# 16 Bit 44100 to 192khz 24 Bit?



## valexnerfarious (Nov 20, 2013)

Ive been thinking about this for a while now and just wondering if it can be done or or make and difference when processing and mixing the sounds...i remember a few people talking about the better the quality the better the wav form handles the processing...so if i record in at 16 bit 44100 and i recorded it back through a preamp or interface that can create 192khz..would that make the quality better when mixing?


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## valexnerfarious (Nov 20, 2013)

kinda the same way you would reamp a guitar


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## Nick Batzdorf (Nov 20, 2013)

No. It doesn't work like that, and in addition you're using words that don't make a lot of sense.


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## KingIdiot (Nov 20, 2013)

for the most part no.

(I re read what you posted, after typing my reply and thinking you were asking something else)


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## Nick Batzdorf (Nov 20, 2013)

The thing is, you can't create missing sound by re-recording - although you might add a different color just like re-amping a guitar does.

If you open your (for instance) 16-bit recordings in a 24-bit session and do nothing, you're just adding zeros. The low-level detail you get with a 24-bit floor rather than a 16-bit one isn't in the original recording.

But if you process at 24-bit resolution, then you'll keep more of the bits you create when you run the 16-bit recording through 24-bit processors.


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## EastWest Lurker (Nov 20, 2013)

No. And 192 is marketing hype anyway IMHO. If you were to do it 88.2 or 96 would be the way to go, but you cannot add what is not in the original recording by upsampling.


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## Nick Batzdorf (Nov 20, 2013)

Oh, I just re-read. Yeah, sample rate and bit depth.

Bit depth is on the subtle side, but raising the sample rate is on the microscopic side. And the argument for high sample rates - because the brick wall filter rings above the audible spectrum - disappears in this context, since you've already recorded the sound.


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## dannthr (Nov 20, 2013)

Basically, you bought a loveseat for your 10x10 room and you're wondering if you move it to a 20x20 room if the loveseat will grow to the size of a sectional and if it will be more comfortable.

No, but if you decide to reupholster the loveseat, there will be more room for you to work in.

If you had started with the 20 x 20 room then you could've bought a sectional instead.


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## synthetic (Nov 21, 2013)

If you mix your tracks through an analog mixer and/or processing and record the mix at 192k, there is quite a bit of benefit. But unless you really know what you're doing it's probably not worth the effort.


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## Beat Kaufmann (Nov 22, 2013)

A picture with a bad resolution doesn't get a better quality with a higher-resolution-screen... 
Same situation with audio. 

But take into account that effects will generate new signals from your 44.1kHz/16-Bit Signal. Most effects with DAWs are calcultaing signals with 32 Bit. So the quality of the new mixed effect signals will be better with more Bit-depth - even if you come in with 16Bit.

Nevertheless, you will not get frequencies out of your 44.1kHz-signal above from 22 kHz with a 192kHz/24Bit"upgrade" for example.
So increasing the Sample Rate doesn't make sense.
My proposal is to increase the Bit amount in any case up to 24 or 32 Bit but keep the 44.1kHz. 

_Beat_


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## Nick Batzdorf (Nov 22, 2013)

> But unless you really know what you're doing it's probably not worth the effort



And unless you have an analog mixer that adds a nice sound! Otherwise you're just degrading the signal.

You know that, but does valexnefarious?


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## synthetic (Nov 22, 2013)

Actually, Allen Sides told me that summing through a Mackie mixer sounds better than summing in Pro Tools. But the difference is subtle enough that its probably not worth it unless you're a professional engineer who has everything else dialed. 

"I have a Canon 5DmkIII, should I upgrade to a Hasselblad H5D-200MS?"
"You? No. You should not."


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## KingIdiot (Nov 22, 2013)

mixer are a different story than re recording through a preamp though. I know you know that but it's gotta be put here.

there's things like crosstalk involved that will probably have something to do with summing as wella s pre amp architecture.

soooo... when did gearzlutz change their color scheme ... oh wait.


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## Nick Batzdorf (Nov 22, 2013)

Does Allen Sides prefer the sound of the 1202 with the rack ears attached or with one removed? Massive difference in punch and warmth; I can only describe it as PHAT.


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## synthetic (Nov 25, 2013)

He has used it for a remote recording, not for final mixdown. But he claimed it was still an improvement over the built-in summing.


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## valexnerfarious (Jun 14, 2014)

So as far as tracking and mixing goes will the higher sample rates benefit the quality of the tracks if i record them lets say at 24bit 192khz rather than 16bit 44.100khz?


