Sample rates.....what's up with 88.2?
Well, sounds like you've got all the bases covered then... I'm pretty much stumped. One more thing though... are the Scope installer .exe's - the ones that you say won't run - residing on your hard drive or are you running them from the CD?valis wrote: I already cleaned everything manually, and it's a different machine. My RME box has dual xeons and certain scope devices dislike that (Vorb for example).
I still think that finding out why the exe's don't run would solve the problem.
Appreciate the help btw.
No, my thoughts were just that maybe if you were running them off of a CD, then the problem might be with the CD drive driver or something like that. Just for the helluvit, though, what if you copied them onto a CD-R & tried running it from that drive? Stupid idea, but just might work... SOMETHING'S preventing them from running where they are.valis wrote:I've been installing from the harddrive and a network share using both a freshly downloaded zip & one that I had lying around on my fileserver.
I don't have a 4.0 cd (2.04 & 3.1c only on CD) and I think that a network share is as good as a cd anyway, especially mapped as a drive.
I would definitely try moving the installer files to a different drive... try it on C & try it copied to a CD-R & run it from your CD drive... that might not tell you "why" it's not running right where it's at, but if it works, then you're golden & it's STILL better then reinstalling Windoze!valis wrote:Well there's actually 2 drives there, and the installer is on D and Scope is on C. Yes you're right something is preventing them, which is what I find so odd. I wish there was some way to enable logging for the installer/Scope so I could troubleshoot better.
Installing from both drives was one of the earlier things I tried & I think that a mapped network drive is as good as a CD for testing purposes (I do it a lot for installing other apps).
I ran & bought a new HD. It's too big but if it solves the problem I'll move an older drive from my DAW over & upgrade the DAW with this one.
I ran & bought a new HD. It's too big but if it solves the problem I'll move an older drive from my DAW over & upgrade the DAW with this one.
A highly appropriate read, considering the topic of this thread.
http://www.gearslutz.com/board/showthread.php?t=93177
Enjoy.
Neil
http://www.gearslutz.com/board/showthread.php?t=93177
Enjoy.
Neil
well, you can't but associate Swiss with Precision 
... and peek the Sharc labels in his box
I learned that filter coefficients for various sample rate conversions have different memory demand - didn't know that before.
btw an article about their standalone samplerate converter made me much more aware of the problem, cool dude - and likes Miyazaki films
cheers, Tom

... and peek the Sharc labels in his box

I learned that filter coefficients for various sample rate conversions have different memory demand - didn't know that before.
btw an article about their standalone samplerate converter made me much more aware of the problem, cool dude - and likes Miyazaki films
cheers, Tom
What? I can't hear you... you don't have enough frequency content above 22k in your voice.garyb wrote:once again though, he says that it's easier to implement a good filter at 96k, but that it's perfectly possible to achieve the same results at 44.1k. i would assume that for cheaper ad/da there's much more of an issue than with better components.

He also affirms that the math IS different going from 96k to 44.1 vs. 88.2 to 44.1... something you & I argued about (errr... DISCUSSED). To me the analogy would be the difference between asking your machine to convert every two samples to a single sample - a nice, even, low-error-potential process, vs. asking your machine to: "please convert every 2.1768 samples to a single sample, and please do that 5,760,000 times for each minute of recorded time... oh, and you can you decide what to do with the extra .8232 of every third sample... feel free to do with it as you will - chuck it into the new even or odd-numbered samples, or mix it up randomly as you see fit. Cheers... I'll be back in five minutes to see how you're doing!"
Neil
sorry Neal - the math itself is identical, just the filter complexity variesNeil wrote:...He also affirms that the math IS different going from 96k to 44.1 vs. 88.2 to 44.1... something you & I argued about (errr... DISCUSSED). ...
imho it's worth quoting Mr. Weiss here, inparticular for the last sentence, as a lot of folks seem to think that the bitdeepth (resolution) is what defines sound - it's the artifacts of the processing ... (lowpass filters are omnipresent in digital circuits)...I guess you are alluding to the 48 kHz vs. 96 kHz discussion from earlier. Many would say that it is easier to scale down from 88.2 kHz to 44.1 kHz than from 96 kHz to 44.1 kHz. It does take more effort to scale down from 96 kHz to 44.1 kHz; as bigger filters are needed.
*A sample rate converter is basically a low-pass filter, with the respective management of the coefficient of the filter(digital filters).*
When converting from 88.2 kHz to 44.1 kHz, one must use a low-pass filter that separates at 22.05 kHz. When converting from 96 kHz to 44.1 kHz, there are more filter coefficients, and, as a result, more memory is required. That is a disadvantage, but not a huge one in and of itself. The conversion from 96 kHz to 44.1 kHz can absolutely be as good as one that converts from 88.2 to 44.1 kHz. A bit more resources are required, but in these days, that is no problem.
There can be sound differences among various converters, depending on the sampling rate used, because, as stated before, the sample rate is converted at the input, by the implementation of filters. These filters vary from brand to brand. With the various filters come artifacts, and different sound results...
couldn't resist to quote this either, as it's not so often that one is confirmed by a well respected expert of this calibre...Some read 24 bit and think that it is TRULY 24 bits! This is not the case. Very good converters do well to have 20 bits of resolution!...

