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SidneyStencil

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  1. Gordon,<br /> <br /> Could you please elaborate on your claim that MM isn't bit true? On the TAS forum too you said that it was "a well known fact that MediaMonkey is not even bit true".<br /> <br /> Chris,<br /> <br /> In one of your earlier articles you gave advice on how to set up a Windows-based system with MediaMonkey. Did you also find that it wasn't bit true?<br /> <br /> Thanks,<br /> <br /> Sid
  2. That's a lot of questions, bball. First off, the Transit doesn't have a volume control, so you'd have to either use the volume knob on an (integrated or headphone) amplifier or control the volume through the player software. In terms of software and format, I can recommend MediaMonkey and FLAC, a combination I've used for three years without a hitch (or any bursts of "white noise" -- I'll come to that). As long as there's only one audio stream and the Windows volume is set to 100%, XP's Kmixer doesn't get involved at all. (See Benchmark Audio's wiki for more details.) So you might actually get away with MediaMonkey's default DirectSound output. But to be absolutely sure that Kmixer doesn't change your data on their way from MediaMonkey to the M-Audio Transit, you can install MediaMonkey's ASIO plugin. (The Transit comes with an ASIO driver, so you don't even need ASIO4ALL.) To avoid glitches, pops and bursts of noise, you might want to: switch off all Windows system sounds (Control panel > Sounds and audio devices > Sounds tab > Sound scheme: No sounds > Apply >OK) switch off the startup/shutdown beep (Control panel > System > Hardware tab > Device manager > View menu: Show hidden devices > Scan list for "Non plug and play drivers" and expand the node > Right click on "Beep" to disable it > Restart) set the latency in the M-Audio Transit's control panel to "Very high" (low latency matters only when you're recording, mixing or editing sound; for mere playback it's irrelevant -- in fact, you want high latency to avoid glitches and dropouts) leave audio settings, sample rates, cables, etc. well alone while you're listening to music (you wouldn't rip out cables, disconnect components or change the speed of a turntable mid-song while you're listening to conventional audio equipment, either) I didn't really understand your last question -- the M-Audio Transit doesn't re/over/upsample -- but I hope this is enough to help get you started.
  3. Oh dear.<br /> <br /> I've stumbled into The Computer Audiophile Jitter Wars, with my very first comment. I'm sorry; I should have spent more time in the forums first to get the lay of the land. <br /> <br /> So, very briefly, here's the AES paper:<br /> <br /> <a href="http://www.aes.org/e-lib/browse.cfm?elib=8354">E. Benjamin, E. Gannon, Theoretical and audible effects of jitter on digital audio quality (1998)</a><br /> <br /> And here's a more recent study published in "Acoustic Science & Technology":<br /> <br /> <a href="http://www.jstage.jst.go.jp/article/ast/26/1/50/_pdf">K. Ashihara et al., Detection threshold for distortions due to jitter on digital audio (2005)</a><br /> <br /> Phew.<br />
  4. I'm not sure I understand this jitter problem. <br /> <br /> Firstly, if the PC sends samples from its memory buffer at 1ms intervals (as per USB specification) to the DAC's own buffer, and the DAC, using its crystal oscillator as a clock, then processes those samples, where do the variations in timing arise?<br /> <br /> Secondly, and more important, does it even matter? According to the AES, jitter below 20ns is inaudible, and even something as dependable, common and cheap (and therefore inherently non-audiophile) as the M-Audio Transit measures a mere 2ns.<br /> <br /> All answers gratefully received.<br /> <br /> Sid
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