# Sample Rate and Rendering Virtual Instruments



## sumskilz (Nov 10, 2017)

It seems most sample libraries are based on 44.1kHz samples. So does that mean when you render the output of a virtual instrument in a 96kHz session, a sample rate conversion takes place in the DAW or does it take place within the architecture of the plugin? It's my understanding that the typical conversion algorithms in a DAW aren't nearly as good as Voxengo or Izotope's algorithms. Obviously the samples themselves won't contain any information above 22.05kHz, but I'm curious if better audio quality could be achieved by rendering at 44.1kHz and then converting to 96kHz using a better algorithm than what is stock in the DAW. Anyone know if that's the case?


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## JohnG (Nov 10, 2017)

This is an interesting question, undoubtedly. I talked about it with Nick Phoenix, who uses 44.1 because that's how the samples are recorded. I run at 48k since every project of mine delivers either at 48k or 96k, but I have long wondered whether I'm putting a lot of unnecessary strain on the PCs I use to serve up samples (and the DAW as well).

I never heard / read anything that definitively says whether that conversion is a meaningful strain or not.

One thing -- the "conversion" that you're alluding to is not such a big deal because it all remains digital -- ones and zeroes. If it were, instead, taking place at the analogue-digital or digital-analogue level, that would indeed be a very big issue, but no D/A-A/D conversion happens when altering the sample rate.

So, not to minimise your concern, but it's not quite as critical an issue, I suspect, as it would be if one were moving back and forth between digital and analogue.


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## storyteller (Nov 10, 2017)

sumskilz said:


> It seems most sample libraries are based on 44.1kHz samples.


I think it is probably more fair to say that most sample libraries are 24/48k (and probably recorded at 24/96k). Many of the “light” versions (e.g. Gold vs Diamond) or libraries targeted at a prosumer vs a professional offer a reduced bitrate/samplerate as a way to bring in more customers at a lower price point. Some will argue they can’t hear a difference... and that might very well be the case for those people... but there is a very audible difference that trained ears can tell. But that is why so many people have 44.1k libraries... it is a great trade off financially and to lower cpu/ram resource consumption. 

44.1k was also a design choice by some designers early on when the cd was the distribution medium of choice. The logic was to record at 88.2k, then cut that sample size in half to 44.1k so a DAW would not have to alter the soundsource when the track was rendered for distribution. This is also why 16bit was chosen in these libraries. Again, the idea was to record at a higher resolution and then distribute samples to customers in the way that would not lose quality at final render. It is flawed logic in the higher sample rate sessions of today, but made sense back then.

Also.... many rhythmic and hit sample libraries “resampled” content from CDs... therefore their source was only as good as a CD’s resolution (16bit 44.1k).

For me, the best way I can describe it is a lack of depth, or a flatness to the sound. I have always said that I’d rather not work on a 16/44.1k session if that is my only option. That may sound elitist or snobby, but “lifeless” is the only way I can describe a 16/44.1k session compared to the depth of 24/48k. What about 24/44.1k? I dislike it, but it is somewhere between lifeless and alive. It is somewhere on life-support. 

A way that is very obvious to learn to detect higher bitrate/samplerates is in decay tails. The subtle natural decay of a sample suffers at lower bitrates. Another way is if you think a sample library sounds lifeless due to a potential low round robin count, you may be surprised to find out you are subconsciously hearing the “lifeless” bitrate/samplerate compared to a higher resolution version. You might truly be hearing low round robin counts... but this is another place you can check and see what it is you are actually hearing.

In a higher samplerate/bitrate session, lower resolution samples are scaled up. Mathematically, there isn’t a difference (at least not noticeable) in their sampled rate versus their scaled rate. In the case of Kontakt and UVI, I think those apps convert the signals internally before they flow into the DAW... I might be wrong on that one though. 



sumskilz said:


> Obviously the samples themselves won't contain any information above 22.05kHz


Yes... and no. @charlieclouser had a great post on sampling at 96k and capturing every sonic possibility of a sound, then slowing the sound down. @Nick Batzdorf also has written about this from time to time. When a sample is slowed down and morphed, the sounds above 20k are audible in the resampled soundsource. So it is a little more complex than an analytical/absolute approach to audio math. For example, the human brain “hears” inaudible frequencies that would be part of a sound in real life because the information in the sound contains enough information to relay the inaudible sound. Lower bitrate/samplerate loses those “pointers” if you will. Some may argue against this... again due to analytical 1s and 0s - but it is true. A trained audio engineer can detect upwards of 40k... not because they hear it, but because their brain knows it is present or absent. This is part of the depth analog gear provided and which has been lost in the digital age. Footnote: I’m in my mid thirties and learned from analog engineers transitioning into the digital age with everyone else. 

The way the brain can hear sounds the ear cannot is the reason why engineers can create great mixes when their hearing has a reduced range with age and tinnitus. If they truly know what they are listening for, the brain knows how to rebuild what the ear cannot hear.



sumskilz said:


> I'm curious if better audio quality could be achieved by rendering at 44.1kHz and then converting to 96kHz using a better algorithm than what is stock in the DAW. Anyone know if that's the case?


I know I’ve written a lot so far, and maybe it hasn’t answered your question directly, but hopefully it indirectly has already answered this question. 96 to 48 is a multiple of 2 so that math is pretty simple. 44.1k is where the math is not quite as easy, but there really is only one equation/solution... so companies won’t have differences. I would argue that upscaling is probably best handled in-engine while using a higher bitrate session. If you mix/render in 44.1 and add reverb/fx/etc, then it will be severely compromised bringing it into a 24/96k session. Finally, there is a concept of IDR in mastering plugins that downsamples while limiting/expanding. There is a reason, and those have a different sound. But scaling up? Nothing is going to add to what is absent in the first place. 

