duke43j;427380 Wrote: > There is a lot of bad information floating around with respect to hi-fi > equipment. A digital cable either works, or it doesn't work. If it > doesn't work you would hear pops, skips or dropouts in the audio. There > is no way you would hear a change in tonality. A digital cable carries > 1's and 0's. If it was faulty, 1's would be mistaken for 0's, and > vice-versa. This would cause the data to fail an error check and the > sample would be discarded. If one or two samples are discarded, the unit > is supposed to keep playing the > same sound as the last good sample; but this can go on for only about > 1/1000 of a second. If it continues, then the unit will mute the output > (a dropout). With this kind of operation, there is no way you would > characterize the resulting sound as "not having enough air", or "losing > detail".
Im afraid you're a ways off the mark here. You need to get caught up on the subject before debunking the "bad information". What jitter does is it "smears" the high frequencies, and this phenomenon is readily observed with just an audio spectrum analyzer, regardless of what anyone thinks it does or doesn't sound like. However, I would say that a loss of detail is a perfectly reasonable description. > The older DACs had a clock that was tied to the rate at which the data > appeared on its input. If the input data appeared at irregular times > (jitter), then the clock in the DAC would tick at irregular intervals > (although it would try to smoothe it out as best it could). Reclockers > try to smoothe the data rate before they get to the DAC unit. This extra > smoothing reduced the jitter even further. > > Newer DACs with an asynchronous rate converter have two clocks; one to > clock in the jittery input data, and a second, very stable clock, to > clock the D/A chip. What you're talking about is properly called ASRC or "Asynchronous sample rate conversion". In the Benchmark DAC1 it is implemented by an AD1896 chip, and prior to this chip's availability I don't think any off the shelf DACs did it. However, it is NOT _generally_ a feature of newer DACs, and it is not simply a means of having a second more stable local clock. In ASRC, the data stream is mathematically resampled to a completely different rate (eg 110KHz), not merely re-clocked. This certainly eliminates susceptibility to the conventional mechanism of s/pdif jitter, but it also completely reconstructs the data stream and the potential audible impact of resampling should not be overlooked. > As long as you don't starve the unit by not feeding it data, or the > opposite problem of feeding it too much data, jitter on the input data > shouldn't be a problem. I'm not sure what you're getting at here. s/pdif uses only a continuous clock signal that is embedded (manchester encoded) in the data. It does not rely on the kind "starving or not" flow control you're imagining. -- seanadams ------------------------------------------------------------------------ seanadams's Profile: http://forums.slimdevices.com/member.php?userid=3 View this thread: http://forums.slimdevices.com/showthread.php?t=63796 _______________________________________________ audiophiles mailing list audiophiles@lists.slimdevices.com http://lists.slimdevices.com/mailman/listinfo/audiophiles
