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the Digital/Analogue time bog
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Digital and Analogue together?
Totally
66%
 66%  [ 10 ]
Mostly
6%
 6%  [ 1 ]
A little bit
26%
 26%  [ 4 ]
No way
0%
 0%  [ 0 ]
Total Votes : 15

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elektro80
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PostPosted: Sun Jun 11, 2006 2:07 pm    Post subject: Reply with quote  Mark this post and the followings unread

mosc wrote:

I know a guy who runs a recording studio. He does very little non-linerar post processing. He professes to use 24 bit 44.1 KHz for his sessions because of the artifacts generated when down sampling. His recordings are superb. .


This is hardly suprising. Very Happy I stick to this rule myself whenever I can. I also do what Kassen is suggesting, sticking with the 88.2 etc., when something needs to get a special treament.

Anyways, it seems like we have been summing up a lot of very relevant issues.. like:

1. It is smart to understand both how the theory works and what is the practical limitations of the gear.
2. Gain management is one huge serious issue no matter if the recording is done using analog or digital gear.
3. A modern studio is best served with a mix of digital and analog gear.
4. Ordinary musicians cannot afford whatever the godlike reviewers in pro audio magazines are pretending is neccessary in order to make decent audio recordings.


All of this is nothing new. It is of course rather telling that the industry is addressing these issues by issuing gear sporting analog warmth inducing leds and whatnot. As if turboinjecting n layers of "analog warmth" on absolutely everything will be of any help at all.

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PostPosted: Sun Jun 11, 2006 2:08 pm    Post subject: Reply with quote  Mark this post and the followings unread

I wonder, IMO this thread has now earned its place in the production forum. Anyone for moving it in there?
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PostPosted: Sun Jun 11, 2006 6:33 pm    Post subject: Reply with quote  Mark this post and the followings unread

Kassen wrote:
I can imagine this working out for a *recording* studio. etc. etc.


What he said ^^^

It feels like this thread is going around in circles. Never have I or anyone else for that matter stated that 96k/24bit will have any advantage over 44.1k/16 or 24bit for recording or when little digital post processing is required.

mosc wrote:
To me, your second non-clipping version (both sample rates) differs from the clipped version more than the change in sample rates


That sounds about right - there's quite a bit of unwanted non-linear clipping going on in the first version! It's good to know that the sample rate difference isn't as bad as the heavy clipping though.....
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PostPosted: Sun Jun 11, 2006 6:36 pm    Post subject: Reply with quote  Mark this post and the followings unread

I think mosc just was thinking out loud.. no intent apart from that.. Very Happy
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PostPosted: Sun Jun 11, 2006 7:19 pm    Post subject: Reply with quote  Mark this post and the followings unread

There is the possibility that the sole reason for the difference, is the downsampling itself. I guess you would need some more files to eliminate that possibility. Many vinyl advocates swear that format produces superior 'sound stage'. This is most likely just phase errors introduced by the mechanical interactions of the x and z motion. Better does not always sound best.
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PostPosted: Mon Jun 12, 2006 2:28 am    Post subject: Reply with quote  Mark this post and the followings unread

As I asked for it, I thought I'd post my experiences with the files.

The difference is quite pronounced: the 44k file sounds more cluttered and muddy - especially in the high end - while the 96k file sounds more defined and, in a way, consonant.
The last, thereby, is significantly more pleasant on my ears.
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PostPosted: Mon Jun 12, 2006 6:41 am    Post subject: Reply with quote  Mark this post and the followings unread

Back to my 44.1 kHz friend with the recording studio. Yes, he mostly records acoustic stuff - folk groups and small choirs - and small rock bands. As I said, he doesn't do much non-linear post processing.

OK with me to move this topic.

As for the topic going around in circles, these analog/digital ones always do, just like the Mac/PC discussions. That's because, IMHO, people make general statements which don't hold up in all instances, or express opinions as if they were facts. Then, there are people, like me, that can get nit picky. My kids hate me for this - maybe they aren't the only ones.

