Post by thread:Using sound card as signal analyzer [OT]

Hello, -sorry for the OT- Looking for an easy way to enter analog data's in my pc, I found that the best system would be to use an existing sound card (cheap, reliable, fast conversion, 16 bit, 44 khz...). This gives me a lot of questions : -I'd like to multiplex the signal : can I use a 4066 to switch quickly between differents inputs and sample the signal each time I change inputs? Is the 4066 fast enough and noise free? (The signals will be in the audible arera since I'm using a sound card) -I want to make the soft working in windows (with borland delphi), because I'd like to be able to use any sound card. Does anyone has pointers to "api programming of windows soundcard inputs"? How to read a single sample from win api's? Thanks for the help! Philippe ------------------------------------------------------------------------- Simple yet usefull Robot stuff goto http://users.swing.be/philippe.jadin Votre site internet en un clin d'oeil : http://clairetnet.cjb.net Belgium, Europe -------------------------------------------------------------------------

>Looking for an easy way to enter analog data's in my pc, I found that >the best system would be to use an existing sound card (cheap, reliable, >fast conversion, 16 bit, 44 khz...). This gives me a lot of questions : >-I'd like to multiplex the signal : can I use a 4066 to switch quickly >between differents inputs and sample the signal each time I change >inputs? Is the 4066 fast enough and noise free? (The signals will be in >the audible arera since I'm using a sound card) Phillipe - IIRC, the most audio ADC's now use inexpensive delta-sigma converters. These converters measure CHANGES in the input signal, and can only accomodate a small change each sample. That means that if you have a sudden large change (as you might have when switching between two input channels), the converter may take quite a few samples before catching up to the change. I don't know how many, but if you want to persue this approach, you'd better find out. The good news is that the 4066 will have no problem switching fast enough to do what you want. Another thing to watch out for in audio ADC's is that they don't really care about absolute accuracy. A gain error of 10% wouldn't really matter, as you can just adjust the volume... Don

Don, What is your source for this info on sigma-delta converters? I am currently working on a project where I am using sigma-delta ADCs and AFAIK, they are not cheap , but really high accuracy (although, I suppose that making 16 or 24 bit R2R ladder ADCs would be more expensive). IIRC, They work by having a capacitor attached to one input of a comparator. The other input is the input to the ADC. If the voltage on the cap is lower than the main input voltage, a short pulse is fed to the cap thru a resistor to increase its voltage. If it is too high, the cap is allowed to discharge for a short bit through a resistor. The pulses fed to the cap are considered to be a fast bitstream,and their average over a thousand or so bits is computed, giving the average value of the input. So, the converter actually "samples" the input a thousand or more times per actual code output. Usually the output rate is around 100Hz and the 1-bit sampling rate is around 20-100kHz. So, if the input were to change suddenly between output codes, it should be able to handle it at least within the converter's error specs. In fact, one of the unusual things about sigma-delta converters is that they often do not require an input anti-aliasing filter (or a very minimal one) because they sample MUCH faster than the output rate. If you try to feed them a signal which is too fast (say, a 1 kHz signal when the output rate is 60Hz) it will automatically be attenuated by the averaging of the bitstream. You only need to ensure that the input doesn't have significant frequency content comparable (in frequency) to half the fast 1-bit sampling rate. Sean At 09:29 AM 2/5/00 -0500, you wrote: {Quote hidden}>Phillipe - > >IIRC, the most audio ADC's now use inexpensive delta-sigma converters. >These converters measure CHANGES in the input signal, and can only >accomodate a small change each sample. That means that if you have a >sudden large change (as you might have when switching between two input >channels), the converter may take quite a few samples before catching up >to the change. I don't know how many, but if you want to persue this >approach, you'd better find out. >Don > | | Sean Breheny | Amateur Radio Callsign: KA3YXM | Electrical Engineering Student \--------------=---------------- Save lives, please look at http://www.all.org Personal page: http://www.people.cornell.edu/pages/shb7 spam_OUTshb7TakeThisOuTcornell.edu ICQ #: 3329174

>I am currently working on a project where I am using sigma-delta ADCs and >AFAIK, they are not cheap A Sigma Delta converter is about as cheap as you can get for any given level of accuracy. As the requisite accuracy increases cost goes up due to the need for precision in fabrication and elimination of scond (& 3rd and 4th ....) order affects. > but really high accuracy (although, I suppose >that making 16 or 24 bit R2R ladder ADCs would be more expensive). Yes, not mainly because of the number of components but because of the required precision of some of them and the need to eliminate undesired affects in more areas. >IIRC,They work by having a capacitor attached to one input of a comparator. The >other input is the input to the ADC. If the voltage on the cap is lower >than the main input voltage, a short pulse is fed to the cap thru a >resistor to increase its voltage. If it is too high, the cap is allowed to >discharge for a short bit through a resistor. The pulses fed to the cap are >considered to be a fast bitstream,and their average over a thousand or so >bits is computed, giving the average value of the input. This is approximately correct. Usually the resistor which drives the capacitor is ALWAYS either being driven high or low - not just in short bursts. Decisions as to whether the capacitor is above the input level (so that the capacitor should be being driven down) or below the input level (when the capacitor should be being drivben "up") are made on a clocked basis. If the transistion occurs between clock pulses the capacitor will continue to be driven in the "wrong" direction for the remainder of the clock cycle. This leads essentially to "pulses" similar to what you described but they occur for the whole of the clock period and effectively the capacitor is always being driven. >So, the converter actually "samples" the input a thousand or more times per >actual code output. Usually the output rate is around 100Hz and the 1-bit >sampling rate is around 20-100kHz. Very roughly the accuracy which can be obtained is proportional to the number of samples. In practice it will be somewhat less. For a 2000 sample conversion you could "hope" to achieve approaching 1 in 2000 accuracy = 11 bits. In practice you wouldn't. I'm getting around 9 or 10 bits at 2000 samples. YMMV (actually YMWV). Accuracy varies depending on actual input level and various other somewhat arcane factors. Capacitor size is more critical than may at first appear. Obviously thedoubling in samples required every time accuracy increases by 1 bit makes high accuracy SD a slowish process. Harris Semiconductors do a nice introduction in Sigma Delta in their AN9504. The LTC2400 from Linear Technology is a 24 bit (! !! !!!!!!!!!) SD converter at a reasonable price considering its accuracy. regards Russell McMahon _____________________________ >From other worlds - http://www.easttimor.com http://www.sudan.com What can one man* do? Help the hungry at no cost to yourself! at http://www.thehungersite.com/ (* - or woman, child or internet enabled intelligent entity :-))

Sean - Russel deftly pointed out that sigma-delta converters are generally the cheapest for a given resolution. However, there's still a big difference between the converters you are familiar with and the ones used in sound cards. Audio applications require high resolution and high linearity, and sample rates in the tens of kHz (note lower case 'k' <G>), but they don't require high accuracy. They don't require any accuracy at all at DC. I suspect that the converters you are using are for instrumentation, and therefor exhibit high accuracy, particularly at DC, but have sample rates in the tens or hundreds of Hz. While these two applications (audio and instrumentaion) may be suited to converters of similar functional design, the requirements of the particular components are radically different, giving rise to very different prices. Another thing affecting price is production volume. Audio converters are made by the tens or hundreds of millions, while instrumentation converters are made by the tens or hundreds of thousands or less. Since you got my curiosity up, I did a web search for "delta sigma converter application". The very first hit was the URL shown below. It links to a page from Crystal Semiconductor division of Cirrus, which has a very informative app note. http://www.edtn.com/scribe/reference/appnotes/md0071fb.htm Don