ORIGINAL: RnRmaChine

Well a sample rate played back at the wrong sample rate would cause it to speed up or slow down in comparision to the original. You can, IF the plugin has the ability to process at double the rate which is really a different thing then the recorded sample rate, run them at double... The reason some plugins have this ability is beause of the "truer" wave reproduction/simulation at higher sample rates. They do sound alot better when you click that button eh??

no, they sound a bit better - and still not as good at when running natively at 88.2khz

This can be tested with a synth like sytrus, although the korg plugs sound pretty much identical at any samplerate. I guess the devil is in the conversion the plug is doing.

Kind regards

Dave Rich

ORIGINAL: RnRmaChine

The sample rate which alot of people tend to disregard as secondary... is defined by a computer in "dot-to-dot" format. (how many samples it takes per second.) It's not a perfect flowing wave like we all have seen on waveform graphs and such.

It is after the reconstruction filter!

Here is a "join the dots" representation of a 20Khz sine wave (curtousy of Sound Forge):


And here is a reconstructed representation of the same sine wave (curtousy of Audition):

Before I blabbler on I will post a link to a good read on this subject. Good for lamen but probably a lil too basic for someone partially educated in the field.

Anyone really should read this before thinking samples rates only matter if you want to hear over 20kHz, there are also well documented attributes to acoustical environments that aren't actually audible in the "human hearing" range. If you are working with mostly samples and adding verbs and such then 44.1 is probably more then adequate. BUT if you are recording an orchestra in the Vienna State Opera you can't possibly think that 44.1 is professional or would get you asked back to record again... do ya?

At least scroll down to the graphs of a square wave and how well defined it becomes as you raise the sample rate... Future Proof Recording Explained

Well f*ck Korg for misleading the unsuspecting public to sell their pointless products.

Their claim that it is a "notoriously difficult “torture test”," is bogus. Removing inaudible harmonics is a practical application of sampling theory unhampered by marketing BS.

When you filter out the inaudible harmonics of a square wave, you end up with a sine wave.

Check out this little graph: http://williams.comp.ncat.edu/Networks/modulate.htm

Start with 64 harmonics, it looks close to a square wave. Lower the number of harmonics untill you end up with a sinewave. That is what the anti-imaging filters in ADCs do. They remove the inaudible frequencies to prevent aliasing fold-back. (Reguardless of wether this happens in the analogue or digital domain).

Korg chose a 20Khz square wave for their "demonstration" because the 20Khz figure will mislead uninformed readers into thinking that it is all in the audible range. In reality only the fundamental is in the audible range (for young people with no hearing damage). All those harmonics are beyond 20Khz.

If they would have chosen a 2Mhz square wave, their graphs would have shown a 2Mhz sine wave after sampling with their product for exactly the same reasons that a 44.1Khz sampling rate will only show a 20Khz sine wave after sampling a 20Khz square wave.

And note, this is very important, that our ears also work as low-pass filters. Even if you play back those inaudible harmonics, they never reach our brains.

1bit (5.6448MHz) recording... It took me by surprise too when I first read about.

Marketing bla bla. High-end converters have moved on from 1-bit sampling due to issues they have. They now use multi-bit oversampling. (4-5 bits at 64 to 128 the base rate). Check out the last graph. It shows how they (Korg) go down to various base rates by using a decimation filter. That is exactly how other converters work except that they have the decimation filter built in to directly hand over at base rates. Other converters have an advantage over Korg's product because they avoid the 1-bit issues by using multi-bit sampling.

Also note in the last graph that they use interpolation to achieve 48Khz sampling rates and multiples of that. That is also exactly what happens in sample rate converters.

Don't believe the marketing hype.

UnderTow

ORIGINAL: UnderTow

ORIGINAL: RnRmaChine

The sample rate which alot of people tend to disregard as secondary... is defined by a computer in "dot-to-dot" format. (how many samples it takes per second.) It's not a perfect flowing wave like we all have seen on waveform graphs and such.

It is after the reconstruction filter!

Here is a "join the dots" representation of a 20Khz sine wave (curtousy of Sound Forge):

<snip>

And here is a reconstructed representation of the same sine wave (curtousy of Audition):

<snip>

Before I blabbler on I will post a link to a good read on this subject. Good for lamen but probably a lil too basic for someone partially educated in the field.

