A2DP spec allow bitpool changes midstream which is why sbc configuration
has a range of values for bitpool that the encoder can use and decoder
must support.
Bitpool changes do not affect the state of encoder/decoder so they don't
need to be reinitialize when this happens, so the impact is fairly small,
what it does change is the frame length so encoders may change the
bitpool to use the link more efficiently.
This is needed to better support out of tree builds (including
distcheck) and to ensure the necessary folders are created in the
build tree on configure and also works around an intl-tools bug
(https://bugs.launchpad.net/intltool/+bug/605826)
The Makefile.am's used are minimal (and in some cases completely
blank). At present they do not include anything interesting
with the majority of the real work still done by the monolitic
src/Makefile.am
It may make sense to start splitting out src/Makefile.am into
smaller chunks but this commit makes the minimum changes to address
the issues that result from using make distcheck and other out of
tree builds.
Note: This 'breaks' the ability to type make in e.g. the src/modules
folder and have all of PA rebuilt accordingly (this is because the
static Makefiles previously present just did a "make -C ..") which
was purportedly for use in emacs. But I'm sure there will be a better
and more robust way to configure emacs to do your builds properly if
this behaviour is still desirable.
This is required to make sure that the source output between
module-echo-cancel and ALSA can't get plugged to the virtual source or
monitor of the virtual sink that we expose. This could be triggered by
changing the profile of the underlying ALSA device.
There is a call to this function where 'skip' variable is NULL. Looks
like this code doesn't get hit very often, probably because a suitable
default sink can be found to move the stream to. However, if we can't
move to the default sink and skip is NULL, there will be a segfault.
Currently if sink base volume differs from 0dB and sync-volume is used,
wrong volume values are written to hw. This patch fixes that.
Signed-off-by: Juho Hämäläinen <ext-juho.hamalainen@nokia.com>
BugLink: https://launchpad.net/bugs/680810
Some laptops have 'Internal Mic 1' exposed as an 'Input Source', e.g., Dell
XPSM 1530, so handle these, too.
Signed-off-by: Daniel T Chen <crimsun@ubuntu.com>
How about this? There are a couple of bugs in sink_write_volume_cb,
by the way. Another patch will be sent once this dB value printing
patch is accepted.
-- 8< --
Earlier, if slave sinks were unlinked in non-automatic mode, their
re-appearance was disregarded. Now they are added back to the list of outputs.
Signed-off-by: Antti-Ville Jansson <antti-ville.jansson@digia.com>
Reviewed-by: Tanu Kaskinen <tanu.kaskinen@digia.com>
The PCM handle is already opened with the SND_PCM_NONBLOCK flag.
This additional call is useless.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@intel.com>
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).
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 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.
