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44,1 or 48 khz?
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Alexi
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#1
11th May 2006
Old 11th May 2006
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44,1 or 48 khz?

hi guys,

short question over here.i usually record in "96khz mode" but have to use a laptop for some recent recordings which won't record 8 tracks at the same time in 96khz.....
48 work fine...........

Is it better to start the whole project in 48khz or should i start 44,1 right away.......



cheers
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11th May 2006
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I personally do everything on 44.1kHz, works fine for me and I have no need to convert afterwards down to 44.1 for burning CDs...

Greetz,

Mike
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11th May 2006
Old 11th May 2006
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Yes, 24bit/44,1...
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11th May 2006
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well I have a different opinion, you can here the difference between 44 and 48 , but only under critical conditions.

I would record in 48, since you normally do 96 I assume you have no problem with converting smaple rates for cd use.

although it really wouldn't make much difference wither way this would be my slight preference

narco
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11th May 2006
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In my expirience, if project is targeted for CD release, resampling from 48kHz to 44.1 does more harm than (small) benefit of recording at 48kHz over 44.1kHz sample rate.
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11th May 2006
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Quote:
Originally Posted by weedmaker
In my expirience, if project is targeted for CD release, resampling from 48kHz to 44.1 does more harm than (small) benefit of recording at 48kHz over 44.1kHz sample rate.
not when mixing or mastering via analog, and arguably not when using a high end conversion.

narco
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11th May 2006
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or could you just get a faster drive for your laptop? (asumming it is the drive that is stopping you recording your 8 tracks)

narco
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Another vote for 24/44.1
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11th May 2006
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My opinion would be to choose the sample rate based on the delivery format. 44.1 if the tracks are intended for CD, and 48k if they are video post production. If youa re mixing analog then you could choose 48k as well.
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11th May 2006
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I would not see any benefit of 48 over 44 for CD use.
I certainly could see the benefit of 88 for CD use, although for ease of use, I keep going back to 44 because I don't hear a significant difference on my Apogee converters and Event ASP8 monitors.
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11th May 2006
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I think it's a big mistake to make assumptions about sample rate frequencies. Some converters sound worse at 44.1 and some of the very best don't. The same is true of dsp software.

This is yet another area where you need to judge with your ears and not your eyes to determine how to get the best results from the particular gear you're working with.
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12th May 2006
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Quote:
Originally Posted by Albert
My opinion would be to choose the sample rate based on the delivery format. 44.1 if the tracks are intended for CD, and 48k if they are video post production. If youa re mixing analog then you could choose 48k as well.
wouldnt that imply that u should do a new recording for every release be it video or cd id just record it as high as possible then bounce to cd quality
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12th May 2006
Old 12th May 2006
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I feel dumb for doing my last 50 projects @ 48khz...should have stuck with 44.1 it seems...
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12th May 2006
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Quote:
Originally Posted by xabiton
wouldnt that imply that u should do a new recording for every release be it video or cd id just record it as high as possible then bounce to cd quality

It's a triage situation. ONE format will get hosed. (Unless you're tracking to DSD/SACD/whatever Sony calls it this week... then they ALL get hosed. Just kiddin'... I guess.)

If the video/DVD version of your product is most important, go with 48, 96 or 192 kHz SR.

If the audio CD version is most important, go with 44.1, 88.2, or -- I always have to do the math on this one -- 176.4 kHz sample rate.

Of course, this triage situation is the reason that Sony trots out the DSD super-oversampled one bit thing every so often. The SACD SR is an even multiple (I'm told) of both 44.1 and 96. (I think that was it. I don't know about 192 and 176.4 kHz...)


Anyhow, if you think about what goes on during an 'uneven' sample rate conversion (say, from 48, 96, or 192 down to 44.1) it's easy to see why it can cause so much damage to your signal. EVERY single sample value in the new digital file/stream will have been estimated via interpolation -- educated guessing. You can try to 'smooth over' the resulting alias error with clever forms of dither -- but there is no way to avoid interpolation error.


