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Posted: Wed Aug 08, 2007 12:53 pm
by Funkstar De Luxe
Amberience wrote:Erm.... maybe I'm an idiot... but why not just work at the same sample rate throughout the whole process, ie: 44.1khz.
Because higher sample rates produce better results when processing. Also, you can't hand in 44.1khz to be mastered - not in a professional environment anyway. And working in real-time at anything higher than 44.1 is very hardware intensive.
So for best results, you work at 44.1, render at a highest rate possible (actually 96k is probably high enough, but there's no reason not to go higher)
Posted: Wed Aug 08, 2007 1:10 pm
by Angstrom
I'm not a mastering engineer, but practically speaking there's no hard reason to deliver a file at a higher sample rate
than you worked at.
bit depth is one thing - sample rate is quite another.
upsampling in your DAW is very very differnt from working at that high sample rate all along.
Many synths / effects will calculate differently at the higher rate and so although in theory you might get a better EQ calculation, in practice you will probably find a mysterious synth lfo is now faster due to bad programming(for example) and some other buried effect has inevitably thrown a wobbler.
That's not to mention the trickiness in interpolating the samples. Remember - you are speeding the sample up but you are asking it to play at the same pitch .. so you are timestretching . This is NOT going to give you better sound.
I would never upsample before handing over to a mastering engineer - I would deliver a render at the samplerate I worked at.
here's the word from the god of mastering
bob katz (who else!) wrote:
My current recommendations are for you to work at the highest possible sample rate and longest practical wordlength available to you (typically 96 kHz. I'm not yet convinced that higher rates than 96K offer a real difference). However, if you are mixing digitally, do not sample rate convert, to avoid additional degrading DSP. In other words, if you are mixing digitally, remain at the same sample rate as the multitrack; we want to see the earliest-generation file possible.
Posted: Wed Aug 08, 2007 1:15 pm
by b0unce
if you're working with 44.1 audio...I think that's right.
but what about working with virtual instruments ?
isnt that where working at 44.1 & rendering audio to 88.2 (for example) is practical
Posted: Wed Aug 08, 2007 1:31 pm
by Funkstar De Luxe
Angstrom wrote:I'm not a mastering engineer, but practically speaking there's no hard reason to deliver a file at a higher sample rate
than you worked at.
bit depth is one thing - sample rate is quite another.
upsampling in your DAW is very very differnt from working at that high sample rate all along.
Many synths / effects will calculate differently at the higher rate and so although in theory you might get a better EQ calculation, in practice you will probably find a mysterious synth lfo is now faster due to bad programming(for example) and some other buried effect has inevitably thrown a wobbler.
That's not to mention the trickiness in interpolating the samples. Remember - you are speeding the sample up but you are asking it to play at the same pitch .. so you are timestretching . This is NOT going to give you better sound.
I would never upsample before handing over to a mastering engineer - I would deliver a render at the samplerate I worked at.
here's the word from the god of mastering
bob katz (who else!) wrote:
My current recommendations are for you to work at the highest possible sample rate and longest practical wordlength available to you (typically 96 kHz. I'm not yet convinced that higher rates than 96K offer a real difference). However, if you are mixing digitally, do not sample rate convert, to avoid additional degrading DSP. In other words, if you are mixing digitally, remain at the same sample rate as the multitrack; we want to see the earliest-generation file possible.
True, but for my purposes (ie VST instruments) rendering at higher rates does give truer sound. And if you need to compensate for bad programming, then you're using the wrong instruments. I don't see why there's such an argument.
However, you seem to have fucked up on your assumption that Ableton was UPSAMPLING. If you recorded something at 96k and are playing it back at 44.1 in Ableton it will sound BAD. Downsampling.
HERE IS WHAT ABLETON DOES POORLY. EVIDENCE PAGE ONE. HERE IS A FREE TOOL TO HELP YOU. LINK PAGE ONE.
