On Jan 7, 2008 11:59 AM, Erik Slagter <erik at slagter.name> wrote: > > I was wondering if anybody was experiencing random cracks and pops (loud > > ones actually :( ) while using pulseaudio with intelHDA? I have the > > following configuration for my inbuilt intelHDA card and will be > > thankful if somebody helped me out here. I haven't observed cracks and > > pops on my other card (Griffin iMic). I am using pulseaudio 0.9.8 which > > is able to acquire realtime scheduling priority and a nice level of -11. > > > If you have true realtime scheduling, you don't need "nice" levels. If a > process in the realtime queue is ready to run, it will run, even if > there are dozens of processes in the "normal" queue ready to run and > even if they are all nice'd -19 (the maximum). > Hmmm... I am sure you are right. Will probably switch off renicing then. > Besides that, a properly designed sound system (software and hardware) > doesn't need a high priority to work smoothly. > You mean even at low latencies?... quite a few people had me convinced otherwise. I was under the impression that realtime sound processing was important for directly interfacing musical instruments with a computer... may be not for day to day audio needs I guess. However, I do recall using ecasound with my guitar... I never prioritized ecasound (with rtlowlatency buffering running directly over alsa hw) though and characterized the cracks and pops I heard then to the same. > So you have another problem here, actually. How is sound reproduction > using purely alsa (without pulse)? > I had no such cracks and pops earlier with alsa+dmix running at 192Khz. I had even added the speex resampler to alsa+dmix and faced no problems. The CPU usage for the audio application used to go up quite a bit (14%) due to the resampling but that's expected. However I did notice a good amount of latency then with games which I played through alsa-oss. I won't say padsp is perfect but it is surely much better latency wise. Please note that the applications I am using send sound to pulseaudio using either gst-pulse or alsa-pulse... and I saw cracks and pops with both. > I have a similar problem in that a pulse stream plays for a few moments, > starts to hiccup and then stops altogether. It doesn't matter which > program I use. Using alsa directly there is no problem. On my other > computer it works like expected, though. I have no need to use pulse > anymore (it doesn't solve my problem) so my problem is gone ;-) > :) I needed low latency mixing (primarily)... and the convenience of switching between audio sinks dynamically (secondarily). I like the idea of per application volume mixing but I would rather keep the UI component tied to the application itself than keep it separate... I am guessing pulseaudio does provide a mixer/sink manipulation API... a little gtk+ embeddable component to control these easily would provide a consistent audio control interface across many applications. What was problem you wanted to solve... (network audio?). May be I can help. _r -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.freedesktop.org/archives/pulseaudio-discuss/attachments/20080107/93f19d92/attachment.htm>