Example: At electro-music 2006 Stuart Diamond was talking about playing the Lyricon, a wonderful vintage analog electronic "wind" instrument. He said it was more expressive than "digital" instruments because analog has infinite control and digital has perceptable steps. After his absolutely superb performance, I talked it over with him. I mentioned that analog was not infinitely adjustable because of the noise. He agreed that he wasn't talking about digital exactly, but the 128 step MIDI wind controllers on the market that he'd tried. We agreed that if they had some digital controller with an unspecified higher degree of resolution, then the digital steps would be below the level of perception and the expressivity would be equivalent.

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Kassen
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PostPosted: Mon Jun 12, 2006 5:14 pm    Post subject: Reply with quote  Mark this post and the followings unread

mosc wrote:

As for the topic going around in circles, these analog/digital ones always do, just like the Mac/PC discussions. That's because, IMHO, people make general statements which don't hold up in all instances, or express opinions as if they were facts. Then, there are people, like me, that can get nit picky. My kids hate me for this - maybe they aren't the only ones.


With all due respect; in this case I think it was going round in circles because you once again brought up a point I had already adressed in detail. I already aksed you to please read carefull what it actually is that I wrote and now I will again.

I've taken considderable care in explaining which of my statements are general and which are speciffic examples of a certain phenomenon. I used some examples because I find these make explaining these things much easier. Not everybody wants to read the sort of tome on DSP that I'm reading. If anyone is doing what you complain about it's you; that 44.1 studio is very interesting in itself but it doesnt realy relate to what you seemed to be replying to at all. I was trying to answer a very speciffic question there about a issue that affects many people in practice and what you did was basically say "If you don't do the sort of thing that causes the problem you don't need the solution", Well; that's great but it's of little help if for some reason somebody needs to perform 50 effects in series on a wave file.

To put it bluntly; I'm realy sorry but I simply strongly disagree with your post here. I don't think that what I'm doing is in any way comparable to the typical pc-mac debate; for one thing I stressed about a million times neither is better and that a carefull choice must be made based on individual needs and the speciffic situation to be dealt with. I'm sure digital systems are perfect to many people in lots of cases but that's not what the topic is about and indeed; that's a lot like the Mac-PC debates. I resent being acused of such things after I spend considderable efford to explain where pitfalls are.

If everybody keeps complaining about this there might be a reason for their complaints.

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PostPosted: Mon Jun 12, 2006 7:42 pm    Post subject: Reply with quote  Mark this post and the followings unread

mosc wrote:
As for the topic going around in circles, these analog/digital ones always do, just like the Mac/PC discussions. That's because, IMHO, people make general statements which don't hold up in all instances, or express opinions as if they were facts. Then, there are people, like me, that can get nit picky. My kids hate me for this - maybe they aren't the only ones.


This discussion shouldn't have gone down this path.... I never intended to start such a thing when I posted my test. I was very careful to state what I used and how I did it, and always said "for me" or something similar when expressing my opinions about the results.

So I'm not sure what you're nit picking about, and I'm not sure what general statements or opinions expressed as facts you're talking about. If you could at least tell me where I've made a general sweeping statement or expressed an opinion as fact I would be happy to explain or apologise.
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PostPosted: Tue Jun 13, 2006 12:48 am    Post subject: Reply with quote  Mark this post and the followings unread

teradacto wrote:
My friend was talking about how when Skinny Puppy made the switch from analogue to digital it was like night-and-day. That's the main reason I made this topic...the other being when, I used an analogue sequencer and tried to synchronize it with a drum machine tempo, it seemed like it was imposoble. But then again I didn't really know what I was doing. I just wanted to know, if there is some dificulties in combining the two...so thanks to you all, even though now I see that this topic is kind of a drag...


