i answered my own stupid question with a little research online. please forgive my ignorance....
sincerely, mentally challanged josh
sincerely, mentally challanged josh
Josh, you should have posted your findings. There is still a lot of discussion about the subject. We have been talking about this a bit on the http://www.newaudiosociety.com site for a while. Let us know what truth you arrived at.
Ok. From what I have gathered I can surmise that the when the digitial signal is upsamapled to a higher frequency the converter can extract more data by interpolating ( using some sort of algorithm i assume ) which will give a smoother/fuller sound, closer to the original recording. But, what is the algorithm used?
It's been a while, but I think it goes like this ....
Upsampling does not increase the detail, nor does it increase the bandwidth of the signal, but it alters the sampling frequency, and so it pushes the aliased signals (see Nyquist's theory of sampled signal) to higher frequencies.
This allows the anti-aliasing low-pass filter to have a slower rolloff. It is easier to design a filter with linear phase and lower passband ripple if the rolloff is less steep.
So the whole point of upsampling was to simplify the anti-aliasing low pass filter, so that it causes less degradation to audible frequencies.
Performing any kind of interpolation would be akin to doing some of the filtering in the digital domain. I believe the interpolation bit may be the difference between oversampling (old technology) and upsampling, but I'm not exactly sure.
Funny you should ask about the algorithm. I was thinking about this last night (because I really have no life !)
A perfect CD player would have an anti-aliasing filter that was a perfect brick wall. Below 22kHz signals would be completely untouched, and above 22kHz would be infinite attenuation.
In the time domain this equates to a sin(x)/x impulse response.
So my guess is that a good place to start for the interpolation algorithm would be the sin(x)/x waveform.
I'm not sure of the science, but it works. I believe that Seandtaylor99 above does a commendable job of an explanation. I recently installed an upsampler on a CD player (AH! 4000) that I was already familiar with and pleased with. The change was dramatic, though it's been taking a while to break in.
my wife would get it because we're both engineers ! In fact we often have nerd type conversations regarding audio, even though she's not interested in hifi, and she calls me a geek whenever I tinker with my system.
By the way I've never known the difference between upsampling and oversampling .... I suspect that they may be the same thing, but that the marketroids decided that upsampling is a more marketable term.
As an engineer I must say .... "Damn those marketroids !".
See Goldy! Not a stupid question after all. The current top of the line Zanden Dac/transport retails for $43,440.00. It doesn't up sample or over sample. Straight red-book only. Seantaylor99 mentions a filter and brick wall. The Zanden claimes better sound with out this filter and brick wall.
It's all above my head. I just listen to what sounds best to me in my price range and go for it.
As far as the audio enthusiast is concerned there is no difference between oversampling & upsampling.
However, from an engineering perspective, there is a technical difference: oversampling is a repeat operation where the input data is simply read at the higher rate again & again (i.e. oversampling ratio). As you well know, this creates an aliased signal that repeats every Fs. In upsampling we zero-stuff & later use a digital filter to estimate what the values of the zero-stuffed samples should be. In this case, too, the operation creates aliased signals that repeat every Fs. So, in both cases we need a digital filter to attenuate the aliased signal & preserve the audio spectrum. This digital filter can be the same for oversampling & upsampling! Thus, these 2 operations really look very similar. FWIW.