Sample Rates (44.1 v 48 v 96 )etc etc

Synapsis

The Mepsyah's Dipsyples
For all those people who compose in the box, using vsti, and vst plugs,,,,

Sample rate does matter.

if you record in 44.1... and use in the box plugs, you will incur a few things...

1. Nyquist filter... if you don't know what that is... research.. and find out...

2. Sample interpolation.... if you don't know what that is... research...

3. Possible sharp high end bite digital distortion...which is not pleasant at all...



The higher sample rate you record in.. the smoother, and less raspy/noisy your audio will be...

The less interpolation and messy the samples will be (layering).. it will also boost the information your vsti puts out..

The nyquist filter will be far less invasive on your high end...


96 khz, to a trained ear, will sound far more HD than a 48 khz... if you are using in the box synths...


This is not the case like I heard a chap post on here about opening one of your favorite tracks in FL studio and setting the sample rate to 96khz, and it sounding high rez... this is about the resolution of synths, and how plugins deal with high density information.. which gives a far more detailed and rich end product...


96khz can tax your cpu... but if you can... use it... and dither down to 44.1 16 bit at the very end... It won't help a shit mix.. but it will make a well mixed production shine as best as it can :)
 
For all those people who compose in the box, using vsti, and vst plugs,,,,

Sample rate does matter.

if you record in 44.1... and use in the box plugs, you will incur a few things...

1. Nyquist filter... if you don't know what that is... research.. and find out...

2. Sample interpolation.... if you don't know what that is... research...

3. Possible sharp high end bite digital distortion...which is not pleasant at all...



The higher sample rate you record in.. the smoother, and less raspy/noisy your audio will be...

The less interpolation and messy the samples will be (layering).. it will also boost the information your vsti puts out..

The nyquist filter will be far less invasive on your high end...


96 khz, to a trained ear, will sound far more HD than a 48 khz... if you are using in the box synths...


This is not the case like I heard a chap post on here about opening one of your favorite tracks in FL studio and setting the sample rate to 96khz, and it sounding high rez... this is about the resolution of synths, and how plugins deal with high density information.. which gives a far more detailed and rich end product...


96khz can tax your cpu... but if you can... use it... and dither down to 44.1 16 bit at the very end... It won't help a shit mix.. but it will make a well mixed production shine as best as it can :)

"dither down to 44.1 16 bit" means nothing.
You're right when advocating for producing at 2 x FS. But, prefering 24 bit to 16 bit isn't a detail. The best production format is 24 bit @ 96 KHz if you have enough CPU, RAM and storage.
A sample rate conversion doesn't require any dithering which is only usefull when lowering the resolution from 24 bits to 16 bits.
My two cents ;-)
 
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If you don't dither when you change bit depth.. it chops all the audio off that didn't make the 'cut' from 24 to 16.. obviously as you know, dithering adds noise to push all the audible information into the 16 bit realm.. so no nasty chopped off sounds from the silence get artifacted... hahaha

I was using the 44.1 16 bit as the stock format for cd... I always dither to 44.1 16... and then any conversions after that (say mp3) have had the bits pushed up into the 16 bit range... anyway :)
 
The accuracy of the time domain is critical for resonance/sampling quality. You need to record into the right DAW software + version using the right gear at max sample rate. When creating synth music you have to research what gear is behind the sample packages and also ensure good end to end tuning. Cheap shortcuts do not exist. It is about getting the gear you need.
 
The accuracy of the time domain is critical for resonance/sampling quality. You need to record into the right DAW software + version using the right gear at max sample rate. When creating synth music you have to research what gear is behind the sample packages and also ensure good end to end tuning. Cheap shortcuts do not exist. It is about getting the gear you need.

Samplers and Synth's are different though.. some are romplers, or hybrid sample/synthesis.. But the good ones are getting really good....

Synths now are coded entirely in the box... we learned max msp a bit in uni.. it is an absolute headfuck (pardon the language)... even max msp had to be coded... how the hell they do it I do not know...

Synth's that run on a daw at 44.1, 16, compared to 96 or above 24 bit, sound quite a lot richer, smoother, less raspy... a sampler like kontakt, using 44.1 16 bit samples, in a daw set to 96 24, sounds to me, sounds the opposite, unless they have up sampled them...

though ITB VST, interacts far better with high sample rates for audio quality, but chews cpu...

