This is an archive of past discussions. Do not edit the contents of this page. If you wish to start a new discussion or revive an old one, please do so on the current talk page. |
Archive 1 |
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)
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)
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)
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)
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.
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)
"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)
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. —Preceding unsigned comment added by 217.40.148.115 ( talk) 16:23, 11 March 2009 (UTC)
Many applications require converting some analog quantity at some remote location to a nice digital display at a more convenient, nearby location. I've seen several people do this with several parts:
The system as a whole is performing analog-to-digital conversion, although it is difficult to say whether "the ADC" is located at the remote site or at the local site.
I think this kind of ADC structure should be mentioned in the Analog-to-digital converter#ADC structures section of this article.
Is there a standard name for this type of ADC structure? "voltage-to-frequency ADC"? "frequency modulation ADC"? -- 68.0.124.33 ( talk) 06:51, 18 August 2009 (UTC)
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)
I've just come to this article, and I'm confused about the apparent inconsistency between exact signal reconstruction and the problems of aliasing. I don't understand the discussion above about what a bandlimited signal is. It would be nice if this could be made a little more layman friendly. Tim Richardson ( talk) 23:43, 2 November 2009 (UTC)
Actually, following around a few links and arriving at the bandlimited page sorted me out. bandlimited signals are a subset of waveforms that don't have higher frequency components when Fourier transformed (eg sinewaves). Tim Richardson ( talk) 23:53, 2 November 2009 (UTC)
What is the minimum voltage that an ordinary, consumer-grade 16-bit ADC can sample?
I have looked at the Wikipedia article "Line Level", and consumer grade audio peaks at 0.447 volts. Since 16 bits equals 65536, dividing into 0.447 you get 0.0000068 volts. That is 0.0068 millivolts, or 6.8 microvolts (edit: was nanovolts).
Is the average consumer card capable of measuring microvolts? Is any card capable of this? Even if you use the "pro audio" specification of 1.737V, you still get 0.0000265 volts, or 0.0265 millivolts, or 26.5 microvolts.
I have come across the phrase ENOB, or Effective Number of Bits (which has its own article). What is the ENOB of the average 16-bit card? What VOLTAGE range is it designed to measure? -mjs 173.68.190.122 ( talk) 04:41, 4 October 2009 (UTC)
I've just applied what I think is the third revert - i.e. we have hit 3RR. To avoid warring, let's discuss this. The text I have (re)removed was:
There is plenty to say about music quality, what a human can/cannot distinguish, and whether CD quality is enough. It would be good to include something sensible on the subject, with sources. An unsourced ramble about 'some people think...' isn't suitable for Wikipedia - we can do better than that. By all means, please add something concrete - it would improve the article - but I will invoke Nyquist if required. 192kHz needs some justification. GyroMagician ( talk) 21:46, 9 March 2010 (UTC)
In the article, fig. 3 depicts ADC transfer function with wide LSB: code length is not of unit and found as Q=2^{n-1}/2^{n}), where n is ADC resolution, bits, code shift is of -1/2Q. My questions are to SpinningSpark: Tell please where did you get this picture? What is it sourced from? Such transfer function is very intresting, but i did not see it in practice and in books/datasheets accessible to me. Did you see ADC with such transfer function in your practice? Beforehand thanks. —Preceding unsigned comment added by 87.253.7.84 ( talk) 12:53, 18 August 2010 (UTC)
There is an obvious problem on Modem that most people think a Modem converts digital signals to analog signals. The modem article is already too long to include a tutorial on the difference between analog and digital signals, and there isn't any good place to reference -- so any suggestions would be welcome. 203.206.162.148 ( talk) 08:03, 8 September 2010 (UTC)
I've just undone an edit by 80.100.243.19. The (good faith) edit suggests that an anti-aliasing filter isn't needed in the ADC uses a V-to-F converter and a counter. I undid for two reasons. First, I think the integration time of the counter is a low pass filter. But also, trying to introduce exception to a clear story, early in the article, is confusing. But I thought I'd bring it up here for discussion - I won't be offended if you all tell me I'm wrong ;-) GyroMagician ( talk) 20:45, 24 September 2010 (UTC)
