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Nyquist theorem and Sampling rate of signals (Newbie)

Started by Deamon October 29, 2010
On 10/30/2010 07:25 AM, Roman Rumian wrote:
> 2010-10-29 18:23, Tim Wescott wrote: > (...) >> If the other links that you've gotten haven't helped, read this: >> http://www.wescottdesign.com/articles/Sampling/sampling.html > > thank you for this article. > These days we were preparing new audio engineering lab examples, and > have noticed and checked that EVERY audio A/D converter from Cirrus, > AKM, Analog Devices, TI and Wolfson has digital decimation filter > (probably half band) introducing aliasing. For normalized Fs/2 frequency > each has attenuation above the level of quantization noise, so for > Fs=48kHz we can easily observe aliasing for frequencies up to 26 kHz. > This aliasing effect starts at 22kHz, not audible for humans, but why > not use filter eliminating aliasing definitively ?
In a general purpose ADC it's because that may not be what the system designer wants -- certainly in the case of an ADC in a close-loop control system, the system designer is going to be very twitchy about phase delay artifacts from anti-aliasing filters, and in other systems the system designer may be intentionally subsampling, and the alias may be exactly the signal that he wants to pass through the ADC. For purpose-designed audio ADCs this becomes more smoky, because presumably the ADCs are designed for systems where anti-aliasing is always going to happen. I can think of a few reasons not to include them in the chip, though: One, because it's hard to make really good mixed-signal chips, and from a system design perspective it may be cheaper to use active filtering before the ADC that use parts from an all-analog process. Two, because every audio system designer is going to have his own opinion about the best anti-alias filtering, and they're going to view anything that you do inside your ADC as "getting in the way". Three, no one's thought of it (or it's only just now becoming practical), and next year that's all you'll be able to get. -- Tim Wescott Wescott Design Services http://www.wescottdesign.com Do you need to implement control loops in software? "Applied Control Theory for Embedded Systems" was written for you. See details at http://www.wescottdesign.com/actfes/actfes.html

Roman Rumian wrote:

> 2010-10-29 18:23, Tim Wescott wrote: > (...) > >> If the other links that you've gotten haven't helped, read this: >> http://www.wescottdesign.com/articles/Sampling/sampling.html > > > thank you for this article. > These days we were preparing new audio engineering lab examples, and > have noticed and checked that EVERY audio A/D converter from Cirrus, > AKM, Analog Devices, TI and Wolfson has digital decimation filter > (probably half band) introducing aliasing. For normalized Fs/2 frequency > each has attenuation above the level of quantization noise, so for > Fs=48kHz we can easily observe aliasing for frequencies up to 26 kHz. > This aliasing effect starts at 22kHz, not audible for humans, but why > not use filter eliminating aliasing definitively ?
The aliasing in the inaudible area is unimportant, especially as it does not affect the specmanship. Eliminating this aliasing will cost more taps of the filter; that could have noticeable impact on the power consumption and the cost of silicon. There is the other problem with delta sigma ADCs: aliasing from the vicinity of the frequency of the modulator (several MHz typ.). There is not much of attenuation there; only a basic RC lowpass. This can create serious problems with EMI succeptibility. Vladimir Vassilevsky DSP and Mixed Signal Design Consultant http://www.abvolt.com
Thanks Guys it was quite trashed out. I loved the analogy of the
string and the number of modes best. Thanks for the help. All the
links also helped .

