# [Explained] Why should you use higher Sample Rates while recording



## xeonblade (Feb 23, 2012)

A - signal amplitude
t - time passed

I will try to make this as simple as possible.
Pictures are exaggerated for simpler explanation.

First 2 signals are analog waveforms (guitar signals) in 44.1kHz and 192kHz sample rate.

When analog signal gets converted to digital it's done the following way:
Depending on sample rate (which is time passed) interface checks for current amplitude and records it. If sample rate is higher, time passed untill next check gets lower. 

time passed = 1/sample rate

This means on 192kHz analog signal gets checked for amplitude ~4 times (192/44.1) more than on 44.1kHz sample rate, giving more accurate digital signal (shown on last 2 pictures).

These are the basics of A/D conversion.

*ADDITION 1*: 44,1kHz is CD audio, 48kHz is DVD audio. Other samplerates are 88.2kHz, 96kHz, 176.4kHz and 192kHz. Downsampling algorithms are easier if they can be divided by 2.
Example: 96kHz down to 48kHz (2x48) or 192kHz to 48kHz (4x48), same goes for 176.4>88.2>44.1.
192>176.4, 88.2 or 44.1 would be harder because it isn't divided by 2.

If anything is unclear, please tell me because English isn't my native language and it was kinda hard for me to explain this simple enough.
Colors are here just to make it easier to see, not connected to picture.

*P.S. BE AMAZED BY MY MSPAINT SKILLS!*

EDIT: I will write even simpler explanation with better pictures and better text when I have time.

Another explanation: http://www.sevenstring.org/forum/re...-rather-than-16bit-depth-while-recording.html


----------



## Larcher (Feb 23, 2012)

i wont lie, your ms paint skills are imppecable


----------



## xeonblade (Feb 23, 2012)

Larcher said:


> i wont lie, your ms paint skills are imppecable



You could atleast comment my explanation. I know my MS Paint skills are amazing and leave you speechless but you could try to say something bro


----------



## Kykv (Feb 23, 2012)

Does it makes huge difference in sound? I would hear some samples some day. Thanks!


----------



## ralphy1976 (Feb 23, 2012)

xeonblade said:


> You could atleast comment my explanation. I know my MS Paint skills are amazing and leave you speechless but you could try to say something bro



No Fourier transform to show why it really works?


----------



## xeonblade (Feb 23, 2012)

ralphy1976 said:


> No Fourier transform to show why it really works?



Uhm, I've said this is only simple explanation to help people understand how it works. I don't even know what Fourier Transform is. I've googled it up and it seems like physics/whatever which I haven't learned yet. I'm in 4th grade of highschool. Where did you learn about it?



KykuPL said:


> Does it makes huge difference in sound? I would hear some samples some day. Thanks!



Not really much. I've said it's an exaggeration on picture just to show how it works. It should add some detail to the sound.
Optimal for human hearing would be between 48 to 96kHz. It's almost impossible to hear difference between 96 and 192kHz in the most cases.


----------



## Larcher (Feb 23, 2012)

xeonblade said:


> You could atleast comment my explanation. I know my MS Paint skills are amazing and leave you speechless but you could try to say something bro



Well, I don't know jack shit about what you explained o_o and even after re-reading it I don't know what it is, but that's because I've never looked into this stuff specifically, but for the average joes who do know what this is, it would seem like it is very a very detailed explanation


----------



## xeonblade (Feb 23, 2012)

Larcher said:


> Well, I don't know jack shit about what you explained o_o and even after re-reading it I don't know what it is, but that's because I've never looked into this stuff specifically, but for the average joes who do know what this is, it would seem like it is very a very detailed explanation



Let me supersimplify it for you.

Sample rate is the rate at which interface check for waveform amplitude and records it.
If sample rate is higher then conversion of analog to digital signal will be more accurate, giving better sound.

Conclusion: Higher sample rate is better because it provides closer sound to real (analog) waveform.
Only CON: It takes more HDD space because it takes more data to detail the sound.


----------



## Larcher (Feb 23, 2012)

oo i see, thank you !


----------



## ralphy1976 (Feb 23, 2012)

xeonblade said:


> Uhm, I've said this is only simple explanation to help people understand how it works. I don't even know what Fourier Transform is. I've googled it up and it seems like physics/whatever which I haven't learned yet. I'm in 4th grade of highschool. Where did you learn about it?



University, for a 4th grader your MS paint skillzzz are Br00tallzz....and do not bother with Fourier now....later if you feel like it do it, but instead get busy getting laid!!!


----------



## thraxil (Feb 23, 2012)

KykuPL said:


> Does it makes huge difference in sound? I would hear some samples some day.



Human hearing tops out at around 20KHz for people with young, undamaged ears. Most of us are well below that. Nyquist showed that you need a sampling rate of twice the maximum component frequency to reproduce a signal. That means 40KHz is enough to guarantee that any human alive will hear a perfect reproduction. CD quality audio pads that a little more just to be safe and so we get 44KHz as the standard. Unless you're a freak of nature, you shouldn't be able to tell any difference between tracks with sampling rates above that. 

Which is not to say that 44KHz is perfectly adequate for all (or even most) recording purposes. Each stage of mixing or layer of effects processing potentially loses information. The higher the sampling rate you start at and work with, the better the results will be overall. The catch, of course, is that higher sampling rates mean larger files, more memory used, and more work for the CPU. So the higher the sampling rate you want to work with, the beefier the computer you'll need. 