Maybe you can elaborate on specific problems you have.
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PostPosted: Tue Jun 13, 2006 12:53 am    Post subject: Re: uh Reply with quote  Mark this post and the followings unread

teradacto wrote:
Embarassed I quit making electronic music. Shocked

i'm going to make Pop music, or Yodeling or something...just so i can find a forum where hopefully everybody isn't so god damn smart. Very Happy

ya frikkin' braniacs


Alllways be carefull what you "poll" for. Very Happy Rolling Eyes
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PostPosted: Tue Jun 13, 2006 6:09 am    Post subject: Reply with quote  Mark this post and the followings unread

Afro88 wrote:

This discussion shouldn't have gone down this path.... I never intended to start such a thing when I posted my test. I was very careful to state what I used and how I did it, and always said "for me" or something similar when expressing my opinions about the results.

So I'm not sure what you're nit picking about, and I'm not sure what general statements or opinions expressed as facts you're talking about. If you could at least tell me where I've made a general sweeping statement or expressed an opinion as fact I would be happy to explain or apologise.


Well, for what it's worth, Afro, I think you are perfectly in the clear. I think it should be very clear that your test wasn't meant to prove some universal truth beyond any doubt and was more of a attempt to see wether you could recreate some of these phenomena yourself. That's a very healthy aditude, I think, and not at all one to be ashamed of.

If you'd like another test you could try the difference between applying effects one by one in a wave editor and as a single chain in a full DAW. If you do that the program will be forced to downsample every time after the effect instead of keeping it all in a high resolution for the duration of the chain. The results may make you look at bouncing tracks of cd quality audio in a different way.

This one indeed is a blanket statement; wherever CPU's and workflow permit using a single chain (or a "batch" in a wave editor) yields a better sound quality then single passes unless your wave file already has the same resolution as the internal resolution of the effects. This is a nice one; it affects nearly everybody who is in digital music, it's easy to test empyrically and easy to prove mathematically.

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PostPosted: Tue Jun 13, 2006 8:16 am    Post subject: Reply with quote  Mark this post and the followings unread

Quote:
a carefull choice must be made based on individual needs and the speciffic situation to be dealt with. I'm sure digital systems are perfect to many people in lots of cases but that's not what the topic is about


There is always a tendency in forums to ask "the big question" hoping that it applies to all circumstances. I agree with Kassen that it is impossible to do this with regards to this topic. Rarely is there an aspect of music where you have one solution, though I've seen hotshots on other websites attempt this all the time.

I am very careful about sound degredation (well, unintentional degrading at least), and I try to keep everything as matched-up as possible. But I think I haven't done enough research when using new plug-ins, especially free "made with SynthEdit" ones.

I don't like mixing different applications on the computer. It's just another one of a million variables that could affect the sound. I use WaveLab and Fruity Loops every time, and I stick to 16-bit 44100 because of what I knew about creating music CDs. This cuts down on some of the variables. MP3s are a different story. Of course, there is a minimum - 8bit 22050 sounds mixed with 16bit 44100 sounds using a sound card set at 48000 and a plan of sending it to a CD makes my head hurt. Anyway, I think it's good to keep these things in mind and stick to a strict setup, whether it's analog or digital, or converting from analog to digital and vice versa. Unfortunately, having all of this on your mind can sometimes hamper your creativity, but it seems to me that once you get into a routine with the technical aspects, know what your end goal is, and are powerfully confident with the tools you use, these complications will fade away.

Knowledge is power! thumleft
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PostPosted: Tue Jun 13, 2006 12:06 pm    Post subject: Reply with quote  Mark this post and the followings unread

Afro88 wrote:
If you could at least tell me where I've made a general sweeping statement or expressed an opinion as fact I would be happy to explain or apologise.


I didn't mean to imply that you make any such statements or opinions. I was talking about myself when I was referring to people being nit pickey.