[EDIT: I did not understand SRC when I wrote this. Good SRC can be 'transparent' within the accuracy limits of the target rate.]
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12th May 2006
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Quote:
Originally Posted by Albert
My opinion would be to choose the sample rate based on the delivery format. 44.1 if the tracks are intended for CD, and 48k if they are video post production. If youa re mixing analog then you could choose 48k as well.

me too

Cheers
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12th May 2006
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Quote:
Originally Posted by theblue1
It's a triage situation. ONE format will get hosed. (Unless you're tracking to DSD/SACD/whatever Sony calls it this week... then they ALL get hosed. Just kiddin'... I guess.)

If the video/DVD version of your product is most important, go with 48, 96 or 192 kHz SR.

If the audio CD version is most important, go with 44.1, 88.2, or -- I always have to do the math on this one -- 176.4 kHz sample rate.

Of course, this triage situation is the reason that Sony trots out the DSD super-oversampled one bit thing every so often. The SACD SR is an even multiple (I'm told) of both 44.1 and 96. (I think that was it. I don't know about 192 and 176.4 kHz...)


Anyhow, if you think about what goes on during an 'uneven' sample rate conversion (say, from 48, 96, or 192 down to 44.1) it's easy to see why it can cause so much damage to your signal. EVERY single sample value in the new digital file/stream will have been estimated via interpolation -- educated guessing. You can try to 'smooth over' the resulting alias error with clever forms of dither -- but there is no way to avoid interpolation error.
very true. however at the end of the day consumers dont care as long as it sounds listenable and the lowest of the low end is still listenable to 90% of the audience. I think sometimes people look too much into quality which is why I think 16/48 is just fine for almost everything its beyond cd quality and most people arent lookin for anything more than that or can even tell for that matter
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12th May 2006
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Sure, if you're careful, you can get good results working at 16 bit -- but working at 24 bits simply makes things much easier, you don't have to keep hustling your levels, you can relax a little and concentrate on getting a good sound without sweating bumping 0 dBfs all the time.


I'll agree that people tend to listen on less than ideal systems -- and today increasingly in less than ideal formats (although at least these days few of us are stuck listening to 24 kbps streams)...

... but I would say that puts a HIGHER burden on the orginal source material.

Really clean, well recorded material makes the best mp3s, AACs, WMAs.

Material with phase or clarity problems just gets trashed all that much harder by the perceptual encoding codecs used to squeeze the material down into a tiny package.


Also... you say 48 kHz... maybe you're mixing out into the analog world and coming back in on a converter set to the CD rate of 44.1 kHz, which is fine. Or maybe you're outputting for video's 48 kHz base rate.

But if you're outputting to CD or conventional 44.1 Mp3 and doing a sample rate conversion from 48 kHz to the target rate of 44.1 you're almost CERTAINLY going to end up with a worse sounding 44.1 file than if you started at 44.1 in the first place, for the 'uneven' sample rate conversion issues hinted at above.
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12th May 2006
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Quote:
Originally Posted by theblue1
Sure, if you're careful, you can get good results working at 16 bit -- but working at 24 bits simply makes things much easier, you don't have to keep hustling your levels, you can relax a little and concentrate on getting a good sound without sweating bumping 0 dBfs all the time.


I'll agree that people tend to listen on less than ideal systems -- and today increasingly in less than ideal formats (although at least these days few of us are stuck listening to 24 kbps streams)...

... but I would say that puts a HIGHER burden on the orginal source material.

Really clean, well recorded material makes the best mp3s, AACs, WMAs.

Material with phase or clarity problems just gets trashed all that much harder by the perceptual encoding codecs used to squeeze the material down into a tiny package.


Also... you say 48 kHz... maybe you're mixing out into the analog world and coming back in on a converter set to the CD rate of 44.1 kHz, which is fine. Or maybe you're outputting for video's 48 kHz base rate.