In a nutshell: record at as high a sampling rate as you can practically achieve (it will be limited by your audio interface or your computer's ability to handle the workload), keep it at a high sample rate through mixing, but when you do a final mix-down (to a .wav, etc) don't bother generating anything higher than 44KHz (unless it's to hand off to someone else who will be doing more mixing and processing on the track).


----------



## xeonblade (Feb 23, 2012)

thraxil said:


> Human hearing tops out at around 20KHz for people with young, undamaged ears. Most of us are well below that. Nyquist showed that you need a sampling rate of twice the maximum component frequency to reproduce a signal. That means 40KHz is enough to guarantee that any human alive will hear a perfect reproduction. CD quality audio pads that a little more just to be safe and so we get 44KHz as the standard. Unless you're a freak of nature, you shouldn't be able to tell any difference between tracks with sampling rates above that.
> 
> Which is not to say that 44KHz is perfectly adequate for all (or even most) recording purposes. Each stage of mixing or layer of effects processing potentially loses information. The higher the sampling rate you start at and work with, the better the results will be overall. The catch, of course, is that higher sampling rates mean larger files, more memory used, and more work for the CPU. So the higher the sampling rate you want to work with, the beefier the computer you'll need.
> 
> In a nutshell: record at as high a sampling rate as you can practically achieve (it will be limited by your audio interface or your computer's ability to handle the workload), keep it at a high sample rate through mixing, but when you do a final mix-down (to a .wav, etc) don't bother generating anything higher than 44KHz (unless it's to hand off to someone else who will be doing more mixing and processing on the track).



Thanks for adding that in. (well, atleast you could quote Wiki since that's what I've read 2min ago on wiki http://en.wikipedia.org/wiki/Sampling_rate#Sampling_theorem)

Most people above 25 won't hear above 16kHz, but that doesn't mean you shouldn't use higher sample rates to capture the sound more faithfully since it applies over whole spectrum of frequencies. 
Best quality possible! Anyways, since 192kHz is 4x48kHz there shouldn't be a great loss downsampling for DVD release (unless you want it on CD and you have to resample by uneven number 192/41.1)

I would go with 96kHz since it's the best combination of file size and quality.

Edit: Sorry, I just noticed that you added link on "showed" 
And thanks for your addition to my humble explanation but it's kinda far from simple. Some rep comming your way!


----------



## xeonblade (Feb 23, 2012)

Does DVD audio support 24bit, 192kHz burning or is 96kHz max?


----------



## Spiff (Feb 23, 2012)

xeonblade said:


> Most people above 25 won't hear above 16kHz, but that doesn't mean you shouldn't use higher sample rates to capture the sound more faithfully since it applies over whole spectrum of frequencies.


I guess that's true...if you're making music for dogs  I kid, I agree with you in general, but somewhere above 44,1khz you are fooling yourself recording with ultra-high sample rates that can never be reproduced to a human ear, and at that point you're just wasting bits (and maybe introducing jitter due to equipment limitations).

Ps. Your mspaint pictures are awesome, don't change them!


----------



## Spiff (Feb 23, 2012)

xeonblade said:


> Does DVD audio support 24bit, 192kHz burning or is 96kHz max?


DVD-Audio - Wikipedia, the free encyclopedia tells it best:


> Different bit depth/sampling rate/channel combinations can be used on a single disc. For instance, a DVD-Audio disc may contain a 96 kHz/24-bit 5.1-channel audio track as well as a *192 kHz/24-bit stereo* audio track. Also, the channels of a track can be split into two groups stored at different resolutions. For example, the front speakers could be 96/24, while the surrounds are 48/20.


----------



## xeonblade (Feb 23, 2012)

Spiff said:


> I guess that's true...if you're making music for dogs  I kid, I agree with you in general, but somewhere above 44,1khz you are fooling yourself recording with ultra-high sample rates that can never be reproduced to a human ear, and at that point you're just wasting bits (and maybe introducing jitter due to equipment limitations).
> 
> Ps. Your mspaint pictures are awesome, don't change them!



I've read somewhere that experiments have shown that people find sample rate between 48 and 96kHz optimal. It was somewhere around 50-60ish kHz. I'll seek for that article and link if I succeed in googling.
(They tested audiophiliacs and didn't tell them which sample rate they were hearing.)

Edit: Thnx for DVD wiki page, I totally forgot to look it up on Wikipedia. I'm retarded.


----------



## Purelojik (Feb 23, 2012)

great paint skill, and great explanation. 

but heres the thing . what i've wabted to know and its what EthererealEntity also did was how this applies when using IR impulses.

if im working at 48k and use 48 k impulses they sound ok. but if i record at 48 k and use 96k impulses the sound is so much better. and i know this is because more data is getting through rather than being cutoff or osmething along those lines but the real question is this....

why the hell would i use lower sample rate IR's if i could simply use the higher ones and get a better sound?


----------



## xeonblade (Feb 23, 2012)

Purelojik said:


> great paint skill, and great explanation.
> 
> but heres the thing . what i've wabted to know and its what EthererealEntity also did was how this applies when using IR impulses.
> 
> ...


Your speakers are probably set to work with 96kHz samplerate so it plays back as it is.
Final product would probably be downsampled to project sample rate.


