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From: Baron on 31 Jan 2010 14:37 CC Inscribed thus: > > "Bob Masta" <N0Spam(a)daqarta.com> wrote in message > news:4b658f7c.1184126(a)news.eternal-september.org... >> On Sun, 31 Jan 2010 12:55:50 +0200, Kari Laine >> <klaine8(a)gmail.com> wrote: >> >>>Hi, >>> >>>I am wondering how AD- and DA-converters are implemented. >>>Any documents about the chips internal configuration? >>>Would it be in theory possible to implement them with discrete >>>components? >> >> Indeed it would be possible, and of course that's >> how it was done originally. Nowadays chips are >> more practical for most applications, but simple >> pulse averaging D/As are still used on the outputs >> of microprocessors that don't have true D/As built >> in. >> >> On older systems with parallel ports, you can make >> a simple 8-bit D/A with a handful of resistors. >> See <http://www.daqarta.com/dw_ggdd.htm> for a >> discussion of simple D/A circuits. >> >> With a D/A in hand, you can turn it into an A/D >> via "successive approximation", where the system >> (computer, or internals of A/D chip) toggles the >> D/A bits as it compares the D/A output to the >> input signal. When they match close enough, that >> bit pattern is the converted A/D value. See >> <http://www.daqarta.com/dw_ggaa.htm> for more >> info. >> >> Best regards, >> >> >> >> > This post just got me thinking about the A/D concept. Just curious > about the basic concept (in 1 or 2 paragraphs). I think I understand > how the process works but just wondering if you sample a microphone > and store its value, that single sample represents the amplitude of > the wave at that time and the frequency is reconstructed by stringing > multiple samples together. > > I hope this is clear. I guess another way of saying this is how are > both the amplitude & freq derived from a single number (please keep it > simple). Google "Nyquist Criterion". -- Best Regards: Baron.
From: John Larkin on 31 Jan 2010 14:52 On Sun, 31 Jan 2010 12:28:09 -0500, "CC" <N(a)NE.nothing> wrote: > >"Bob Masta" <N0Spam(a)daqarta.com> wrote in message >news:4b658f7c.1184126(a)news.eternal-september.org... >> On Sun, 31 Jan 2010 12:55:50 +0200, Kari Laine >> <klaine8(a)gmail.com> wrote: >> >>>Hi, >>> >>>I am wondering how AD- and DA-converters are implemented. >>>Any documents about the chips internal configuration? >>>Would it be in theory possible to implement them with discrete components? >> >> Indeed it would be possible, and of course that's >> how it was done originally. Nowadays chips are >> more practical for most applications, but simple >> pulse averaging D/As are still used on the outputs >> of microprocessors that don't have true D/As built >> in. >> >> On older systems with parallel ports, you can make >> a simple 8-bit D/A with a handful of resistors. >> See <http://www.daqarta.com/dw_ggdd.htm> for a >> discussion of simple D/A circuits. >> >> With a D/A in hand, you can turn it into an A/D >> via "successive approximation", where the system >> (computer, or internals of A/D chip) toggles the >> D/A bits as it compares the D/A output to the >> input signal. When they match close enough, that >> bit pattern is the converted A/D value. See >> <http://www.daqarta.com/dw_ggaa.htm> for more >> info. >> >> Best regards, >> >> >> >> >This post just got me thinking about the A/D concept. Just curious about the >basic concept (in 1 or 2 paragraphs). I think I understand how the process >works but just wondering if you sample a microphone and store its value, >that single sample represents the amplitude of the wave at that time and the >frequency is reconstructed by stringing multiple samples together. Yes. In theory, to reconstruct the signal, you need to sample at a rate at least twice the highest frequency in the signal. In practise, 3x or 4x works well. The resulting ADC samples may look ratty to the eye, but if you later run them through a DAC and a lowpass filter, you can almost-perfectly reconstruct the original smooth signal. http://en.wikipedia.org/wiki/Sampling_theorem People who don't understand this post elaborate web pages proving that DVDs grossly distort music, which of course they don't. John
From: CC on 31 Jan 2010 14:59 "John Larkin" <jjlarkin(a)highNOTlandTHIStechnologyPART.com> wrote in message news:qfnbm5p34mi980l03fglbuf16s9i8hefl3(a)4ax.com... > On Sun, 31 Jan 2010 12:28:09 -0500, "CC" <N(a)NE.nothing> wrote: > >> >>"Bob Masta" <N0Spam(a)daqarta.com> wrote in message >>news:4b658f7c.1184126(a)news.eternal-september.org... >>> On Sun, 31 Jan 2010 12:55:50 +0200, Kari Laine >>> <klaine8(a)gmail.com> wrote: >>> >>>>Hi, >>>> >>>>I am wondering how AD- and DA-converters are implemented. >>>>Any documents about the chips internal configuration? >>>>Would it be in theory possible to implement them with discrete >>>>components? >>> >>> Indeed it would be possible, and of course that's >>> how it was done originally. Nowadays chips are >>> more practical for most applications, but simple >>> pulse averaging D/As are still used on the outputs >>> of microprocessors that don't have true D/As built >>> in. >>> >>> On older systems with parallel ports, you can make >>> a simple 8-bit D/A with a handful of resistors. >>> See <http://www.daqarta.com/dw_ggdd.htm> for a >>> discussion of simple D/A circuits. >>> >>> With a D/A in hand, you can turn it into an A/D >>> via "successive approximation", where the system >>> (computer, or internals of A/D chip) toggles the >>> D/A bits as it compares the D/A output to the >>> input signal. When they match close enough, that >>> bit pattern is the converted A/D value. See >>> <http://www.daqarta.com/dw_ggaa.htm> for more >>> info. >>> >>> Best