Upsampling. Truth vs Marketing

Thanks for the feedback Sean. Putting your orig. & 2nd post together, I see what you were trying to say.

El said: "In Sean's explanation the second set of 20 dots in set B should not be random. Those dots should lie somewhere between the two dots adjacent to them".

By placing the "extra" dots ( sampling points ) "mid-point" between the previously adjoined dots, the end result would look MUCH smoother and far more predictable. While this "could" be the case if playing back sine waves of varying amplitude and duration, music is anything but "sinusoidal" by nature. There are very rapid peaks and dips that take place, sometimes completely changing the direction that the signal was previously headed just a split second previous. These peaks and dips can can switch randomly back and forth across the "zero line" or they can remain above or below the "zero line" for extended periods of time. On top of that, these waveforms may not be symmetrical at all i.e. much bigger peaks on the positive side than there are dips on the negative side or vice-versa. It is for this reason that "industry standard test tones" aren't quite as revealing as we would like as far as revealing how a component performs during normal use reproducing musical waveforms. This is why several different types of tests have to be used in order to obtain any type of meaningful relationship between test bench performance and real world performance.

If music was more like a sine wave i.e. with predictable amplitudes, polarities and durations, error correction algorithms could be much simpler and far more accurate. However, musical notes are anything but predictable in terms of amplitudes, polarities, durations or patterns. As such, the potential to read an error from anything but a perfect disc is not only high, but the potential for further errors to take place when data is lost and the machine is trying to "fill in the blanks" becomes even higher.

Somewhere in one of the old IAR's ( International Audio Review ), Moncrieff covered quite a bit on the flaws of how "Redbook" cd was designed and how their "error correction" and / or "interpolation" techniques were far from all-encompassing. Then again, this was all newer technology at the time, so they were kind of winging it as they went along. As such, the potential for a newer, much better digitally based format is definitely there, especially if we learn from past mistakes and take advantage of the more recent technology that we have.

Germanboxer: As far as certain manufacturers supporting / slagging specific design attributes, did anyone ever expect a manufacturer to support a design / type of product that they themselves didn't already take advantage of? Would you expect a company that didn't use upsampling to say that upsampling was superior or a company that did use upsampling to say that the technology that they were using was a poor choice?

Bombay: Glad that you were able to see where i was coming from after further explanation. Hopefully, others could follow along here too.

As a side note, read the description of this DAC as listed on Agon. You'll see that the designer not only played with various types of filtering, but gave the end user the option to accomodate their personal preferences / system compatibility at the flip of a switch. Bare in mind that this unit was out long before Philips came out with their SACD 1000, which also gave users the options of various filter shaping and cut-off frequencies, etc... Sean
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Eldartford's sentence: "In Sean's explanation the second set of 20 dots in set B should not be random. Those dots should lie somewhere between the two dots adjacent to them".

is exactly correct. One possible location of "somewhere between" could be legitimately the midpoint. There is no problem with that at all. If the waveform looks smooth then what's the issue with that??? How, in the world, do you know that the waveform at this point in the CD is not supposed to be smooth?? There could be a consistently low volume passage or a consistently loud volume passage of 1 particular instrument that creates a smooth area. Entirely possible.

Anyway, the thing to remember in your 2nd example is that when you placed that "random" set of points, you were looking at the output of the digital estimation filter. The output of digital estimation filter is very deterministic & it is designer created. The o/p simply cannot be random - no way!! It lies "somewhere between" the actual sampled data points off the CD along a line determined by the algorithm of the digital estimation filter. This is that (digital) filter that creates all those signature sounds (like Wadia's house sound, Sim Audio's, dCS's, etc, etc) that many love & equally many hate.

In Eldartford's example, I think, that he used a smooth waveform only to illustrate the point. This is the way that it is usually introduced in DSP 101 classes. His particular example is pertains to oversampling. When he shows the repeating of numbers, he has considered a 12X oversampling & when he does the div-by-4, he is considering 4X oversampling. The div-by-4 most probably represents the digital FIR that follows any over (or up) sampling operation.
My only question here is why did the example consider an oversampling of 12X then later decimate to 4X?? Should have just started of with a 4X DAC. Anyway.....

You mentioned "error correction" for the 2nd time. Error correction in redbook CD playback has nothing to do w/ upsampling or oversampling. Error correction is NOT designed to correct the music written on the CD. It is designed to compensate for high-speed read & transmission of the bits where read errors will occur (owing to the high speed read operation). I think Eldartford's succinct explanation is exactly what error correction is all about. Any other idea of it is a mistaken impression.

I have read the recent upsampling verbose text by Moncrieff on IAR. IMHO, I have not read more bull**** anywhere that filled up so many pages. Very little of what he has written is correct. AFAIK, Moncrieff is very lost when it comes to up & oversampling. If you are taking your lessons from him, then I can see why you are mistaken too. Get hold of a DSP text (like Oppenheim & Schaeffer or Rabiner & Gold) & read that. You'll get the correct explanation of upsampling & oversampling.

Sean...The sampling (your first set of dots) is at 44.1KHz. The highest audio information that exists at this sampling rate is around 20KHz, and at this frequency the music signal amplitude is very small. Therefore, unless the signal is momentarily a constant (two adjacent points the same) the in-between points will lie between adjacent points. Of course this is all overlaid with random noise that will blur the quantization staircase.

The CD recording protocol has been cited as an everyday example of the application of CRC error correcting technology, and I have seen descriptions of the CD protocol as having interpolation as a "fall back" procedure when the CRC error correction fails. Of course the second "fall back" is to abort playing the disc, and this ought to be the only time that the process is easily heard.

To tell the truth I have never actually read this infamous "Red Book" which defines the CD spec, and so am relying on what others have reported. How would I get a copy?

Bombaywalla...Your check is in the mail :-)