Linked by Thom Holwerda on Thu 8th Oct 2009 19:09 UTC, submitted by MadMAtt
Lennart Poettering, creator of open source sound server PulseAudio, was recently interviewed at this year's Linux Plumbers Conference. In this Q&A he details the latest PulseAudio developments and addresses some of PA's critics. Thanks to PulseAudio, the Linux audio experience is becoming more context-aware. For example, if a video is running in one application the system should now automatically reduce the volume of everything else and increase it when the video is finished.
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RE[6]: Not Anyone? I'm someone!
by kaiwai on Thu 8th Oct 2009 22:02
in reply to "RE[5]: Not Anyone? I'm someone!"
Member since:2005-07-06
Which is why professional linux audio is doomed.
Umm, you realise that you need applications to make professional linux audio a reality; you know things from Propellerheads like Reason, ReBirth, ReCycle are missing.
Considering that they make up a niche of the market - don't expect it happening anytime soon.
OSS4 just works, i can mix low latency and regular latency streams and it works perfect. ALSA+dmix can't do it and Pulseaudio becomes a hungry beast and eats most of my CPU when i try to do that. Every OS but linux does it the OSS4 way (primary and secondary buffers and in-kernel resampling), The future of audio floating point? please what a waste.. for processing inside an app maybe, but for sending an mp3 to the soundcard? YOU DONT NEED FLOATING POINT. Also you don't need floating point for resampling in real world usage.
So you ignore all the information given as the causes of problems and you also give no information on your setup. You sound like a person grasping onto straws as your one last lynch to linux hatred evaporates before your eyes.
Edited 2009-10-08 22:03 UTC
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Member since:
2006-02-25
Which is why professional linux audio is doomed. OSS4 just works, i can mix low latency and regular latency streams and it works perfect. ALSA+dmix can't do it and Pulseaudio becomes a hungry beast and eats most of my CPU when i try to do that. Every OS but linux does it the OSS4 way (primary and secondary buffers and in-kernel resampling), The future of audio floating point? please what a waste.. for processing inside an app maybe, but for sending an mp3 to the soundcard? YOU DONT NEED FLOATING POINT. Also you don't need floating point for resampling in real world usage.