----------



## Winspear (Feb 23, 2012)

Purelojik said:


> but heres the thing . what i've wabted to know and its what EthererealEntity also did was how this applies when using IR impulses.
> 
> why the hell would i use lower sample rate IR's if i could simply use the higher ones and get a better sound?



I'll keep bumping that Gearslutz thread up until somebody replies haha. Funny, they are usually all over sample rate threads. 

An answer as to why the hell would you want to...
Depending on the answer we get to our question over there, it may be the same case as with other plugins and oversampling - doesn't always work. Certain combinations of up and down sampling work better than others...Some plugins don't do it well. Etc.

I suggest doing a Gearslutz search on plugin oversampling in general. Also applying it to synths is a good read.


----------



## xeonblade (Feb 23, 2012)

A bit offtopic from this, but can be concidered somehow connected: One thing that annoys me more than bad quality production is GREAT production with slight but noticable clipping (which can be heard with studio headphones).
I just want to cry when I hear Dream Theater clipping, even slightly


----------



## Konfyouzd (Feb 23, 2012)

xeonblade said:


> Let me supersimplify it for you.
> 
> Sample rate is the rate at which interface check for waveform amplitude and records it.
> If sample rate is higher then conversion of analog to digital signal will be more accurate, giving better sound.
> ...


 
How do you compensate for the latency caused by upping the sample rate? Just a better comp?


----------



## xeonblade (Feb 23, 2012)

Konfyouzd said:


> How do you compensate for the latency caused by upping the sample rate? Just a better comp?



I'm not using 192kHz, I was only comparing because 44.1 and 192 are the opposites. I'm using 96kHz atm, but I'd try using 192kHz if I had interface that supported it.
I don't think any PC made for studio recording should have any problems with 192kHz (read as "minimal 4GB RAM and Quad core over 2.5GHz")
Altho, do note that you'd probably notice some problems if you try to record many tracks at once using 192kHz since USB or Firewire don't have enough bandwidth to support it.

P.S. Upping the sample rate lowers the latency but increases CPU usage and increases interface bandwidth which lowers the amount of simultaneous tracks available to be recorded at the same time without bugs.


----------



## Purelojik (Feb 23, 2012)

im using some old altec lansing pc speakers but for mixing im using a variety of different headphones too. and honestly the lower sample rates sound harsh as hell. even when i mix down everything seems to be preserved. i'll do a test clip one of these days.

the only thing i have is the clip i recorded using the 96 impulses from Gods Cabs Sm7 and Sm57.

this clip is double tracked with next to no Eq applied except the usuall high and low pass stuff. and some drums, no bass.

http://dl.dropbox.com/u/35982007/Alex%20Myla%20-%20Gods%20Cab%20test%20Ganesh%20Rao.mp3

*EDIT*: sorry that above link has some compression

heres the uncompressed track. 

http://dl.dropbox.com/u/35982007/Alex%20Myla%20-Gods%20Cab%20Impulse%20Test%20w_Ganesh%20Rao.mp3


----------



## Konfyouzd (Feb 23, 2012)

xeonblade said:


> I'm not using 192kHz, I was only comparing because 44.1 and 192 are the opposites. I'm using 96kHz atm, but I'd try using 192kHz if I had interface that supported it.
> I don't think any PC made for studio recording should have any problems with 192kHz (read as "minimal 4GB RAM and Quad core over 2.5GHz")
> Altho, do note that you'd probably notice some problems if you try to record many tracks at once using 192kHz since USB or Firewire don't have enough bandwidth to support it.
> 
> P.S. Upping the sample rate lowers the latency but increases CPU usage and increases interface bandwidth which lowers the amount of simultaneous tracks available to be recorded at the same time without bugs.


 
Hmm... I'll have to give this a try when I get home. For some reason every time I increase the sample rate, my sound is better (lowering past a certain point makes it sound like a robot giving birth) but the sound I get back when I play gets more and more delayed as the sample rate goes up.


----------



## xeonblade (Feb 23, 2012)

Explanation is just what I tried to teach here. Higher sample rate = more sound detail which seems to be really important with IRs.


----------



## xeonblade (Feb 23, 2012)

Konfyouzd said:


> Hmm... I'll have to give this a try when I get home. For some reason every time I increase the sample rate, my sound is better (lowering past a certain point makes it sound like a robot giving birth) but the sound I get back when I play gets more and more delayed as the sample rate goes up.



Well, DAW says latency is directly proportional to sample rate, meaning 96 to 192 should half the latency... Maybe it's something wrong with the drivers or you are running USB and it's troubling somehow?
Try increasing buffer if you get increased latency. It might seem against all logic, but who knows, it might help. Because it's double faster and have bigger flow of data it might need a bit bigger buffer to keep it all together.
But it's weird unless you hear crackling and noise.


----------



## Konfyouzd (Feb 23, 2012)

It may very well be the drivers. I can play around with it and get the latency down eventually but something about my setup seems like it may still be wrong. After I get off work I'll open up a project and PM you some numbers if that's cool...?


----------



## xeonblade (Feb 23, 2012)

OFFTOPIC: 


Purelojik said:


> heres the uncompressed track.
> 
> http://dl.dropbox.com/u/35982007/Alex%20Myla%20-Gods%20Cab%20Impulse%20Test%20w_Ganesh%20Rao.mp3



Fine sounding stuff I have to say 

Back on topic:



Konfyouzd said:


> It may very well be the drivers. I can play around with it and get the latency down eventually but something about my setup seems like it may still be wrong. After I get off work I'll open up a project and PM you some numbers if that's cool...?


