Talk:Analog-to-digital converter

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exact reconstruction ?

This article says that a sampled bandlimited signal can be "EXACTLY" reconstructed, but I think there are some who would dispute this. There is some talk going on at Talk:aliasing that suggests that the Nyquist sampling theorem is only an approximation. The maths is beyond me, so I'm hoping that someone will explain the problem in words. I'm guessing that it has to do with the twin impossibilities of building a perfect brick-wall filter and taking an instantaneous sample. -- Heron 09:16, 2 Apr 2004 (UTC)

A bandlimited signal CAN be EXACTLY reconstructed. The problem is that it is impossible to have a truly bandlimited signal. This is because a signal that is truly bandlimited (i.e. has exactly zero power above a given frequency) cannot be simultaneous limited in time, and is therefore non-causal. Think of this as Heisenburg's Uncertainty Principle for signals. The amount of aliasing can always be controlled, however, to be below the SNR of the signal in question. So, calling Nyquist an approximation is silly, and not useful. It is more accurate to say that the Nyquist's criterion can only be approximately satisfied, rather than the theory itself is an approximation.

Think about band-limited periodic signals. They are composed by a group of impulses in the frequency domain, so it's easy to see that they are band-limited. If you multiply this periodic (therefore unlimited) signal by a step function, we convolve the impulses in the frequency domain by an unlimited 1/s function, and the signal is not band-limited anymore.

It's good to think about periodic signals, because the values of the impulses in the time domain will be different depending on the delay between the sampling and the signal. Even so, the reconstruction will bring the same wave shape, delayed.

You can even have semi-periodic samplings, where the sampling frequency is not a rational multiple of the signal frequency. The reconstruction will bring the original signal, but you need an unlimited number of time samples, what implies in infinite frequency resolution.

Every band-limited function is unlimited in time. -- nwerneck 22:10, 1 Dec 2005 (UTC)

Regarding musical application

This part should talk about the difference between sample rate and resolution regarding music recording and playback. Sample rate will increase the bandwith of the recording and nowadays 44kHz is too low. Theorist might call Nyquist theorem to the discussion, saying the normal person can't hear above 15-17kHz and that 44kHz is enough. The problem is that sound waves are produced mechanically by a speaker and if you simply dont send the hi-frequencies, it will vibrate differently, changing the sound. BTW, DVD is 96kHz. BTW, there is a musical producer that has detected a 95kHz frequency. The guys knew something was bad with his equipment, called in the tech guys, they didnt believe he could hear that high (experts usually hear 20kHz-22kHz) and made a series of test to see if the guy was really feeling it, and HE WAS!

Regarding resolution, 16bits are insufficient for today's standarts. Not only regarding the low SNR, but also regarding the depth, the dynamics of the music. Simply put, between 16bits and 20 or 24 bits, YOU WILL NOTICE the difference, even if you are not an expert. Once again, DVD quality is 24bits.

--- My two cents: we should isolate this discussion in another article, about psychoacoustics. This article would list scientific references about sound perception. I do know one or two articles about high and low frequency perception. Signal quantization is something, hearing quantized signals is another thing.

Now, I don't know what you mean when you say that DVD has 24bits and 96kHz. DVDs are MPEG2 streams, and not raw signals like CDs!!...

AFAIK, sampling rates and bit depths larger than 44100/16 are only required when the signals are going to be digitally processed. But this means "in consumer applications"... The real limits of human perception is a current area of research, an open question. -- nwerneck, 01 Dec 2005 20:25:49 -0200

Stubness

I agree this article needs some rewriting and re-engineering, but why was it marked as a stub? -- NIC1138 21:23, 17 December 2005 (UTC)[reply]

Resolution

Updated wikilink from Resolution to Resolution (logic) this might need to point to Resolution (music) or even another page. feel free to change it if you know better. --STHayden 22:35, 6 August 2006 (UTC)[reply]

Merge with Digital-to-analog converter

I've posted a merge tag on the article. It would make sense to have the opposite technologies in one article. If you agree or disagree, please post. --Davidkazuhiro 13:03, 9 February 2007 (UTC) I Take that back. The articles are large and well established as it is. --Davidkazuhiro 13:05, 9 February 2007 (UTC)[reply]

Resolution and number of levels

Josecampos did some changes regarding the number of levels of the digitalization. I agreed only with some of his modifications... There are 256 levels in a 8 bit digitalization, from 0 to 255. But the difference from level to level is actually (V+-V-)/(2^8 - 1). So I fixed the article making a distinction between the number of levels and the number of "intervals"... Comments? -- NIC1138 18:44, 1 May 2007 (UTC)[reply]

12.47.224.7 15:57, 26 September 2007 (UTC) Jim Bach 12.47.224.7 15:57, 26 September 2007 (UTC) From: James.C.Bach@Delphi.com Date: 26-SEP-2007[reply]

The difference from level to level is (V+ - V-) / 2^N . . . . NOT (V+ - V-) / 2^n -1 . . . for a 3-bit ADC that is (V+ - V-)/8 not V+ - V-)/7!

