Relay-Version: version B 2.10 5/3/83; site utzoo.UUCP Posting-Version: version B 2.10.1 6/24/83; site alice.UUCP Path: utzoo!watmath!clyde!burl!ulysses!allegra!alice!jj From: jj@alice.UUCP Newsgroups: net.analog,net.audio Subject: On CD oversampling, setting the record straight again. Message-ID: <5422@alice.uUCp> Date: Thu, 8-May-86 09:44:19 EDT Article-I.D.: alice.5422 Posted: Thu May 8 09:44:19 1986 Date-Received: Sat, 10-May-86 20:00:51 EDT Organization: New Jersey State Farm for the Terminally Bewildered Lines: 105 Xref: watmath net.analog:809 net.audio:8381 Todd at NBI says > There are a few misapprehensions involved in his article. > In a CD player utilizing digital filtering, the digital filter can be > considered the oversampling device (oversampler). The stream of 16 bit > ... > times by the digital filter such that the input stream of the digital > filter is formatted as follows (read right to left): > > W4 W4 W4 W4 W3 W3 W3 W3 W2 W2 W2 W2 W1 W1 W1 W1 > > **OR** > > Each word of the 44.1K stream is sampled one time and zeros are entered > as the remaining three samples for that given period. The digital filt- > er input stream would appear as follows: > > 0 0 0 W4 0 0 0 W3 0 0 0 W2 0 0 0 W1 > > I won't discuss much of the differences in results when using each of > these methouds. I don't know which of these is commonly used by manufac- > turers of CD players. I do prefer the first approach because it does give > some interpolation between inputted samples (in addition to the filtering Sorry to disappoint you, but the second approach is used. It provides a much simpler sin x / x correction, and also provides just as much interpolation after things are put through the filter (which is in the neighborhood of 96*4 samples long after interpolation). It's not just adjacent samples that are interpolated, it's a considerable history of the signal that's interpolated, and zeros are put between the known samples for several other reasons as well... > function) due to the implementation of the digital filtering. The digital > filter's output stream is at a 176.4 K rate. Some are at 2x instead of 4x > This process does not correct errors, but will help some in filtering > high frequency components of 'ticks' resulting from poor error conceal- Not really. Since the interpolation FILTER is flat across the entire original band (0-20K), no "audible" energy is filtered out of the click, so nothing is disguised! > ment. Unrecoverable errors (mis-read words off of disc that cannot be > corrected) must undergo error concealment attempting to substitute accep- > table values for missing words. Some methouds used in error concealment > are 1) muting (convert missing value to 0), 2) repeating [?] (convert > missing value to most recent correct value), 3) interpolation (substitute > missing value with the average of the previous and following values), > and 4) use of polynomial algorithm. The methouds of concealment are > listed in order from most to least likely in causing audible 'ticks'. > Error concealment is performed while the data stream is still at the > 44.1 K rate and before oversampling. True. > Although I do not know of application of this is any product, a data > interpolator could be considered an effective oversampling device. > Consider a case in which the interpolator takes four samples of each > word in the 44.1 K stream. The interpolator could take the mean > average of each group of four inputs from the effective inputted > 176.4 K rate. Note the following diagram illustrating this: > > W4 W4 W4 W4 W3 W3 W3 W3 W2 W2 W2 W2 W1 W1 W1 W1 > {M--------M}{J--------J}{G--------G}{D--------D}{A--------A} > ------N}{K--------K}{H--------H}{E--------E}{B--------B} > --O}{L--------L}{I--------I}{F--------F}{C--------C} > > A=(W1+W1+W1+W1)/4 > B=(W1+W1+W1+W2)/4 > C=(W1+W1+W2+W2)/4 > : > : > : This is a very simple interpolator, wiht a very simple filter, and very little filtering of frequency components above the original nyquist rate. One of the major reasons for digital interpolation is to make the analog filters simpler. This example wouldn't do it. > The algebraic expressions above represent the values of each word > outputted from the interpolator. Notice that there will be four > ... > wide, square impulse response. ___+---+___ As it's impulse response > is symetrical, no phase distortion will be introduced.) > > Oversampling in itself, does nothing for us. The oversampling > device is what gives us advantages. Oversampling digital filters as > well as interpolators can be considered to minimize quantitization > distortion (giving smoother transistions between any two originally > sampled values). Not minimize, but reduce. The fact that transitions are smoother (which isn't true anyhow) is unrelated. The reason that noise is reduced is that the quantization noise that does exist is spread over a wider bandwidth, and some of it is filtered out by the analog filters. Please, people, if you're going to make an attempt at explaination, be ACCURATE. Rabiner and Shaffer, or Rabiner and Gold, or Oppenheim and Shaffer, have all written good texts that will explain this to you. Please go to the library and read one of them. -- TEDDY BEARS UNITE! HUG A SHY PERSON TODAY! "I wish I was home again, back home in my heart again, ..." (ihnp4;allegra;research)!alice!jj