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In Reply to: RE: Design goal is to minimize mobo traffic to just the audio samples posted by cics on July 10, 2008 at 20:17:53
"cMP, cMP² and other players that load into RAM before playback would eliminate this conflict as no wav data traffic occurs. It however doesn't end here as some devices and applications have a polling nature. This would create such conflict and is audible."
Aren't hardware interfaces that send the audio samples out of the box buffered? Assuming the samples make it across the PCI bus to the interface before the buffers underrun, why would a slight delay affect the sound? Do you have a physical mechanism in mind and, if so, have you been able to verify this?
(I am talking about the occasional delay, not the situation when there is lots of conflict for the PCI bus in which catchup from small delays does not happen and the buffer fill level gradually depletes.)
"This all happens at frantic speeds but for ultra low latency playback, this conflict will have impacts."
A related question is why the need for ultra low latency playback? Given that some processing or data movement has to be done, why does it matter if it is done earlier rather than later? (I am talking about processing that is done while the music is playing, not processing that could be moved before playback starts, such as loading RAM.) I can understand the need for low latency in pro-audio applications where additional tracks are being laid down by musicians who have to stay in sync, but why does this matter for end-user playback?
Tony Lauck
"Perception, inference and authority are the valid sources of knowledge" - P.R. Sarkar
Follow Ups:
"why the need for ultra low latency playback?"
Lower latency settings on my soundcard lead to better sound quality. Do you not find this to be so? Especially noticeable when using kernel streaming. Why this should be so is the good question.
Yes, that's the question: Why do latency settings affect the sound quality? And how does this effect depend on particular hardware and software?
Tony Lauck
"Perception, inference and authority are the valid sources of knowledge" - P.R. Sarkar
A good programmer would argue that latency is a non-issue for playback. "Just set it to highest level as we get less context switches which is more efficient...". This is not correct for best SQ.
At a software, firmware and hardware level, PCI prefers small payloads. See here for more details.
"Latency jitter" as in variations in latency was thought to be the cause for why latency affects SQ. I've scrapped this idea.
From a Jitter viewpoint, when a soudcard's buffer is populated (whilst the other buffer is converted to SPDIF or whatever), there's a burst of electrical activity. The idea is to keep this burst as short as possible thereby reducing interference to soundcard's XO, i.e. reduce J pp . We achieve this by setting latency to lowest possible level. Of course, using such a low latency would mean more frequent buffer loads. I called this the ASIO frequency (or ASIO Hz). At 32 samples latency for 96k output, ASIO Hz is 3kHz. This is now periodic in nature and is digitally induced. We now have Periodic Jitter - the worst kind which exists for all digital playback systems. ASIO gives us control over this.
I prefer higher ASIO Hz and you definitely want to avoid anything less than 1kHz. Why? Soundcard's PLL or PLLs down the chain will be able to further reduce this periodic jitter as the frequency is likely to be above PLL's cut-off.
cMP² designs for this. Here's an explanation of cMP's software. That's why I recommend ASIO together with Optimize set to Critical. cPlay offers an additional level of refinement by implementing an extremely efficient ASIO interface which is also the reason for not supporting other ASIO data types except for 32 bits.
Thank you. This is an excellent answer to my question. Also, thank you for the link to your January 6th post, which I missed.
Tony Lauck
"Perception, inference and authority are the valid sources of knowledge" - P.R. Sarkar
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