24 bit/96khz vs 24 bit/192khz
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#1
16th August 2013
Old 16th August 2013
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24 bit/96khz vs 24 bit/192khz

Please what's the difference between the 24 bit/96khz and the 24 bit/192khz sound resolutions. And how does it affect my recording and production. Thanks for your replies.
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16th August 2013
Old 16th August 2013
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To 99.999% of people, the audible difference is 100% undetectable. For the tiny number of people who can actually hear a difference, the jury's still out on which sounds better, and even which is technically better. There are LOTS of threads you can read about why 96 or 88.2 are technically preferable, and people far more knowledgable than I argue both sides at length. For your purposes, and the purposes of everyone else for that matter, the only difference is the system resources used to record, edit, and mix at 192... there, the difference is very much real, and very much what you'd expect: it takes twice the system power to process the same number of tracks.

Bottom line: unless you have an incredibly powerful, stable system, or you record only a few tracks at a time and mix without too many plugins, or you can't sleep at night unless you've recorded everything to DSD but that's not an option here, or you're a glutton for punishment, or you think the most important advertising you can do is to say "recorded at 192kHz!", or all of the above, record at 88.2 for music, or 96 for film and TV.
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#3
16th August 2013
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I can hear a difference with my decent but modest setup, but I always prefer the 96k when capturing a mix. I think it has to do with modern converters behaving better when they are at or near their optimal clock speed. So it really depends on the converter.

However, capturing a mix to DSD (Tascam DV-RA1000HD) and converting to 24/176.4 (Saracon, no dither) for pre-mastering, then to 16/44.1 (Saracon, TPDF dither) for Redbook sounds significantly better than an 16/44.1 ITB bounce-to-disk.

-d
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16th August 2013
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There are simply twice as many samples per second in each channel it may or may not sound better. But if you ar doing it ITB you will need more than double the proccessing power. It could really effect your workflow recreation time and bank account.
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16th August 2013
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Do you have a bazillion dollar studio?

No? Didn't think so.

Then just lock everything down on 24bit/44.1k and forgetaboutit....
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#6
16th August 2013
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192K is junk for music production...archival...classical...mastering...fine....

My Music production...NO!!!!!

....96K? I replaced with a good converter at 44.1khz..Lost interest in recording MORE high frequencies.

High sample rates are not for me. To my ear, Music production at 44.1khz is preferable for several reasons both in sound and in functionality.
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#7
16th August 2013
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Totally agree!
It is really a question of economics than audibility.
192 is heavy on resources; that is the trade off if you can sense a quality difference.
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16th August 2013
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Thanks Everyone. You really helped a bro in need. Very grateful. Guess I'll just stick to 96khz and less then.
#9
16th August 2013
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The biggest argument for higher sampling rates for me is the increased transparency of time stretching

44.1 is VERY unforgiving when it comes to time stretching. 48 is a huge improvement and is a good compromise of using CPU. 96k is great, but too taxing on my system to have a usable workflow.
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17th August 2013
Old 17th August 2013
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Quote:
Originally Posted by Mixtree Audio View Post
I can hear a difference with my decent but modest setup
we have supernaturals in the house ...
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17th August 2013
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Simple rule: 88.2 if the system can. 48 if it can't.

You can trust my ears. Dan Larvy's EE converter design experience. 44.1 has NEVER been an acceptable recording resolution....until the very recent ITB digital trend where people believe it's better to take the sonic hit up front. I will never track at 44 again.
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#12
17th August 2013
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For alongtime now, I still used the 24/48khz even though my converters is capable of doing at 192khz. I focus on good songs rather than kilohertz...
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#13
17th August 2013
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Its in the stretchablility. If you have to stretch a track and record it at 192 k then you can time stretch this much more than 44.1 without getting heavy artifacting sooner.

Its the same iwth with recording with a mic that goes up about 20k. When you pitch shift down your program material will still have top end.
#14
17th August 2013
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As we have the ball rolling on sample rate here, converting from a lower sample rate, to a higher one. Upsampling. Would that give one better pitching and stretching of audio? Is it even a thing?

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17th August 2013
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Quote:
Originally Posted by kmoneybts View Post
The biggest argument for higher sampling rates for me is the increased transparency of time stretching.
Particularly noticeable with ProTools.

And with most DAWs, you get better latency times with 96.

Some older converters (e.g. 888) were not very well clocked and caused some HF distortion which made them sound 'harsh' and higher sampling rates pushed that distortion out of the human hearing range.

But assuming you have good modern converters and you do not have to worry about latency times and are not going to stretch bits to a silly level, 44.1 or 48kHz will do just fine.
#16
17th August 2013
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Its not about super high frequencies...it's all about shoving the aliasing up an octave or two when processing using plugins.

And no, oversampling is not the solution really as we all know that real time sample rate conversion if done properly would bring a computer to it's knees in a daw environment.

So why not work at 88khz, get rid of a lot of the aliasing in plugins. EQ's, Compressers and Soft Synths sound a lot better.

