I think Goddard's point is that it is impossible for higher sampling rates to have a negative impact on the audible range.
 
In post #253 I show 2 examples of how higher sampling rates (and hence the presence of more ultrasonic energy) can impact the audible range. Since he knows everything, I'd like him to tell me how those two examples are flawed. His main point and those two points can't be correct at the same time.

John T
 
Let's suppose there is some interaction in sound waves, as described. So we play a note on a trumpet, and there's some stuff going on at, say, 40khz that's completely inaudible, but interacts with the other movements in the air and has some influence on what's going on at, say 5khz, which is audible.
 
As an aside, I can't imagine that this isn't the case. You're pushing air around, and it's a complex system.
 
However, if we're capturing all the audible range, then we are by definition capturing every effect that occurs in the audible range, even if we aren't capturing the cause.

John, what you are referring to is intermodulation distortion. However for intermodulation distortion to occur, you need a non-linear transfer medium, whether air, a microphone, a signal path, a plugin or whatever. At the air pressure levels that audio waveforms occur, it is essentially linear, so there will be virtually no intermodulation distortion.
 
The problems start when you record the ultrasonic frequencies and then let them loose in your DAW, where we love to add non-linear transfer functions to the signal. I posted two such examples: a compressor and a tube emulator. These perform their "magic" by behaving differently depending on the level of the signal. If you have ultrasonic energy present, these signal paths will respond to it, which will then also alter the audible frequencies (unless they themselves first filter out the HF in their algorithms).
 
So unless anyone can come up with actual proof for why having ultrasonic energy present is beneficial, it is best to avoid it altogether.

There can be situations where there may be energy present that does not seem to be very obvious hearing it but if you filter or don't capture that energy there can be audible differences. I feel this is a very audible situation. I am from that school too. You can have a track that sounds Ok but insert a LPF filter on it and roll off everything (slowly) above say 15KHz and that track suddenly will sound better and clearer. It does to me and this happens more often than not. Same applies to the low end and interesting we are not talking about low end much here. An instrument may never go below (with fundamental that is) 80 Hz. Yet put a HPF filter on that and set it so nothing at 80Hz is touched but below that the signal is removed and that too can sound better. That is really audible. Maybe these concepts support the theory that we simply don't need to capture stuff we cannot hear so much at all.
 
Audible situations are much more interesting to talk about. Things that are very minimal are a waste of time to a certain extent. The reality is if you listen to a 24 Bit 382Khz mix and a 16 Bit 44.1 K mix, if the music is great, the engineering great then the mix will survive and still smell as sweet as a rose right at the end. Emotional impact equal for both mediums. That is what is important. My point about very high quality sources being bottle necked down to 16 bit 44.1K and no one hearing the difference really highlights the uselessness of this stuff. All fun but not that important at the end of the day.
 
Learning how to EQ, compress and limit a mix perhaps is way more important and in that situation a mix can be destroyed. Things that do effect 100% of the sound to me are much more interesting than stuff that might only have an effect of 0.1% or less.
 
I was thinking about analog as well. What is different about analog is the way the frequency response dies off at the high end. It is slow and gradual up there. My Revox B77 reel to reel machine specs at 25KHz or so as the top limit of its freq response (-3dB) but it still produces output at 35KHz! The graph is slow and easy from 25KHz up to 35KHz. It is a beautiful thing. The old Ortofon moving coil pickup cartridges are flat to 50KHz from memory. Now that is top end!
 
Final mixes I think need all the frequency range they can get. Individual tracks not so much so perhaps. I believe there is difference between tracks, buses and final mixes frequency response demand wise that is. It is the complex interaction of tracks that may only be limited out to 15 KHz or far less that might result too in final harmonics that extend up higher. High frequencies interact. If we want that nice analog top end than all we need to do is to go to 24 Bit 96KHz and that is easy to do now. We don't really need anything above that. Some experts say we only need to go to 60KHz digitally obviously to get the response out to 30KHz perhaps. Our nearest rate to that is 88.2Khz which does at least give us a nice 40Khz response.
 
