Philip.Belben(a)pgen.com wrote: After you do the D/A, you have a huge amount of sampling noise at Fs/2 (and its harmonics). For a CD player, Fs is 44.1 KHz, so the sampling noise is at 22.05 KHz. Since the desired audio frequency response extends to 20 KHz, a non-oversampled player needs "brick wall" analog filters, with a pass band to 20 KHz, and a stop band starting below 22.05 KHz. Digital oversampling is used to move the sampling noise to a higher frequency. 4x oversampling moves it to 88.2 KHz. The oversampling filter is a low-pass filter, so the audio content still has a frequency response to somewhere above 20 KHz. But now your analog filter can have a much shallower slope, which is easier to implement and introduces much less phase distortion. Your analog oversampling scheme does not eliminate the noise at Fs/2, and in fact introduces more sampling noise at higher frequencies. It does not help reduce the requirements of the analog filter. Further notes on oversampling: The digital oversampling filter can effectively implement a tradeoff between the quantization and time domains. For instance, many early CD players used 14-bit D/A converters with 4x oversampling. It can be shown that this does not lose any of the data on the disc, and ignoring the analog filtering issues, is fully equivalent to a non-oversampled 16-bit D/A. But due to two factors, the 4x 14-bit converter actually produces a *more* accurate reproduction than the 1x 16-bit converter: 1) The reduced requirements on the analog filter, 2) It is easier to get precision in the time domain (using a crystal oscillator clock source) than in the quantization. D/A converters have inherent problems with linearity of output across changes at power-of-two boundaries. This is most noticable at the midpoint (or zero) crossing, where it is crucial that a change of the most significant bit must have exactly the same weighting as the change of all of the lower order bits plus one LSB. Most audiophiles seemed not to believe it, and thought that it was important to get 16-bit converters, even if they weren't oversampled. Of course, a 4x 16-bit converter is even better than a 4x 14-bit converter. The logical extreme of oversampling is to use such a high factor that you only need one bit of data. This is called a delta-sigma D/A converter. For 16-bit source data this would require at least 65536x oversampling, but there are some tricks to reduce this to e.g. 128x. This is the basis of the so-called "1-bit" D/A converters. The advantage is that you only need a single-pole analog filter (and in practice can get by without even that), and that the tradeoff from quantization to time domain has been maximized, which is good because it is much easier to get high precision in the time domain as described above. Note that many of the same audiophiles that thought that they were being cheated by 4x 14-bit converters have been completely "taken in" by 1-bit converters, even though their own reasoning should suggest that they are being cheated 7.5 times as much as by the 4x 14-bit converter.