Avatar

Please consider registering
guest

sp_LogInOut Log In sp_Registration Register

Register | Lost password?
Advanced Search

— Forum Scope —




— Match —





— Forum Options —





Minimum search word length is 3 characters - maximum search word length is 84 characters

sp_Feed Topic RSS sp_TopicIcon
Chapter 1 - DAC question on reconstruction filter
Avatar
SonicScholar
Member
Members
July 12, 2017 - 4:10 pm
Member Since: July 11, 2017
Forum Posts: 5
sp_UserOfflineSmall Offline

So in the book under chapter 1, the reconstruction of the signal involves passing the signal through a low-pass filter AKA 'reconstruction filter'. . .

"The output filter is called the reconstruction filter and is responsible for re-creating the seemingly missing information that the original sampling operation did not acquire - all the inter-sample fluctuations"

Ok, so wait here one sec! How do we re-create the intersample fluctuations, if we didnt have that information to start with???? Sersiously whats going on here? Maybe i'm just thinking about this too much, but my reasoning is thus:

Say, we have a signal which consists of many complex waveforms. Our sampling fequency is 44,100Khz, so we can aquire and quantize the signal at 22.7 microseconds. What if the signal had fluctuated inbetween those sampling intervals, and when we recreate say, the middle of a particular intersample fluctuation, the real sample is 0.876 (for example) but our 'averaging' done with each sample (with the sin(x)/x function) will really average out between the two quantized samples (of which the intersample part inbetween we are looking at) and spit out a value as such.  How then can we re-create this accurately? Surely if intersample 'signals' are in fact higher than nyquist in frequency, we cant hear them but they effect the signal as well, if they were below nyquist we would sample them?

Confused

٩(̾●̮̮̃̾•̃̾)۶

Avatar
W Pirkle
Admin
July 13, 2017 - 10:17 am
Member Since: January 29, 2017
Forum Posts: 344
sp_UserOfflineSmall Offline

Ok, so wait here one sec! How do we re-create the intersample fluctuations, if we didnt have that information to start with???? Sersiously whats going on here? Maybe i'm just thinking about this too much, but my reasoning is thus:

This was the same kind of argument Nyquist faced, and is the founding basis for digital audio and why it actually works - and yes we can recover the missing intersample fluctuations that weren't sampled. See Figure 1.5 which graphically displays how the signal is reconstructed, with the time-domain sinc() functions adding up to recreate the missing intersample fluctuations as a result of convolution with the output LPF. Google it and you'll find a zillion papers about it. Here's a pretty good one:

http://www.spot.pcc.edu/~ghech.....cument.pdf

- Will

Avatar
SonicScholar
Member
Members
July 13, 2017 - 4:37 pm
Member Since: July 11, 2017
Forum Posts: 5
sp_UserOfflineSmall Offline

Thanks Will, I will take a look at that paper!

٩(̾●̮̮̃̾•̃̾)۶

Avatar
W Pirkle
Admin
August 26, 2017 - 5:28 pm
Member Since: January 29, 2017
Forum Posts: 344
sp_UserOfflineSmall Offline

Here's another excellent resource by Nigel Redmond:

http://www.earlevel.com/main/t.....?order=asc

- Will

Forum Timezone: America/New_York

Most Users Ever Online: 55

Currently Online:
10 Guest(s)

Currently Browsing this Page:
1 Guest(s)

Top Posters:

Skyler: 48

Derek: 46

Frodson: 45

Peter: 41

clau_ste: 39

TheSmile: 37

JimmyM: 33

Gwen: 32

EZB: 24

lppier: 23

Member Stats:

Guest Posters: 1

Members: 565

Moderators: 1

Admins: 4

Forum Stats:

Groups: 12

Forums: 36

Topics: 581

Posts: 2326

Newest Members:

frankthetank, dhodgson, Wanderer, mister1234, Evan Galvanek, Rowan Fraser, patrickbarr1984, Ray C., Jon R., Pat

Moderators: W Pirkle: 344

Administrators: Tom: 69, JD Young: 80, Will Pirkle: 0, W Pirkle: 344