But, to be nit pickey, you did say in the post where you first presented your test results, [quote="Afro88']I think that's convinced me to start using higher sample rates whenever effects are going to applied. [/quote]Interesing because you did this test because Kassen said in an earlier post,
Kassen wrote:
What's going on here? The errors added up. You can now see you could prevent this by using higher resolution files and only going down to cd quality at the end; in that case the rounding errors do get in the file but in the bits below the 16th (hopefully!) and still get discarded.


Your test changed both the sample rate and the bit depth. A test is rarely conclusive if two variables are changed. I point this out because not to criticise you personally, but to try to contribute to the knowledge of the topic. I'm not sure if the degradation was do to sample rate (which is related to aliasing distortion) or the bit depth.

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PostPosted: Tue Jun 13, 2006 1:07 pm    Post subject: Reply with quote  Mark this post and the followings unread

mosc wrote:
Your test changed both the sample rate and the bit depth. A test is rarely conclusive if two variables are changed. I point this out because not to criticise you personally, but to try to contribute to the knowledge of the topic. I'm not sure if the degradation was do to sample rate (which is related to aliasing distortion) or the bit depth.


Yes, that's true, these do add noise (except for increasing bit depth).

Generally you can say that for finite wordlengths (so any practical system) nearly every operation will cause a rounding error. The big quetion is how large those will be and wether they'll ever get into the final data.

The people that implement plugins aren't totally clueless (most of the time....) and there are very good reasons for jumping sample rate and bit depth (and even switching from integers to floats and back). The idea is that you do get noise when going down again but not as much as you would get if you did all the operations in the same bit depth and sample frequency as the file is. In this case I think it's safe to say the degradition caused by switching formats was caused exclusively by switching sample rates. Just going up then down in bit depth isn't something that should affect your data, at least not compared to you original data, it may well cause a decrease relative to the plugin's internal data. The largest amount you could possibly lose is half a LSB of the original bit depth.


About the two variables; you are right, scientiffically speaking but here bit depth and sample rate affect how the calculations work in entirely different ways. These are not things that are changed at random, there are very sound reasons for both and if you could get your daw to only change either one independantly this should have a positive effect and changing both also has (larger) benefits. The tests to prove this have already been done and don't realy need to be repeated because the math adds up nicely.

I agree that if we were charting some unknown teritory you should only change one variable at a time, then change groups looking for interaction but these tests have already been done. Ideally plugins and DAWs will already have been optimised for this, the big question is how to make the most of the optimisation and prevent those situations where "optimal" isn't "inaudible" within the overall context.

reinventing the wheel may also be fun and educational but this wheel involves realy, realy hard math and I don'think there's any substitute for starting with a very serious cource on DSP if that's what you want.

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PostPosted: Tue Jun 13, 2006 4:23 pm    Post subject: Reply with quote  Mark this post and the followings unread

mosc wrote:
I didn't mean to imply that you make any such statements or opinions. I was talking about myself when I was referring to people being nit pickey.

But, to be nit pickey, you did say in the post where you first presented your test results, " think that's convinced me to start using higher sample rates whenever effects are going to applied." Interesing because you did this test because Kassen said in an earlier post, "What's going on here? The errors added up. You can now see you could prevent this by using higher resolution files and only going down to cd quality at the end; in that case the rounding errors do get in the file but in the bits below the 16th (hopefully!) and still get discarded."


Yes, fair enough, but a little higher he says the words downsample and upsample. In earlier posts he also talks about using higher sampling rates when treating audio repeatedly. I thought "alright, lets try 96k/24bit, the value they chose for dvd audio" - it seemed like the next logical step up resolution wise.

mosc wrote:
Your test changed both the sample rate and the bit depth. A test is rarely conclusive if two variables are changed. I point this out because not to criticise you personally, but to try to contribute to the knowledge of the topic. I'm not sure if the degradation was do to sample rate (which is related to aliasing distortion) or the bit depth.


