You're probably the only one guy on Earth who works hard.
If your life is fantastic,cool for you.But everyone hasn't the same experience of life."Guts and balls" don't do everything,as you think.

 

I came from reel-to-reel recording to DAW (and sister site's treasure trove) about two years ago and really up until that time had no higher use for the PC than to run Flightsimulator at a decent frame rate. Since I got Pro Tools under the hood my approach of it remains pretty much as a "Studio Simulator". With real (hardware) tube pre-amps, EQ's and compressors on inserts I can make just about everything sound "right" for my kind of music. The thing I find really hard to swallow about DAWs is latency. I find that in a lot of cases I can all but eliminate that however by simply ratcheting all down to 44.1 KHz. In my view the downside of very high resolutions like 96 and 192 is (just my opinion!) that the above mentioned clarity and detail can make for tiresome listening. The often praised "smoothness" in (real) tube processing lies, in my view, exactly in the absence of some detail. Again, I see this all from the viewpoint that the DAW is a studio simulator and from there I am very satisfied with the vintage-y results crude processing has as a consequence.

 

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Converters have gotten better and better. It's not a must to use extremely high sample rates and large file sizes to impart a negligible difference at the end of the day. If you want to go that route.. don't use obscure division. Follow the principles:

Film: 96 to 48 conversion
Album: 88.2 to 44.1 conversion

 

Our human ear hates perfection. In nature, there are only sounds distorted by air, wind, cold, heat. When you speak loudly, your voice is saturated.
Our ear likes soft or powerful saturation because it creates contrasts. We like echoes, reverbs because it is part of our daily life.
The hardest part of a good mix is to try to understand the artist and his relationship to the sound, to the life of the sounds he hears.

 

No, sorry, this is the whole number rate conversion myth. When audio signals are converted between rates the converter uses an internal DAC conversion to render the signal in 'analog' form and then resamples the analog version of that signal at the new rate using ADC. Any relationship between the rates is irrelevant.

 

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interesting point about latency Aileron..i had a thought, what about a process whereby you can track at 44.1 and a copy is somehow made at the same time at higher res, so when you go back to mix offline, you have the higher res files to listen to (kind of seamlessly so no having to load them in?..
are you' saying re: fatigue....i cuuold be wrong, but i get the impression that high resolution is by nature like needles..very sharp sounding..almost sounds like hi-res digital is too good..like pointless and counterproductive in a nature sense...whereas the sag and "buffering" of candle gear imparts a kind of elasticity and taut bounciness to the sound. what does a candle do (if possible) to a digital stream of zeroes and ones? ..does it do it wrongly? could one make a thermionic valve with two sets of cathode/anode/grid sets so they are fired positively for 1's and 0's? a valve designed specifically for digital?

 

i don't even want to think about that....it's disgusting

 

Wow, @Pronto , you sure come up with a noteworthy suggestion, re copy at higher res while tracking, real time. Yes, that could be done to a second system from an Adat out on my mixer.
Interesting.

 

I agree completely. And with the misleading internet trend about the use of hp filters, and never a mention of lp filters.. it's the recipe for a lot of the painful music we can stumble upon.

What I've found is using a steep lp at 16kHz on the 2 bus when mixing is a golden reference. At that point you can hear exactly which of your tracks need hp filters, and easily use the hp filters to nail the low frequency sweet spot for every sound in the song. Then afterwards, disengage or tailor the lp frequency on the 2 bus to set the global level of brightness.

When all that collective shrilling shit above 16kHz is still there when mixing, it makes it hard to know where the hp filters should truly be. And what people wind up doing is cutting too much ass off everything until they're left with a number of tracks that don't have the right low-freq foundation, so they serve as a high-freq ice pick on the ears.

 

Best first post ever, except for point one, it's useless, but the marketing guys will explode the weaknesses in consumers

 

To be frank, the high sample rate conversion (higher sounds better) is the subjective "myth". But the raw principles of math and division are not a myth. Why do you think the higher rates are always introduced in precise multiples of 2, and for both audio/video solutions? Research it. In short, even if technology offered a sample rate as high as 2304kHz, it's why 2116.8kHz would surely be included with it.

