Phil Leigh;238608 Wrote: > I must be missing something...the ORIGINAL sampling frequency is a > given...let's say it's 44.1 kHz. > So all you need to do is read those frames out at that frequency. Why > exactly is that so hard? Assuming you never run out of frames to read. > As far as I can understand things, the whole "clocking" problem comes > about if - and only if - there is no buffer. If the design is > synchronous end-to-end I can see how the clocks and clock drift and > jitter can really mess things up.
It's just that no two oscillators give you exactly the same frequency. So suppose the DAC clock is slightly faster. Then when you hit play, maybe there's a slight pause to let the buffer fill a little, and then away you go. Since the DAC clock is faster after a while the buffer is completely empty, and then you have a problem. Listener;238610 Wrote: > > 2. How big is the effect? For example how far off would an SB3 clock > be from an perfect clock? > I looked this up once, and don't remember the numbers - but the answer is that it's big enough to cause problems, particularly on long tracks. Making the buffer bigger doesn't help you when the local clock is faster - unless you're willing to have a long pause before the music starts. And you have to remember that a good commercial DAC should be able to deal with quite a wide variety of digital sources. It wouldn't be a very good design if it depended crucially on how accurate the clock was in the source. How audible any of this is - that's another question :-). -- opaqueice ------------------------------------------------------------------------ opaqueice's Profile: http://forums.slimdevices.com/member.php?userid=4234 View this thread: http://forums.slimdevices.com/showthread.php?t=33146 _______________________________________________ audiophiles mailing list [email protected] http://lists.slimdevices.com/lists/listinfo/audiophiles