Yeah ofc, I'll be glad if I can help you any way.


----------



## xeonblade (Feb 23, 2012)

Analog-to-digital converter - Wikipedia, the free encyclopedia

Sampling rate - Wikipedia, the free encyclopedia

Sample rate conversion - Wikipedia, the free encyclopedia

If anyone is interested in further reading.
I'm writing about Audio Bit Depth atm, will be released within an hour.

Edit, it's written: http://www.sevenstring.org/forum/re...-rather-than-16bit-depth-while-recording.html


----------



## xeonblade (Feb 23, 2012)

Whoever wrote "MS Paint master" as comment in reputation have made my day


----------



## Ryan-ZenGtr- (Feb 23, 2012)

@Xeonblade; It was me!


_... A new maestro has arrived!!!_


 
Nice MS paint by the way, I see a Picasso influence. 


Of course, we can all agree that higher quality is better, but I can't see the practical benefits of extreme high quality. After all, everything is going to get smashed (compression) and limited, then dithered down to 16 bit CD quality for lossy Mp3 online distribution.



Sound on Sound magazine have this debate all the time; the engineers create high quality product and sign it off for delivery to the label but the final conversion to online distribution format is done by student work placement schemes somewhere behind the scenes of apple music store and the others.

The mastering engineers were seeking to promote a signing off protocol, much like in the vinyl days when a test pressing was sent back to the mastering engineer for final approval before production.

All fun and games though. Here's looking forward to when we can actually get hold of high quality formats of our favourite artists. You can be sure they recorded in high quality, they just got dithered/****** by the distribution formats.+-


----------



## xeonblade (Feb 23, 2012)

Well, no way I'm going to press my releases in lossy 16 bit 44,1kHz mp3 format 
DVD-A for physical releases and 24bit 96kHz FLAC and 24bit 96kHz 320kbps mp3 (for newbies that don't know to open FLAC) for downloads.

DVD audio should be a new standard as the technology progresses and it's enough big to hold 24bit 96kHz material.


----------



## The_Mop (Feb 23, 2012)

Nice explaination. It's nice to see home grown informative stuff here  If I'm honest, there's not much more that needs to be explained when it comes to sampling rates. Any more complex stuff than this and you're really entering into DSP territory (which I do for my uni course!)... which admittedly can be boring as shit XD

One thing that probably should be mentioned is, what makes sampling rate like 44.1 kHz this acceptable is the fact that the audio can be considered to be continuous. If it's not, to make sure that the audio is represented properly, the sampling rate needs to be about 96 kHz. Funny that..  I'd say myself 192 kHz is kinda overkill for just normal listening, especially with the considerations you mentioned before (getting bumped down to different formats, physical media, e.t.c).

And don't worry about Fourier transform - it's not necessary when talking about AD/DA conversion fundamentals. It's just for transferring audio from the time domain to the frequency domain (like turning an audio signal into a spectrum graph). There's all sorts of shit you can do with it after then, but for sampling you don't really need to go into it.


----------



## xeonblade (Feb 23, 2012)

The_Mop said:


> Nice explaination. It's nice to see home grown informative stuff here  If I'm honest, there's not much more that needs to be explained when it comes to sampling rates. Any more complex stuff than this and you're really entering into DSP territory (which I do for my uni course!)... which admittedly can be boring as shit XD
> 
> One thing that probably should be mentioned is, what makes sampling rate like 44.1 kHz this acceptable is the fact that the audio can be considered to be continuous. If it's not, to make sure that the audio is represented properly, the sampling rate needs to be about 96 kHz. Funny that..  I'd say myself 192 kHz is kinda overkill for just normal listening, especially with the considerations you mentioned before (getting bumped down to different formats, physical media, e.t.c).
> 
> And don't worry about Fourier transform - it's not necessary when talking about AD/DA conversion fundamentals. It's just for transferring audio from the time domain to the frequency domain (like turning an audio signal into a spectrum graph). There's all sorts of shit you can do with it after then, but for sampling you don't really need to go into it.



Thnx for those kind words 
Like I've said, I would record in 192kHz and downsample to 96kHz (all in 24bit) for DVD-A release because downsampling to 96kHz is acceptable in both size and quality and algorithm for 192>96 is as simple as it could be.


----------



## The_Mop (Feb 23, 2012)

Admittedly I would say that diagram is a little misleading though. To sample at ~2 times per cycle at 44.1 kHz, you'd have to be dealing with frequencies at 22 kHz and up... 44.1 kHz just simply isn't that bad  I know this is just for demonstration, and I'm sorta reading into it a little too much, but hey, this is science and we must be accurate now


----------



## xeonblade (Feb 23, 2012)

Bro, it's kinda misleading for you because you've studied that. I'm writing for people who have no clue about it  I just drew a random signal and took a random size for 44.1Hz cycle. I've said it's an exaggeration anyways.
P.S. I've added some minor stuff about other sample rates and downsampling.


----------



## Rational Gaze (Feb 23, 2012)

Just adding my two cents:

Sampling rate should never be mentioned without bit resolution however. Sampling rate is directly proportionate to frequency response. Meaning that the higher the sampling rate, the more dynamic range you have. Lowering the sampling rate begins cutting off high frequency information. The computer is basically able to squeeze more samples into a given amount of time. 

