|
 Kyma Forum
 Tips & Techniques
 Why do frequencies near HalfSampleRate
|
| next newest topic | next oldest topic |
| Author | Topic: Why do frequencies near HalfSampleRate |
|
mathis Member |
posted 11 October 2004 02:49
 
get modulated? This is probably due to digital physics, since itīs not only in Kyma. I want to generate a 22k5 waveform inside 44k1 samplingrate, which should result in an analog sine, right? But it is amplitude modulated with a period length of 49 samples. How come? And watching the oscilloscope this is relevant for all high frequencies from 500 Hz on already. Clearly visible amplitude modulation. Very curious. Can anybody explain also for a non-connaisseur? Sidenote: I just discovered the analog wafeforms sampled from Minimoog, Jupter8 and alike. Amazing! Since when do we have them? Bests, IP: Logged |
|
SSC Administrator |
posted 11 October 2004 11:04
This has more to due with the algorithm for Kyma's OscilloscopeDisplay than it does with the analog to digital converters. You won't see the same effect if you look at the *output* of the Capybara on an external analog oscilloscope. Here's why. Imagine a sine wave whose frequency is exactly half the sample rate. In other words, you will take two samples of this sine wave on each cycle. What if you happen to sample it on the two zero crossings that a sine wave has on each cycle? That is why the Nyquist Theorem says that the highest frequency you can represent in a digital sampled system is *less than* half of the sample rate. OK, so if you can't get exactly half the sample rate, let's take a sine wave that is a little bit less than half the sample rate: (SR/2 - 1) hz. On the first sample, you might get unlucky again and hit right on the zero crossing. But on the next sample is going to come a little bit sooner in the cycle than the zero crossing. And on subsequent cycles, your sample points will slowly drift with respect to the waveform. After one second, you have drifted all the way through the waveform and end up back at the initial zero again. What does the result of this sampling look like? It looks like a square wave that repeats (SR/2-1) times per second and has an amplitude that grows and shrinks at a rate of once per second. NOT a sine wave a full amplitude. So does this mean that Nyquist was wrong?! Does it mean that you cannot represent analog signals digitally? Discussion continued at http://www.symbolicsound.com/cgi-bin/bin/view/Learn/WhyDoI SeeAmplitudeModulationForFrequenciesNearTheHalfSampleRate [This message has been edited by SSC (edited 12 October 2004).] IP: Logged |
|
Jadie Member |
posted 11 October 2004 21:59
This is also one reason why some argue that analog (and 96K/192K for that matter) sounds "warmer" than a CD-quality recording. The harmonics at those higher frequencies are less accurately represented than, say, a 1000 Hz sine wave. No matter how good it is, the DAC cannot accurately convert two samples per cycle into a sine wave. IP: Logged |
|
SSC Administrator |
posted 12 October 2004 10:16
Sorry to anyone who tried to follow the link yesterday, it had an extra parenthesis on the end of the URL. Here is the real link: http://www.symbolicsound.com/cgi-bin/bin/view/Learn/WhyDoISeeAmplitudeModulationForFrequenciesNearTheHalfSampleRate IP: Logged |
All times are CT (US) | next newest topic | next oldest topic |
  |
|
Contact Us | Symbolic Sound Home
This forum is provided solely for the support and edification of the customers of Symbolic Sound Corporation.
Ultimate Bulletin Board 5.45c