts. Again, no disrespect, it's a lovable quirk if nothing else. Engineers like their toys. It's like the fisherman who has a zillion rods and reels and lures and flies and so forth, and is always on the lookout for the the latest (new or vintage) k3wl toy. And it's human nature to defend or justify esoterica, even if the justification is not necessarily there.

Add to that the "gear list factor" in recording studios; the documentable fact that half the gear in big studios rarely ever actually gets used, but if they didn't have it in their gear list, they would not attract as many customers. And as John ("Massive") said correctly back in the beginning of this thread, the majority of industry professionals actually do NOT use higher sample rates most of the time.

As far as the question about flipping the switch and hearing a difference, there's a couple of things that often can account for that. The fist is that a given converter may actually sound better at that rate. The second is psychosomatic, which is why truly double-blind testing is the only way to know for sure what one is really hearing. The third is that any given converter design may have a "sweet spot", for lack of a better term. To go back to the car analogy, it's kinda like where an engine's power band may be located, with this engine delivering more power (or maybe being more efficient) at this gear ratio and RPM, but that engine performs better at a different range. As was said near the beginning, some converters may sound better at 44.1 because they just are not designed to work well at 88.2, or it could also be vice versa.

The problem with the "math" of A/D and information theory (which is a whole field that covers a whole lot of stuff that has nothing necessarily to do with music or audio) is that it is very difficult for the non EE/non info theory specialist, and it's not easy even for the specialists. Things like Fourier transforms and the Nyquist theorem and such sound like fairly simple concepts on the surface, and that can be very misleading, because the actual math behind them is way over the heads of most audio engineers, even the pros.

I don't claim to understand it all myself, either. But I understand enough to know that many of the textbook representations and common understandings of how it all actually does work are off the mark. There's a famous saying about quantum theory in science, that says something along the lines of, "there are only five people on this planet that truly understand quantum theory, and four of those are probably faking it." Information theory and the actual science and mechanics of A/D//D/A is kind of like that in that the reality of it is not very intuitive, and that the pictures and concepts we imagine to help us understand are not really right.

The fact remains that the Nyquist theorum states that in order to reproduce a signal at a given frequency within a limited bandwidth signal, you need to have a sample rate of at least twice that frequency. It does NOT state that the higher the frequency, the more accurate the representation at that frequency, it says that at a sample rate of twice the frequency, that frequency is *reproduced*. "Reproduced" here means an exact copy, i.e. a lossless representation (remember this is *information* that Nyquist is really referring to, and any loss of information would not be a reproduction.)

The stair step/connect-the-dots/thinner slice model usually used to teach about sample rate does not really work on a real level because there's no way to connect the dots at only twice the sample rate to make the original waveform with any certainty or accuracy. That model is not really an accurate description of what's actually going in.

Just like with many concepts in high science, such as it is impossible to accelerate a body to a speed faster than the speed of light, unless one can comprehend the math themself - which most folks can't - they're best off just accepting it and moving on to the consequences of it, otherwise they'll just be another one of those crackpots who ignore all the real math and real evidence and write nonsense letters to some scientist claiming that they have a simple alternative to Relativity.

Any difference in sound between sample rates on any given converter is virtually certain to NOT be a function of the actual sample rate itself, because that's what Nyquist shows. Any actual difference heard - real or imagined - will be due to some other factor or factors than the sample rate itself.

G.
 
There's a famous saying about quantum theory in science, that says something along the lines of, "there are only five people on this planet that truly understand quantum theory, and four of those are probably faking it."

So....are you saying we need Stephen Hawking to weigh in on the samle rate issue before we have a definitive answer? ;)

:D


The fact remains that the Nyquist theorum states that in order to reproduce a signal at a given frequency within a limited bandwidth signal, you need to have a sample rate of at least twice that frequency. It does NOT state that the higher the frequency, the more accurate the representation at that frequency, it says that at a sample rate of twice the frequency, that frequency is *reproduced*. "Reproduced" here means an exact copy, i.e. a lossless representation (remember this is *information* that Nyquist is really referring to, and any loss of information would not be a reproduction.)