Maybe I should do a demo and see if people can hear the difference :)
 
But.... and a big one.... But, IMD can shoot through the roof on 96k projects when mixing. So quite often 96k sounds noticeably worse than 44.1k. A lot of problems start creeping in with plugins when you run your project at 96k. You also have to wonder if it makes any sense since virtually all plugins now upsample to 96k or 192k or higher automatically. A few things to consider:

1) in a proper band limited system there is no conversion difference between 44.1k and 96k
2) With linear math, the sample rate doesn't matter
3) virtually all plugins are running at 96k or higher internally
4) lots nonlinear plugins don't properly filter the incoming 96k signal and will perform better with a 44.1k signal and simply upsampling.

In other words, the old-fasioned "more is better" doesn't apply to sample rate. It's far more complicated than that.
 
But.... and a big one.... But, IMD can shoot through the roof on 96k projects when mixing. So quite often 96k sounds noticeably worse than 44.1k. A lot of problems start creeping in with plugins when you run your project at 96k. You also have to wonder if it makes any sense since virtually all plugins now upsample to 96k or 192k or higher automatically. A few things to consider:

1) in a proper band limited system there is no conversion difference between 44.1k and 96k
2) With linear math, the sample rate doesn't matter
3) virtually all plugins are running at 96k or higher internally
4) lots nonlinear plugins don't properly filter the incoming 96k signal and will perform better with a 44.1k signal and simply upsampling.

In other words, the old-fasioned "more is better" doesn't apply to sample rate. It's far more complicated than that.

Running your software synths at higher sample rates - Gearslutz Pro Audio Community

Yea, but nah... I can hear it in the audio... vsti sound way better, and retain a lush character, even once bounced and dithered to 44.1...

But each to their own :)
 
also with the phase accuracy and alignment of the ppq in the daw are far more accurate.. as in accurate maths and joining the dots..
 
dithered to 44.1...
Please
icon_facepalm.gif
 
But.... and a big one.... But, IMD can shoot through the roof on 96k projects when mixing. So quite often 96k sounds noticeably worse than 44.1k. A lot of problems start creeping in with plugins when you run your project at 96k. You also have to wonder if it makes any sense since virtually all plugins now upsample to 96k or 192k or higher automatically. A few things to consider:

1) in a proper band limited system there is no conversion difference between 44.1k and 96k
2) With linear math, the sample rate doesn't matter
3) virtually all plugins are running at 96k or higher internally
4) lots nonlinear plugins don't properly filter the incoming 96k signal and will perform better with a 44.1k signal and simply upsampling.

In other words, the old-fasioned "more is better" doesn't apply to sample rate. It's far more complicated than that.

Yes it is very complicated and to some degree you are right, but one thing that needs your attention here is the impact of latency and its impact on the typical ITB setup. You need to know precisely what DAW software + version to combine with what audio interface and it is the highest sample rate that in general also gives the lowest latency. Part of the complexity is in that combo. If you use plugins you can account for a certain amount of inaccuracy in the figures fed to the internal delay compensation, so the only way of getting truly low latency is to both limit or minimize it at whatever that has a potential impact on it. One way of doing this is to not use any digital processing at all. If you on top of that use the right software and hardware for the recording, then it is mostly the clock accuracy + sample rate combination that is going to be a concern. Now because of the limits in the amount of available headroom you have in various audio interfaces, it becomes this tradeoff you need to solve mathematically. But from blind tests I have done I know the clock accuracy is of high importance. Various DAW software and plugins on certain platforms can also have long term latency performance drops. I know for instance there are long term latency issues in Pro Tools on the PC platform.

There are certain signal processing routes you can take to manage it with a certain amount of latency tolerance, but in general if you are really serious you skip the digital processing altogether. In this way eventhough there is a certain amount of long term latency increase natively in your particular setup as you play your mix over and over, it stays within a manageable range.

My overall recommendation is to put music first, become very hardware aware and use that awareness to improve the quality. This is to some degree also a matter of finding your true signature, that type of vibe that you personally love. To have that integrity I think is key. When using software I think it is best to use an integrated approach in which you distribute the processing.

You should love what you hear. It should feel very "right" to you...
 
