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Music servers and other computer based digital audio technologies.

RE: "network rendering, streamers, music servers, mini computers"

You have to look at peak processing, not average processing. If you transfer the samples one at a time to the DAC then it processes them one at a time. The processing cycles at the sample rate. If you transfer samples a buffer at a time, then the processing takes place each time a buffer arrives. This results in a busy period followed by an idle period at the buffer rate. Given typical packet sizes the cycle time will be in the middle of the audio range, e.g. 1 kHz for USB. The numbers will be similar for Ethernet. Noise associated with this processing may be audible. This is similar to the situation that many people have observed in computer audio where reducing the buffer size improves the sound quality up to the point where buffer overruns start happening and audible glitches appear.

These are real problems. There is a lot of debate about why these effects happen and what the best (or most economical) way of dealing with them might be. However, careful listeners all notice that bits are not bits, or at least usually do not appear to be such. There is spirited debate as to what to do about it. Of course there are people who deny this can happen, but you will find them on other forums.

Tony Lauck

"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar


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