Anyone really should read this before thinking samples rates only matter if you want to hear over 20kHz, there are also well documented attributes to acoustical environments that aren't actually audible in the "human hearing" range. If you are working with mostly samples and adding verbs and such then 44.1 is probably more then adequate. BUT if you are recording an orchestra in the Vienna State Opera you can't possibly think that 44.1 is professional or would get you asked back to record again... do ya?

At least scroll down to the graphs of a square wave and how well defined it becomes as you raise the sample rate... Future Proof Recording Explained

Well f*ck Korg for misleading the unsuspecting public to sell their pointless products.

Their claim that it is a "notoriously difficult “torture test”," is bogus. Removing inaudible harmonics is a practical application of sampling theory unhampered by marketing BS.

When you filter out the inaudible harmonics of a square wave, you end up with a sine wave.

Check out this little graph: http://williams.comp.ncat.edu/Networks/modulate.htm

Start with 64 harmonics, it looks close to a square wave. Lower the number of harmonics untill you end up with a sinewave. That is what the anti-imaging filters in ADCs do. They remove the inaudible frequencies to prevent aliasing fold-back. (Reguardless of wether this happens in the analogue or digital domain).

Korg chose a 20Khz square wave for their "demonstration" because the 20Khz figure will mislead uninformed readers into thinking that it is all in the audible range. In reality only the fundamental is in the audible range (for young people with no hearing damage). All those harmonics are beyond 20Khz.

If they would have chosen a 2Mhz square wave, their graphs would have shown a 2Mhz sine wave after sampling with their product for exactly the same reasons that a 44.1Khz sampling rate will only show a 20Khz sine wave after sampling a 20Khz square wave.

And note, this is very important, that our ears also work as low-pass filters. Even if you play back those inaudible harmonics, they never reach our brains.

1bit (5.6448MHz) recording... It took me by surprise too when I first read about.

Marketing bla bla. High-end converters have moved on from 1-bit sampling due to issues they have. They now use multi-bit oversampling. (4-5 bits at 64 to 128 the base rate). Check out the last graph. It shows how they (Korg) go down to various base rates by using a decimation filter. That is exactly how other converters work except that they have the decimation filter built in to directly hand over at base rates. Other converters have an advantage over Korg's product because they avoid the 1-bit issues by using multi-bit sampling.

Also note in the last graph that they use interpolation to achieve 48Khz sampling rates and multiples of that. That is also exactly what happens in sample rate converters.

Don't believe the marketing hype.

UnderTow

Excellent, excellent post UnderTow. Nothing new here except for those that have never seen it before. Nicely done.

ORIGINAL: UnderTow

ORIGINAL: RnRmaChine
1bit (5.6448MHz) recording... It took me by surprise too when I first read about.

Marketing bla bla. High-end converters have moved on from 1-bit sampling due to issues they have. They now use multi-bit oversampling. (4-5 bits at 64 to 128 the base rate). Check out the last graph. It shows how they (Korg) go down to various base rates by using a decimation filter. That is exactly how other converters work except that they have the decimation filter built in to directly hand over at base rates. Other converters have an advantage over Korg's product because they avoid the 1-bit issues by using multi-bit sampling.

Also note in the last graph that they use interpolation to achieve 48Khz sampling rates and multiples of that. That is also exactly what happens in sample rate converters.

Don't believe the marketing hype.

Yes! Thank you for helping clear up the mis-information on 1 bit sampling. (DSD, SACD et al.) I think it's just dishonest of them to tout it as something new and better when it is a generation behind and the same technology that was used since the early 90s. Which is not to say that it doesn't sound good. It can when done properly. I just object to the blatant misleading hype that surrounds it.

Some more interesting reading:

There's Life Above 20 Kilohertz! - A Survey of Musical Instrument Spectra to 102.4 KHz by James Boyk, California Institute of Technology


Here is one more theorem regarding the samplerate matter:

ORIGINAL: saecollege.de

Bit Rate

Digital sound is made up of words of 0's and 1's and 00, 11, 01, 10 are the four possibilities in a two bit word. A three bit word can be made up with 000, 111, 001, 010, 100, 101, 011, 110, which means there are eight possibilities. You see - 2 bit gives 4, 3bit gives 8, 4 bit gives 16, 5 bit gives 32, and so on. Now if we were to use the bit words to express volume with a four bit word we could give 16 different values for volume. So the higher the bit rate the more accurate the resolution be it volume, digital pictures or digital sound. So 24 bit digital sound has more resolution and accuracy than 16bit digital sound.

i.e.