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## wst3 (Jun 14, 2014)

there has been quite a bit of research in this area recently. The answers have been all over the map, but more for pragmatic reasons than anything else.

As you ascend from 44.1 kHz to 192 kHz the bandwidth required to transport and/or store the data increases.

Lets see if this works:
Fs WL	Ch	Kbit/Sec	Mbyte/min
-------- ---- ---- ---------- ------------
44100	16	1 0.67 5 
48000 16	1 0.73 5 
44100 24	1 1.01 8 
48000 24	1 1.10 8 
88200 16	1 1.35 10 
96000 16	1 1.46 11 
88200 24	1 2.02 15 
96000 24	1 2.20 16 
176400 16	1 2.69 20 
192000 16	1 2.93 22 
176400 24	1 4.04 30 
192000	24	1 4.39 33 

CD Quality requires approximately 5 MByte per minute for storage, and just under 0.7 kbit/sec for transport.

We increase this to 1.10 kbit/sec and 8 MByte/min by bumping the sample rate up to 48 kHz and the word length to 24 bits.

Which pales in comparison to the 4.39 kbit/sec for transport, and 33 Mbyte/min for storage for 24 bit at 192 kHz.

That's 66 Mbytes per stereo minute just for storage, versus 10 MBytes per stereo minute for CD quality.

For finished work that might not sound like much, but think about how many tracks you have in a typical project!!

So why work at higher sample rates or longer word lengths?

Word length is easy - there isn't a 16 bit converter chip out there that gives you an honest 16 bits, so at least with a 24 bit chip you have a shot at 20-22 honest bits, and yes, the difference in detail at low signal levels is real. And yes, you can now work at a lower "0 VU" and not give up S/N ratio, while increasing headroom and apparent dynamic range.

Sample rate - there isn't a lot to suggest that we can hear anything over 20 kHz, and for any of us that spent any time on stage, in front of drummers and stupid loud guitar amps, well that number is probably a little lower.

So 44.1 kHz might be shading it a bit close for the audible range, but 48 kHz is probably fine. All other things being equal... and that's the catch - all other things are not equal.

There is a great deal of proof to suggest that we are sensitive to the phase relationships between harmonics and fundamentals, or even harmonics and harmonics. So we can't hear the first harmonic above 16 kHz, but we probably care quite a bit about the first harmonic above 8 kHz. And the phase relationships between them.

A brick wall filter is a terribly ugly thing to hear. It makes a mess of those relationships! So there is something to be gained by doubling the sample rate, and using a much better behaved filter. We also push aliasing up out of the audible range, which is a good thing.

Doubling it again probably doesn't have an audible effect, but it will still create measurable differences.

So 24 bits at 96 kHz seems like a good standard for recording.

Oddly enough, it probably is not sufficient for modern digital signal processing. There is a lot to be gained by converting the fixed length samples to floating point for processing. And there might (MIGHT) even be a benefit to upsampling to 192 kHz for processing, the jury is still hearing testimony on that one<G>!

And then we get to deliver - but delivery where, and to whom? If you are listening on ear buds then even CD quality is wasted on you. If you are listening on a great monitoring system then you can benefit from longer word lengths and higher sample rates.

Which brings us around (finally) to the original question...

There is absolutely no benefit to "re-amping" or re-recording something sampled at CD quality to a higher bit rate or word length. As many here have pointed out, you can not increase the resolution, you can't put something into the signal that isn't there in the first place. It's a lot like using a filter to try to restore lost signal.

Upsampling - which may be at the root of the question - is a mathematical process used to preserve the original signal while doing all sorts of terrible things to it... mathematically speaking. It is not creating anything new, but rather preventing signal loss.

UNLESS, as has also been said, you are recording through a specific device for a specific effect.

I know that was kind of rambling, but hopefully some of that background info will put the answer into perspective.


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## Daryl (Jun 14, 2014)

synthetic @ Fri Nov 22 said:


> Actually, Allen Sides told me that summing through a Mackie mixer sounds better than summing in Pro Tools. But the difference is subtle enough that its probably not worth it unless you're a professional engineer who has everything else dialed.
> 
> "I have a Canon 5DmkIII, should I upgrade to a Hasselblad H5D-200MS?"
> "You? No. You should not."


There's nothing wrong with degrading the sound, if it sounds good.

D


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## Nick Batzdorf (Jun 14, 2014)

Good thing you used the word "apparent," Bill - apparent dynamic range - or I would have had to start showing off by explaining that the dynamic range is the same with a 32-bit and 2-bit system.

That was a close call. Be thankful.