cheers, Tom
First of all, if bigger filters are needed, and as he says "a bit more resources are required", then the math must indeed be different. It's a machine, you can't expect it to behave the same way adding 1+1 as when it adds 1+3, otherwise you'd get the same answer for all equations.astroman wrote:sorry Neal - the math itself is identical, just the filter complexity variesNeil wrote:...He also affirms that the math IS different going from 96k to 44.1 vs. 88.2 to 44.1... something you & I argued about (errr... DISCUSSED). ...
imho it's worth quoting Mr. Weiss here, inparticular for the last sentence, as a lot of folks seem to think that the bitdeepth (resolution) is what defines sound - it's the artifacts of the processing ... (lowpass filters are omnipresent in digital circuits)...I guess you are alluding to the 48 kHz vs. 96 kHz discussion from earlier. Many would say that it is easier to scale down from 88.2 kHz to 44.1 kHz than from 96 kHz to 44.1 kHz. It does take more effort to scale down from 96 kHz to 44.1 kHz; as bigger filters are needed.
*A sample rate converter is basically a low-pass filter, with the respective management of the coefficient of the filter(digital filters).*
When converting from 88.2 kHz to 44.1 kHz, one must use a low-pass filter that separates at 22.05 kHz. When converting from 96 kHz to 44.1 kHz, there are more filter coefficients, and, as a result, more memory is required. That is a disadvantage, but not a huge one in and of itself. The conversion from 96 kHz to 44.1 kHz can absolutely be as good as one that converts from 88.2 to 44.1 kHz. A bit more resources are required, but in these days, that is no problem.
There can be sound differences among various converters, depending on the sampling rate used, because, as stated before, the sample rate is converted at the input, by the implementation of filters. These filters vary from brand to brand. With the various filters come artifacts, and different sound results...
couldn't resist to quote this either, as it's not so often that one is confirmed by a well respected expert of this calibre...Some read 24 bit and think that it is TRULY 24 bits! This is not the case. Very good converters do well to have 20 bits of resolution!...
cheers, Tom
Dunno why you included the thing about bit depth, though - I don't have a fight with what you're saying there. I go to 32-bit float for most mixdowns, but not for tracking - 24-bit is plenty for that, IME.
Neil
the bitdepth sidenote was in fact off topic in this context, and referred to some older discussions where these figures were slightly overestimated, so to say...
the context between the 2 quotes is Mr. Weiss' statement about the origin of converter 'sound' - I've written the very same a couple of years ago
not that I wanna brag or tell he's copied his wisdom from an internet post, but I've been called a few names recently (not by you) ...
cheers, Tom
the context between the 2 quotes is Mr. Weiss' statement about the origin of converter 'sound' - I've written the very same a couple of years ago
not that I wanna brag or tell he's copied his wisdom from an internet post, but I've been called a few names recently (not by you) ...

cheers, Tom
no, you misunderstand me. i never said there was no difference in the math. i argued that it was not just a matter of throwing away every other sample(going from 88.2k to 44.1k), and that there was just as high a potential for atifacts(as when going from 96k to 44.1k). i understand your logic in using 88.2k. i just know that you can do work of equal quality at 44.1k, and i just don't think that sample rate, in itself, is the best place to put your resources. it's just an opinion based on my experience. i'm not claiming it's the word of god....Neil wrote:He also affirms that the math IS different going from 96k to 44.1 vs. 88.2 to 44.1... something you & I argued about (errr... DISCUSSED).
What was it? A 24-bit vs. 16-bit discussion or a 24-bit vs. 32-bit discussion? I've always felt that 24-bit is fine for tracking, FWIW; for Mixdowns I sometimes use 32-bit... if it's something I'm going to be doing "pre-mastering" on, or "pseudo-mastering" myself like a demo project, for example, because I use Izotope's Ozone & that'll process at 32-bit, so may as well take advantage of its capabilities therein.astroman wrote:the bitdepth sidenote was in fact off topic in this context, and referred to some older discussions where these figures were slightly overestimated, so to say...
OK - well, I missed that missive, link me to it if it's still on the web somewhere... I'd be interested to see it, you rat-bastard (seemed like you felt left-out there since I hadn't called you a name yet).astroman wrote: the context between the 2 quotes is Mr. Weiss' statement about the origin of converter 'sound' - I've written the very same a couple of years ago
not that I wanna brag or tell he's copied his wisdom from an internet post, but I've been called a few names recently (not by you) ...
cheers, Tom

