Linked by Thom Holwerda on Thu 7th Jul 2011 17:36 UTC, submitted by vivainio
Linux Linux.FR has an interview with Lennart Poettering of PulseAudio and systemd fame (among others). Regarding PulseAudio: "I can understand why people were upset, but quite frankly we didn't really have another option than to push it into the distributions when we did. While PulseAudio certainly wasn't bug-free when the distributions picked it up the majority of issues were actually not in PulseAudio itself but simply in the audio drivers. PulseAudio's timer-based scheduling requires correct timing information supplied by the audio driver, and back then the drivers weren't really providing that. And that not because the drivers were really broken, but more because the hardware was, and the drivers just lacked the right set of work-arounds, quirks and fixes to compensate for it."
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RE[7]: Broken audio hardware
by _txf_ on Fri 8th Jul 2011 09:25 UTC in reply to "RE[6]: Broken audio hardware"
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From your link, the main difference from traditional audio processing is that "pulse audio glitch-free" opts not to use sound card timers/IRQs and use system timers/IRQs instead to provide even lower latency than is available through the sound card.

From my understanding the goal here is generally the opposite (to allow application that are not affected by latency to provide sound output at the highest latency possible yet have the flexibility for low latency).

I'm not understanding the technical reason that pulse audio couldn't revert to using sound card IRQs as they were intended instead of breaking sound support all together? Can someone clear this up?

The thing is that using the sound card irqs would result in generating an interrupt for every fragment of sound that is sent to the soundcard.

The fragment size is fixed by the soundcard and not the system, hence not dynamically reconfigurable. Also note that the soundcard has no information on the system and as such cannot make any judgements on the best solutions.

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