Yes, this is true. I understand that for a truely scientific test, I should only change one variable. I will do some more comparisons with 16 and 24 bit renders later on, and if this doesn't produce a similar quality difference than the previous test, I will do 44.1 and 96 (and probably 88.2) at 16 bit. Now you're being nitpicky, and I like it, because it's relevant and challenges what I think is right. If only you said this from the beginning! It got very frustrating when you kept on talking about what's needed for recording and mixing. But we got to this point in the end, so all the heated posts are worth it anyway.

This is great, now we're getting productive again. Thanks for posting this mosc Smile I'm off to do some more tests...

edit: bah, the quotes in mosc's quote didn't work. I've just put in these instead: " "
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PostPosted: Tue Jun 13, 2006 5:23 pm    Post subject: Reply with quote  Mark this post and the followings unread

Afro88 wrote:

This is great, now we're getting productive again. Thanks for posting this mosc Smile I'm off to do some more tests...

Thanks... I feel releaved.

BTW, I think it was interesting that you found one program that sounded the same at the two sample rates. Like Kassen said, good code developers can deal with non-linearities, round off errors, and other artifacts.

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PostPosted: Tue Jun 13, 2006 5:26 pm    Post subject: Reply with quote  Mark this post and the followings unread

Ok, I've now done 44.1k at 24bit and 96k at 16bit, as well as 88.2k at 16 and 24bits so we can compare the range. To my ears there is no difference between 16 at 24 at 44.1, 88.2 and 96. This is probably because the initial signal has only about 1 db of headroom, and there really are no low (volume wise) parts during any phase. I would test these at -48db or something, but we all know the difference between 16 and 24bits when it comes to low volume passages.

The difference between 96k and 44.1k rendering is to my ears exactly the same at both 16 and 24 bits. Also, I can't tell the difference between the 88.2k file and the 96k file. At first I thought I could, but as soon as I closed my eyes and did a random test I couldn't tell at all.

So from this I'd say using the Classic chorus plugin at 96 or 88.2k is a wise choice no matter what. I would guess that using most other plugins capable capable of higher sample rates would yeild similar results, but I don't know this absolutely for sure. I do know that I'm going to use 88.2k or 96k whenever I can just to be on the safe side. I'm not sure which, because I use alot of my 96k G2 in tracks, but I'll be bouncing to 44.1k. I guess it's six of one and half a dozen of the other. As dvd audio becomes more popular maybe it's best to stick with 96k... but then I guess no one would be able to tell the difference if I upsample the final mixdown to 96k anyway Wink

By the way, does anyone know if the built in Cubase SL3 plugins are 96k capable? Or do they just work at 44.1?


test 2 44k 24bit.wav
 Description:
Multiple chorus plugin test rendered at 44k 24bit, then truncated to 44k 16bit

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 Filename:  test 2 44k 24bit.wav
 Filesize:  1.35 MB
 Downloaded:  1047 Time(s)


test 2 96k 16bit.wav
 Description:
Multiple chorus test rendered at 96k 16bit, converted to 44.1k 16bit

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 Filename:  test 2 96k 16bit.wav
 Filesize:  1.35 MB
 Downloaded:  995 Time(s)


test 2 88.2k 16bit.wav
 Description:
Multiple chorus test, rendered at 88.2k 16bit, converted to 44.1k 16bit

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 Filename:  test 2 88.2k 16bit.wav
 Filesize:  1.35 MB
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PostPosted: Tue Jun 13, 2006 5:37 pm    Post subject: Reply with quote  Mark this post and the followings unread

And the last ones...