On the picture of the adc on the sound card, the card is labelled as a Xi-Fi Fatal1ty Pro. I get the feeling this is incorrect, so i figured i'd point it out. However it also happens on that particular sound card's page. —Preceding unsigned comment added by 121.209.39.143 ( talk) 07:11, 28 September 2010 (UTC)
What does E_FSR mean? The voltage at the frequency sampling rate? What does that even mean? —Preceding unsigned comment added by 99.159.44.37 ( talk) 18:26, 26 January 2011 (UTC)
An editor has asked for a discussion to address the redirect Digital number which was recently added pointing to this article. Editors here might want to participate in the redirect discussion (if you have not already done so). SpinningSpark 20:40, 4 March 2011 (UTC)
The "Resolution" section has diagrams of three possible transfer functions, and describes two of them mathematically (mid-tread with a small 0 and a large max code, and mid-rise with offset with all steps equal width). It states that “The exception to this convention seems to be the Microchip PIC processor, where all M steps are equal width, as shown in figure 1.”. There are painfully few references here, especially no references for the claim that PICs' ADCs use mid-rise-with-offset; interestingly, a Google search for the phrase “mid-rise with offset” seems to turn up nothing more than a lot of verbatim copies of the text shown on this page. What's the deal? Where did this information come from? Hawk777 ( talk) 04:49, 26 March 2011 (UTC)
In the example table the headline says input frequency: e.g. 44.1kHz (typical sample rate for an audio stream). Then the largest permissable jitter is worked out correctly for an input frequency of 44.1kHz. But due to the sampling theorem you must not have frequencies beyond 22kHz in your audio stream, hence the permissable jitter is a factor of 2 larger for an audio stream. To have more realistic examples, one should say input frequency 22kHz for the audio stream, etc... It would be helful to mention that this requires a sample rate of 2x the input frequency... — Preceding unsigned comment added by 86.3.141.168 ( talk) 17:46, 6 August 2011 (UTC)
There should be history of analog-to-digital converter development. — Preceding unsigned comment added by 76.17.115.184 ( talk) 02:32, 4 November 2011 (UTC)
Was it Sir Denys Wilkinson? — Preceding unsigned comment added by 71.191.185.32 ( talk) 01:13, 22 June 2012 (UTC)
The claim that it's not worth using 24 bits for sound recording if jitter is not ultra low is somewhat clueless. One reason for using 24 bits is to have extra headroom before clipping. If the recording level is reduced, then the effect of jitter is also reduced: more jitter is needed to make a one-bit error in a quiet signal than in a loud one! So for instance if you record using only 16 bits out of the available 24, then the jitter requirement is that for 16 bits. But there is 8 bits of headroom for transients. 192.139.122.42 ( talk) 20:08, 30 July 2012 (UTC)
This article badly needs an opening section on the basic idea of sampling, even if just a continuous waveform overlaid with a stepped waveform. Seriously. As it is now, it just starts out with a discussion of resolution and the reader who knows nothing of A/D converters (and this is probably a large proportion of people who would be using Wikipedia to read about them) is rudely thrown into a discussion of resolution. Resolution of what, he/she might ask, before clicking on to something else more interesting. — Preceding unsigned comment added by Oscarruitt ( talk • contribs) 03:56, 31 July 2012 (UTC)
Teapeat ( talk · contribs) has added a brief section on metastability. I don't have access to the ref provided. That ref appears to be narrowly focused. Here is a more general discussion of metastability issues in ADCs [1]. -— Kvng 05:17, 29 March 2013 (UTC)
"One effective bit of resolution changes the signal-to-noise ratio of the digitized signal by 6 dB" -- wouldn't it be 3 dB? That is 10 log_10(2). — Preceding unsigned comment added by 91.137.20.132 ( talk) 18:20, 20 March 2014 (UTC)
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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)
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
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)
That formula is wrong...Q = (span)/(number_of_levels) ... —Preceding unsigned comment added by 87.16.20.206 ( talk) 10:34, 3 May 2010 (UTC)
Every literature says that Q = (span) / (number_of_levels). Mistake is because you consider that max voltage can read from ADC is Vref (and not Vref -1LSB). Then, for your example, when you have and ADC with a resolution of 1bit and a full range of 0..1V, you have 1LSB=1/2=0.5V -So the voltage that corresponds to D=1 is 0.5V (of course always +/- 1/2LSB in ideal case) not 1V Sorry for my english -- Frank Rossi —Preceding unsigned comment added by 87.16.20.206 ( talk) 14:12, 3 May 2010 (UTC)