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## gsilbers (Jun 14, 2014)

EastWest Lurker @ Wed Nov 20 said:


> No. And 192 is marketing hype anyway IMHO. If you were to do it 88.2 or96 would be the way to go, but you cannot add why is not in the original recording by up sampling.


+1

you will hear a small difference from 44/48 vs 88/96. but its impossible to hear 88 vs 192. 
when tracking. (not up converting)

with that said, there is like 1000+ things more important to make it sound good or things to worry about than high sample rate.


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## wst3 (Jun 14, 2014)

Ah Nick - be careful yourself!!

It is true that I can cover the same dynamic range with 2 bits that I can cover with 32 bits - but the resolution will suffer!


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## Nick Batzdorf (Jun 14, 2014)

I know you know this stuff, Bill.

What I'm really getting at is that a lot of people think more bits = more dynamic range = I don't have to worry about overloading.

No. The advantage to the extra bits is that you have more low-level detail.


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## wst3 (Jun 15, 2014)

And I know you know - but I can always try to trip you up... you'd do the same for me!


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## waveheavy (Jun 16, 2014)

From an analog audio perspective, i.e, how we hear, the more digital bits and higher the sample rate used to record will reproduce more of the natural 'analog' sound of the instrument or vocal. It's as others here have said.

Once you do it at 16bit/44.1kHz you can't get any more of the natural analog sound of what you recorded. Upsampling, re-amping, summing, preamp, etc., may affect the post audio, but it won't increase the natural analog of what you first captured when recording. The bare minimum studio digital standard for recording and mixing is 24bit/44.1kHz.

What you will get with converting 16bit/44.1kHz audio to something higher is more 'headroom' when using plugins and general mixing chores. The extra bits added during conversion will be dropped off at the dithering stage.


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## Nick Batzdorf (Jun 16, 2014)

Something like that, but you wouldn't want to use those words at a cocktail party of mastering engineers.


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## wst3 (Jun 16, 2014)

Nick Batzdorf @ Mon Jun 16 said:


> Something like that, but you wouldn't want to use those words at a cocktail party of mastering engineers.



possibly your best post ever Nick!


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## JohnG (Jun 16, 2014)

I've read many opinions about all this. I think I'm seeing most who can afford it tracking at 96k.

Dan Lavry wrote a white paper in which he makes a case that anything above 96k is not only a waste of effort, but actually worse, if I understand it at all. 

Here's a sample (ha!) of his writing:

"Some ear people may have been hopeful that given that 48KHz is better then 44.1KHz, and 96Khz may be better then 48KHz, then lets just keep going faster. Of course thinking that would be naive, because at some point one needs to slow down, because other factors will come in... In other words, one needs to find the optimum rate (the best rate) and it is not too slow, nor is it too fast...

...There is some "left over" vocabulary such as "filter steepness" or "gentler filter" that you mentioned here. I have not seen anyone quantify what is too steep, or what is gentle enough. Those words are just being "thrown around" to just "say something. At 96KHz, a filter is gentle enough! In fact, it is a digital up sampling or down sampling filter only that you care about, because the analog filtering is done "way up there" because ALL of todayâ€™s AD and DA gear is up sampling and over sampling, and almost all of it at 64-1024 the sample rate. 

I can design a digital filter that will be transparent at 96KHz. There is no single golden ear that would be able to tell the difference between my 96KHz digital filter and no filter at all. I could even do it in analog, and tests like that were done, with real good golden ears, and with well designed filters down to 24KHz, not to mention 96KHz. 

So yes, how gentle does count when you are near 20KHz, but not near 60 or 88 or 96KHz. Yes, one could hear an FIR filter "preshoot" when the filter is near 20KHz, but not at 96KHz...

So the use of "gentle" sounds better should be restricted to when the filter is entering the range where people can hear. Not when we are working at sample frequencies higher then twice what anyone can hear...

Regards
Dan Lavry"


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## JohnG (Jun 16, 2014)

Here is the location of Dan's White Papers:

http://www.lavryengineering.com/lavry-white-papers

Paper on the Optimal Sample Rate for Quality Audio:

http://www.lavryengineering.com/pdfs/lavry-white-paper-the_optimal_sample_rate_for_quality_audio.pdf (http://www.lavryengineering.com/pdfs/la ... _audio.pdf)

He writes near the conclusion:

"Proper use of technology begins with understanding the specific goals. Technology makes it possible for factories to manufacture pills with 10 times the dosage required by humans. Needless to say it is a bad idea, until the day that humans are 10 times larger. Similarly, technology makes it possible to convert audio at 10 times the speed. That too is a bad idea."


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