In a 3-bit ADC there might only be 7 transitions from code-to-code . . . but the input voltage range (i.e. V- to V+) has 8 regions in it, representing codes "0" thru "7" ("000" to "111" for bit-bangers :-) ). These regions can be equal-width (at 1/8th of Vsupply) like the MicroChip PIC controllers, or they can have a 1/2-width "0" code and a 1.5-width "7" code like just about everyone else's ADC on the planet. Look at the diagrams (and in some cases, equations) provided in the URLs cited below.

Think about it this way . . . let's say the width of your hand represents the voltage range your A/D covers (i.e. HandWidth = V+ - V-) . . . and each finger represents an output code ("0" thru "4", you pick which hand and whether the thumb is LSB or MSB :-) ) . . . you have (presumably) 5 fingers . . . how many cracks (i.e. voltage transitions) do you have between the fingers? 4, right? So, would you estimate that the width of each of your fingers is HandWidth/(N-1) (i.e. HandWidth/4) OR HandWidth/N (i.e. HandWidth/5)? Obviously it is HandWidth/5, which is HandWidth/N. And, if you used your left hand your pinky reasonably approximates the "0" code of a real-world ADC (i.e. 1/2-width) and your thumb reasonably approximates the "4" ("max") code of a real-world ADC (i.e. 1.5-width).

Checkout: http://www.freescale.com/files/microcontrollers/doc/app_note/AN2438.pdf?fsrch=1 http://www.embedded.com/columns/technicalinsights/60403334?_requestid=213222 http://www.maxim-ic.com/appnotes.cfm/appnote_number/1080/ http://zone.ni.com/devzone/cda/tut/p/id/3016

12.47.224.7 15:57, 26 September 2007 (UTC) Jim Bach 12.47.224.7 15:57, 26 September 2007 (UTC)[reply]

Analog != Audio

This article is linked to from some general articles on electronics (I got here via the oscilloscope article), yet some authors appeared to have assumed that it was predominantly about audio applications of A to D conversion. While the audio information seems good, its organization is confusing. In several spots the topic shifts without warning from general discussion of an analog signal to very audio-specific applications. General A-to-D material should be edited to keep the discussion general, and audio-specific information should be moved to its own section or article. Chriscorbell 23:26, 11 May 2007 (UTC)[reply]

Ramp converter with microcontroller

Adding this here, since it's a bit too long to insert into the body of the article. Given a microcontroller (eg PIC) which does not have any analog IO ability, and something we need to measure (eg the resistance of a potentiometer), we can get a fairly decent measurement (perhaps 5%) by using an RC circuit. The PIC's I/O pin is connected to the junction of R/C; C is grounded; R is taken to +V. Then, the I/O pin is initially set up as an output, at logic 0. The I/O pin is then converted back to an input, which (being CMOS) is fairly high-impedance, and probably switches at Vsupply/2. We poll this pin until it goes high. Within its limitations, this technique works extremely well.

That kind of converter is now briefly mentioned in this article as a "ramp-compare ADC". Would wikibooks:Analog and Digital Conversion be a better place to put detailed information on this technique? --76.209.28.72 18:23, 6 July 2007 (UTC)[reply]

Merge from Analog-to-digital conversion with SAR

Would it be worth merging the content at Analog-to-digital conversion with SAR here? J ɪ m p 17:22, 8 January 2008 (UTC)[reply]

inappropriate example

"For example, to sample audio at 44.1 kHz with 32 bit resolution, a clock frequency of over 1.4 MHz would be required". Implying 32 bit resolution audio in an example that seems to be simple math is misleading and inappropriate. 32 bit audio has never been a goal of any converter manufacturer, it's plain silly. The example uses the number 32 in order to inflate the required clock frequency. Suggest removing or rewriting this paragraph. —Preceding unsigned comment added by 65.115.107.210 (talk) 22:44, 19 December 2008 (UTC)[reply]

Oversampling

The section states

a 20 bit ADC can be made to act as a 24 bit ADC with 256x oversampling

, but I believe that it should either read

a 20 bit ADC can be made to act as a 24 bit ADC with 16x oversampling

or

a 16 bit ADC can be made to act as a 24 bit ADC with 256x oversampling

. The sum of 256 (2⁸) 20 bit samples (i.e. 256x oversampling) would require a 28 bit number to represent all possible values.