OK, but that's not "smoothing", that's just low-pass filtering, which ideally removes ALL effects of the "stair-steps".
However, here we are not talking about analog processing BEHIND the DAC, we talk about digital processing BEFORE the DAC. And here "smoothing" is not simple and any kind of arithmetic "interpolation" doesn't necessarily have to create a "better" reproduction but instead - just as likely - can create a worse one. Look at these images: 14972 Here, for a low sampling frequency, e.g. 44.1 kHz, there are two samples A and B which - just looking at these two samples, might result in a waveform as the one shown. Now if you go to a higher sampling frequency, the usual process of oversampling would just duplicate the samples A and B resulting in A' and B'. A resulting waveform then might be as shown in the second image which looks much worse. So one might be tempted do calculate some "intermediate" values A" and B" and hope that the result might more closely match. Now here's the problem: unless you've looked at the whole signal (which you can't do due to time and processing requirements), you don't know whether the first waveform given above is correct, it could just as well be one like this: 14973 What you see here is that now your "intermediate" steps A" and B" are a much worse match for the original waveform than A' and B' - you get more noise. Now, what might come as a surprise is that all of this does NOT matter at all as long as in the end you filter at your cutoff frequency again. The reason for this is that all the noise and distortion added above actually happens in higher frequency ranges, it's only in harmonics to the original waveform. Since these higher frequencies are inaudible (as in: definitely, completely, undisputedly inaudible; not even a bat will hear them), you can filter them and the result of this is, again, the perfect copy of the first waveform. For BOTH of the right-hand side curves. Actually this would still hold if you throw in random samples in place of A', A", B' and B". This is what you do with oversampling. The reason for this is that the filters you use to do this are not perfect. They will either distort your audible signal or not perfectly filter out the higher frequencies but they get better the higher the sampling frequencies you use are. So you accept the added noise due to oversampling because the end result will still be less distorted due to simpler and better filters. Now what John is talking about is something else. The problem with these images above is that while the anaolg low-pass filter behind the DAC can perfectly reproduce the signal, the same unfortunately is NOT true of the DAC itself. The DAC itself is a filter, too, and depending on the waveforms above those higher frequencies CAN actually have an impact on the way the DAC reproduces you AUDIBLE signal (imaging). To avoid this, you usually try to filter your signal even BEFORE it reaches the DAC to avoid having your artifacts distort your signal. These are the filters John and I are talking about. And the question it all boils down to is: what makes the filter in sox better than the ones in the DAC? Sox is a very simple interpolation software so if it can do better: why don't the DAC makers just do the same thing? +-------------------------------------------------------------------+ |Filename: waveform2.jpg | |Download: http://forums.slimdevices.com/attachment.php?attachmentid=14973| +-------------------------------------------------------------------+ --- learn more about iPeng, the iPhone and iPad remote for the Squeezebox and *New: Logitech UE Smart Radio* as well as iPeng Party, the free Party-App, at penguinlovesmusic.com ------------------------------------------------------------------------ pippin's Profile: http://forums.slimdevices.com/member.php?userid=13777 View this thread: http://forums.slimdevices.com/showthread.php?t=99088 _______________________________________________ audiophiles mailing list [email protected] http://lists.slimdevices.com/mailman/listinfo/audiophiles