Neil
well, I feel ok - since I was supposed to convince Rupert Neve and some other 'big boys'. Much appreciated style 
at my command unleash hell
cheers, Tom

I can start you a new one, if you like - just needs a 3 letter word and some insights about marketing and what you pay for a square inch of mag space, make it a new topic and ...Neil wrote:...OK - well, I missed that missive, link me to it if it's still on the web somewhere... I'd be interested to see it...
at my command unleash hell
cheers, Tom
Yes, you were supposed to convince Rupert, Geoff, Dan Lavry, me; and now, apparently, Daniel Weiss... were you going to do this by providing logical arguments, or simply by saying: "i declare it, therefore it must be so".astroman wrote:well, I feel ok - since I was supposed to convince Rupert Neve and some other 'big boys'. Much appreciated style
I can start you a new one, if you like - just needs a 3 letter word and some insights about marketing and what you pay for a square inch of mag space, make it a new topic and ...Neil wrote:...OK - well, I missed that missive, link me to it if it's still on the web somewhere... I'd be interested to see it...
at my command unleash hell
cheers, Tom
Dunno where you're coming from... Unleashing hell is not an audio-related term I recognize.
Neil
strange that you've added Mr. Weiss, as there's nothing at all in that interview that I wouldn't agree to - it would be a stupid attempt to convince him about his own conviction 
maybe there's a missunderstanding from that early post you answered so gently.
I have frequently 'posted against' the assumption that the reason for 88.2 to be less demanding (and easier to implement) is it's an even multiply of 44.1. Since the main processing is in the high order filter (also reflected in the interview you quoted), that's bare nonsense - still it's repeated again and again every couple of month.
This is surely not my cup of math, but the load increase you report does in no way corelate to a 'slightly higher demand of resources'.
Imho it reads like someone cheated at the implementation, even more as in reviews 'sequencer builtin' sampleraterate converters were more or less all considered crap - but that could of course be one of those marketing tricks, too.
I dunno because I cannot compare.
There is no question at all that in a frequency range between 64 and 96 khz you can run a system almost alias-free and thus gain clarity - but it's effective only if the sources are adequate(ly) recorded - and processed.
You certainly know much better than me that there's a ton of things you can do wrong that have a much higher impact on sound quality than the samplerate.
Btw increased transparancy isn't always an advantage - I once bought the Curtis Mayfield 'Superfly' soundtrack on CD (one of those nice price things).
Remastered crisp as hell - someone had mixed the funk out of it entirely.
On the original topic I'm as shocked as GaryB that the system doesn't sync at 88.2 with wordclock. The original manual says 'any rate', they may have modified it...
cheers, Tom

maybe there's a missunderstanding from that early post you answered so gently.
I have frequently 'posted against' the assumption that the reason for 88.2 to be less demanding (and easier to implement) is it's an even multiply of 44.1. Since the main processing is in the high order filter (also reflected in the interview you quoted), that's bare nonsense - still it's repeated again and again every couple of month.
This is surely not my cup of math, but the load increase you report does in no way corelate to a 'slightly higher demand of resources'.
Imho it reads like someone cheated at the implementation, even more as in reviews 'sequencer builtin' sampleraterate converters were more or less all considered crap - but that could of course be one of those marketing tricks, too.
I dunno because I cannot compare.
There is no question at all that in a frequency range between 64 and 96 khz you can run a system almost alias-free and thus gain clarity - but it's effective only if the sources are adequate(ly) recorded - and processed.
You certainly know much better than me that there's a ton of things you can do wrong that have a much higher impact on sound quality than the samplerate.
Btw increased transparancy isn't always an advantage - I once bought the Curtis Mayfield 'Superfly' soundtrack on CD (one of those nice price things).
Remastered crisp as hell - someone had mixed the funk out of it entirely.
On the original topic I'm as shocked as GaryB that the system doesn't sync at 88.2 with wordclock. The original manual says 'any rate', they may have modified it...
cheers, Tom