| Commit message (Collapse) | Author | Age | Files | Lines |
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We need to resume audio devices even for streams that are created in
corked stat, so that the latency ranges of the audio device are known
during the initial latency negotiation. If we don't the latency
negotiation will be based on placeholder data and changed later on which
clients do not expect.
This should fix issues with Skype.
https://bugzilla.redhat.com/show_bug.cgi?id=554929
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Move the mainloop to monotonic based time events.
Introduces 4 helper functions:
pa_{context,core}_rttime_{new,restart}(), that fill correctly a
timeval with the rtclock flag set if the mainloop supports it.
Both mainloop-test and mainloop-test-glib works with rt and timeval
based time events. PulseAudio and clients should be fully functional.
This patch has received several iterations, and this one as been
largely untested.
Signed-off-by: Marc-André Lureau <marca-andre.lureau@nokia.com>
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start the move, delete the original sink, create a new sink, finish the move; similar for source outputs
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tsched buffer size of 2s
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hook subsystem while doing so.
git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@2516 fefdeb5f-60dc-0310-8127-8f9354f1896f
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git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@2445 fefdeb5f-60dc-0310-8127-8f9354f1896f
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- Add new PA_STREAM_FIX_CHANNELS, FIX_RATE, FIX_FORMAT, DONT_MOVE, VARIABLE_RATES to pa_sream_flags_t adn implement it
- Expose those flags in pacat
- Add notifications about device suspend/resume to the protocol and expose them in libpulse
- Allow changing of buffer_attr during playback
- allow disabling for remixing globally
- hookup polkit support
git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@2067 fefdeb5f-60dc-0310-8127-8f9354f1896f
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that in the module loader
git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@2043 fefdeb5f-60dc-0310-8127-8f9354f1896f
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git-svn-id: file:///home/lennart/svn/public/pulseaudio/trunk@1971 fefdeb5f-60dc-0310-8127-8f9354f1896f
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