But it is also important to note how bit resolutions affect the process, not just sampling rates. The number of bits determines resolution -> more bits, more details. Each additional bit gives a 6db increase in your dynamic range. These things become pretty well linked. So it's the same principal. You could record at 44k/16 bit because that's the CD resolution, but you'll get a lot more out of it by doubling your bit depth, even at that rate. 

And while the CPU has a larger workload, with higher resolutions/bit depths, I would say having a large harddrive, or several, is even more important. The breakdown is like this:

1 minute of audio, on 1 mono channel = 5MB at 44.1/16
7.5mb at 44.1/24 bit

The figures become pretty high the more you push the sampling rate.

Session Sample Rate ---------Session Bit Depth------MB/Track Minute (mono)----------(stereo)
44.1khz 16 5mb 10mb
--------- 24 7.5mb 15mb
48khz 16 5.5 11
--------- 24 8.2 16.4
88.2khz 16 10 20
------- 24 15 30
96khz 16 11 22
-------- 24 16.5 33
176.4khz 16 20 40
-------- 24 30 60
192khz 16 22 44
-------- 24 33 66

Higher sample rates can be chosen for demanding projects to capture a greater frequency response from the source audio and to minimize sound degradation throughout the project lifecycle. But I feel that unless you are working in a professional environment, with gear that will actually show when you are using 192khz sample rates, there really is no need for such resolutions.

Just thought I'd add to the discussion


----------



## Winspear (Feb 24, 2012)

xeonblade said:


> A bit offtopic from this, but can be concidered somehow connected: One thing that annoys me more than bad quality production is GREAT production with slight but noticable clipping (which can be heard with studio headphones).
> I just want to cry when I hear Dream Theater clipping, even slightly



Isn't it dreadful!?


----------



## xeonblade (Feb 24, 2012)

EtherealEntity said:


> Isn't it dreadful!?



Fuck this loudness war  I have a volume knob on my speakers, you know...


----------



## xeonblade (Feb 25, 2012)




----------



## xeonblade (Feb 27, 2012)




----------



## oddcam (Feb 29, 2012)

My computer lets my choose the Default Sample Rate playback... I can hear a difference between 48000 and 192000Hz. Is this what the OP is talking about, or is this a software issue?


----------



## xeonblade (Mar 8, 2012)

It would make difference if you listened to 96kHz recording but you set it at 48kHz and then if u moved to 192kHz you would hear the detail of original 96kHz.


----------



## Winspear (Mar 10, 2012)

EDIT: This was meant to be addressed to Purelojik  , 
I never did get a reply to that GS thread - I guess it's quite a niche within a niche question, haha!

Anyway, I stumbled across some other information by accident lastnight that I wish to share with you;

A thread in which someone discusses the SR of impulses 
iZotope SRC Settings for Downsampling Impulse Responses: - Gearslutz.com

One post implies you should use a loader that properly deals with SR, such as LiquidSonics - Reverberate | Convolution Reverb for VST, RTAS and AU . This made me curious about the 'options' I posted in my GS post and which may be the correct answer. I ended up asking Redwirez, here is their response;

Hi,

The reason it sounds smoother is probably because the resampling from 96 to 
48kHz rolls of the highs slightly. You are not really using the higher sample 
rate IR because the plug-in does not upsample the audio stream to match the IR 
but instead resamples the IR to match the sample rate of the project. If it 
sounds better then go for it, but it is not as accurate as using the 48kHz IR. 
Of course, accuracy is not always the most important thing.

I believe most convolvers work the same way, but don't quote me on it 

So there we are. That covers the MixIR plugin by Redwirez, at least. I guess it varys. 

Of course, use whatever sounds best! Just good to read a few of these threads and be aware of the potential issues.

I never did care too much about the project sample rate itself...but I'm now becoming aware that oversampling within other plugins can make quite a difference. Soft synths using algorithms are also greatly affected.
Have a look at this
Running your software synths at higher sample rates - Gearslutz.com

If you don't have time to read the whole thing..just check out the very last post with audio clips. Wow, extreme. Even the much more subtle examples are going to add up over a mix..

This alone was enough to convince me to begin running my projects at 88.2 rather than just using oversampling in the plugins that allow it. Seen as Sonar doesn't allow me to bounce the synths down in the way described in that thread. That means I'll now be using the 88.2's from Redwirez, too.

It's going to blow up my CPU some..but this other plugin that got linked appears to be an absolute gem. 
GPU Impulse Reverb VST | NVidia/ATI GPUs used as DSP for convolution reverb calculation

Uses the GPU instead of the CPU. Haven't had time to try the demo yet but I like this idea a LOT, seen as almost all of the CPU usage I get during a mix that screws me over comes from my multiple IRs.

It's also a 64 bit loader which is another godsend to me, finally! I'm sick of how my guitar chain is all 32 bit. This will help a lot. Just need some 64 bit amp sims now...

The other loader linked above is 64 bit, too. The tweakability and routing options on it look intense.

Neither of these are free but are very cheap!

Blabber over


----------



## xeonblade (Mar 11, 2012)

Mother of all bumps


----------



## Winspear (Mar 11, 2012)

Xeon, I didn't mean to address that post to you haha!