No, fuck it. Ableton does everything perfectly and is the best appliaction in the world. Ever. I am incorrect, the graphs I posted are incorrect, VST programmers are wrong, 44.1 is the magic number for processing, all professional studios use 44.1. My apologies.
I'm out. Bye.
[edit] this is tongue in cheek - but you get my point

Posted: Wed Aug 08, 2007 1:34 pm
by Angstrom
IMO 88.2 would be the sensible frequency yes.
And yes synths
should render better at those higher frequencies due to reasons I can't be bothered to type. basically the calculations are done better.
but - remember that many VST programmers can forget to make processes sampling rate independent. That means when the sampling rate goes up the sound goes crazy!
So that means anything from LFO speeds, to filter calculations, to in extreme cases oscillators (!) This sometimes happens with the 'big name' vst instruments as well as the freebie ones. These sampling rate dependency errors are more common than you might think.
anyway - if you have a load of vsti running at a higher sample rate to file - check the 88khz mix closely to make sure there isn't a crazy high speed LFO, or an odd sounding distortion, or a slightly more open filter buried in the resulting mix somewhere . You don't want to get the master done and notice it then!
essentially you would gain from the VSTi running at a higher rate - but you also have to make sure the output is what you actually want. Many sounds become worse when they are calculated 'better'
Alongside that VSTi calculation improvement though, you have the downside of any samples being affected adversely.
Until I can work at 88 all the time and my samples are there too, I am sticking to a 24bit 44.1 workflow.
Posted: Wed Aug 08, 2007 1:38 pm
by Funkstar De Luxe
Angstrom wrote:IMO 88.2 would be the sensible frequency yes.
And yes synths
should render better at those higher frequencies due to reasons I can't be bothered to type. basically the calculations are done better.
but - remember that many VST programmers, even of paid vst, can forget to make certain processes sampling rate independent. So that means anything from LFO speeds, to filter calculations, to in extreme cases oscillators (!)
These sampling rate dependency errors are more common than you might think.
anyway - if you have a load of vsti running at a higher sample rate to file - check the 88khz mix closely to make sure there isn't a crazy high speed LFO, or an odd sounding distortion, or a slightly more open filter buried in the resulting mix somewhere . You don't want to get the master done and notice it then!
essentially you would gain from the VSTi running at a higher rate - but you also have to make sure the output is what you actually want. Many sounds become worse when they are calculated 'better'
Alongside that VSTi calculation improvement though, you have the downside of any samples being affected adversely.
Until I can work at 88 all the time and my samples are there too, I am sticking to a 24bit 44.1 workflow.
Actually, despite what may appear to be a mathematical problem, the difference between upsampling from 44.1 to 88.2 is no more difficult that from 44.1 to 96. It's all to do with aliasing filters, not really raw number crunching.
Posted: Wed Aug 08, 2007 1:39 pm
by Funkstar De Luxe
Angstrom wrote:
Alongside that VSTi calculation improvement though, you have the downside of any samples being affected adversely.
Yes, this my whole point of this topic. You absolutely need to make sure your samples are recorded at your render rate. Up or down sampling in Ableton sucks.
Posted: Wed Aug 08, 2007 1:40 pm
by ChiDJ
Posted: Wed Aug 08, 2007 1:43 pm
by Angstrom
Funkstar De Luxe wrote:
No, fuck it. Ableton does everything perfectly and is the best appliaction in the world. Ever. I am incorrect, the graphs I posted are incorrect, VST programmers are wrong, 44.1 is the magic number for processing, all professional studios use 44.1. My apologies.
I'm out. Bye.
[edit] this is tongue in cheek - but you get my point

what the hell are you talking about.
where , anywhere in this thread does anyone say that.
specifically me- who you have quoted.
I have said
1: lots of daws have this problem with upsampling and downsampling
2: live also has this problem, but amazingly not as bad as some other daws, which ought to have performed better.