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Yes it is very complicated and to some degree you are right, but one thing that needs your attention here is the impact of latency and its impact on the typical ITB setup. You need to know precisely what DAW software + version to combine with what audio interface and it is the highest sample rate that in general also gives the lowest latency. Part of the complexity is in that combo. If you use plugins you can account for a certain amount of inaccuracy in the figures fed to the internal delay compensation, so the only way of getting truly low latency is to both limit or minimize it at whatever that has a potential impact on it. One way of doing this is to not use any digital processing at all. If you on top of that use the right software and hardware for the recording, then it is mostly the clock accuracy + sample rate combination that is going to be a concern. Now because of the limits in the amount of available headroom you have in various audio interfaces, it becomes this tradeoff you need to solve mathematically. But from blind tests I have done I know the clock accuracy is of high importance. Various DAW software and plugins on certain platforms can also have long term latency performance drops. I know for instance there are long term latency issues in Pro Tools on the PC platform.

There are certain signal processing routes you can take to manage it with a certain amount of latency tolerance, but in general if you are really serious you skip the digital processing altogether. In this way eventhough there is a certain amount of long term latency increase natively in your particular setup as you play your mix over and over, it stays within a manageable range.

My overall recommendation is to put music first, become very hardware aware and use that awareness to improve the quality. This is to some degree also a matter of finding your true signature, that type of vibe that you personally love. To have that integrity I think is key. When using software I think it is best to use an integrated approach in which you distribute the processing.

You should love what you hear. It should feel very "right" to you...

I have a Steinberk UR MK II, which they say has zero latency for vst.. as I think that is it's design.. it does a pretty good job...

I Master in Pro Tools.. and signal everything to busses, and then the main mix bus.. I use SSL compression on the parallel compression bus, and send that back to the mix bus, I send it to two mono tracks and split them to stereo to widen the image with no effects, I send that back to the mix bus..

On the mix bus, depending on what it needs, but generally I have mixed it, and a lot of the work has been done there, so i will glue it with linear phase EQ, as if linear phase is all on the main mix bus, it send the whole signal out linear to the next thing, like tape saturation, and multi band limiting.. not much, just touching the threshold... then a limiter oversampled with a graphic display, so see what is happening there, then another limiter at 0, just to stop any possible peaks after that.. I have achieved the best sound yet on my latest track... it's close to professional reference tracks now..

I learn pretty quick these days, but the learning curve is steep.. and a long journey to it's peak so to speak ;P
 
Oh, and another little treasure, if you are limited to mixing/mastering in headphones, a handy plug in called redline monitoring creates stereo speakers and room ambience in your headphones, and does a fantastic job, you can check the phase and everything :)
 
480 is the most common, but by no means a standard.

Here's what I've found so far:

Logic - 480
DP - 480
Cubase - defaults to 480, but adjustable
Sibelius - defaults to 256, but adjustable from 96 to 960
Finale - 1024
Ableton Live - 96

Note that these are for exported MIDI files only, almost all of these programs run at a different internal resolution. I'm not sure if all of these numbers are 100% accurate, some of them were found on forum posts, not in manuals.

I'm really surprised at how difficult it is to find this information. I have a way to figure this out, but thought that searching the internet would be faster. Guess I was wrong. If no one confirms this I will post my findings here after I run some tests.

I think that PPQ, has something to do with sample rate somehow... I mean, it's 96 ppq, 960, 480.. I wonder.. I think I will stay above 44.1, just for my own sake..



 
I've been recording, mixing, mastering at 44.1/24 for several years and would like to move to 96/24, but I'm afraid my laptop won't handle it well, especially wrt latency. My processor is i7-4500U @ 1.8GHz, with 16GB RAM, which handles up to 50 tracks fine in my DAW with no noticeable latency at 44.1/24. Does anyone know what I'd need to support a similar load at 96/24? I've seen some articles which say you need Quad Core 3.0 or better, but I'd like to hear from someone who has actual experience...
 
I've been recording, mixing, mastering at 44.1/24 for several years and would like to move to 96/24, but I'm afraid my laptop won't handle it well, especially wrt latency. My processor is i7-4500U @ 1.8GHz, with 16GB RAM, which handles up to 50 tracks fine in my DAW with no noticeable latency at 44.1/24. Does anyone know what I'd need to support a similar load at 96/24? I've seen some articles which say you need Quad Core 3.0 or better, but I'd like to hear from someone who has actual experience...

For anyone who is serious you need a dual cpu solution, with 4 cores on each processor at a minimum. Then on top of that, you want as much as possible of the plugins to do their processing on their own dedicated hardware. Then on top of that you also ensure you have as much RAM and SSD drive capacity as possible, especially RAM. I think Mac Pro computers are great for DAW use.
 
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