The analog value represented by the all one codes is VREF -1LSB , so with 1bit of resolution and VREF=1V you have 1LSB = 1/2 = .5V and infact VREF - 1LSB= (1-.5) =.5V while,according your formula this "all-one-codes" value would be 1V and not .5V (source: http://www.analog.com/library/analogDialogue/archives/39-06/Chapter%202%20Sampled%20Data%20Systems%20F.pdf - page 2.4) -- Dontronix ( talk) 07:15, 5 May 2010 (UTC)
Have a look at: http://www.national.com/appinfo/adc/files/ABCs_of_ADCs.pdf too. -- Dontronix ( talk) 07:56, 5 May 2010 (UTC)
Regarding discussion on number of levels in an 8 bit A/D or D/A. The first coded level is zero. Therefore there are (256-1)or 255steps. The word steps is often used in A/D, D/A talk and is less confusing terminology than talking about intervals. — Preceding unsigned comment added by Alan Bate SEA ( talk • contribs) 13:47, 22 February 2012 (UTC)
The use of 2^M-1 is incorrect. Even the source that is used for defining the resolution in bits uses 2^M [1]. Another way to think of A/D conversion is the division of the full-scale range in half where a comparator is used to determine if the MSB is 1 or 0; then the next range is divided in half again, and again, until M times; thus the full-scale range is divided a total of 2^M times to give a final resolution size of the full-scale range divided by 2^M. 75.174.53.165 ( talk) 05:13, 24 January 2017 (UTC)
I just did some research into this, because I had the same issue myself when I saw the formulation in practice. Range/2^n is correct. The reason is that each bit doesn't represent a discrete analog value. It represents a small range of value and generally sets at the center of that range. For instance, if you're converting 3-bit to a range of 0-8 volts, then: 000 -> 0V-1V (0.5V) 001 -> 1V-2V (1.5V) 010 -> 2V-3V (2.5V) 011 -> 3V-4V (3.5V) 100 -> 4V-5V (4.5V) 101 -> 5V-6V (5.5V) 110 -> 6V-7V (6.5V) 111 -> 7V-8V (7.5V)
LordQwert ( talk) 17:21, 13 March 2017 (UTC)
References
This is an archive of past discussions. Do not edit the contents of this page. If you wish to start a new discussion or revive an old one, please do so on the current talk page. |
Archive 1 |
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)
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)
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)
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)
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.
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)
"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)
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. —Preceding unsigned comment added by 217.40.148.115 ( talk) 16:23, 11 March 2009 (UTC)
Many applications require converting some analog quantity at some remote location to a nice digital display at a more convenient, nearby location. I've seen several people do this with several parts:
The system as a whole is performing analog-to-digital conversion, although it is difficult to say whether "the ADC" is located at the remote site or at the local site.
I think this kind of ADC structure should be mentioned in the Analog-to-digital converter#ADC structures section of this article.
Is there a standard name for this type of ADC structure? "voltage-to-frequency ADC"? "frequency modulation ADC"? -- 68.0.124.33 ( talk) 06:51, 18 August 2009 (UTC)
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)
I've just come to this article, and I'm confused about the apparent inconsistency between exact signal reconstruction and the problems of aliasing. I don't understand the discussion above about what a bandlimited signal is. It would be nice if this could be made a little more layman friendly. Tim Richardson ( talk) 23:43, 2 November 2009 (UTC)
Actually, following around a few links and arriving at the bandlimited page sorted me out. bandlimited signals are a subset of waveforms that don't have higher frequency components when Fourier transformed (eg sinewaves). Tim Richardson ( talk) 23:53, 2 November 2009 (UTC)
What is the minimum voltage that an ordinary, consumer-grade 16-bit ADC can sample?
I have looked at the Wikipedia article "Line Level", and consumer grade audio peaks at 0.447 volts. Since 16 bits equals 65536, dividing into 0.447 you get 0.0000068 volts. That is 0.0068 millivolts, or 6.8 microvolts (edit: was nanovolts).
Is the average consumer card capable of measuring microvolts? Is any card capable of this? Even if you use the "pro audio" specification of 1.737V, you still get 0.0000265 volts, or 0.0265 millivolts, or 26.5 microvolts.