I have just mastered a very high quality Jazz CD. The mixes came to me at 24 Bit 88.2 Khz and yes they sound great. But even after mastering and dithering down to 16 Bit 44.1Khz the difference between the two sounding formats is so miniscule. It is really inaudible in fact. I did a little test and just changed down the original mixes from their original format to 16 Bit 44.1KHz and did no mastering, just compared the two and believe me I would challenge anyone to hear anything between them! The issue is the music is just so good and the moment I hear either of those formats I am off listening to the playing so once that happens I think there is no hope of hearing anything much else!

John T
Everyone's entirely aware that this stuff has been being discussed for years.

 
And yet rather than reading up on past discussions (some of which I even provided links to) and getting up to speed on their own, people prefer to remain oblivious to all that's been raised and debated at length already elsewhere, and then, having too little knowledge of the topic to understand what I've posted or its relevance, feel entitled and justified in complaining about me being obscure or circular or elliptical or not rigorous enough and accuse me of posting misinformation or of being flat out wrong or of being a douche or of being confused, when they still can't even understand how 32-bit floating-point data represents 24-bit integer PCM data after I've not only explained it but also pointed them to a color-coded diagram showing exactly which bits do what, and then still had to provide further even more basic explanation, and even another poster had to post a color-coded diagram?
 
I'm sorry, but if you can't even make an effort to educate yourself to get up to speed, don't act like babies expecting to be spoon-fed info and complain that it's obscure or irrelevant or whatever if it goes over your heads. Do some homework on your own, consult more reliable and knowledgeable sources of info than forum posts or blogs of dubious expertise or validity and maybe you might just acquire some quality knowledge on your own which you can share, which you may then find a far greater reward than any "platinum" post count and which can serve you and your production prowess well into the future.
 
Here's a simple way anyone might check their level of knowledge, a 1998 article by Martin Walker of SOS on the options available for "Optimising PC Digital Audio Quality In Software". If you don't already know and understand the stuff Martin talks about in that article, which remains in large part just as valid and vital today, then you definitely need to do your homework, because this thread has been cluttered with far too many very basic questions about stuff which anyone who's been using DAWs and on this forum for years really should already know, especially if they are working professionally in this field.
 
I can see now that it was a mistake to even join this thread, and that I may have erred by overestimating the general comprehension level of many of the participants, but I was growing concerned due to that Noel's original post referred to that blog as a "great article" and as "well worth the read" and a number of people including Garrigus had then concurred that it was a good article/info, when it was quite obvious to me that it was written by a less-than-knowledgeable poser and riddled with misinfo and scaremongering and that there was a serious risk of folks less knowledgeable being taken in by it.
 
But now, having seen once again the snideness and drive-by insults and bandwagon jumping and outright schoolyard bullying attempts that have become far too common on the CW forums, let the gullible and uninformed find their own way along the path to knowledge without me from here on. I've already given enough precious time to this thread, and although I do appreciate that some folks have expressed their thanks along the way, I only participate in the forum nowadays as my way to give back for what others were kind enough to share with me in the past, and I sense that my karma cup has been filled as concerns this thread.
 

Who farted?

@Goddard,
 
So - you are chastising folks for cluttering the forum by asking 'basic questions'?
 
Really?
 
There are lots of levels of knowledge in this forum, and while I am a decent keyboard player, and very good with computers,  I certainly am no expert in the details of audio fidelity, nor for that matter am I an expert in several areas of using Sonar.  
 
I would like to think I could ask a question without being judged for doing so.  This forum exists for folks to ask questions and have discussions.
 
All I can say is 'wow'.
 
Bob Bone
 
 
 
 
 

Noel Borthwick [Cakewalk]
Also a high quality converter at 44.1 will sound way better than a realtek 96k convertor :)

Yeah, I raised an eyebrow at the notion that you could effectively test this stuff by tweaking the settings on a ropey motherboard converter. But by that point, my eyebrows were in danger of popping off the top of my head, so I thought I'd be best off not even getting into that one.

