Currently when rewinding alsa, a fixed value of 256 bytes is used,
which represents 1.33ms @ 48kHz (2ch, 16bit). This is typically fine
and due to DMA constraints we would not want to rewind less than this.
However with more demanding sample specs, (e.g. 8ch 192kHz 32bit)
256 bytes is likely not sufficient, so calculate what 1.33ms would
be and use which ever value is bigger.
Discussed with David Henningsson and Pierre-Louis Bossart here:
http://thread.gmane.org/gmane.comp.audio.pulseaudio.general/7286
Rewinding the ring buffer completely causes audible issues with DMAs.
Previous solution didn't work with tsched=0, and used tsched_watermark
for guardband, which isn't linked to hardware and could become really high
if underflows occurred.
Added separate parameter that can be tuned to hardware limitations and size
of DMA bursts.
The smoother is paused when the device is suspended but never resumed on
unsuspend. Pass the paused = FALSE flag to the pa_smoother_reset() call to make
it unpause when unsuspending. This patch improves source timings quite a bit.
This commit mostly converts the X11 handling to XCB. There are still
some uses of XLib to deal with the X11 session handling modules, however all
client-side code should now be free of XLib and thus this should fix Bug #799
This is required to when playing on a52: device, rewind is broken
in those plugins.
Credits to Michael Rans <mcarans@yahoo.co.uk> for finding this
workaround, and Tanu Kaskinen <tanuk@iki.fi> for providing
valuable feedback.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@intel.com>
The syntactically correct error meant that the timestamp was always
marked as found and only the first header was checked.
In the case where the timestamp was the first header, things
would have worked as expected.
Thanks to pino for reporting via bug refs #818
On sink unlinking, existing sink inputs are moved, which in turn calls
a get latency callback, which references the jack client. Therefore,
make sure the sink is unlinked before the client is closed. Failure to
do so might lead to SIGSEGV.
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
All seeks/flushes that depend on the playback buffer read pointer cannot
be accounted for properly in the client since it does not know the
actual read pointer. Due to that the clients do not account for it at
all. We need do the same on the server side. And we did, but a little
bit too extreme. While we properly have not applied the changes to the
"request" counter we still do have to apply it to the "missing" counter.
This patch fixes that.
This is not 100% ideal as we have not way to tie specific boosts to specific
inputs and this particular chipset (as noted in #772) appears to
support just that.
For the time being incorporate it into the normal boost logic.
See http://pulseaudio.org/ticket/772
When an GetProperties() reply arrives after we already deleted the
device structure for it make sure we don't accidentaly touch the
invalidated object.
https://bugzilla.redhat.com/show_bug.cgi?id=543205
If m-s-r sets the device we let it do so. Otherwise we handle the routing. We run before
module-intended-roles as the priority list will likely be configured appropriately
to do the same job, albeit with manual setup.