But if you're outputting to CD or conventional 44.1 Mp3 and doing a sample rate conversion from 48 kHz to the target rate of 44.1 you're almost CERTAINLY going to end up with a worse sounding 44.1 file than if you started at 44.1 in the first place, for the 'uneven' sample rate conversion issues hinted at above.
i never understood uneven sample rates as far as how they make ur material sound worse tho i just go with whatever i feel like at the moment. sometimes i like that harsh sound on my recordings but I usually go to 44.1 or 48 Ive tried 88.2 but in the end I havent heard a major difference when encoded back to 44.1 whats the major change there
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12th May 2006
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[EDIT: Oh, heavens, I REALLY STEP INTO THE STINKY STUFF HERE. Ignore all this nonsense, I really did NOT know what I was talking about!]

Here's my best attempt to explain the issues around bit depth and "uneven" sample rate conversion:
__________________________________

Bit Depth (digital word length)

Definitely work at the greatest bit depth feasable during production.

Even though you'll eventually be going to 16 bit for CDs or Mp3s, do all your production, processing, etc, at the highest bit depth practical (typically 24 bit for most of us) helps preserve as much of each track's individual accuracy/detail as possible. Mixing, EQ, compressing, and other processing is also best performed at the highest bit depth.

Look up bit depth on google and pay attention to the fact that each additional bit added to the word length of a digitally stored value allows a doubling of the possible number of values.

So a 16 bit number can store something over 64,000 possible values (actually 65,536). A 20 bit word length can store over 1,000,000 possible values. A 24 bit format can store over 16 million values (actually 16,777,216) -- thats's a 256-fold increase in potential dynamic resolution by using 24 bit over 16 bit format audio.

Lowering bit depth can be as easy as simply truncating bits off the digital word.

But we typically add a very, very small amount of noise (before truncation) to dither the sonic image to soften any sonic "jaggies" that may be revealed by the truncation.

Interestingly, this aspect of audio is very parallel to working with bitmapped images, particularly resizing such images.

Because of the nature of what we're doing with bit depth, truncating bit depth is analogous to resizing a picture by an even amount. Going, say, from 16,000 x 16,000 pixels to 8,000 x 8,000 pixels is as easy as throwing out every other column and every other row of pixels. The resolution is not as high -- but you haven't remapped any pixels or the 'shapes' they tend to form.

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Sample Rate Conversion

And downsampling to a given sample rate from a sample rate which is an even multiple of the target -- iow, downsampling from 192 kHz to 96 kHz or 48 kHz is also parallel to this -- we lose resolution -- but we don't have to remap any values. In essence, we simply discard every other sample. (The same applies to, say 88.2 kHz down to 44.1 kHz, etc.)


BUT when we downsample from an uneven multiple of the target frequency -- say from 96 kHz down to 44.1 kHz -- THEN we end up having to remap ALL our values -- guessing (interpolating) what those values MIGHT have been, often introducing far greater alias error (distortion, perhaps subtle, perhaps not so) than if we had started from 88.2 or even 44.1 in the first place (and not downsampled at all.)


To go back to the graphics example:


Img 1 - The image above was originally created as a 100 x 100 pixel graphic.


Img 2 - This image was the result of downsizing that same image down to a 50 x 50 pixel image. Note that resolution goes down -- but the general shape is retained fairly well... or at least comparatively well. Take a look at this:


Img 3 - This image was what happened when we downsized the image from 100 x 100 down to 57 x 57.

Despite the fact that the resolution of Img 3 above was 14% greater than Img 2, Img 2 creates a much more faithful representation of the image, despite its lower resolution.


[You may have to back away from your monitor quite a bit to see what I mean. I apologize for the size of these, they're already up on my server. But I think they clearly show what's going on when you have an 'uneven' downsample, whether it's a bitmap or a PCM recording.]


And audio's just like that... Or at least close enough for us to analogize...


Anyhow, the increase in dynamic resolution offered by increasing bit depth is as close to win-win as we'll be getting here. That 256-fold increase in dynamic resolution only costs us about an extra 50% in processing and storage overhead.

OTOH, doubling our sample rate -- or quadrupling it -- increases overhead by double (or quadruple).

And, while most folks can fairly easily discern 24 bit sound from 16 bit sound, listening tests have been considerably less persuasive that higher sample rates result in the same kind of perceived improvement.

