On Tue, Sep 25, 2007 at 10:37:17AM +0200, Fons Adriaensen wrote: > On Mon, Sep 24, 2007 at 09:41:39PM -0700, Maitland Vaughan-Turner wrote: > > > > > Intuitively, one could also say that more sample points yield a > > > > waveform that is closer to a continuous, analog waveform. Thus it > > > > sounds more analog. > > > > > > This is completely wrong. Sorry to be rude, but such a statement > > > only shows your lack of understanding. > > > > Why is it wrong? If I drew some dots on a waveform and then connected > > the dots, to try to reconstruct the waveform, wouldn't I get a better > > result with more dots? > > If you just connect the dots, yes. But that's not how an analog waveform > is reconstructed from PCM samples. 'Connecting the dots' is not even part > of that process. > > > > > Thanks for the link. My whole point of digging up this old thread > > > > though, was to say that I've tried it, and my ears tell me that the > > > > papers are incorrect. > > > > > > Then please point out the errors in the paper by Lipshitz and Vanderkooy. > > > > my ears tell me that... that's all; it's just subjective. haha, I see > > subjective reports don't get you far around here. > > You said the papers were incorrect. Then point out the errors. > And indeed, a subjective evaluation is useless if not the result > of a double blind test. > > If you have ever been involved in organising a controlled listening > test you should know how easy it is to get completely invalid results > and to fool yourself into believing things that are just an illusion. > > > > I'm not saying that DSD is crap. It sounds well. But it doesn't meet > > > the claims set for it (as shown by L&V - you need at least two bits > > > to have a 'linear' channel) and as a storage or transmission format > > > it's inefficient compared to PCM. That means that if you use PCM with > > > the same number of bits per second as used by DSD, you get a better > > > result than what DSD delivers. > > > > well, what do you mean by better? It seems like 24 bit is already > > better in terms of dynamic range at any sample rate, but if you mean > > more detailed representation of a waveform (in time), it seems like > > you necessarily need to have the highest possible sample rate. > > There is a point where more detail becomes irrelevant because it's > way below the noise. For a properly dithered PCM signal it can be > shown that the error that remains is in a strict mathematical sense > indistinguishable from noise. Hence if it's below the analog noise > floor it doeasn't matter any more. BUt you need at least _two_ bits > for this to work. > > For 'detail in time' the situation is even simpler. If the waveform > is bandlimited, then *every* detail is captured by sampling it at a > rate equal to twice the bandwidth. Each sample does not only represent > the waveform at the time it was taken, but in fact contains information > about the entire waveform. The samples completely describe the waveform > in the same strict sense as three points are sufficient to define a > circle. It's not an approximation.
Here is a good document on sampling theory, how it works perfectly in a mathematical sense, and how differences from ideal (lack of infinitely narrow impulses, ideal lowpass filters, quantization error) are made insignificant in practical implementations: http://www.lavryengineering.com/documents/Sampling_Theory.pdf _______________________________________________ Linux-audio-dev mailing list Linux-audio-dev@lists.linuxaudio.org http://lists.linuxaudio.org/mailman/listinfo.cgi/linux-audio-dev