16-bits --> 2^16 = 65 536 possibilities
24-bits --> 2^24 = 16 777 216 possibilities

ORIGINAL: saecollege.de

Sampling Rate

Digital sound is produced by sampling a sound (or should I say the electrical version of it) in real time and expressing it in bit words. Once you start sampling or recording digital sound a clock starts and progressive samples of what the sound is are taken. The rate at which the samples are taken is called the sampling rate.




The drawing above shows a wave of a sound being sampled. If the time in the drawing is 1 second, then there are 6 samples (the last one is the first in the next second) of the sound in one second or a sampling rate of 7. So obviously the higher the sample rate the more accurate the resolution. So when we say that the sound is 16bit, 44.1Khz it means that the sound is being sampled at 44.1 thousand times a second and it is being measures with 16 bit accuracy. In the above waveform the sampling volume levels given would be 0,2,2,0,-2,-2, Not a very accurate version of a simple waveform. But 44.1kHz, now that's fast, or is it? Lets look at sound in seconds.




In this chart you can see the relationship between the sampling rate and the waveforms it's sampling. 1kHz will have 44.1 samples taken of each of it's waveform as its oscillating at 10,000 waveforms a second. 100Hz will have 441 samples taken of each of its waveforms. But 10kHz will have 4.41 samples taken of each of it's waveforms. Now look at the first waveform we drew. In that drawing we took 6 samples of the waveform and got an amplitude reading saying 0,2,2,0,2,2. imagine how inaccurate 4.41 samples are of a complex waveform. That is why digital high frequencies sound harsh!! The industry has constantly denied this factor and even gone to the extent of saying the hear can't distinguish between a square wave and a sine wave above 7kHz. Pigs Bum.

At a sampling rate of 96kHz you get 9.6 samples of a 10kHz wave and believe me, you can hear it.

In an article by Rupert Neve, I read recently, he said that we should aim for 24bit resolution and 192kHz sampling rate if we want to equal the quality of high quality analogue recording. We will get there. DVD is already up to 24 bit 96kHz sampling so we are on the way. But if your 16bit, 44.1kHz CD sounds bright, consider what makes it bright and you will see that it's a false bright created by the high frequencies sounding like square waves!

Why 44.1kHz Sampling Rate?

Why not 44, or a nice round number like 50. When the first engineers were inventing digital sound they had worked out the on/off, 0/1, idea and needed a way to represent it. The idea came to use white dots on a TV screen where a white dot was on and a black dot was off. Neat. So you record it like a video picture on a video recorder. That was fine, but the engineers had been caught out before. What about PAL (the European video standard) and NTSC? (the American and Japanese standard.) They weren't going to get caught up in that again, no way, so they configured a number that was compatible between the 528 line NTSC and 625line PAL and the number was 44.1kHz. Just a piece of useless info you might want one day!

Junski

just in case you folks didn't realise sonar can actually run at any samplerate - just type it in to the dialog box.

I tried running at 64khz before,- which to me seems a sensible figure.

Quite a few plugins didn't like it though - let alone my hardware ;)

Kind regards

Dave Rich

Well whoever wrote that SAE article is wrong and does not understand sampling theory. And unfortunately so is Rupert Neve. The other article has allready been posted and commented on.

UnderTow

edit...

ORIGINAL: UnderTow

Well whoever wrote that SAE article is wrong and does not understand sampling theory. And unfortunately so is Rupert Neve. The other article has allready been posted and commented on.

UnderTow

indeed.

theres no way you could represent a square wave at 44.1khz ever.

Kind regards

Dave Rich

ORIGINAL: UnderTow

Well whoever wrote that SAE article is wrong and does not understand sampling theory. And unfortunately so is Rupert Neve. The other article has allready been posted and commented on.

UnderTow

It's so easy to give that type statements ... it would be good to read some lines of arguments. Could you explapain what's wrong there in SAE article/conclusions (or in R. Neve's statements (are there sources available))? Not that I would know the thrut but, let us learn from others mistakes!

Also ... I didn't noticed the link for James Boyk's article you say been linked and commented already (I checked all sub links too before posting it) ... I noticed that this article was used as a resource (ref.) in that Ultrasonic article (written by Andrew Hon) which was linked in post #18 made by Roflcopter. Boyk's article is much deeper (since it's the original paper) than the summary made by Hon ... that's why the link given (though, it won't change the world but ...).

Here is the link (again): http://www.cco.caltech.edu/~boyk/spectra/spectra.htm

Junski