This statement is dead wrong. I wish I had more time right now, if I did, I'd get into more detail, but I felt I had to chime in anyway. Jitter is a problem when transmitting digital signals. Basically, due to imperfections in the cable, or on longer cable runs, or due to impedance mismatch, or a variety of other reasons, each "sample" of data doesn't arrive at it's source at precisely the same interval from the last.
for lack of time, I'll just post this link, which has been posted here somewhere before:
http://www.digido.com/portal/pmodule_id=11/pmdmode=fullscreen/pageadder_page_id=28
But the link you posted says:
>>The top waveform represents a theoretically perfect digital signal. Its value is 101010, occuring at equal slices of time, represented by the equally-spaced dashed vertical lines. When the first waveform passes through long cables of incorrect impedance, or when a source impedance is incorrectly matched at the load, the square wave can become rounded, fast risetimes become slow, also reflections in the cable can cause misinterpretation of the actual zero crossing point of the waveform. The second waveform shows some of the ways the first might change; depending on the severity of the mismatch you might see a triangle wave, a squarewave with ringing, or simply rounded edges. Note that the new transitions (measured at the Zero Line) in the second waveform occur at unequal slices of time. Even so, the numeric interpretation of the second waveform is still 101010! There would have to be very severe waveform distortion for the value of the new waveform to be misinterpreted, which usually shows up as audible errors--clicks or tics in the sound. If you hear tics, then you really have something to worry about.
If the numeric value of the waveform is unchanged, why should we be concerned? Let's rephrase the question: "when (not why) should we become concerned?" The answer is "hardly ever."
The only effect of timebase distortion is in the listening; as far as it can be proved, it has no effect on the dubbing of tapes or any digital to digital transfer (as long as the jitter is low enough to permit the data to be read. High jitter may result in clicks or glitches as the circuit cuts in and out). A typical D to A converter derives its system clock (the clock that controls the sample and hold circuit) from the incoming digital signal. If that clock is not stable, then the conversions from digital to analog will not occur at the correct moments in time. The audible effect of this jitter is a possible loss of low level resolution caused by added noise, spurious (phantom) tones, or distortion added to the signal.<<
Bob Katz also added this on another email list I'm on:
>>We have to remember that digital audio stores data, NOT clock.
However, the interfaces transmit both clock and data. At the receive
side, the receiver modules (PLLs) separate the clock from the data,
and then work on the data. So, as long as the receiver is not
glitching and is locked to the incoming clock, then the extracted
data will be just fine, and accurate, whether the interface is
AES/EBU, SPDIF, ADAT, Toslink, Glass fiber, or plastic...
If you hear differences in a playback (reproduction) system (D/A
converter) where the data is identical, then blame it on the
clocking. The data is still fine. Repeat after me: "THE DATA IS
FINE". You can equalize it, process it, and work on it with no
concerns that the clock jitter will affect the audio. All the
processors IGNORE the clock and work on the data.<<
Topic: rosendahl recorder (Read 4157 times)