 

I refer you to my earlier answer. Raw conversion at higher rates offer little benefit. Audio effect processing at higher rates sounds significantly better, which is why many good plugs offer internal oversampling.

You appear not have read what I actually wrote. When converting from any rate to any other, the conversion is done by reconstructing the analog signal and resampling it. You cannot 'fill in' the gaps between samples by doubling or averaging. You have to reconstruct tnhe entire signal across the time domain, and resample it.



You cannot get from sample set 1 to sample set 2 by using simple integer interpolation, you have to recreate the analog curve, and resample it.

 

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The main 'problem' with lower sample rates are the freqs that are thrown back at the Nyquist frequency (sample rate/2).

Every kind of distortion, saturation and waveshaping produces additional harmonics. If these harmonics exceed the Nyquist freq they are thrown back. So, the higher the sampling rate the higher the Nyquist freq is and the less level these harmonics have when they reach the Nyquist freq and are thrown back. And when they reach the audible freq range they probably are not audible levelwise anymore.

But, if this is audible or not depends a lot on what you are doing. If, for instance, all your pluggies that produce harmonics work with high oversampling factor like 8, 384kHz or 44.1kHz sampling rate probably won't make any difference. If they don't use oversampling, a higher sample rate could be audible and therefore useful.

And btw, higher sample freqs reduce the latency because it's calculated in samples and the same amount of samples need less time with higher sample rates.

 

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As I said earlier, what "sounds better" is subjective.. regardless how high the sample rate is, or whatever. Choosing to have oversampling turned on/off with effects processors, x2, x4, x8, etc. yields a different weight (usually heavier) and color that's going to be desirable to the users ears in that instance, or not.

What you're missing about sample rate conversion is the conversion point itself is not even sound.. it's "math". That's why it fairs best when dealing with equivalency in the process. Audio/video technology is built to deal with it that way. The analog you hear is the "converted". Digital is totally inaudible (no such thing as digital sound).. and that's what does the actual "conversion".. the math.

The conversion is about the handling of numbers. It's the same reason a mono file doesn't tax the process as much as a stereo file in simple DA conversion= playback. Because the digital interpretation of it is not as heavy a process. The same principles pertain to complexity of division when rendering audio and video. That's why I recommended 88.2 to 44.1, and 96 to 48.. it's a more efficient process in digital terms. I said nothing whatsoever about it having to do with something sounding better.. whatever that is.

 

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No, it really isn't. Engineering is done using measurement. Audio engineering is no different.

On the basis of which argument this conversation doesn't exist, the music you record on your device doesn't exist and your DAW doesn't exist.

 

In audio engineering you measure to hear what you "like", and you assume others will "like". It's not about no damn engineer saying "Hey, switch on the oversampling on all channels and render that out for me would ya?". One has to "listen" to determine what sounds better to them. Audio engineering calls for one who "listens" more meticulously than the average music fan.

I appreciate you showing up to begin with, just looking to argue and go out of your way to say something on the contrary to someone else. I don't give a fuck about it. Have a nice day.

 

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Thomas Edison, 1889: “Fooling around with alternating current (AC) is just a waste of time. Nobody will use it, ever.”

“The phonograph has no commercial value at all.”
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"There's no chance that the iPhone is going to get any significant market share." Steve Ballmer (2007)

 

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I have to wonder how many of you can tell the difference in a real blind test

 

You can do what you'd like, but I promise you, working with 24bit, 44.1 is all you'll need in our lifetime. Don't stress yourself and your computer with extra work.

 

...but when a vendor is saying ..sounds more transparent and rich...they are saying it invoking some kind of "objective truth"...therefore, engineering or the scientific method (see Ethan Wiener's Null Tester as a reliable source to measure differences) is required...fair game at this point..it helps to combat the blind-faith buying of exorbitantly expesnvie cables etc.
I do agree that if buying something makes you feel good..thats enough and its no one elses business, but when you're being promised something where its not clear that its just someone's opinion..numbers and measurement are important.
In actual fact, i think comparison is the only way to gauge something....one of the salesmen COULD conceivably make the argument ridisculous if they said something like..ahh but nothing is objective, since your gear is set up by poeople looking for a result and its your version of reality..is your reality more correct than mine..no one knows..

 

Last edited: Feb 15, 2019