Julian, you are right in part...but wrong in as much. If the bursts you talk about are short enough then you are right about being able to correct more bits that way, but you are more likely to be able to correct a given (low) error incidence if it's spread randomly across multiple frames than if they are all localised within a single frame - hence the levels of interleaving put into the system (i.e. deliberately scattering burst errors across many different data words). If anyone wants to find out about how the error correction is done etc people should read something like
this well-written article. And as adb has said repeatedly in the past, the error-correction doesn't "kick in" and "work harder", resulting in some analoguy sag or distortion - if works at the same rate continuously while the player is playing.
I would be very surprised if audio players do re-read data - misreads IME just result in a click. Re-reading would also not help with damaged media, only external influences (as in e.g. anti-jog)
Originally posted by Lowrider
For instance...
OK..just a couple of examples
For one of these two waveform dimensions, the vertical amplitude axis, the CD contains some information, but the data coming from the CD are emphatically not fully or adequately descriptive of the music waveform's ever changing amplitude, especially for musical frequencies above 2 kHz or so. Instead, these data coming from the CD are only sketchy clues about the music waveform's ever changing values along the vertical amplitude axis.
1) These "sketchy clues" are the data bits being read off the disc which are are read off essentially perfectly in all but the worst cases. While CD-ROMs in your PC have a little more circuitry to do an even better job, think what would happen to your PC if if couldn't. The whole point of digital transmission is that the exact value of the signal amplitude has NO SIGNIFICANCE. The only important thing is whether it is greater or lower than a certain value - if it exceeds it then there it is IRRELEVANT by how much. 2) That figure of 2kHz is complete crap - maybe they meant 20kHz. 44.1 kHz can distinguish fully (to a resolution dependent on the bitdepth of the digitisation) any frequency below 22.05kHz.
But the other half of that same music waveform, its horizontal time axis, is not encoded digitally onto the recording, neither in full nor as sketchy clues. And it is not even encoded onto the recording in analog. Actually, it is not encoded onto the recording at all! Instead, it is merely assumed.
This assumption is what's called a standard! If the data is not stored at 44.1kHz then all bets are off. But this is not a difficult standard to meet - crystals are avaiable for less than a couple of quid that provide standard frequencies to better than 1 part in 10e7. That's many orders of magnitude better than you'll get off the best TT. (And to accommodate drift or physical tolerances on the disc it appears there's 3 sync bytes in every 36 on the disc)
With this amount of misinformation, distortion and obfuscation how can we work out what is useful information (which it looks as if there may be scattered amongst the garbage) and what is more crap?
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