IPB

Welcome Guest ( Log In | Register )

Representing frequency of n Hz needs sampling rate >2n Hz, not =2n, From: Badly drawn waveforms vs. the audio that’s actually output/93496
icstm
post Feb 16 2012, 11:56
Post #1





Group: Members
Posts: 121
Joined: 25-January 12
Member No.: 96698



QUOTE (Wombat @ Feb 14 2012, 16:47) *
Hehe, shortly after posting here there was a nice picture posted that tells the whole story smile.gif
where is the nice picture? That is really important to share!
The whole point of FFTs is that using if I am drawing what the FFT has stored in the time domain (ie if I am drawing sine waves) then I only need a couple of points to draw a PERFECT reproduction!
That is one of the biggest issues that people need to first get their head around when understanding digital audio.

when you see an irregular wave, what you need to ask your self is "what regular wave would I have to add together (ie what frequencies would have to be simultaneously present) to create the irregular wave I see”

so an irregular sine wave is in effect a regular one augmented with higher or lower frequency wave to varying amplitude.

Each of the additional waves only needs a couple of points to represent their frequency. As per usual it appears that CA article really doesn’t get what the Nyquist theorem is saying…

However it is right when it says that it is possible for a perfect 8kHz signal sampled at 96kHz to sound better than sampled at 192kHz if the DAC has trouble with 192kHz (ie is a cheaper DAC). The whole point is that a perfect DAC will produce a perfect 8kHz signal when sampled at 16kHz. But the quality of the output has little to do with the conversion and more to do with the analogue output. One of my earliest posts on HA was asking about DACs in AVRs as I am keen to understand the practicalities on consumer grade stuff.

QUOTE (my earlier post)
What determines a good DAC?
I assume the main issues are post the conversion to analogue and that these issues are the usual ones that impact an amp (pre or power), such as clean power supply, suitable analogue filters, etc.
However are there any differences in the digital part of the amp (such as up/over sampling circuits, or the conversion itself)?
Go to the top of the page
+Quote Post
 
Start new topic
Replies
DonP
post Feb 16 2012, 17:13
Post #2





Group: Members (Donating)
Posts: 1471
Joined: 11-February 03
From: Vermont
Member No.: 4955



QUOTE (icstm @ Feb 16 2012, 06:56) *
Each of the additional waves only needs a couple of points to represent their frequency. As per usual it appears that CA article really doesn’t get what the Nyquist theorem is saying…
....
The whole point is that a perfect DAC will produce a perfect 8kHz signal when sampled at 16kHz.


This key mistake in citing the Nyquist theorem leads to no end of potential trouble. The sampling frequency has to be greater than 2x the maximum signal frequency you want to reproduce.
Go to the top of the page
+Quote Post
[JAZ]
post Feb 18 2012, 10:12
Post #3





Group: Members
Posts: 1783
Joined: 24-June 02
From: Catalunya(Spain)
Member No.: 2383



While I agree that this is a scientific-based forum and that it is adequate to correct this small imprecision that we're used to say (as in saying only 2x instead of >2x)....

... can we stop arguing about it at last? There's been what, already 5 posts about it?

Back on topic, the number was mentioned in relation to the fact that the frequency can be properly reconstructed from the sampled one, which is something that we all agree on. For this, the images posted by xnor are quite representable of that fact.

We can talk about the needed time in case of frequencies nearing half the sampling rate, like pdq's last post, but let's stop talking about ">" versus ">=".
Go to the top of the page
+Quote Post

Posts in this topic


Reply to this topicStart new topic
1 User(s) are reading this topic (1 Guests and 0 Anonymous Users)
0 Members:

 



RSS Lo-Fi Version Time is now: 19th September 2014 - 21:18