JITTER REDUCTION BY USING AN ASYNCHRONOUS UPSAMPLER
Jitter reduction has become a bit controversial. There used to be nice and affordable after market devices that you simply insert inbetween source and dac, that claimed reduced jitter.
These days many cd players and computer devices still use a variety of jitter reduction techniques and most consist merely of inserting an asynchronous upsampler in the signal path. The manufacturer may choose to use the upsampling on offer from the IC or choose to use the same output samplerate.
Many people nowadays believe that the technique used is not the right answer to fight jitter. The process reads in digital data and, through a series of calculations, converts it to a sample rate unrelated to the original. The resultant stream is independently clocked, thus forming a kind of barrier between the original and the output signals. The problem is that the process merely converts timing errors in the incoming signal to amplitude errors in the output signal. The output bitstream may indeed test low for jitter but that can be misleading, and non-correlative to listening results. Using jitter reduction devices may result in better sound, but probably not because of lower jitter but because of a variety of other reasons. The original timing errors are still embedded within the data, but in a form that is not being measured.
Many so-called jitter reduction devices simply use an asynchronous upsampler to make for a measurably better performance. But as so often in audio, things are not so simple and initial impressions may be deceiving.