The stair step/connect-the-dots/thinner slice model usually used to teach about sample rate does not really work on a real level because there's no way to connect the dots at only twice the sample rate to make the original waveform with any certainty or accuracy. That model is not really an accurate description of what's actually going in.


......


Any difference in sound between sample rates on any given converter is virtually certain to NOT be a function of the actual sample rate itself, because that's what Nyquist shows. Any actual difference heard - real or imagined - will be due to some other factor or factors than the sample rate itself.

I wasn't opposing the Nyquist theorem, and I do understand what it is saying (at the fundamental level)...
...so maybe the "slice" analogy was a bad choice, but as you admit, it DOES get used often to give folks a quick "visual".

And while there may be many factors other than sample rate contributing to what someone hears coming from a converter...the point I was making is that many pros seem to tout the higher rates (even if privately, they don't use them :eek:). Do they say what they say maybe just for “show and sell”? :)
But open any audio magazine and try to find one place where anyone is openly promoting the use of LOWER sample rates...and maybe that’s what leads to the "psychosomatic" responses from others…???

One thing though…it is impossible for any one person to know what another person is hearing...exactly.
Scientific hearing tests and all other things being equal…there is still a large amount of personal, subjective interpretation involved…and that tends to often trump the math (in many cases).
You know…people write-n-read gear specs all day…but in the end, everyone asks, “how does it sound?”…and the answers are often not the same ones given in a room full of people (with or without blindfolds). :D

So IMO…its very easy, and not unusual, for sound quality decisions NOT to be made solely on the math…which may be why folks don’t always believe the math when it comes to audio....???

I dunno...there's just lots of variables!

I am curious how many people here are tracking at 44.1/48 for reasons other than CPU/drive overload...? How many are using lower sampling rates because the prefer the sound or are convinced anything higher is a waste of HD space...?
I may be wrong...but I get the feeling more are going for the 88.2/96 rates if they have robust DAWs.

Personally...I don't think it's that much of an issue at the home/project studio level...so no real need to debate the minutia.
I guess when you really get up in the stratosphere...and all your gear is at the same relative high-end level...then it's probably more important to leave no stone unturned when seeking the purest audio quality.

What do you think?

Me…my target is “good sound”…and not so much any kind of “laboratory precision” or some such thing…which is why tape still works for me AFA any math is concerned, even though digital can claim greater precision and dynamics.
But like I said…I may try to do some “scientific” sample rate comparisons with my rig at some point…just out of curiosity...to see if my 88.2 choice is wrong.
 
So....are you saying we need Stephen Hawking to weigh in on the samle rate issue before we have a definitive answer? ;)
No, I'm saying we already *have* a definitive answer, thanks to Nyquist. There's no ambiguity in it. Meaning that any perceived differeince is for other easons. Period.

The problem is that people refuse to accept that point on faith alone - which they must do because they don't have the level of understanding required to accept it on understanding.
the point I was making is that many pros seem to tout the higher rates
Why is that so important? F___ "the pros". When you know the facts, and you know what you hear, what else do you need? Also, the number of pros that tout the higher rates is not necessarily a majority of them.
One thing though…it is impossible for any one person to know what another person is hearing...exactly.
Scientific hearing tests and all other things being equal…there is still a large amount of personal, subjective interpretation involved…and that tends to often trump the math (in many cases).
But you can't engineer for that. You can only engineer for what *you* hear, and hope your ears are up to the task.

And it also proves the point; it's yet another example of there being something else going on other than the sample rate itself.
So IMO…its very easy, and not unusual, for sound quality decisions NOT to be made solely on the math…which may be why folks don’t always believe the math when it comes to audio....???
Anybody that's been around here long enough knows that I'm a guy who always argues against following the numbers and following the math. So it humorously ironic that you're trying to put me into that corner.

This is different. I have said a few time already, and will say it one more time, if 88.2 sounds better to you, then by all means use it, and I'll support your use of it to the death. I'm just saying that it's not because of the sample rate itself *because it's against the laws of physics* for it to be because of the sample rate. It's not just math, it's how the world is built.

