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11-03-2015, 06:31 PM
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#201
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Human being with feelings
Join Date: Aug 2013
Location: Bowral, Australia
Posts: 1,643
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A bit off topic perhaps, but if your looking for examples of the effects of aliasing, you really can't go past the highly amusing VSTi, 'Mr Alias 2', which can be found here: http://www.thepiz.org/mralias2/pro.php
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11-03-2015, 07:29 PM
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#202
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Human being with feelings
Join Date: Aug 2011
Location: Near a big lake
Posts: 3,943
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Quote:
Originally Posted by pipelineaudio
So is it possible my video is showing two different things? The bend thingy could be IMD, and the aliasing shown by the sweep is a separate issue?
I wonder what a good test to tell for sure would be
That's what I get for being sure about something
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The bendy thing is normal, for "real" analog guitar amps and distortion pedals etc., so unless you can compare the two and see what differences there are, I wouldn't even worry about it. (As far as I can hear anyway, the guitar amp sims are doing what they're supposed to and nothing unusual.)
About the compressor: if I had hardware compressors here to test, that would be interesting. I don't though. I only use compressors ITB. Maybe compare other compressor plugins with similar basic settings?
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11-03-2015, 09:21 PM
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#203
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Mortal
Join Date: Jan 2006
Location: Wickenburg, Arizona
Posts: 14,051
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I did a bunch of DSP compressors, they all do some pretty serious aliasing
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11-04-2015, 12:50 AM
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#204
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Human being with feelings
Join Date: Aug 2011
Location: Near a big lake
Posts: 3,943
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Quote:
Originally Posted by pipelineaudio
I did a bunch of DSP compressors, they all do some pretty serious aliasing
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Do you have any hardware compressors to compare? I've never tried something like this with one to see what might happen.
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11-04-2015, 06:17 AM
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#205
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Human being with feelings
Join Date: Mar 2014
Location: Louisville, KY, USA
Posts: 1,075
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I was speculating that analog hardware compressors should not cause aliasing problems because the ADC will filter out any ultrasonic frequencies that are generated, if any.
Also, hardware units have noise ratings measured as THD (total harmonic distortion) which may only refer to the measurement method. But, the term implies that the distortion will be harmonic. Thus describing the saturation that all emulating plug-ins are attempting to mimic. And, which is considered to be non-offensive unlike aliasing.
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11-04-2015, 08:04 AM
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#206
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Human being with feelings
Join Date: Apr 2010
Location: Cloud 37
Posts: 1,071
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Yes, aliasing is unique to digital, as I guess there's no sample-rate with analog. Though I don't see how (digital) hardware would be any different in regards to aliasing, unless it simply ran at a higher sample-rate.
I think digital in which aliasing has been eliminated would be superior to analog, as there's less overall distortion (if I want distortion, I'll add it.) Of course that's more a matter of opinion. I'm into super 'clean' sounding audio. E.G. I'd like my drums to sound the same as when I'm standing in a room and listening to them (I know, unattainable).
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02-14-2016, 02:10 PM
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#207
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Human being with feelings
Join Date: Nov 2015
Posts: 1,566
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I'd like to start working at 88.2 project rate (ITB stuff with soft synths), I noticed on quick testing that a few of my plugins sound much better than at 48, I wasn't really expecting such a massive difference in reverb soundstage for example.
However not all plugins seemed to handle 88.2 properly, so I wonder if there's any sort of tester program or a standard test procedure for evaluating plugin behaviour?
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02-15-2016, 01:47 AM
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#208
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Human being with feelings
Join Date: Aug 2014
Posts: 11,052
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Quote:
Originally Posted by Mr. PC
Yes, aliasing is unique to digital, as I guess there's no sample-rate with analog. Though I don't see how (digital) hardware would be any different in regards to aliasing, unless it simply ran at a higher sample-rate.
I think digital in which aliasing has been eliminated would be superior to analog, as there's less overall distortion (if I want distortion, I'll add it.) Of course that's more a matter of opinion. I'm into super 'clean' sounding audio. E.G. I'd like my drums to sound the same as when I'm standing in a room and listening to them (I know, unattainable).
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The human auditory system, as well as the gases we tend to hear pressure waves propagated through, are far from being a "clean" system. Especially the pumping compression given to something as loud as a drum kit in a small room!
There is often a huge disconnect between what we think things sound like and what they actually do. Not many people record drum kits using a binaural dummy 10' away... mainly because a blend of different mic positions sounds better, and "cleaner" if that's what you're shooting for.
