| Commit message (Collapse) | Author | Age | Files | Lines |
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The structure itself will contain various bits of info so exposing this fully to the client is a bad idea.
By keeping to a rename operation we keep what we do store abstracted from the clients.
Also fix some doxy comments.
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This allows clients to edit the priroity order. What is not yet in place is the initialisation of that priority list
when new devices are detected or the cleaning (remove holes) when devices are removed.
In order to keep the storage transparent I will likely remove the write functionality and replace it with a
simple rename method.
I also still need to expose the priority itself when reading the data.
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necessary
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device-priority routing.
The routing logic itself does not yet exist, but the command currently will unload/load module-stream-restore as approriate.
(module-stream-restore would conflict with the role-based priority-routing).
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of devices.
This fixes a few bugs in the copy+pasted implementation of apply_entry()/get_name().
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This is effectively copied from the stream restore extension.
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of various sinks.
This will be used as the basis for a queryable system for past and present devices, initially for use in KDE.
Currently all this module does is save lists of sinks/sources and their descriptions, so it needs to
gain a protocol extension to make this queryable.
As things stand it will save the device descriptions of all sinks and restore them if they differ from whats on record.
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Original patch supplied by 'adi'
http://pulseaudio.org/ticket/669
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In virtual machines sound card clocks and OS scheduling tend to become
unreliable, adding various 'uneven' latencies. The adaptive algorithm
that handles drop-outs does not handle it this well: in contrast to
drop-outs on real machines that are evenly distributed, small and can
easily be encountered via the adpative algorithms, drop-outs in VMs tend
to happen abruptly, and massively, which is not easy to counter.
This patch simply disables timer based scheduling in VMs reverting to
classic IO based scheduling. This should help make PA perform better in
VMs.
https://bugzilla.redhat.com/show_bug.cgi?id=532775
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http://pulseaudio.org/ticket/702
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On some cards line-out is independant of Sepaker and it is a good idea
to cover that so that they can independantly be activated.
https://bugzilla.redhat.com/show_bug.cgi?id=520884
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As seen on some HDA chips:
https://bugzilla.redhat.com/attachment.cgi?id=359804
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As seen on some drivers:
https://bugzilla.redhat.com/show_bug.cgi?id=498612
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As used by some HDA devices:
https://bugzilla.redhat.com/attachment.cgi?id=365290
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As used by some HDA chips:
https://bugzilla.redhat.com/attachment.cgi?id=366816
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I checked the source code, and latency_msec is missing from the list
of valid module arguments. Attaching a patch to add it.
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sessions does not create spew in syslog
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users.
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That's what you get for applying your stash without fully checking if anything else was committed in the mean time :p
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s/Sourround/Surround/
Spotted by Colin Guthrie
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This is how things should have worked orinally, but I obviously misunderstood things a bit.
This seems to prevent the random hangups that were happening previously :)
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Loading module-cli could have the effect of setting O_NDELAY on stderr,
because it was just a dup'ed fd of stdin which module-cli sets O_NDELAY
for and which flag is shared between all dupes.
Instead of using stdin/stdout directly we now open a new file descriptor
for the controlling terminal, which is equally useful as stdin/stdout
but gives a new file that does not share O_NDELAY with stdin/stdout.
This solves a problem where when running pulseaudio -C resulted in
log output being truncated since stdio does not really handle O_NDELAY
that well in on its fds.
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If an element does not control some channels assume they are 0dB in
comparison to the other elements, i.e. do not influence the volume at
all. Previously we were assuming they were as high as the highest of the
channels we do control.
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vertically and horizontally
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metrics so that we don't accidently set a buffer size that is suitable for tsched where we don't use tsched
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broken drivers apparently need that
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way we do it for initial opening
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- As discussed on alsa-devel it's probably better to initialize the
buffer size first, followed by the period size. If that fails try the
other way round. If that fails try to configure only buffer size. If
that fails try to configure only period size. Finally, try to
configure neither.
- Don't require integral periods anymore.
Both of these changes should help improving compatibility with various
weirder sound devices, such as TV cards.
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