It's obviously far too complex to discuss in depth, here, but I have found in the last 10 years I've been dealing with computer based recording that these seemingly parallel but very different issues continue to confuse large numbers of people.

So, to summarize:
  • use the greatest bit depth practical for production and reduce bit depth of your finished mix for output
  • if you want to work at high sample rate resolution, it's best to work at an even multiple of your target rate
  • this only applies if you are 'mixing in the box' -- keeping your audio in the digital realm -- for that reason, you may actually get better results running your (for instance) 96 kHz mix out into the analog realm and then back into another digital interface running at 44.1 -- try it that way and compare it with a full ITB downsample.
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13th May 2006
Old 13th May 2006
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thats some good info thanks
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13th May 2006
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>> And audio's just like that... Or at least close enough for us to analogize...

No it isn't. It would be if your poured the bits directly into your ears the way you are pouring these pixelated images directly into your eyes. But we run digital audio through D/A converters that create continuous voltages from the bits before listening to it. Downsampling algorithms should be and can be every bit as faithful as good DACs in reducing the bit rate.

>> you may actually get better results running your (for instance) 96 kHz mix out into the analog realm and then back into another digital interface running at 44.1 -- try it that way and compare it with a full ITB downsample.

Why do you think that a software downsampler cannot do as well or better than the combined effect of DAC and ADC, even the highest quality ones?

-synthoid
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13th May 2006
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44.1
moving (slowly) to 88.2
I have to convert my whole sample/IR library.
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13th May 2006
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Quote:
Originally Posted by synthoid
>> And audio's just like that... Or at least close enough for us to analogize...

No it isn't. It would be if your poured the bits directly into your ears the way you are pouring these pixelated images directly into your eyes. But we run digital audio through D/A converters that create continuous voltages from the bits before listening to it. Downsampling algorithms should be and can be every bit as faithful as good DACs in reducing the bit rate.

>> you may actually get better results running your (for instance) 96 kHz mix out into the analog realm and then back into another digital interface running at 44.1 -- try it that way and compare it with a full ITB downsample.

Why do you think that a software downsampler cannot do as well or better than the combined effect of DAC and ADC, even the highest quality ones?

-synthoid
It's not an ideal analogy.

I've seen graphic analogies that make the relationship more clear, to be sure. The best one I've seen shows a continuous wave form with two sets of 'samples rate divisions' superimposed. In that example, it's quite easy to see how -- once the continuous wave form has been digitized at one rate, trying to interpolate new values for a different rate* without reference to the original continuous wave form causes easy-to-grasp alias error.

I apologize for a somewhat deficient graphical analog.


But I stand by the bottom line: sample rate conversion to a target from a rate that is not an even multiple of the target will, by the very nature of the process, force an interpolatoin (educated guess) of ALL the 'new' sample values -- since the 'new' samples correspond to a different position on the x (time) axis. We don't know -- can NEVER know -- the precise dynamic value of those samples -- because they weren't taken. They MUST be guessed at.

Now... audio quality is a subjective thing, to be sure. Maybe YOU are not bothered by the alias error imparted by 'uneven' sample rate conversion.

But -- by every definition possible -- the results of such conversion must, of necessity, involve a guessed approximation of pretty much every sample value in the whole stream/file.

And I think you will find a consensus among seasoned professionals that (down) sample rate conversion from a rate which is not an even multiple of the target should be avoided if at all possible.


And, in answer to your final question, I think I see where you're going. What, you might ask, is the difference between the alias error implicit in a round of D/A in one sample rate to A/D in another sample rate -- and the alias error inherehent to the interpolation used by sofware-based SRC?

I'll have to beg off on the underlying technical/mathematical issues. But THAT carefully qualified suggestion (you'll recall the important qualifier "may" -- yes?) is based on my own real world experience -- which was pretty clear to me -- as well as the gleaned common wisdom of those I've read and respect.

But don't take it from me or my second hand common wisdom. Get out there and research it yourself and see what you find. If your mind doesn't get changed, I'd love to hear your thinking when you come back.