Hope this helps! I know it was long winded, but there aren’t a lot of discussions where this is clearly explained in one place. Hopefully others will chime in and fill in any holes I missed!


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## sumskilz (Nov 10, 2017)

JohnG said:


> One thing -- the "conversion" that you're alluding to is not such a big deal because it all remains digital -- ones and zeroes. If it were, instead, taking place at the analogue-digital or digital-analogue level, that would indeed be a very big issue, but no D/A-A/D conversion happens when altering the sample rate.
> 
> So, not to minimise your concern, but it's not quite as critical an issue, I suspect, as it would be if one were moving back and forth between digital and analogue.


It seems like it shouldn't be a big deal at least in increasing sample rate, but while I haven't done an A-B test with VI renders at different sample rates yet, I have tested two different conversion algorithms in Cubase 8.5 and one was pretty bad, a very noticeable degradation. The other sounded okay, but I haven't tested to see if maybe there are cumulative negative effects when several tracks are up sampled. I've been told that the algorithm in Izotope RX 6 is completely transparent, but I don't own it to test it.

@storyteller

Interesting post, thanks. I hear some of the differences you hear. One of my main reasons for working at a higher sample rate is also avoiding aliasing fold back on saturation plugins. There are some otherwise great sounding plugins that don't have oversampling to deal with this. I haven't noticed anything negative about just rendering VIs in a higher sample rate session, but my question just comes from wondering if it's possible there's a better way.


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## JohnG (Nov 10, 2017)

sumskilz said:


> I have tested two different conversion algorithms in Cubase 8.5 and one was pretty bad, a very noticeable degradation



I am intrigued -- Cubase offers the user more than one choice of how to convert samples that were recorded at a rate different from that of the overall session? And this is the _digital_ conversion, not analogue? 

Even VE Pro doesn't do that.


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## Nick Batzdorf (Nov 10, 2017)

sumskilz said:


> It's my understanding that the typical conversion algorithms in a DAW aren't nearly as good as Voxengo or Izotope's algorithms. Obviously the samples themselves won't contain any information above 22.05kHz, but I'm curious if better audio quality could be achieved by rendering at 44.1kHz and then converting to 96kHz using a better algorithm than what is stock in the DAW. Anyone know if that's the case?



It's possible you could preserve more by storing the samples as 32-bit floats and then using Izotope.

My personal opinion: the original recording quality of the samples certainly makes a difference, but by the time you're using a sample library in a sequence... I mean, this is *so* far down the list of things to be concerned with that it's almost a medical condition. 

These are samples, not live acoustic instruments in an audiophile recording. Never mind that some individual samples are often tweaked while the library is being developed, we're combining libraries, running things through fake spaces, compressing and processing the sounds... really, it just doesn't matter.

To put my opinion in perspective, I use tweak audio cables, have my room set up pretty seriously for monitoring, and care a lot about audio subtleties!



storyteller said:


> A trained audio engineer can detect upwards of 40k... not because they hear it, but because their brain knows it is present or absent.



Other people, notably Rupert Neve, have said that. I believe we do sometimes hear - or maybe sense - things that by all rights we shouldn't be aware of, but that particular one... well, it's my religion not to tell anyone they don't hear something they say they hear, but that claim does raise one of my eyebrows and lower the other one.


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## Nick Batzdorf (Nov 10, 2017)

Whether to work at 96kHz is another question. I don't, because I don't want to have to use twice as many computer systems.


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## sumskilz (Nov 10, 2017)

JohnG said:


> I am intrigued -- Cubase offers the user more than one choice of how to convert samples that were recorded at a rate different from that of the overall session? And this is the _digital_ conversion, not analogue?
> 
> Even VE Pro doesn't do that.


It's digital conversion, but it doesn't seem very useful, because one is obviously significantly superior to the others. I'm actually sort of confused by its existence as a feature. There is "resample" and then there is sample rate conversion in "musical mode" which seems to be based on whatever automatic time-stretching algorithm is currently set to default. Some emphasize accuracy in pitch, some in timing. If you change your session's sample rate while you have audio files in "musical mode", it offers you this option, but compared to "resample", they all sound bad, and by "bad", I mean grainy and noticeably unlike the original. It literally asks "Do you want to convert the sample rate of the clips in musical mode?" Why that would be the default in any circumstance, I don't know.

Here's comparison of the accuracy of different conversion algorithms: http://src.infinitewave.ca/

They evidently vary a lot in quality. Although that's obviously the least simple conversion commonly made (96 down to 44.1) being used as a benchmark.



Nick Batzdorf said:


> Other people, notably Rupert Neve, have said that. I believe we do sometimes hear - or maybe sense - things that by all rights we shouldn't be aware of, but that particular one... well, it's my religion not to tell anyone they don't hear something they say they hear, but that claim does raise one of my eyebrows and lower the other one.


I know that I don't consciously hear anything above 18kHz anymore, but to me the higher sample rates seem to have a clearer stereo image and a more 3D quality.


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## JohnG (Nov 10, 2017)

Sounds like that applies to sound bites in the sequence, but not too clear on whether it also affects the samples themselves -- I guess you're saying it does.

I don't hear anything bad at all running at 48k. Neither do the people I write for, so I am going to forget about it until further notice!


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## babylonwaves (Nov 12, 2017)

Don’t forget that you’re talking about _sampling_ libraries and those are no necessarily sampled chromatically. In fact, many aren’t. So in effect when you think of true „fidelity“ pitched samples have a much more drastic influence on the timbre and quality. On top there are round robin algorithms which borrow from zones in the neighborhood. And that’s real time pitch shifting again.


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