Reply by Jon Harris April 12, 20052005-04-12
"Al Clark" <dsp@danvillesignal.com> wrote in message
news:Xns9636B5ED48E0Baclarkdanvillesignal@66.133.129.71...
> Andre <no_spam@fischer-zoth.de> wrote in news:d3gkoo$57c$02$1@news.t- > online.com: > > > Additionally, you might have to deal with delays. Typical delay times > > between analog input and digital output of delta-sigma ADs are around > 15 > > samples! This might make synching mux timing and sampled data hard to > do. > > > > This delay is exactly why SAR converters are used in feedback systems. > > A very big advantage to sigma delta converters is that the analog > antialiasing filters are trivial since the actual sample rate is much > high than the "effective" sample rate. > > The internal digital filters may have stopband attenuation of 90dB at > something near 1/2 the "sample rate" This of course is a major reason for > the added delay. The better converters have better filters which have > more delay.
I have noticed a trend recently where some of the higher-end audio converters (especially DACs) are offering lower latency outputs, either standard, or as an option. Before, no one seemed to care too much or talk about it, but the manufacturers are starting to make a bigger deal about latency/group delay. Just thought I'd mention this.
Reply by Lasse Langwadt Christensen April 12, 20052005-04-12
John Larkin wrote:
> On Tue, 12 Apr 2005 10:33:59 -0500, Rob Gaddi > <rgaddi@bcm.YUMMYSPAMtmc.edu> wrote: > > >>Ville Voipio wrote: >> >>>Actually, it would be more precise to talk about "oversampling" >>>converters, because that's what makes the difference. Not the >>>actual converter topology or modulator order. >>> >>>- Ville >>> >> >>Not exactly. The pole in the feedback loop of a delta-sigma converter >>serves to take the nominally white quantization noise power and blue >>shift it into the higher frequencies, which if your mixed signal system >>is designed correctly will then be out of band from your signal, such >>that the quantization noise can be filtered and decimated out. This is >>in contrast to just taking say an SAR, oversampling by N, filtering, and >>decimating, which doesn't perform this noise shaping because it doesn't >>have the feedback. > > > > I've never understood that. You'll have to explain it to me some day. > > John >
the simple way: If you model the quantization as as adding white noise with a power proportional to the number of bits, you'll see that for a delta-sigma modulator that noise will show up at the output high-passed, the power is still the same but "pushed" towards higher frequencies. When you then lowpass and decimate you remove the higher frequencies and thus much of the the noise. quatization noise before lowpass and decimate: ^ | - - | / | / | / +==------+> f fs/2 quatization noise after lowpass and decimate: ^ | | | | +==+------> f fs/2 With a SAR the noise is flat at the output, so lowpass and decimate will remove less of the noise power. quatization noise before lowpass and decimate: ^ | | | |--------- +---------+> fs/2 quatization noise after lowpass and decimate: ^ | | | |-- +--+-----> f fs/2 -Lasse
Reply by Al Clark April 12, 20052005-04-12
Andre <no_spam@fischer-zoth.de> wrote in news:d3gkoo$57c$02$1@news.t-
online.com:

> Additionally, you might have to deal with delays. Typical delay times > between analog input and digital output of delta-sigma ADs are around
15
> samples! This might make synching mux timing and sampled data hard to
do.
>
This delay is exactly why SAR converters are used in feedback systems. A very big advantage to sigma delta converters is that the analog antialiasing filters are trivial since the actual sample rate is much high than the "effective" sample rate. The internal digital filters may have stopband attenuation of 90dB at something near 1/2 the "sample rate" This of course is a major reason for the added delay. The better converters have better filters which have more delay. You may find that low cost sigma delta converters are still a better choice than a muxed SAR. You can find 4 channel devices that smaple at 100kHz or 200kHz for reasonable cost. -- Al Clark Danville Signal Processing, Inc. -------------------------------------------------------------------- Purveyors of Fine DSP Hardware and other Cool Stuff Available at http://www.danvillesignal.com
Reply by John Woodgate April 12, 20052005-04-12
I read in sci.electronics.design that John Larkin 
<jjlarkin@highNOTlandTHIStechnologyPART.com> wrote (in 
<8j6o51t1hn361rf2usqr5bvtte41dhfmrp@4ax.com>) about 'Delta-sigma ADC 
question', on Tue, 12 Apr 2005:
>I've never understood that. You'll have to explain it to me some day.
Look for Audio Engineering Society papers by S Lipshitz and J Vanderkooy. If you are like me, you still won't understand it. (;-) -- Regards, John Woodgate, OOO - Own Opinions Only. There are two sides to every question, except 'What is a Moebius strip?' http://www.jmwa.demon.co.uk Also see http://www.isce.org.uk
Reply by John Woodgate April 12, 20052005-04-12
I read in sci.electronics.design that Randy Yates 
<randy.yates@sonyericsson.com> wrote (in 
<xxpfyxwb6cn.fsf@usrts005.corpusers.net>) about 'Delta-sigma ADC 
question', on Tue, 12 Apr 2005:

> Back in my old school/analog days, I was taught that settling time is >inversely proportional to bandwidth.
That applies to minimum-phase networks. Anything with a delay in it is in principle not minimum-phase and there is no general relation between settling time and bandwidth. -- Regards, John Woodgate, OOO - Own Opinions Only. There are two sides to every question, except 'What is a Moebius strip?' http://www.jmwa.demon.co.uk Also see http://www.isce.org.uk
Reply by Mark Borgerson April 12, 20052005-04-12
In article <xxpfyxwb6cn.fsf@usrts005.corpusers.net>, 
randy.yates@sonyericsson.com says...
> Hi Mark, > > Maybe you can enlighten me a bit here. Back in my old school/analog > days, I was taught that settling time is inversely proportional to > bandwidth. If I have a Fn Hz channel (from sampling at 2*Fn samples > per second), then why wouldn't the settling time of an input be the > same whether I used delta sigma or flash converter techniques? >
The simple explanation seems to be that there are internal digital filters that have to settle before the output is valid. The number and kind of internal filters determines the response to a step input on the signal. I'm sure someone else has (or will) explain in greater detail. If not, look into the data sheet on the CS5534 (a sigma-delta converter with multiplexe inputs).
> I've heard this flavor of argument for years (decades?) against > using delta sigma converters in multi-channel systems. It must > be true - the folks who have used them would know (I have not). But > as I've just queried, there's something that doesn't seem to add up, > in my view. >
If it was easy, no one would pay us the big bucks for solving these problems! ;-) <SNIP> Mark Borgerson
Reply by John Larkin April 12, 20052005-04-12
On Tue, 12 Apr 2005 10:33:59 -0500, Rob Gaddi
<rgaddi@bcm.YUMMYSPAMtmc.edu> wrote:

>Ville Voipio wrote: >> >> Actually, it would be more precise to talk about "oversampling" >> converters, because that's what makes the difference. Not the >> actual converter topology or modulator order. >> >> - Ville >> > >Not exactly. The pole in the feedback loop of a delta-sigma converter >serves to take the nominally white quantization noise power and blue >shift it into the higher frequencies, which if your mixed signal system >is designed correctly will then be out of band from your signal, such >that the quantization noise can be filtered and decimated out. This is >in contrast to just taking say an SAR, oversampling by N, filtering, and >decimating, which doesn't perform this noise shaping because it doesn't >have the feedback.
I've never understood that. You'll have to explain it to me some day. John
Reply by Jim Granville April 12, 20052005-04-12
Randy Yates wrote:
> Hi Mark, > > Maybe you can enlighten me a bit here. Back in my old school/analog > days, I was taught that settling time is inversely proportional to > bandwidth. If I have a Fn Hz channel (from sampling at 2*Fn samples > per second), then why wouldn't the settling time of an input be the > same whether I used delta sigma or flash converter techniques? > > I've heard this flavor of argument for years (decades?) against > using delta sigma converters in multi-channel systems. It must > be true - the folks who have used them would know (I have not). But > as I've just queried, there's something that doesn't seem to add up, > in my view.
The data sheets will usually show the settling time, and there ARE SD chips, designed for MUX drive. (I think Linear offer some?) As is typical in design, it is a trade off.... -jg
Reply by April 12, 20052005-04-12
OK.

--RY

CBFalconer <cbfalconer@yahoo.com> writes:

> Please don't toppost.
- Randy Yates Sony Ericsson Mobile Communications Research Triangle Park, NC, USA randy.yates@sonyericsson.com, 919-472-1124
Reply by Jerry Avins April 12, 20052005-04-12
CBFalconer wrote:
> ... The delta system generates a one > bit comparison, at a higher clock rate.
Then that gets low-pass filtered, smoothing the choppy result and adding precision -- extra bits -- by averaging. The filter adds delay and needs to be flushed and refilled whenever the MUX selects a new input. Jerry -- Engineering is the art of making what you want from things you can get. &#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;&#2013266095;