_______

* by 'different rate' in the first para, I mean, as elsewhere, going from a sample rate which is not an even multiple of the target -- or, for that matter, in upsampling, going to a target rate which is not an even multiple of the source rate.
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13th May 2006
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Quote:
Originally Posted by Albert
My opinion would be to choose the sample rate based on the delivery format. 44.1 if the tracks are intended for CD, and 48k if they are video post production. If youa re mixing analog then you could choose 48k as well.
I could not have said it better myself. Use the flavor that matches your end product's needs. All video production is done at 48K (i.e. movie scores, songs written specifically for a movie, etc...). So...record at 48K so there will be no conversion to deal with.

If CD only is your end product, 44.1K is fine.
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14th May 2006
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hey guys,


i get you.......i`ll do this thing at 44,1.....recording starts on monday, heads up



cheers
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14th May 2006
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Its a bit over simplified saying 44.1 output just discards every second sample. If you think about what happens in the A->D stage you'll realise that every second sample of 88.2 won't be the same as a 44.1 recording of the same source.

Even in 88.2 -> 44.1 every sample is going to be (should be) extrapolated under most situations

although overall I agree. Best solution is to record at least at 48 and master or mix in analog, or stay at 44 for all digital

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14th May 2006
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Not this old horny chesnut again..

First, I have a big problem with any kind of graphical analogy. Audio just doesnt work like that because the ears dont work like that and sound and light are 2 completely different phenomena.

Second, doesn't it depend on your recording environment, equipment and source as to what sample rate is best for you? The LSO at Abbey Road you might want a higher sample rate, but electric guitar in your bedroom.. is there any point?

Plus, Apogee's at 44.1 are much better than M-Audio at 96k.. as an example.. It just comes down to converter design as to what the best sample rate is, imo.

Third, when downsampling the audio is upsampled first to a common multiple.. so its not a case of simply dividing DOWN until you reach 44.1 - losing whatever you need along the way.

Me, I use 44.1 cos its just SO much easier and sounds just as good as anything else. I get really annoyed when project turns up at 48k because I have to change my setup and then the next few projects are all sped up cos I forgot about the 48k settings..
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14th May 2006
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i defenitly go for 88 or 44. just because you wanna have the least and less complicated math going on on your computer. 88 its just half of 44 (duh) but thats samples per second so for a project doing 48+tracks 3-5 min each at 48 or 96 or 192 to later be dithered into 44.1 thats a hell a lot of math IDB that doenst mean ill do it 100% correclty and can create some imperfection on the audio quelity. but if its 88.2 (which is enough to sample those upper harmonics we will never hear but still need as a natural ocurrance and to avoid aliasing) then to dither it to 44.1 is just every sample in half while 96 to 44 is every sample divided into 2.18xxxxxxxxxx so thats a lot of extra change in the endng result that gets chopped of at 24 bits, and even more at 16 bit wordlegnth. if you work to make music for film/TV then yes use 96 and 48 cause thats the format its used.
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14th May 2006
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I just stick with 44.1kHz throughout. The difference in quality between 48 and 44.1 is so small that it's not worth the hassel.
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Quote:
Originally Posted by gsilbers
i defenitly go for 88 or 44. just because you wanna have the least and less complicated math going on on your computer. 88 its just half of 44 (duh) but thats samples per second so for a project doing 48+tracks 3-5 min each at 48 or 96 or 192 to later be dithered into 44.1 thats a hell a lot of math IDB that doenst mean ill do it 100% correclty and can create some imperfection on the audio quelity. but if its 88.2 (which is enough to sample those upper harmonics we will never hear but still need as a natural ocurrance and to avoid aliasing) then to dither it to 44.1 is just every sample in half while 96 to 44 is every sample divided into 2.18xxxxxxxxxx so thats a lot of extra change in the endng result that gets chopped of at 24 bits, and even more at 16 bit wordlegnth. if you work to make music for film/TV then yes use 96 and 48 cause thats the format its used.
Wow.
I don't think I agree with a single point of this.

Except that 48k is used for film/TV.....
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u b k / So much gear, so little time!
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markknopfler66 / Music Computers
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