----------



## xeonblade (Mar 11, 2012)

It's cool, I've read it anyways


----------



## Purelojik (Mar 11, 2012)

EtherealEntity said:


> EDIT: This was meant to be addressed to Purelojik  ,
> I never did get a reply to that GS thread - I guess it's quite a niche within a niche question, haha!
> 
> Anyway, I stumbled across some other information by accident lastnight that I wish to share with you;
> ...





this makes a lot of sense to me. thanks [email protected]!


----------



## Eric Christian (Mar 12, 2012)

xeonblade said:


> A - signal amplitude
> t - time passed
> 
> I will try to make this as simple as possible.
> ...


 

Can you make another MS Paint chart that shows what happens at the end of the mixing process when all your 192kHz 24 bit recordings are downsampled to 44.1 16 bit audio to make a CD master?


----------



## pipelineaudio (Mar 12, 2012)

For those who really want an intellectual excercize into the sample rate science, understand this, and why it relates:

If you tell me there is a perfect circle, and give me the location of any two points on that circle, I can tell you how big the circle is


----------



## Nirob (Mar 12, 2012)

i used to record in 24bit/44.1khz but these days i'm going with 24bit/48khz and getting a much better result from my pod ux2...


----------



## Winspear (Mar 12, 2012)

Eric Christian said:


> Can you make another MS Paint chart that shows what happens at the end of the mixing process when all your 192kHz 24 bit recordings are downsampled to 44.1 16 bit audio to make a CD master?



Exactly the same deal, but trying to draw a digital signal with another digital signal with less resolution, rather than starting at analogue.



pipelineaudio said:


> For those who really want an intellectual excercize into the sample rate science, understand this, and why it relates:
> 
> If you tell me there is a perfect circle, and give me the location of any two points on that circle, I can tell you how big the circle is



Nice


----------



## xeonblade (Mar 12, 2012)

Eric Christian said:


> Can you make another MS Paint chart that shows what happens at the end of the mixing process when all your 192kHz 24 bit recordings are downsampled to 44.1 16 bit audio to make a CD master?



I could try these days when I catch some free time.


----------



## ElRay (Mar 12, 2012)

thraxil said:


> Nyquist showed that you need a sampling rate of twice the maximum component frequency to reproduce a signal. That means 40KHz is enough to guarantee that any human alive will hear a perfect reproduction.


Wrong, wrong, unbelievably wrong.

Print out the fine drawings. Connect the ends of the lines in the bottom two graphs. Which one looks more like the original signal?

Draw a "good" sine wave. If you sample at the peaks/troughs, you'll get a decent representation of the original signal. Now, simulate a 180 degree phase shift by lining-up all your samples with the zero crossings. Guess what? Based on your Nyquist samples, there's no signal content, even thought in reality there is.

Draw a sine wave. Over that, draw a square wave with the same frequency and zero phase shift. Over that, draw a sawtooth wave with the same frequency and zero phase shift. Over that, draw a triangular wave with the same frequency and zero phase shift. Now lay-out your Nyquist frequency samples. How good do they do at capturing the very audible differences in the waveforms?

Draw a sine wave and "under sample" (sample under the Nyquist frequency). Take your data points, and connect them. Surprise. You've now got a sine wave of a different frequency.

Nyquist is all too often used incorrectly -- People put the cart before the horse. All that it really does is give you the MINIMUM sampling rate given the HIGHEST frequency in your data. If you sample below the Nyquist frequency, you'll have artifacts from under-sampling the higher frequencies that will create frequencies that are not in your original signal.

If your sampling rate is only 40 kHz, then you need to have your signal low-passed, very steeply, at 20 kHz. If not you'll have artifacts.

In grad school, we typically used a minimum of 8 samples per cycle of the highest frequency we were interested in.

Ray


----------



## thraxil (Mar 12, 2012)

ElRay said:


> Wrong, wrong, unbelievably wrong.
> 
> Nyquist is all too often used incorrectly -- People put the cart before the horse. All that it really does is give you the MINIMUM sampling rate given the HIGHEST frequency in your data. If you sample below the Nyquist frequency, you'll have artifacts from under-sampling the higher frequencies that will create frequencies that are not in your original signal.



Yes, it gives you the minimum sampling rate for the highest frequency in your data. There will be artifacts, but you can't hear those artifacts as they're at a higher frequency than the highest frequency in your source data and probably well below the noise floor of human perception, which is what this is all about.

This paper: Audibility of a CD-Standard A/DA/A Loop Inserted into High-Resolution Audio Playback from Boston Audio Society covers extensive listening tests and concludes that 
44.1kHz/16 bit provides highest-possible fidelity playback with not a single one of the listeners in 554 trials able to distinguish between 44.1/16 and higher rates with a higher reliability than pure guessing.


----------



## ElRay (Mar 16, 2012)

thraxil said:


> Yes, it gives you the minimum sampling rate for the highest frequency in your data. There will be artifacts, but you can't hear those artifacts as they're at a higher frequency than the highest frequency in your source data and probably well below the noise floor of human perception, which is what this is all about.


Sorry, wrong again. The artifacts are due to frequency content above 1/2 the Nyquist frequency and they fold-back on to the audible spectrum. They are VERY audible. Read that Wikipedia article you cited. It's very clear about folding around 1/2 Fs and the need to filter and over-sample.