regards, >>> >>> >>> >>> >>This post just got me thinking about the A/D concept. Just curious about >>the >>basic concept (in 1 or 2 paragraphs). I think I understand how the process >>works but just wondering if you sample a microphone and store its value, >>that single sample represents the amplitude of the wave at that time and >>the >>frequency is reconstructed by stringing multiple samples together. > > Yes. In theory, to reconstruct the signal, you need to sample at a > rate at least twice the highest frequency in the signal. In practise, > 3x or 4x works well. The resulting ADC samples may look ratty to the > eye, but if you later run them through a DAC and a lowpass filter, you > can almost-perfectly reconstruct the original smooth signal. > > http://en.wikipedia.org/wiki/Sampling_theorem > > People who don't understand this post elaborate web pages proving that > DVDs grossly distort music, which of course they don't. > > John > got it. thanks
From: whit3rd on 31 Jan 2010 15:23 > This post just got me thinking about the A/D concept. Just curious about the > basic concept (in 1 or 2 paragraphs). Hah! Excuse for a math digression! The vast majority of 'analog' signals are time-varying functions which are smooth (continuous and with continuous and well-defined finite time derivatives). That description excludes some kinds of phenomena, like truly uncorrelated 'white' noise (thus, we extremely pedantic types like to refer to 'pink' noise). One measurement of a signal isn't enough to tell its development in time, so a series of measurements is made; there are some kinds of signals (periodic repeating ones) where that series can be finite and still not 'miss' aspects of the underlying analog signal, thus we speak of the series as 'oversampled' if it suffices to determine the interesting part of the signal, and 'undersampled' if it does not. A nonperiodic signal can be packaged like a string of sausages into a periodic one. Then there are mathematical treatments (like Fourier analysis) that can represent a signal in multiple ways, either a series of timed voltage measurements or a spectrum of amplitude-and-phase of frequencies. There's an inversion theorem that says your 1000-measurements time series and the 500-frequencies-and- phases spectrum are interchangeable representations (holding the same information content). Then it gets complicated; the picture on your TV/video screen is a large array of changing hue/brightness/saturation triple quantities, and it too can be re-represented in lots of ways so a digital description can be squirted through the RF airwaves and reconstructed. The screen can be made into a mosaic of patches, each with some information that changes (and is retransmitted often) as well as other information that doesn't change (and is retransmitted less often). Patches with high spatial frequency (lots of detail) can be rendered with extra accuracy, patches with high time-variance can be given faster updates (but color accuracy becomes less important). Background stationary objects get the most color refinement but the slowest re-transmission of edge positions. The important thing to remember, is that this is all manipulation of measurements, of the kind of information that has an error estimate attached to it: as long as your re-doing of the info doesn't amplify the errors, it's 'legitimate' for whatever purpose drives you. The use of a logarithm scale for plotting a function, or a frequency scale for representing music, it's all just a mathematical rethinking of the same information, in possibly more useful forms.
From: Bob Masta on 1 Feb 2010 08:33
On Sun, 31 Jan 2010 17:52:01 +0200, Kari Laine <klaine8(a)gmail.com> wrote: >Bob Masta wrote: >> On Sun, 31 Jan 2010 12:55:50 +0200, Kari Laine >> <klaine8(a)gmail.com> wrote: >> Bob Masta >> >> DAQARTA v5.00 >> Data AcQuisition And Real-Time Analysis >> www.daqarta.com >> Scope, Spectrum, Spectrogram, Sound Level Meter >> Frequency Counter, FREE Signal Generator >> Pitch Track, Pitch-to-MIDI >> DaqMusic - FREE MUSIC, Forever! >> (Some assembly required) >> Science (and fun!) with your sound card! > >Thanks Bob! > >Your product is interesting. How about Linux version? Sorry, it's one of those "maybe some day" kinds of things. Most of the coding effort in a big project goes into the user interface, which would mean a major effort to convert from the Windows API to something for a Linux GUI. >My hobby at the moment is to write Linux software for the Velleman >PCSGU250 scope. It is not much yet. My math is poor so I have to study >little to be able to do the Fourier and sin(x)/x stuff. >I don't even yet to know what that sin(x)/x stuff is for... You might want to check out my "Gut-Level Fourier Transforms" series at <http://www.daqarta.com/author.htm>. That should provide you with the basics, though not the complete math to actually write a full FFT. You *could* do a Discrete Fourier Transform (DFT) with no more than this, but DFTs are pretty slow by comparison. I don't have a good Web link for basic FFT code, but I haven't looked. I learned about FFT code back in the days of paper, when dinosaurs walked the Earth. A good source, if it's still available, is Hal Chamberlin's "Musical Applications of Microprocessors", which contains explanations and complete working code for FFTs. The code is in old-style BASIC, but you can easily convert to your language of choice. Best regards, Bob Masta DAQARTA v5.00 Data AcQuisition And Real-Time Analysis www.daqarta.com Scope, Spectrum, Spectrogram, Sound Level Meter Frequency Counter, FREE Signal Generator Pitch Track, Pitch-to-MIDI DaqMusic - FREE MUSIC, Forever! (Some assembly required) Science (and fun!) with your sound card! |