3: be careful upsampling on render - it may give you better sound in some areas, but worse in others
4: upsampling vsti nees to be checked carefully, many popluar vsti have issues.
now you seems to be seriously mis-representing what I have said, or willfully mis-understanding it.
Posted: Wed Aug 08, 2007 1:49 pm
by Funkstar De Luxe
Angstrom wrote:Funkstar De Luxe wrote:
No, fuck it. Ableton does everything perfectly and is the best appliaction in the world. Ever. I am incorrect, the graphs I posted are incorrect, VST programmers are wrong, 44.1 is the magic number for processing, all professional studios use 44.1. My apologies.
I'm out. Bye.
[edit] this is tongue in cheek - but you get my point

what the hell are you talking about.
where , anywhere in this thread does anyone say that.
specifically me- who you have quoted.
I have said
1: lots of daws have this problem with upsampling and downsampling
2: live also has this problem, but amazingly not as bad as some other daws, which ought to have performed better.
3: be careful upsampling on render - it may give you better sound in some areas, but worse in others
4: upsampling vsti nees to be checked carefully, many popluar vsti have issues.
now you seems to be seriously mis-representing what I have said, or willfully mis-understanding it.
People in this forum appear to be completely undisciplined in any form of audio theory. It's more of an Ableton fan club.
Guys, do what you want.
Posted: Wed Aug 08, 2007 1:55 pm
by Angstrom
hey,
well refuted there upsampling boy. I haven't said anything you haven't said . Al I have said that was even vaguely fan-boy ish is repeated here
Funkstar De Luxe wrote:Also, I am not slating Ableton, most DAWs are totally bad at resampling. This is why it should be done by a dedicated application. As far as native DAW resampling goes, Ableton is about mid table. You can see more test here
http://src.infinitewave.ca/
so, I really fail to see your point
firstly, as you know so much I'm quite keen to know why interpolation isn't more of an issue in a 44.1 upsample to 96 than in a 44.1 -> 88.2. I would say that 44.1 -> 96 is a much more difficult upsample to get right, you say it isn't ?
I'd be interested to read that, as a fan club member learning from a DSP master such as yourself. please explain the simplicity of calculating that interpolation without transient blurring.
Posted: Wed Aug 08, 2007 2:07 pm
by wilxon
WHO CARES?
CAN YOU HEAR THE DIFFERENCE IF YOU ONLY RESAMPLE ONCE
I Think the key is to not resample the same file over and over again so that the effects are not audible later on.
Posted: Wed Aug 08, 2007 2:18 pm
by Tarekith
Funkstar De Luxe wrote:People in this forum appear to be completely undisciplined in any form of audio theory.
Amen, there's a distinct lack of understanding about basic audio engineering and software implementation concepts among many people here. The whole "if you can't hear a difference it must be fine" argument is all well and good when you're only comparing audio that has been processed once. But this stuff is all cumulative in the digital world, and definitely can be audible later in the production process.
Posted: Wed Aug 08, 2007 2:30 pm
by Angstrom
yes, but he addressed that comment at me.
Posted: Wed Aug 08, 2007 2:34 pm
by wilxon
Tarekith wrote:Funkstar De Luxe wrote:People in this forum appear to be completely undisciplined in any form of audio theory.
Amen, there's a distinct lack of understanding about basic audio engineering and software implementation concepts among many people here. The whole "if you can't hear a difference it must be fine" argument is all well and good when you're only comparing audio that has been processed once. But this stuff is all cumulative in the digital world, and definitely can be audible later in the production process.
I hear what your saying - but i just wonder why?
Why do you need to re-process a signal by re-sampling time and time again - to the point that it makes a small difference?
I wherever i can i always try to leave things in an organic state as possible untill i finish the mixdown.
The only 2 point that i will use the resample feature is when making dance music i like the blend of a few sounds and ill record it - or when i use the render feature when mixing down.
Even when im in an SSL studio - i record using minimal subtractive EQ and then everything else i try to leave until i mix it down.