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02-15-2016, 04:50 AM
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#209
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Banned
Join Date: Jun 2015
Location: Lower Rhine Area, DE
Posts: 964
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point is, that we dont hear with our ears. we hear with our brains.
the sound waves are coming in into the ear and are interpreted in the brain. emphasis on "interpreted". and this interpretation depends on lots of factors, that a music producer cant take into account. nearly none of them.
so I am not posting again the Monty Montgomery video and the Lawo-papers and Ethan Winer explanation and so on and on and on but I am saying it again: there is no point to go beyond 24bit/44.1kHz. that is proven time after time after time but there are some, who still dont know or dont want to know or to be the kickass-wiseasses at every cost. ah, yes, and there is the snakeoil-industry...
everything else is lying into the own pocket or a bias towards "bigger numbers are better".
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02-15-2016, 08:10 AM
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#210
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Human being with feelings
Join Date: Aug 2014
Posts: 11,052
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Quote:
Originally Posted by LightOfDay
point is, that we dont hear with our ears. we hear with our brains.
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This is true.
This is why, when people here a drum kit in a room these days, it will tend sound more like drums on a record than what the drums actually sound like in the room. Expectation is a huge part of how we perceive sound.
Quote:
Originally Posted by LightOfDay
the sound waves are coming in into the ear and are interpreted in the brain. emphasis on "interpreted". and this interpretation depends on lots of factors, that a music producer cant take into account. nearly none of them.
so I am not posting again the Monty Montgomery video and the Lawo-papers and Ethan Winer explanation and so on and on and on but I am saying it again: there is no point to go beyond 24bit/44.1kHz. that is proven time after time after time but there are some, who still dont know or dont want to know or to be the kickass-wiseasses at every cost. ah, yes, and there is the snakeoil-industry...
everything else is lying into the own pocket or a bias towards "bigger numbers are better".
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This is just you getting religious again
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09-13-2022, 10:59 AM
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#211
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Human being with feelings
Join Date: Jun 2006
Posts: 22,572
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I see a lot of replies on this thread, and a lot of discussion -
but I have a few questions, this is my current workflow:
record in 44.1/24 bit
switch audio device to dummy audio at 96k (there isnt much a point in switching to my interface ((ff800)) at that level I think to listen, because I can't really play the project back... maybe I could adjust some latency or other things to get this to work better, but for now I just switch to dummy audio
and render
then I take that file and insert it into a "master" project with all of the songs from the "Album."
I listen to that in 96k and render again - to 96k. I could also spit out 44.1 versions, but I figure this is the way to get the highest possible quality. Not for my recorded audio, but for the plugins being used, like ampsims, limiters, eq;s, etc...
does this seem worthwhile to you?
I will eventually upgrade my PC and do everything in 96k - but I also wonder how worth it that is for recording, etc.
I think for timestretching and other things it could be beneficial to have recordings at a high sample rate, but I really would love to read some objective shit on this.
thank you!
made a thread here where im asking some questions and wondering if my workflow is causing an issue:
https://forum.cockos.com/showthread....82#post2595382
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09-13-2022, 04:56 PM
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#212
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Human being with feelings
Join Date: Jul 2010
Location: Silicon Valley, CA
Posts: 2,787
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Some plug-ins may work better at higher sample rates but theoretically it should make no difference since you can't add (usable) information.
If up-sampling can make a plug-in work better the programmer could make the plug-in up-sample, do the DSP, and then down-sample back the original without you ever knowing about it.
When you play it back the DAC makes a continuous analog signal which is similar to "infinite" up-sampling.
The exception would be an "exciter" effect that adds new high frequencies. But that would only be helpful if the original audio is 8 or 12kHz, or something that doesn't cover the full audio range. 44.1kHz already handles the full audio range so it wouldn't help to up-sample to a format that goes into the ultrasonic range.
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09-13-2022, 10:58 PM
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#213
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Human being with feelings
Join Date: Aug 2014
Posts: 11,052
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Quote:
Originally Posted by Jae.Thomas
I see a lot of replies on this thread, and a lot of discussion -
but I have a few questions, this is my current workflow:
record in 44.1/24 bit
switch audio device to dummy audio at 96k (there isnt much a point in switching to my interface ((ff800)) at that level I think to listen, because I can't really play the project back... maybe I could adjust some latency or other things to get this to work better, but for now I just switch to dummy audio
and render
then I take that file and insert it into a "master" project with all of the songs from the "Album."
I listen to that in 96k and render again - to 96k. I could also spit out 44.1 versions, but I figure this is the way to get the highest possible quality. Not for my recorded audio, but for the plugins being used, like ampsims, limiters, eq;s, etc...
does this seem worthwhile to you?
I will eventually upgrade my PC and do everything in 96k - but I also wonder how worth it that is for recording, etc.
I think for timestretching and other things it could be beneficial to have recordings at a high sample rate, but I really would love to read some objective shit on this.
thank you!
made a thread here where im asking some questions and wondering if my workflow is causing an issue:
https://forum.cockos.com/showthread....82#post2595382
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You might be avoiding some aliasing. That would be the only advantage to doing this. Only way to tell would be to listen to a 96kHz and 44.1kHz version.
You're not doing any harm though, especially if you are resampling with the R8brain algorithm when you render.
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