> > >Roman Rumian wrote: > >> 2010-10-29 18:23, Tim Wescott wrote: >> (...) >> >>> If the other links that you've gotten haven't helped, read this: >>> http://www.wescottdesign.com/articles/Sampling/sampling.html >> >> >> thank you for this article. >> These days we were preparing new audio engineering lab examples, and >> have noticed and checked that EVERY audio A/D converter from Cirrus, >> AKM, Analog Devices, TI and Wolfson has digital decimation filter >> (probably half band) introducing aliasing. For normalized Fs/2 frequency
>> each has attenuation above the level of quantization noise, so for >> Fs=48kHz we can easily observe aliasing for frequencies up to 26 kHz. >> This aliasing effect starts at 22kHz, not audible for humans, but why >> not use filter eliminating aliasing definitively ? > >The aliasing in the inaudible area is unimportant, especially as it does >not affect the specmanship. Eliminating this aliasing will cost more >taps of the filter; that could have noticeable impact on the power >consumption and the cost of silicon. > >There is the other problem with delta sigma ADCs: aliasing from the >vicinity of the frequency of the modulator (several MHz typ.). There is >not much of attenuation there; only a basic RC lowpass. This can create >serious problems with EMI succeptibility.
You need to take care with those RC filters. Assuming the ADC has a differential input, you filter with two resistors in the leads, and one capacitor across them. Then you use 2 capacitors of some small enough value to still have reasonable RF properties - maybe 22pF or 33pF - to ground from the two leads. Make sure the ground side of these two capacitor is grounded at the same point in space, to avoid injecting differential noise. Now you have a simple RC filter that rolls off the response nicely, before the horrors at the modulator frequency, and which also has few problems with EMI susceptibility. Steve
On Oct 29, 5:23&#2013266080;pm, Tim Wescott <t...@seemywebsite.com> wrote:
> On 10/29/2010 12:26 AM, Deamon wrote: > > > Please I am a complete newbie in Signal analysis and modelling and I > > am quite confused &#2013266080;about the use of &#2013266080;Nyquist theorem in sampling .Does > > this Nyquist criterion affect the &#2013266080;rate at which data is transferred > > that is &#2013266080;the data rate ? > > No more than quantum theory affects how electrons behave in a > semiconductor -- it's rather the other way around. > > > I have &#2013266080;read the wikiopedia saying that I > > must sample at twice the bandwidth to be able to recontruct a signal > > perfectly But I don't understand the concept . If I sample at 2B then > > it is bigger than the wave itself . > > If the other links that you've gotten haven't helped, read this:http://www.wescottdesign.com/articles/Sampling/sampling.html > > I believe it shows pretty early on why you need to sample at over 2B to > completely capture a signal.
Don't know how early it appears in the article (I didn't browse the whole thing) but the key is figure 2. Understand that figure, and you understand the sampling theorem. Rune

steveu wrote:
>> >>Roman Rumian wrote: >> >> >>>2010-10-29 18:23, Tim Wescott wrote: >>>(...) >>> >>> >>>>If the other links that you've gotten haven't helped, read this: >>>>http://www.wescottdesign.com/articles/Sampling/sampling.html >>> >>> >>>thank you for this article. >>>These days we were preparing new audio engineering lab examples, and >>>have noticed and checked that EVERY audio A/D converter from Cirrus, >>>AKM, Analog Devices, TI and Wolfson has digital decimation filter >>>(probably half band) introducing aliasing. For normalized Fs/2 frequency > > >>>each has attenuation above the level of quantization noise, so for >>>Fs=48kHz we can easily observe aliasing for frequencies up to 26 kHz. >>>This aliasing effect starts at 22kHz, not audible for humans, but why >>>not use filter eliminating aliasing definitively ? >> >>The aliasing in the inaudible area is unimportant, especially as it does >>not affect the specmanship. Eliminating this aliasing will cost more >>taps of the filter; that could have noticeable impact on the power >>consumption and the cost of silicon. >> >>There is the other problem with delta sigma ADCs: aliasing from the >>vicinity of the frequency of the modulator (several MHz typ.). There is >>not much of attenuation there; only a basic RC lowpass. This can create >>serious problems with EMI succeptibility. > > > You need to take care with those RC filters. Assuming the ADC has a > differential input, you filter with two resistors in the leads, and one > capacitor across them. Then you use 2 capacitors of some small enough value > to still have reasonable RF properties - maybe 22pF or 33pF - to ground > from the two leads. Make sure the ground side of these two capacitor is > grounded at the same point in space, to avoid injecting differential noise. > Now you have a simple RC filter that rolls off the response nicely, before > the horrors at the modulator frequency, and which also has few problems > with EMI susceptibility.
The 1st order filter will make for only ~50dB of alias attenuation at the very best case. I had to make 2nd order filters. Vladimir Vassilevsky DSP and Mixed Signal Design Consultant http://www.abvolt.com