The fact that you think your converter sounds better at 88.2 I'm not contesting. I have no way to know one way or the other, because I have never heard it. I'll take your word for it, quite frankly. All I'm saying is that is not evidence in any way, shape or form that the sample rate itself is the reason for the change, that the rules of the world we live in tell us in fact that it's not, and that there is no logical reason to use that as a basis for recommending a higher sample rate to anyone else.
miroslav;3201326 I may be wrong...but I get the feeling more are going for the 88.2/96 rates if they have robust DAWs.[/QUOTE said:
Again, even if that were the case, so what? Why does what anybody else believes or does change anything? Truth is not a matter of public opinion.
I guess when you really get up in the stratosphere...and all your gear is at the same relative high-end level...then it's probably more important to leave no stone unturned when seeking the purest audio quality.

What do you think?
Then the principle of keep the signal chain short kicks in. The better quality my overall the signal chain, the less I want to do to it to mess it up. This will include extra stages of sample rate conversion. Which is why I said in the previous post, that the increase in audio quality (for whatever reason) by running the converter at the different speed had better more than offset the decrease caused by the extra processing to get it back to the target rate, otherwise I'm making things worse, not better.
But like I said…I may try to do some “scientific” sample rate comparisons with my rig at some point…just out of curiosity...to see if my 88.2 choice is wrong.
You'll need to include more than one make and model of converter to tell you anything more than whether it's just your particular converter that sounds better at that speed. It'll have to be double-blind, and you'll need to include at least one control group of data. Additionally, you'll have to run the identical tests on more than just yourself if you want to find out whether it's just your ear's preference or not.

G.
 
I think you’re mistaking my comments as some form of counter-argument to what you are saying.
Not at all.
Rather I was giving my opinion on why *people* often end up thinking/believing some of the stuff about sample rates and converters, and making the point that when higher rates are “pushed” by pros or ads or whatever, and few if any are openly saying that higher rates maybe be pointless…
…the higher numbers tend to win out, maybe to the point where people DO hear things….even if they are not there, and yeah, check out the Emperor’s new clothes. ;)

I understand your point about Nyquist…and agree that the “slices” analogy was bad…so I’m not pushing you into a corner where you must debate the validity of the math! :)

While I don’t buy my equipment or make my recording decisions based just on what I read in magazines or on forums….:D ….like you said, there are times when you have to go on faith...’cuz who the F**K wants to spend all their studio time running double-blind tests and comparing specs. We mostly want to make music that sounds good, regardless how the math works out, and I’m sure you will agree that things don’t have to be that complicated for anyone to do that.
Like the sig I stole from the Joe Meek quote…”if it sounds right, it is right”…so while it may be that my converter or some other portion of the signal chain is the main cause of what/how I hear things…I don’t want to worry about it too much, and the whole 44/48 VS 88/96 debate may not really be all that critical in the big scheme of things. IOW…everyone has to take their ENTIRE signal chain into account…and not so much the just the mic or just the converter/sample rate…especially at the home/project studio level where few people have million-dollar studios with all top-notch gear in their signal chain.

AFA the tests I was thinking of doing…I only would want to compare how the different sampling rates sound on my converters….so I don’t think I need another/different converter for that.
I just need a good blindfold and someone to switch the rates for me…right? :D
 
I think you’re mistaking my comments as some form of counter-argument to what you are saying.
Not at all.
Well, in that case I'm apparently guilty of being slow on the uptake again; in which case, my apologies.
AFA the tests I was thinking of doing…I only would want to compare how the different sampling rates sound on my converters….so I don’t think I need another/different converter for that.
Not if you're just testing your converter, no. Just as long as everyone understands all all such tests will demonstrate is the audible difference between sample rate settings (and not necessarily the sample rates themselves) on that specific serial numbered box of that make and model of converter, and will contain no useful information for anybody except yourself ;).
I just need a good blindfold and someone to switch the rates for me…right? :D
You need to have an absolutely identical sound source first of all. Unless you have three models of converter into which you can run a live source through three different sample rates simultaneously, you'd need to use pre-recorded sound in some form (the wider the bandwidth, the better, of course.)

Then you need to record each of these paths, ensuring that the volume and RMS of each recording remain absolutely identical to each other.