I have come across the phrase ENOB, or Effective Number of Bits (which has its own article). What is the ENOB of the average 16-bit card? What VOLTAGE range is it designed to measure? -mjs 173.68.190.122 ( talk) 04:41, 4 October 2009 (UTC)
I've just applied what I think is the third revert - i.e. we have hit 3RR. To avoid warring, let's discuss this. The text I have (re)removed was:
There is plenty to say about music quality, what a human can/cannot distinguish, and whether CD quality is enough. It would be good to include something sensible on the subject, with sources. An unsourced ramble about 'some people think...' isn't suitable for Wikipedia - we can do better than that. By all means, please add something concrete - it would improve the article - but I will invoke Nyquist if required. 192kHz needs some justification. GyroMagician ( talk) 21:46, 9 March 2010 (UTC)
In the article, fig. 3 depicts ADC transfer function with wide LSB: code length is not of unit and found as Q=2^{n-1}/2^{n}), where n is ADC resolution, bits, code shift is of -1/2Q. My questions are to SpinningSpark: Tell please where did you get this picture? What is it sourced from? Such transfer function is very intresting, but i did not see it in practice and in books/datasheets accessible to me. Did you see ADC with such transfer function in your practice? Beforehand thanks. —Preceding unsigned comment added by 87.253.7.84 ( talk) 12:53, 18 August 2010 (UTC)
There is an obvious problem on Modem that most people think a Modem converts digital signals to analog signals. The modem article is already too long to include a tutorial on the difference between analog and digital signals, and there isn't any good place to reference -- so any suggestions would be welcome. 203.206.162.148 ( talk) 08:03, 8 September 2010 (UTC)
I've just undone an edit by 80.100.243.19. The (good faith) edit suggests that an anti-aliasing filter isn't needed in the ADC uses a V-to-F converter and a counter. I undid for two reasons. First, I think the integration time of the counter is a low pass filter. But also, trying to introduce exception to a clear story, early in the article, is confusing. But I thought I'd bring it up here for discussion - I won't be offended if you all tell me I'm wrong ;-) GyroMagician ( talk) 20:45, 24 September 2010 (UTC)
On the picture of the adc on the sound card, the card is labelled as a Xi-Fi Fatal1ty Pro. I get the feeling this is incorrect, so i figured i'd point it out. However it also happens on that particular sound card's page. —Preceding unsigned comment added by 121.209.39.143 ( talk) 07:11, 28 September 2010 (UTC)
What does E_FSR mean? The voltage at the frequency sampling rate? What does that even mean? —Preceding unsigned comment added by 99.159.44.37 ( talk) 18:26, 26 January 2011 (UTC)
An editor has asked for a discussion to address the redirect Digital number which was recently added pointing to this article. Editors here might want to participate in the redirect discussion (if you have not already done so). SpinningSpark 20:40, 4 March 2011 (UTC)
The "Resolution" section has diagrams of three possible transfer functions, and describes two of them mathematically (mid-tread with a small 0 and a large max code, and mid-rise with offset with all steps equal width). It states that “The exception to this convention seems to be the Microchip PIC processor, where all M steps are equal width, as shown in figure 1.”. There are painfully few references here, especially no references for the claim that PICs' ADCs use mid-rise-with-offset; interestingly, a Google search for the phrase “mid-rise with offset” seems to turn up nothing more than a lot of verbatim copies of the text shown on this page. What's the deal? Where did this information come from? Hawk777 ( talk) 04:49, 26 March 2011 (UTC)
In the example table the headline says input frequency: e.g. 44.1kHz (typical sample rate for an audio stream). Then the largest permissable jitter is worked out correctly for an input frequency of 44.1kHz. But due to the sampling theorem you must not have frequencies beyond 22kHz in your audio stream, hence the permissable jitter is a factor of 2 larger for an audio stream. To have more realistic examples, one should say input frequency 22kHz for the audio stream, etc... It would be helful to mention that this requires a sample rate of 2x the input frequency... — Preceding unsigned comment added by 86.3.141.168 ( talk) 17:46, 6 August 2011 (UTC)
There should be history of analog-to-digital converter development. — Preceding unsigned comment added by 76.17.115.184 ( talk) 02:32, 4 November 2011 (UTC)
