emrath@uiuccsb.UUCP (02/05/84)
#N:uiuccsb:5700026:000:635 uiuccsb!emrath Feb 4 02:36:00 1984 I have read here and in the mags that some players use 14-bit converters and some use 16-bit converters and some use oversampling. I have not read, however, what the actual format on the disc is. How many (data) bits per sample and how many samples per second are actually on the disc? (I do know the fundamental sample rate is 44.1kHz per channel) Does anybody have a reference? (I don't know much about digital signal processing, but I had this one idea that digital filtering and oversampling could be similar - interpolate extra numbers between the known numbers and feed them to the D2A). BTW: How do you prefer to spell disk?
gregr@tekig1.UUCP (Greg Rogers) (02/09/84)
************************************************************************* The following is a reply to a specific question about the techniques used with CD players and is a simple statement of facts. I will express no opinions about the sounds of CD's in general nor about differences in sound (if any) between techniques. I believe far too much subjective opinion on these issues has already nearly destroyed the usefulness and pleasure of participating in this newsgroup. I hope any followup to this article will refrain from subjective comments as well unless supported by some scientifically acceptable test data. Thank You *************************************************************************** In answer to your question about types of CD players, 14 bit vs 16 bit etc. There are at the moment two distinctly different approaches being used in CD players to transform the digital data to an analog signal. As you gathered one method (initially used by Phillips) utilizes a 14 bit D/A converter, digital filtering, and oversampling for conversion, while the other method (initially used by Sony) utilizes a 16 bit D/A converter and analog filtering only. The digital data on the CD disc has a resolution of 16 bits, and a sample rate of 44.1 kHz on all CD's and is independent of the players digital-to-analog conversion technique. The actual data on the disc is encoded in a highly complex manner to ensure the integrety of the data even if flaws in the disc surface destroy some of the information. In most cases "lost data" can be completely reconstructed "on the fly" in real time, in severe cases interpolation will be used if necessary (rarely needed), or in the cases of last resort controlled muting is used (very rare, analogous to the tick and pop eliminaters used in analog systems). Just for information the technique used is to first encode the data using a system known as CIRC (Cross Interleaved Reed Solomon Code), add C&D (Control and Display) bits, and then store the data on the disc using a technique known as EFM (Eight to Fourteen Modulation) which is very analogous to techniques used with computer floppy disks to simplify the hardware requirements to read and write the disks. The reverse process is used when playing the CD which results in a complete recovery of the 16 bit/44.1 kHz data ready to be transformed by either of the D/A conversion systems mentioned above. Before describing those a little further it should be noted that although the techniques described above would appear to be very complicated to people unfamiliar with them, in reality the of CD's over many years. What I'm trying to say is just because you ain't heard of them don't assume they are unproven/experimental or even new. They ain't. At this point let me give you a few figures that actually describe the capability of the error detection and correction system in terms of the physical disc. Maximum COMPLETELY correctable burst length 4000 data bits = 2.5 mm on disc Maximum Interpolatable burst length 12,300 data bits = 7.7 mm on disc This means that scratches or imperfections in the disc less than 2.5 mm long in the direction of the track (around the disc, on the track) will be TOTALLY corrected!! Imperfections longer than 2.5 mm but less than 7.7 mm will be interpolated. Imperfections longer than this will be muted in a very well controlled manner. Returning now to the final step - Digital to Analog Conversion. The 16 bit D/A conversion method (Sony) is conceptually the easiest. The 16 bit data at a 44.1 kHz sample rate is passed to a 16 bit digital-to-analog converter. Because the signal has been digitized the resulting output of the D/A contains frequency components above the original audio signal. These need to be filtered out since they were not part of the original audio signal, and could overload amplifiers, create intermodulation products and so forth. In the Sony system a high-order analog Bessel filter follows the D/A converter to filter out these unwanted frequencies (I believe it is a 9th order filter, anyone positively know for sure?). The signal is then ready for your amplifier. This simple straightforward approach seems rather expensive. The cost of the precision 16 bit D/A should be higher than a 14 bit D/A as used by Phillips and the multiple section analog filter must be trimmed to very high tolerances which is also rather expensive. In the Phillips system the digital data is filtered prior to the D/A converter using a digital transversal filter. Although the 16 bit data is only available at a 44.1 kHz rate, three additional values of 0 are inserted between each actual sample within the digital filter so that data leaves the digital filter at a 176.4 kHz rate. (Don't let the fact that the inserted values are 0 bother you too much, without an understanding of the mathematics of digital filtering this simply can't be explained. It does however work and again nothing is really new about this.) At this point the data is rounded of to 14 bits in a rather fancy way and presented to a 14 bit D/A converter that operates at 176.4 kHz. The output of the D/A is then filtered using an analog Bessel filter of the 3rd order. The net result of the digital filtering, the fancy 14 bit round off, and the 4 times higher sample rate D/A, is to produce a signal/noise ratio of about 97 dB, essentially identical to the 16 bit D/A method. Since the digital filter and the round off circuit are both contained in a single custom integrated circuit and the D/A is 14 bits (although has to operate at 4 times the speed) instead of 16 bits I would expect the Phillips approach to be considerably cheaper. If the Phillips system is really cheaper why didn't Sony use it also? Well I'm only speculating about cost and perhaps Sony can build and trim their high order analog filter very inexpensively using automated means? Perhaps Sony had their 16 bit D/A already available to them from other projects (PCM tape decks) and Phillips would have needed to develop their own 16 bit D/A? Perhaps it was easier for Phillips to design their digital filter IC than for Sony? If anyone else on the net has any insight as to why Sony went one way and Phillips went the other please speak up. In any event I guess thats what makes engineering interesting, more than one way to skin a cat. More interesting from the listeners point of view is if either company really (I don't care what they say in their advertisements) believes one approach SOUNDS better than another. The Phillips approach does give a better high end phase response although we can debate the audibility of this parameter forever. (No correct that - You can debate it, I give up that uselessness.) What other measureable parameters might one approach favor over another? I haven't thought this out yet, maybe someone else more knowledgeable than I in this field has some ready answers. Well I hope this has answered the original question. There is plenty more that can be said but it all gets real technical real fast and probably doesn't belong here. If anyone finds out more technical details on Sony's filter please let me know. Again please DON'T take this opportunity to open a debate over which system sounds better unless you got some real scientific evidence to offer. Keep hoping something useful is going on here but no longer convinced, Greg Rogers