All files for the tests are in the same place now. If a mod feels like putting them in a proper order that would be great. If not, I'm sure people can figure it out Smile


test 2 88.2k 24bit.wav
 Description:
Multiple chorus test, rendered at 88.2k 24bit, converted to 44.1k 16bit

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 Filename:  test 2 88.2k 24bit.wav
 Filesize:  1.35 MB
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test 2 44k 16bit.wav
 Description:
Multiple chorus test, rendered at 44k 16bit, no conversion necessary

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 Filename:  test 2 44k 16bit.wav
 Filesize:  1.35 MB
 Downloaded:  1056 Time(s)


test 2 96k 24bit.wav
 Description:
Multiple chorus test, rendered at 96k 24bit, converted to 44.1k 16bit

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 Filename:  test 2 96k 24bit.wav
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PostPosted: Tue Jun 13, 2006 6:02 pm    Post subject: Reply with quote  Mark this post and the followings unread

Kassen wrote:
About the two variables; you are right, scientiffically speaking but here bit depth and sample rate affect how the calculations work in entirely different ways. These are not things that are changed at random, there are very sound reasons for both and if you could get your daw to only change either one independantly this should have a positive effect and changing both also has (larger) benefits. The tests to prove this have already been done and don't realy need to be repeated because the math adds up nicely.


The result is greater than the sum of it's parts, sure. I can't hear it in my single audio file with one type of plugin at one volume level, but it makes sense. I guess it's sort of like screen resolution - increasing the horizontal number of pixels will give you finer detail in one way, and the same with vertical number of pixels, but doing both at the same time results in a much more readable picture, more than one or the other could do on their own.

Kassen wrote:
I agree that if we were charting some unknown teritory you should only change one variable at a time, then change groups looking for interaction but these tests have already been done. Ideally plugins and DAWs will already have been optimised for this, the big question is how to make the most of the optimisation and prevent those situations where "optimal" isn't "inaudible" within the overall context.

reinventing the wheel may also be fun and educational but this wheel involves realy, realy hard math and I don'think there's any substitute for starting with a very serious cource on DSP if that's what you want.


This is why I'm going to leave my tests here (unless I get really really bored Wink). My interest was whether upping the sample rate and bit depth would affect my productions, and the answer is yes. I always do a a fair bit of post processing (I guess the style demands it), and upping the sample rate aids clarity. I also record audio a fair bit, do a few fades and use delay and a bit of reverb, so an increased bit depth would help here. Doing both at the same time will have larger benefits - great! I'll never be able to test for these benefits given time and boredom restraints, but as it's a bonus to making the changes for different reasons, I'm open to accepting it.
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PostPosted: Tue Jun 13, 2006 6:06 pm    Post subject: Reply with quote  Mark this post and the followings unread

mosc wrote:
Thanks... I feel releaved.


So do I Smile

mosc wrote:
BTW, I think it was interesting that you found one program that sounded the same at the two sample rates. Like Kassen said, good code developers can deal with non-linearities, round off errors, and other artifacts.


Indeed! They just cost so much! I can only afford the Waves Q series at the moment, but I would love their stereo imager and compressor range too...
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elektro80
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PostPosted: Tue Jun 13, 2006 6:07 pm    Post subject: Reply with quote  Mark this post and the followings unread

The signal used is complex enuff, but depending on the DAW used doing the same with a fairly complex choir piece with reverb could turn out differently . One of the reasons I gave up on C b ase was aftersuch a session.
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PostPosted: Tue Jun 13, 2006 6:16 pm    Post subject: Reply with quote  Mark this post and the followings unread

Well, your approach makes sense. I can't listen to your latest tests at this time because I'm running on a Mac laptop. I will listen when I get a chance. As for picking multiples of 44.1, that's a common approach and it is logical, but from what I've read there are very good algorithms for downsampling non-integer multiples.
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PostPosted: Wed Jun 14, 2006 7:43 am    Post subject: Reply with quote  Mark this post and the followings unread

mosc wrote:
from what I've read there are very good algorithms for downsampling non-integer multiples.


There are, but not all software suites use those. There are some pages that compare the different DAWs using frequencyplots after downsampeling. Tghe results are quite startling. Some of the realy expensive stuff looks terrible, some of the cheaper programs (like renoise) have nearly perfect algorithems....

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PostPosted: Wed Jun 14, 2006 7:55 am    Post subject: Reply with quote  Mark this post and the followings unread

Got a link to those? Sounds very valuable.
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