Dbphd - I am no expert, but here goes anyway. Corrections welcome.
Bass management is just what you think it is. It simply separates the frequencies into separate channels, as you describe. If you set your mains to crossover at 80hz, the frequencies below 80 go to the sub and those above 80 go to the mains.
The issue Kal is addressing is where that frequency information is contained in a digital signal. Each digital signal for a given channel is just a single amplitude number (usually a 16 bit number), with 44,100 of these numbers per second for CDs or 48,000 (and some 96,000) for Blu Ray sound tracks. Each individual number does not have a frequency associated with it. The frequency information comes when analyzing a series of these digital numbers. You can mathematically determine the frequency of a set of LCPM numbers by doing a mathematical transformation of the numbers. The digital filter in the receiver/player does these calculations and reconstructs the digital signal in such a way as to remove the low freqencies from the digital signal for the mains and add it to the subwoofer digital signal. All of this is done on the digital data, before it is converted to analog.
There are chips available that do these conversions for LPCM signals. The manufacturer just selects which one to use or has a special one made for them. The different "filters" that are available in some DACS, for example, as just different designs of these digital filters, somewhat like the different analog crossover designs found in speakers.
Similarly,time delays can be added in the digital data to account for the distance of the listener from the speakers and room correction can added.
Similar chips are not generally available for DSD signals, which consist of 1 bit of data, but at 2.8 MHz (2,822,000 samples per second), or 64 times as fast as CD (44.1 KHz) signals. (128 DSD is also becoming available) To do bass management, time delay and room correction on DSD signals they are typically converted to LPCM signals, usually at 88,200 samples per second, and then standard LPCM digital filter chips can be used.
As far as I know, there is no inherit reason, other than maybe computational speed, why a digital filter could not be developed for DSD signals. When SACD was initially designed by Sony they did not develop such a chip and assumed people would just send the full signal to each speaker. Of course, in the mastering, some of the low frequencies were already moved to the subwoofer channel. So, it was up to the mastering engineer to do any bass management.
Many DSD users are purist, who think that changing the signal to LPCM degrades the sound. Jdub39, on the other hand, thinks the 8801 bass management and room correction is a great improvement over the straight DSD sound.
That is a long winded overview. Fortunately, there are lot of very good engineers who can make all this magic happen. The mathematics of it is pretty complicated.
Hope I got most of that correct.
