On Fri, 21 Dec 2007 16:19:55 -0500, Jerry Avins <jya@ieee.org> wrote:>Eric Jacobsen wrote: >> On Fri, 21 Dec 2007 11:37:01 -0500, Jerry Avins <jya@ieee.org> wrote: >> >>> Eric Jacobsen wrote: >>>> On Fri, 21 Dec 2007 09:47:03 -0600, Vladimir Vassilevsky >>>> <antispam_bogus@hotmail.com> wrote: >>>> >>>>> Jerry Avins wrote: >>>>> >>>>>> There are commercial 96KHz systems. There are technical reasons that >>>>>> make it desirable to exceed 48KHz for some processing operations, but >>>>>> none for reproduction. >>>>> If you look at the output spectrum of an audio DAC, you will see the >>>>> huge amount of noise at the frequencies above the sample rate. The >>>>> fairly decent filter is required to get rid of that. The noise is the >>>>> artifact of the noise shaping. It can possibly affect the quality due to >>>>> the nonlinear effects, and it causes the EMC problems, too. But what is >>>>> more important this noise shows up on the A-curve noise measurements, >>>>> spoiling the otherwise nice figures of SINAD. So there is some sense in >>>>> using the higher sample rates. >>>> That makes pretty good sense, actually. Use a high output sample >>>> rate so that you have some unused spectrum to which to move the shaped >>>> noise, then remove that with the reconstruction filter. >>> I don't get it. The noise before filtering always goes above the sample >>> rate, and the reconstruction filter is designed to remove it. A >>> higher-than-needed sample rate allows some noise to be lower than the >>> sample rate, but it needs to be removed anyway. What's the advantage for >>> playback? >>> >>> Jerry >> >> How do you move the shaped noise above the sample rate without it also >> being within the Nyquist region? >> >> As has been discussed, the higher the output sample rate the more room >> there is spectrally for the shaped noise and filter transition band. > >I understand why getting more bits worth of significance needs faster >sampling -- there's more than one way to make that trade -- but once the >low-noise signal is acquired, why keep the superfluous samples? > >JerryI think we're talking about a DAC application here and not an ADC. The increase in the sample rate before DAC conversion provides some benefits in noise shaping and reconstruction filter design. That can be done regardless of the original sample rate, e.g., after upsampling. There seems to be a disconnect here of some kind, because I know you know that. ;) Eric Jacobsen Minister of Algorithms Abineau Communications http://www.ericjacobsen.org
Oversampling
Started by ●December 21, 2007
Reply by ●December 22, 20072007-12-22
Reply by ●December 22, 20072007-12-22
Eric Jacobsen wrote: ...> There seems to be a disconnect here of some kind, because I know you > know that. ;)Yeah. Actually, the OP wants to sample fast in order to localize certain events in time. I told him that merely oversampling won't do more than interpolating after the signal is captured, but that using a higher cutoff in his anti-alias filter and sampling accordingly would indeed help. That's sampling faster, but not oversampling in the usual sense. If you can suggest better, I'm sure he would appreciate it. Jerry -- Engineering is the art of making what you want from things you can get. �����������������������������������������������������������������������
Reply by ●December 22, 20072007-12-22
On Sat, 22 Dec 2007 11:13:01 -0500, Jerry Avins <jya@ieee.org> wrote:>Eric Jacobsen wrote: > > ... > >> There seems to be a disconnect here of some kind, because I know you >> know that. ;) > >Yeah. Actually, the OP wants to sample fast in order to localize certain >events in time. I told him that merely oversampling won't do more than >interpolating after the signal is captured, but that using a higher >cutoff in his anti-alias filter and sampling accordingly would indeed >help. That's sampling faster, but not oversampling in the usual sense. >If you can suggest better, I'm sure he would appreciate it. > >JerryWe were responding to different things, then. Eric Jacobsen Minister of Algorithms Abineau Communications http://www.ericjacobsen.org
Reply by ●December 22, 20072007-12-22
George wrote:> > True about not adding more usable data. But in this application what we're > looking for is accurate measurement of the time when the analog waveform > satisfies a certain condition. The more frequently we sample it, the more > accurately we can time-tag the input condition when it occurs. We'd like to > obtain an accuracy of better than the sampling interval at 352.8 > ksamples/sec. (Hope I'm not missing something here.)You might be missing something. It all depends on what "measurement of the time when the analog waveform satisfies a certain condition" means. If that waveform is fairly low frequency sampling at an absurdly high frequency may not give you better accuracy in making your measurement and could even make it less accurate under certain conditions. OTOH if your so called waveform is some sort of sudden event like a step or a pulse that has high frequency components then the higher sampling rate might make sense. Without any explanation of the details the answer to your question is going to be of the form that your elephant is a wall or rope or tree trunk. -jim> > How much of a processing load is created at this speed still has to be > determined however. The integer ratio isn't important - was just using it > to be brief.