OK here is a simple test...
 
Create a project with the highest possible sample rate your setup supports. Then have some tracks that contain audio feed into different busses that each have various processing including compression, tube emulators, guitar effects like fuzz, distortion etc.
 
Now create a track with an ultrasonic signal in some way (I'm sure some plugins that generate test signals would be able to create ultrasonic signals). Now feed this HF signal into each of the busses, one at a time. Dollars to doughnuts your audible signal is noticeably altered in many cases.

Very few people would own speakers that can recreate subsonic frequencies to the extent that they can be "felt", so I wonder if it is even something worth worrying about at all?

Noel Borthwick [Cakewalk]
We had a very thought provoking discussion at NAMM with Henry Juszkiewicz, Gibson CEO about how much more important the lower frequencies were rather than the HF that people obsess about. He's really clued in to this stuff and approaches it from a perspective that factors in the physiology of the human ear.

I had to laugh since I threw this out way back in this thread someplace and it got trundled. Not only do the Fletcher–Munson curves show you need a pretty wild gain to perceive the high frequencies in the first place, but as people age they started getting "deafer than a fencepost" above 16kHz anyway (not to mention the added element of what something is played on).
 
On the flip side, I love this FabFilter Pro-Q video just because the narrator uses the term "low frequency rubbish" several times (the stuff people can and will hear readily, regardless of age).
 
It is amazing how people can argue the crap out of a part of a system, and totally ignore the part of the system that makes that debate carry weight (or not). Machines and science do not hear... the human ear hears!

mettelus
I had to laugh since I threw this out way back in this thread someplace and it got trundled. Not only do the Fletcher–Munson curves show you need a pretty wild gain to perceive the high frequencies in the first place, but as people age they started getting "deafer than a fencepost" above 16kHz anyway (not to mention the added element of what something is played on).

I do a test on this about once a year, and at the age of 41, I'm currently doing ok up to about 17.5k, which I gather makes me genuinely lucky and privileged.
 
Warming to the theme of practical mixing, really most of what's going on above 15k is pretty nasty. The old RIAA vinyl mastering standard was simply to cut above 15k, with a brutally steep filter. OH NOES, THAT MUST MEAN VINYL RECORDS SOUNDED DULL. Or maybe it tells us something else.

I'm going to ramble a bit more on that high end point, feel free to ignore me.
 
One of the big A-HA! moments for novice mixers, I think, is when they realise that just brutally high-passing all of the mud out of most of your tracks makes your mixes instantly better. Certainly, this is one of the most commonly discussed Gearslutz Pro Tipz.
 
Then, they progress a bit, and get a bit more nuanced about that, of course. But still, it's a valid insight. Get rid of stuff that's not helping.
 
What's surprising to me is how rarely people seem to talk about low passing, or cutting the highs on given sounds. If everything is bright, then you're in for an ear-piercing 20-car-pileup horrorshow in the high end. Lots and lots of high end energy is aesthetically displeasing to almost everyone, and doesn't aid clarity; quite the opposite.
 
 
 
 

If your camera doesn't capture gamma rays, your photographs are a joke.

An instructive exercise is to load up a final commercially released track, that you think sounds great, and stick a low  pass filter on it. Set it playing, and roll the filter up and down the frequency range. You'll hear a difference immediately, as soon as you're cutting anything, but stick with it and take your time. Roll it down to say 18k, and have a full listen through. Then roll it down to maybe 15k and have a full listen through. Give yourself time to really soak up what you're hearing.
 
I'm not suggesting that thoughtlessly cutting all the high stuff is a good methodology; it's not. But training your ears in how much of significance is up there is a worthwhile exercise.
 
It's also counter-intuitive. I often find myself cutting a lot of high end from the drum overheads (which you'd assume to keep the high end of), and letting the guitars (which you'd assume to be more mid range) run rampant, for example. We're not in the business of making things sound perfect in solo, we're in the business of creating a listenable and engaging mix.