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digital format question, please
Old 27th July 2005
  #1
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digital format question, please

here is what i am doing,
summing from protools hd thru cranesong
spider to different pc
into samplitude. i arrive digitally
the conversion taking place into the spider.

my current project is at 24/48
my question is -
am i better off sampling down to 44/16
adding cranesong dither and end up w
a final sampled doodad in samplitude, or

up sampling to 96/24 then going down w software?

the final limiting would be done in samplitude

or are there other options?

the reason i ask is, i feel if i record into samplitude at 44/16 as i monitor i am really hearing the final product

as to upsampling or staying at present sample rate
there is a "school of thought"
that perhaps any limiting or corrective eq, i do would be better at 96/24

any thoughts on this topic appreciated in advance, thanks
Old 27th July 2005
  #2
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Old 27th July 2005
  #3


do as few sample rate conversions as you can
do processing at as high a resolution/sample rate as practical



-tINY

Old 28th July 2005
  #4
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 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.

________________________

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 values, 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:


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


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:


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 this third image was 14% greater than the second one -- the second one creates a much more faithful representation of the image.

[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... heh 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 9 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.
Old 28th July 2005
  #5
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thanks for the reply.

as i said my current project is 24/48
i just bussed it out at that rate, then reconverted in spider to 44.1
i applied some high end from my avalon 2055
somewhat brittle

next i brought the 24/48 mix into the spider (same eq setting) however, now i upsampled in the spider to 24/96
the difference was enormous, especially in the high end

a difference to me of being able to add "air"
or have to apply high shelving

also there was much more depth

this method obviously is superior, or is it?
what will happen if i now convert w software to 16/44.1
some people must have tried this

also i was under the impression
that modern sample rate conversion was less
concerned w divisable multiples
than sample rate conversion of yore

if indeed this upsampling allows for less aliasing,
is it not a solution of sorts to working at 24/48 instead of 24/96 and all that entails (storage, processing)

comments please,
while i search for the best software conversion

to me this is a very important discussion
it is not about which sounds better, i have no doubt that
24/96 sounds better, i want to know how best to arrive at usable 16/44

the upsampling thing is quite dramatic

thanks
Old 30th July 2005
  #6
Sample rate conversion from uneven multiples of the target rate produces problematic alias error, no matter how fancy your gear is. Avoid doing it whenever possible.

As far as getting out of your current situation, let your ears be your guide -- at least assuming you have a decent monitoring situation. You can either use software SR conversion or you can go out into the analog world and resample in at 44.1 kHz. Use the one that sounds best on your material.


Gotta go.
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