luciyuspax,
The process of upsampling does not inherently improve sound quality during D/A conversion. However, Benchmark converters re-sample for a very specific reason: jitter immunity.
Benchmark converters use a proprietary clocking system (we refer to it as UltraLock). It works like this...
The incoming digital signal is immediately re-sampled by an ASRC (asyncronous sample rate converter). The ASRC, as the name implies, is not syncronized to the clock of the incoming digital signal.
Therefore, its performance is independant of the quality of that clock. In other words, it doesn't matter if the signal came from a cheap transport with cheap cables, or from a $10,000 signal chain. The large amount of jitter caused by the cheap transport and cheap cable will be moot with respect to the ASRC process. The output of the ASRC is then clocked to an on-board clock with extremely low jitter and strategic sheilding and board traces.
The output of the ASRC can be configured to any sample rate that we choose, including the original sample rate. However, we dictated the re-sample rate as 110 kHz because it is the highest sample-rate at which the digital interpolation filter of the D/A chip will operate optimally.
The ill-effects of the digital interpolation filter at higher-then-110 kHz include pass-band ripple (non-linearities in frequency response) and inferior attenuation of stop-band frequencies (which results in aliasing). Therefore, the D/A performance is optimized by maintaing 110 kHz.
Many converter designers have since employed similar topologies, but use lower re-sampling frequencies, such as 96 kHz. By resampling to 110 kHz, the low-pass filter of the ASRC and D/A are moved as far up as possible as to not infringe on the analog bandwidth of the audio.
I hope I explained this clearly...
Thanks,
Elias