Can I hear the difference between a normal unprocessed recording at 44khz and 88khz....nope probably not at 38khz....but I can hear a huge difference in the plugin processing.

We need to get over this putting 88khz = Bat hearing and understand why those that like it more like it for the better quality in processing.
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#17
17th August 2013
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Originally Posted by Captain Proton View Post
Can I hear the difference between a normal unprocessed recording at 44khz and 88khz....nope probably not at 38khz....but I can hear a huge difference in the plugin processing.

We need to get over this putting 88khz = Bat hearing and understand why those that like it more like it for the better quality in processing.
Bingo!

Compare audio to video and you begin to understand what is happening. Why is hi-def and 2K material filmed in 4K? Simple, because every processing step detracts from quality/resolution.
#18
18th August 2013
Old 18th August 2013
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Quote:
Originally Posted by Captain Proton View Post
Its not about super high frequencies...it's all about shoving the aliasing up an octave or two when processing using plugins.
If a plugin oversamples or upsamples before it processes, does that negate the need for higher sampling rates for the material being processed?
#19
25th August 2013
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I'm not an expert on a/d conversion or anything, but I've recently been experimenting with 192 recording. I'm using an apogee ensemble, and i can hear a difference between 96k and 192k. Its definitely subtle, but to me 192 sounds more real.

Here's a thought: The Nyquist theorem says that you need a sample rate of at least double the highest frequency you want to reproduce right? And from what I understand the reason we use 44.1khz instead of just 40khz is because we need some room for the slope of a good cutoff filter to prevent aliasing. So that being said, in theory, what if we had an a/d converter that ran at 40khz sampling rate, and we wanted to record a sine wave at 20khz. The Nyquist theorem says that we should be able to do this accurately right? Well what if the sine wave and the a/d converter were running 90 degrees out of phase from each other? So the samples were being taken at zero crossings instead of at peaks and dips.

If that is possible, then I have to believe that 44.1khz isn't as perfect as everyone seems to think, and recording at 96k or 192k isn't just about recording higher frequencies, but also about capturing lower frequencies more accurately. In a perfect world, everything would be in phase with your a/d converter and transients would happen at the exact right point in time that your a/d converter is taking a sample. But in the real world, 44.1k gets pretty darn close, but doesn't always capture everything as perfectly as we think it does.

Like I said, I'm no expert. This is just a thought I've had and was wondering if its a valid argument for 192k or not. Can anyone else shed some light on this?
#20
25th August 2013
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If your doing instruments ITB, there is a great noticeable difference when changing sample rate, add distortion on guitars to your 44.1 sample rate and you'll see a mix that will be 1 in 100 that are actually good. Sample rate does matter but on certain things more than others. Recording a track, not as much, ITB, hell of a time (in a bad way).
#21
25th August 2013
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Quote:
Originally Posted by Captain Proton View Post
Its not about super high frequencies...it's all about shoving the aliasing up an octave or two when processing using plugins.

And no, oversampling is not the solution really as we all know that real time sample rate conversion if done properly would bring a computer to it's knees in a daw environment.

So why not work at 88khz, get rid of a lot of the aliasing in plugins. EQ's, Compressers and Soft Synths sound a lot better.

Can I hear the difference between a normal unprocessed recording at 44khz and 88khz....nope probably not at 38khz....but I can hear a huge difference in the plugin processing.

We need to get over this putting 88khz = Bat hearing and understand why those that like it more like it for the better quality in processing.
Finally someone addresses the real story....when I've heard folks say they can "hear" the difference between 96 & whatever, my guess is in fact what they are actually saying is they hear the plugs working better or the plugs are stuffing it all up . 44 is fine as is 48 , 88 allows daw and associates to play nice and share their toys. 96 on some vocals but never a whole project, what's the point? I also agree in taking the hit up front instead of on the arse end, it all cracks back down to 44/16 for replication any hoo...

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25th August 2013
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Quote:
Originally Posted by Aaron Giese View Post
I'm not an expert on a/d conversion or anything, but I've recently been experimenting with 192 recording. I'm using an apogee ensemble, and i can hear a difference between 96k and 192k. Its definitely subtle, but to me 192 sounds more real.

Here's a thought: The Nyquist theorem says that you need a sample rate of at least double the highest frequency you want to reproduce right? And from what I understand the reason we use 44.1khz instead of just 40khz is because we need some room for the slope of a good cutoff filter to prevent aliasing. So that being said, in theory, what if we had an a/d converter that ran at 40khz sampling rate, and we wanted to record a sine wave at 20khz. The Nyquist theorem says that we should be able to do this accurately right? Well what if the sine wave and the a/d converter were running 90 degrees out of phase from each other? So the samples were being taken at zero crossings instead of at peaks and dips.

If that is possible, then I have to believe that 44.1khz isn't as perfect as everyone seems to think, and recording at 96k or 192k isn't just about recording higher frequencies, but also about capturing lower frequencies more accurately. In a perfect world, everything would be in phase with your a/d converter and transients would happen at the exact right point in time that your a/d converter is taking a sample. But in the real world, 44.1k gets pretty darn close, but doesn't always capture everything as perfectly as we think it does.