Was it Sir Denys Wilkinson? — Preceding unsigned comment added by 71.191.185.32 ( talk) 01:13, 22 June 2012 (UTC)
The claim that it's not worth using 24 bits for sound recording if jitter is not ultra low is somewhat clueless. One reason for using 24 bits is to have extra headroom before clipping. If the recording level is reduced, then the effect of jitter is also reduced: more jitter is needed to make a one-bit error in a quiet signal than in a loud one! So for instance if you record using only 16 bits out of the available 24, then the jitter requirement is that for 16 bits. But there is 8 bits of headroom for transients. 192.139.122.42 ( talk) 20:08, 30 July 2012 (UTC)
This article badly needs an opening section on the basic idea of sampling, even if just a continuous waveform overlaid with a stepped waveform. Seriously. As it is now, it just starts out with a discussion of resolution and the reader who knows nothing of A/D converters (and this is probably a large proportion of people who would be using Wikipedia to read about them) is rudely thrown into a discussion of resolution. Resolution of what, he/she might ask, before clicking on to something else more interesting. — Preceding unsigned comment added by Oscarruitt ( talk • contribs) 03:56, 31 July 2012 (UTC)
Teapeat ( talk · contribs) has added a brief section on metastability. I don't have access to the ref provided. That ref appears to be narrowly focused. Here is a more general discussion of metastability issues in ADCs [1]. -— Kvng 05:17, 29 March 2013 (UTC)
"One effective bit of resolution changes the signal-to-noise ratio of the digitized signal by 6 dB" -- wouldn't it be 3 dB? That is 10 log_10(2). — Preceding unsigned comment added by 91.137.20.132 ( talk) 18:20, 20 March 2014 (UTC)
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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)
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
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)
That formula is wrong...Q = (span)/(number_of_levels) ... —Preceding unsigned comment added by 87.16.20.206 ( talk) 10:34, 3 May 2010 (UTC)
Every literature says that Q = (span) / (number_of_levels). Mistake is because you consider that max voltage can read from ADC is Vref (and not Vref -1LSB). Then, for your example, when you have and ADC with a resolution of 1bit and a full range of 0..1V, you have 1LSB=1/2=0.5V -So the voltage that corresponds to D=1 is 0.5V (of course always +/- 1/2LSB in ideal case) not 1V Sorry for my english -- Frank Rossi —Preceding unsigned comment added by 87.16.20.206 ( talk) 14:12, 3 May 2010 (UTC)
The analog value represented by the all one codes is VREF -1LSB , so with 1bit of resolution and VREF=1V you have 1LSB = 1/2 = .5V and infact VREF - 1LSB= (1-.5) =.5V while,according your formula this "all-one-codes" value would be 1V and not .5V (source: http://www.analog.com/library/analogDialogue/archives/39-06/Chapter%202%20Sampled%20Data%20Systems%20F.pdf - page 2.4) -- Dontronix ( talk) 07:15, 5 May 2010 (UTC)
Have a look at: http://www.national.com/appinfo/adc/files/ABCs_of_ADCs.pdf too. -- Dontronix ( talk) 07:56, 5 May 2010 (UTC)
Regarding discussion on number of levels in an 8 bit A/D or D/A. The first coded level is zero. Therefore there are (256-1)or 255steps. The word steps is often used in A/D, D/A talk and is less confusing terminology than talking about intervals. — Preceding unsigned comment added by Alan Bate SEA ( talk • contribs) 13:47, 22 February 2012 (UTC)
The use of 2^M-1 is incorrect. Even the source that is used for defining the resolution in bits uses 2^M [1]. Another way to think of A/D conversion is the division of the full-scale range in half where a comparator is used to determine if the MSB is 1 or 0; then the next range is divided in half again, and again, until M times; thus the full-scale range is divided a total of 2^M times to give a final resolution size of the full-scale range divided by 2^M. 75.174.53.165 ( talk) 05:13, 24 January 2017 (UTC)
I just did some research into this, because I had the same issue myself when I saw the formulation in practice. Range/2^n is correct. The reason is that each bit doesn't represent a discrete analog value. It represents a small range of value and generally sets at the center of that range. For instance, if you're converting 3-bit to a range of 0-8 volts, then: 000 -> 0V-1V (0.5V) 001 -> 1V-2V (1.5V) 010 -> 2V-3V (2.5V) 011 -> 3V-4V (3.5V) 100 -> 4V-5V (4.5V) 101 -> 5V-6V (5.5V) 110 -> 6V-7V (6.5V) 111 -> 7V-8V (7.5V)
LordQwert ( talk) 17:21, 13 March 2017 (UTC)
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