The reason that 44.1 kHz is suitable and people don't hear any improvement with higher sampling rates is not because human hearing tops-out at around 20 kHz, it's because music tops-out at about 4 kHz. Even allowing for the 3rd harmonic on that highest key on a piano only bumps you up to 16 kHz. 

Nowhere does that article say anything close to:


thraxil said:


> Nyquist showed that you need a sampling rate of twice the maximum component frequency to reproduce a signal. That means 40KHz is enough to guarantee that any human alive will hear a perfect reproduction.


That other article does indicate that for practical reasons, you don't need to sample about 44.1 kHz, but that's not equivalent to "40KHz is enough to guarantee that any human alive will hear a perfect reproduction" either.

Ray


----------



## thraxil (Mar 18, 2012)

I feel like we're talking past each other here. 

The sampling theorem as expressed by Shannon, based on Nyquist's work says precisely: "If a function x(t) contains no frequencies higher than B hertz, it is completely determined by giving its ordinates at a series of points spaced 1/(2B) seconds apart."

Yes, you do need to bandlimit your signal to filter out the frequencies between B and 2B to avoid aliasing. 

So what? Does that invalidate the sampling theorem or make it not relevant to digital audio? To me it seems to be basically baked into the definition. 

My logic for stringing things together goes: 1) humans can't hear anything above 20kHz-ish (and our sensitivity decreases quite a bit as it approaches that) 2) The Nyquist-Shannon Sampling Theorem says that with a signal that doesn't contain any frequencies over B Hz, sampling at 1/2B is enough to reproduce it 3) so we take a signal with nothing over 20KHz, sample it at twice that, and you're golden. 

The math behind it says that that holds up. The real world limitations on how impractical it is to make really steep filters is why the CD audio standard adds a fudge factor and we end up with 44.1kHz. That's probably even overkill, but I trust that there was good research done to pick that as a reasonable compromise. It doesn't invalidate the sampling theorem.

My original post may have been glib, mainly because I get tired of the "I only listen to 192kHz/24bit, anything less sounds like crap" cork-sniffing attitude that's so common on internet forums when the reality is that all the actual listening test research that's been done keeps showing that 44.1/16bit is perfectly adequate (and I've got some Monster HDMI cables to sell you). I don't want to re-write an entire signal processing textbook each time, so my kneejerk response is to basically say "it's good enough and here's a link to some information on the Nyquist frequency which is the core concept behind the why so you can read more on the details if you actually care."

If you want to take me to task for not providing every single step and explaining every detail of the history of digital audio, than fine. But "Wrong, wrong. unbelievably wrong."?


----------



## pipelineaudio (Mar 18, 2012)

thraxil said:


> The sampling theorem as expressed by Shannon, based on Nyquist's work says precisely: "If a function x(t) contains no frequencies higher than B hertz, it is completely determined by giving its ordinates at a series of points spaced 1/(2B) seconds apart."



Plus



> Yes, you do need to bandlimit your signal to filter out the frequencies between B and 2B to avoid aliasing.



Means the out of bounds info is theoretically irrelevant to the information in the passband

Forget aliasing except as a practical matter of filter design.

Luckily at 44khz they gave us an extra 2k for filter implementation


----------



## Winspear (Mar 19, 2012)

I think I linked this in here before but will put it here again 
Running your software synths at higher sample rates - Gearslutz.com

Pretty interesting read and explanations/examples of aliasing occuring in synthesis.


----------



## xeonblade (Mar 19, 2012)

Well, I'm acquiring Tinnitus in right ear so fuck audiophillia and studio recording for me...


----------



## pipelineaudio (Mar 19, 2012)

EtherealEntity said:


> I think I linked this in here before but will put it here again
> Running your software synths at higher sample rates - Gearslutz.com
> 
> Pretty interesting read and explanations/examples of aliasing occuring in synthesis.



And compressors...big time

Thats why we oversample in so many dynamic range plugins. Look at scott stillwell's stuff...put it on a cymbal with a fast release and try it with and without the oversampling on

But this is a whole different game than recording at a high sample rate


----------



## xeonblade (Apr 2, 2012)

Big time bump


----------



## QueeZeR (Apr 3, 2012)

I found this link a while ago:
http://people.xiph.org/~xiphmont/demo/neil-young.html

It disagrees about the whole sample rate thing (but agrees about 24 bit, during mixing).
I don't know how accurate the info is, but found it an interesting read.


----------



## ElRay (Apr 12, 2012)

TL; DR: You're right about oversampling in the final product, but for the wrong reason.




thraxil said:


> The sampling theorem as expressed by Shannon, based on Nyquist's work says precisely: "If a function x(t) contains no frequencies higher than B hertz, it is completely determined by giving its ordinates at a series of points spaced 1/(2B) seconds apart."


That may be "adequate" for mere recording of the signal, but it's painfully obvious to anybody that has real experience in discrete signal processing that there's an infinite number of analog signals that will produce the exact same two discrete data points, yet sound completely different. The Nyquist frequency is a sufficient condition on the MINIMAL sampling rate to avoid aliasing, nothing more.

In addition, that's just regarding the basic digital samples, that says nothing about the psychoacoustics of hearing. Here's a simple example: Modulate the envelope of a high frequency sine wave with a low frequency signal and then high-pass the result with a cut-off that will guarantee that none of the modulating signal is present in the output. If you listen to that single sine wave, you will hear the modulating signal, even though it is not present. Now, if you were to sample that sine wave at merely the Nyquist rate, you won't hear the modulation. Sample it at 8x the Nyquist rate, and you'll perceive most of the modulating signal.