Like I said, I'm no expert. This is just a thought I've had and was wondering if its a valid argument for 192k or not. Can anyone else shed some light on this?
I'm with you! I hear a difference too but I rarely bring it up because doing so is the quickest way to start an internet argument. Is the difference subtle? Yes, and subtly is what I've been trained to listen for.
#23
27th August 2013
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I've been recently using 88.2 and love it... Think it sounds great... My computer handles it well... I think anything more then that is overkill...
Im sure you could hear a difference with other sample rates but different doesn't always mean better!
#24
29th August 2013
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Quote:
Originally Posted by Ric4001 View Post
If a plugin oversamples or upsamples before it processes, does that negate the need for higher sampling rates for the material being processed?
As I said in my previous post.... real time ample rate conversion is not....NOT... going to be of the quality needed really. A good converter like R8Brain uses up some serious cpu just to do a single stereo wav file. A real time up sampler then down sampler is not going to be of a decent quality it just can't be....just run the machine at 88khz and bang no need for band limited oscillators or real time up sampling.
#25
6th January 2014
Old 6th January 2014
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Quote:
Originally Posted by Aaron Giese View Post
I'm not an expert on a/d conversion or anything, but I've recently been experimenting with 192 recording. I'm using an apogee ensemble, and i can hear a difference between 96k and 192k. Its definitely subtle, but to me 192 sounds more real.

Here's a thought: The Nyquist theorem says that you need a sample rate of at least double the highest frequency you want to reproduce right? And from what I understand the reason we use 44.1khz instead of just 40khz is because we need some room for the slope of a good cutoff filter to prevent aliasing. So that being said, in theory, what if we had an a/d converter that ran at 40khz sampling rate, and we wanted to record a sine wave at 20khz. The Nyquist theorem says that we should be able to do this accurately right? Well what if the sine wave and the a/d converter were running 90 degrees out of phase from each other? So the samples were being taken at zero crossings instead of at peaks and dips.

If that is possible, then I have to believe that 44.1khz isn't as perfect as everyone seems to think, and recording at 96k or 192k isn't just about recording higher frequencies, but also about capturing lower frequencies more accurately. In a perfect world, everything would be in phase with your a/d converter and transients would happen at the exact right point in time that your a/d converter is taking a sample. But in the real world, 44.1k gets pretty darn close, but doesn't always capture everything as perfectly as we think it does.

Like I said, I'm no expert. This is just a thought I've had and was wondering if its a valid argument for 192k or not. Can anyone else shed some light on this?
You are right on the money. Imo(educated), if you can't hear it, then you don't know what your missing. Therefore 192k isn't useful.
#26
6th January 2014
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Quote:
Originally Posted by Aaron Giese View Post
I'm not an expert on a/d conversion or anything, but I've recently been experimenting with 192 recording. I'm using an apogee ensemble, and i can hear a difference between 96k and 192k. Its definitely subtle, but to me 192 sounds more real.

Here's a thought: The Nyquist theorem says that you need a sample rate of at least double the highest frequency you want to reproduce right? And from what I understand the reason we use 44.1khz instead of just 40khz is because we need some room for the slope of a good cutoff filter to prevent aliasing. So that being said, in theory, what if we had an a/d converter that ran at 40khz sampling rate, and we wanted to record a sine wave at 20khz. The Nyquist theorem says that we should be able to do this accurately right? Well what if the sine wave and the a/d converter were running 90 degrees out of phase from each other? So the samples were being taken at zero crossings instead of at peaks and dips.

If that is possible, then I have to believe that 44.1khz isn't as perfect as everyone seems to think, and recording at 96k or 192k isn't just about recording higher frequencies, but also about capturing lower frequencies more accurately. In a perfect world, everything would be in phase with your a/d converter and transients would happen at the exact right point in time that your a/d converter is taking a sample. But in the real world, 44.1k gets pretty darn close, but doesn't always capture everything as perfectly as we think it does.

Like I said, I'm no expert. This is just a thought I've had and was wondering if its a valid argument for 192k or not. Can anyone else shed some light on this?


The theorem is correct. 44.1khz sample rates will capture and reproduce the human range of hearing perfectly. Saying "192khz sounds better 44.1khz" is a little shaky as a blanket statement, I think.

The flaw comes in the implementation. The sinc filter (the low pass filter applied upon converting the digital signal to analogue again) is idealistic in theory but in application it is more pragmatic. That is to say, the better the design of the converters and the sinc filter, the more negligible the difference between 44.1khz and higher sample rates becomes.


Quote:
Originally Posted by jreatonit View Post
You are right on the money. Imo(educated), if you can't hear it, then you don't know what your missing. Therefore 192k isn't useful.

IMO, if you can hear it... you don't know what you are missing...
...that would be converters with closer to idealistic implementation.
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