Then, when you play back the sources for testing, not only should you not know which is which beforehand, but whoever or whatever is playing them back should not know which is which either, and also should not be directly visible by you. This is what double-blind means; neither the tester nor the subject should know the answer, because subtle visual and vocal clues from the tester can otherwise subconsciously clue in the test subject.

Ideally you'd run a computer program that would randomly play back equal parts of the various sample rate recordings at least three times each in random order, and allows you to pick relative quality of each one by mouse or keyboard. This would be a very simple program to write; if you know any even casual programmers, they should be able to hack up a program like that in just a matter of minutes.

Finally, a control data group would make it perfect, but that might be harder to pull off. In this case that would mean including the original analog performance/recording in with the digital versions. This would make the test truly scientifically robust, but I think most of us would accept the non-control group tests as being robust enough if you couldn't get the control group seamlessly into the test.

G.
 
I get into this type of argument.. er... spirited debate with my friend all the time. He's always ragging on me for not using the "highest numbers" on everything while recording and I keep trying to explain to him that it's not going to make the slightest perceivable difference; at least--as Glen says--not strictly due to the fact that the numbers are higher. But this particular friend is the type that buys inch-thick $15 audiophile magazines and just gobbles up all of that marketing bullshit typically spewed in such publications (it's all in the specs, right?!). Sorry, but I question the gullibility of somebody who can't see that spending $2500 on a 3 foot HDMI cable is ludicrous from a cost/benefit perspective.

I've digressed a bit here, but I definitely see how the "more is always better" philosophy pervades all manner of digital media, without the hard evidence to back it up (i.e., detectable improvements in product quality).

I'm fairly new to recording but I worked as an NVH engineer for several years, which dealt with a lot of digital signal processing, and that knowledge has come in handy in understanding recording in the digital realm. I fully agree with Glen on the Nyquist stuff, in that using a sample rate that is double the max frequency of a limited bandwidth source guarantees that you will capture the max frequency, although the accuracy of the amplitude representation might suffer a bit if you don't have a bit of "headroom" in the sampling rate. But if you look at the sensitivity of the typical human ear in the frequency domain, by the time you're getting into the 20k range it all goes to hell anyway so it's of little consequence much beyond that.

Bottom line, the math is one thing, but whatever sounds better to your ears should be the guide.
 
I've digressed a bit here, but I definitely see how the "more is always better" philosophy pervades all manner of digital media, without the hard evidence to back it up (i.e., detectable improvements in product quality).
Not just digital media. How else can you explain triple-cheesburgers? 1000+ horsepower cars designed for street (ooo, THAT would be fun though :D )? etc.
 
I've digressed a bit here, but I definitely see how the "more is always better" philosophy pervades all manner of digital media, without the hard evidence to back it up (i.e., detectable improvements in product quality).
Damn, Noisewreck already beat me to the point that it's not just digital media that suffers from this marketing ploy. :eek::p;)

But - as much as I would LOVE to - I can't place all the blame on the marketing slugs. Just as much blame has to land on the consumer for falling for this crap. I can understand your average Joe Home Recorder not getting Nyquist, because that's tough stuff. But there's very little excuse, just for example, for their not being able to enough basic study and thinking from first principles to figure just how bullsheite the whole Loudness thing is. Information theory is rocket surgery, dynamics ain't.
...although the accuracy of the amplitude representation might suffer a bit if you don't have a bit of "headroom" in the sampling rate.
True, which is a good part of the reason why we sample at 44.1 and not at 40. But again, that headroom is due to physical realities in implementing the science (there's no such thing in today's technology as a brick wall band pass filter, for example), not because of a fault in the science itself.
Bottom line, the math is one thing, but whatever sounds better to your ears should be the guide.
Absolutely, always true. :).

G.
 
Bah, triple cheeseburgers are for girlie-men these days ;). Check out the Quadruple Bypass Burger at the Heart Attack Grill. No joke! It's been all over the news shows. This place is real, right down to the Lucky Strikes and the nurses who wheel you in and out of the place in wheelchairs.

G.


Ok, I just threw up in my mouth a little bit. Note to self: don't look at pictures of foot-high hamburgers on a full stomach. :p
 
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