To make concurrent use of SW and HW volume glitchles their application
needs to be synchronized. For accurate synchronization the HW volume
needs to be applied in IO thread. This patch adds infrastructure to
delay the applying of HW volume to match with SW volume timing. To
avoid synchronization problems this patch moves many of the volume and
mute related functions from main thread to IO thread. All these
changes become active only if the sync volume flag for a sink has been
set. So, for this patch to have any effect it needs to be taken into
use by sink implementor.
Signed-off-by: Jyri Sarha <jyri.sarha@nokia.com>
Reviewed-by: Tanu Kaskinen <tanu.kaskinen@digia.com>
Reviewd-by: Colin Guthrie <cguthrie@mandriva.org>
This decrease PA_VOLUME_MAX to be less than 2^31. We want to do this in
order to simplify signed arithmetic when applying software volume
scaling (since the volume can now always be safely treated as a signed
number).
This ensures that we always clamp the volume to PA_VOLUME_MAX. While
this currently has no effect, it will be required for making sure we
don't exceed PA_VOLUME_MAX when its value changes in the future.
This adds a PA_VOLUME_IS_VALID() macro for checking if a given
pa_volume_t is valid. This makes changes to the volume ranges simpler
(just change PA_VOLUME_MAX, for example, without needing to modify any
other code).
In the assembly optimized versions of SSE, a noise could occur when the
number of channels were 3,5,6 or 7. For MMX and ARM, this could occur
when the number of channels were 3.
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
A good many of the header files are broken into a function
reference page and an overview page. These changes add
a direct link from each function reference page to their
overview page if one exists.
stream.h, simple.h
The words drain and flush are a little ambiguous, make it explicit as
to what happens to any existing audio.
*mainloop.h
reword *_free and *_get_api for grammar
Mostly change "Set the volume of all channels" to
"Set the volume of the first n channels" as the first is incorrect,
it doesn't set all the channels and doesn't explain what n was for.
When looping through the streams on a given device checking to see if the
stream is 'active' there should be no assert if the stream is not linked, it
should simply be ignored.
This assert can be hit if a sink and a sink input are both created and setup
but the final put calls are left to the end as is done in module-ladspa-sink.
While the order of the calls in module-ladspa-sink could be altered, we should
deal gracefully with the way it is now and not complain about ending up
in this state.
A trigger case was trivial:
1. Load a ladspa-sink.
2. Play a stream and move it to it.
3. Unload the module, then reload it.
4. Due to module-stream-restore and module-suspend-on-idle, the hook callbacks
will ultimately hit this assert.
Thanks to Kim Therkelsen for highlighting this issue.
This forces us to get native-endian samples in the adrian module so that
we can rely on the existing endianness conversion mechanisms instead of
doing it in the module.
The adrian module was using home-brewed endianness conversion instead of
the appropriate mactos, and speex assumed a little-endian host. This
fixes both of these.
Optimises the core inner-product function, which takes the most CPU. The
SSE-optimised bits of the adrian echo canceller only if the CPU that PA
is running on actually supports SSE.
This retains CPU information (processor type and supported features) in
pa_core, so that this information can be used by modules at init time to
figure out what optimisations may be used.
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
Note that this removes the screen-based changes by Leszek Koltunski
in 65e80, however this will be restored in due course.
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
Make new defines for the smoother window size and adjust time constants instead
of reusing some unrelated constant.
Increase the smoother window size even more because the bigger it is, the
better. Since we have a 200ms max update interval and the max smoother history
is 64 entries, 10seconds is a good default.
Decrease the smoother adjust time to 1 second. The previous value of 4 seconds
was too much to adapt quickly after a resume.
Use snd_pcm_avail_delay() in pa_alsa_safe_delay() so that we can check the delay
value against the avail value and patch it up when it looks invalid. Only do
this for capture.
Move the code to start the capture and the smoother closer together to improve
smoother accuracy.
Rework things to look more like the alsa sink where the device is started in
only one place.
Since all algorithms will need to specify a block size (the amount of
data to be processed together), we make this a common parameter and have
the implementation set it at initialisation time.