Finally, if there is any processing that will need to be performed before the final "mix down", the information loss by sampling merely at the Nyquist Frequency is tremendous. Remember, the sampling frequency affects not just the resolution of the input signal, but also the resolution of the filters. As another example, my PhD dissertation was on auditory processing of hearing in noise. If I ran my simulations at anything less than about 160 kHz, they were unstable and/or did not replicate observed phenomena.

You can read a textbook and mis-apply a theory all you want, but that doesn't change reality. The value of the Nyquist frequency is merely as (A) The lower bound on the sampling rate, given the frequency components of a signal, or (B) The upper bound on the frequency component of a signal, given the sampling rate, in order to prevent aliasing in a discrete signal. That's in, nothing more. It says nothing about the quality of the sample, nor the accuracy of the analog signal recreated from the discrete samples.

The reason you can get by with lower sampling rates for audio is not due to the Nyquist theory, it's due to the signal components of music. The 3rd harmonic of the highest note on an 88-key piano (C8) is only 12 kHz. Sampling at 48 kHz gives you four samples per cycle. Considering that the highest note on a 24-fret, standard tuned, guitar is E6, you're getting about 37 samples per cycle at standard CD sampling rates. Take that out to the 5th harmonic, and you're still getting 5 samples per cycle.

Ray


----------



## All_¥our_Bass (Apr 12, 2012)

A really good analogy is framerate count in animation.
More frames = more fluid movement, less frames is jerky.
Works the same way for sound. If you keep downsampling even from the best recorded tone ever you get this really murky lo-fi kind of tone.

You can try this in your DAW record something or import some really clear hi-fi recording you know very well, select the whole thing and get your daw to downsample it from 44k to 22k, listen, then go to 11k and listen, 5.5k, etc. keep going and notice how the quality degrades, there's less treble and bass and even the midrange gets progressively more blurry and fuzzy.


----------



## thraxil (Apr 14, 2012)

ElRay said:


> Finally, if there is any processing that will need to be performed before the final "mix down", the information loss by sampling merely at the Nyquist Frequency is tremendous. Remember, the sampling frequency affects not just the resolution of the input signal, but also the resolution of the filters.



Oh, you mean like in my first post in this thread where I say the same thing?: 



thraxil said:


> Which is not to say that 44KHz is perfectly adequate for all (or even most) recording purposes. Each stage of mixing or layer of effects processing potentially loses information. The higher the sampling rate you start at and work with, the better the results will be overall. The catch, of course, is that higher sampling rates mean larger files, more memory used, and more work for the CPU. So the higher the sampling rate you want to work with, the beefier the computer you'll need.
> 
> In a nutshell: record at as high a sampling rate as you can practically achieve (it will be limited by your audio interface or your computer's ability to handle the workload), keep it at a high sample rate through mixing, but when you do a final mix-down (to a .wav, etc) don't bother generating anything higher than 44KHz (unless it's to hand off to someone else who will be doing more mixing and processing on the track).


----------



## tr0n (Apr 14, 2012)

thraxil said:


> That means 40KHz is enough to guarantee that any human alive will hear a perfect reproduction. _CD quality audio pads that a little more just to be safe and so we get 44KHz as the standard._ Unless you're a freak of nature, you shouldn't be able to tell any difference between tracks with sampling rates above that.


This is from the first page of the thread.

CDs are at 44.1kHz not 'to be on the safe side' but because decades ago when engineers were looking for the best way to archive data digitally they decided on VHS cassette tapes, by recording an audio signal as if it was a video signal.

The below information I have re-written from: http://www.cs.columbia.edu/~hgs/audio/44.1.html (this also briefly explains why we have 48kHz and implies why higher sample rates are necessary).

Because there were two video standards back then (PAL & NTSC), if data needed to be shared between continents that used different VHS playback devices then there needed to be a common multiple that allowed the audio to be read on any machine.

The PAL standard is 625 lines per 25 frames (2 frames per field = 50Hz), and with 37 blanking lines per field:

294 lines per field x 50 fields per second x 3 samples per line (RGB) = 44.1kHz

The NTSC standard is 525 lines per 30 frames (60Hz) with 35 blanked lines:

245 lines per field x 60 fields per second x 3 samples per line (RGB) = 44.1kHz

So 44.1kHz was therefore the lowest common multiple between both standards and happened to be a little above the 40kHz required sample frequency.

As the article says, CDs were recorded using the same technology, so that is why they became 44.1kHz.







I have a ton of articles from when I studied audio engineering a couple of years ago. I'd suggest anyone genuinely interested in sample rates read this essay: http://www.lavryengineering.com/documents/Sampling_Theory.pdf

And for DSP in general: The Scientist and Engineer's Guide to Digital Signal Processing

Edit: As for 192kHz I recall reading something that suggested it's down to the brain's ability to distinguish between separate events. 1/192kHz = ~5x10^-6s. But I'm not entirely sure about this.


----------



## thraxil (Apr 14, 2012)

Nice. I didn't know the history behind exactly why 44.1 was chosen. Though I probably should have since my computer engineering degree is from Columbia and Henning Schulzrinne (who wrote that page) was one of my profs.


----------

