I'd rather not have to do this, as I don't really see the point in
duplicating what is done in pkgconfig, but this is likely the
easiest way to avoid nasty hacks.
pa_sink/source_used_by() ignores corked/monitor streams, but we need to
make sure there aren't any of these while updating rate (at least for
now -- this is a restriction that would be nice to get rid of).
Basically adds code to handle passthrough sources. This isn't a tested
path at the moment, but in the future, when we do wish to support these,
it'll save us the trouble of having to sync all the code again.
This is where the actual changes happen.
Some additional checks would be required to make sure the
rate is actually supported
Tested with both PCM and passthrough streams
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
Avoid resampling or use integer resampling when supported by the
sinks/sources
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
This adds the WebRTC echo canceller as another module-echo-cancel
backend. We're exposing both the full echo canceller as well as the
mobile echo control version as modargs.
Pending items:
1. The mobile canceller doesn't seem to work at the moment.
2. We still need to add bits to hook in drift compensation (to support
sink and source from different devices).
The most controversial part of this patch would probably be the
mandatory build-time dependency on a C++ compiler. If the optional
--enable-webrtc-aec is set, then there's also a dependency on libstdc++.
The new module argument can be used to provide a custom
directory for loading alsa path configuration files. This is
useful for testing: no need to be root to create test
configuration files.
module-stream-restore and modile-filter-apply can get into an infinite
loop if m-s-r is called before m-f-a (m-s-r rescues a stream and
attaches it to a sink/source, which then triggers m-f-a to move it back
to the filter sink/source, and so on). The purpose of the m-f-a hooks is
to beat m-s-r, so moving them to be run first.
This removes the nasty side-effect that a call to
pa_namereg_get_default_{source,sink}() will also *set* the default
source/sink.
This is a more complete fix for commit 766dbc68 ("conf: Make sure
module-dbus-protocol is loaded after module-default-device-restore")
https://bugs.freedesktop.org/show_bug.cgi?id=40897
This makes sure that we only perform any processing (resync or actual
cancellation) after the source provides enough data to actuall run the
canceller.
When a source-output isn't connected to our virtual source, we skip echo
cancellation altogether. This makes sense in general, and makes sure
that we don't end up adjusting for delay/drift when nothing is
connected. This should make convergence faster when the canceller
actually starts being used.
This increase the threshold for difference between the playback and
capture stream before samples are dropped from 1ms to 5ms (the
cancellers are generally robust to this much and higher). Also, we make
this a module parameter to allow easier experimentation with different
values.
Watermark level and latency values are not restored when
resuming, the values used prior to suspending are reused.
This leads to side effects when underruns happen and buffer
sizes are updated, PulseAudio can never meet lower latency
requirements.
Solution: keep track of watermark and latency values on sink or
source creation, and reapply them on resume to start with
a clean slate.
Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart@linux.intel.com>
This problem was found when tracing down a crash coming from the
esound protocol, which does not set a channel map.
BugLink: http://bugs.launchpad.net/bugs/864071
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
N.B.: As Colin notes, this is because commit 117c7145 was incomplete
("format: Fix channel map handling")
If module-jackdbus-detect failed in the later part of initialization,
the ma variable was freed twice.
BugLink: http://bugs.launchpad.net/bugs/867444
Signed-off-by: David Henningsson <david.henningsson@canonical.com>
Instead of relying on the snd_mixer_* functions failing, we check for
POLLERR and POLLNVAL first. After this, any errors in handling the mixer
events are deemed fatal (that is we cause the ALSA source/sink thread to
terminate).
The case where POLLERR is set but POLLNVAL is not does not actually
occur, but we're making this a soft failure (stop polling the mixer, but
don't kill the I/O thread). If other conditions where POLLERR occurs
turn up, we need to handle them explicitly.
Thanks to Linus Torvalds for helping get this right.
Loading between a sink and its monitor causes a deadlock (while sending
messages for latency snapshots). It isn't a case that has any real
conceivable use, so let's just disallow it.
This improves the error handling in the mixer rtpoll callback. It avoids
a crash if an error occurs (the rtpoll_item is freed but still
referenced), and specifically makes sure we don't continue trying to
poll the device if the card is disconnected.
This makes set_formats() put PCM formats lower down the list than
compressed formats since we negotiate by picking the first format in
this list that is also in the client-provided list of possible formats
during sink input creation.
This will be incorrect if we ever decide to do encoding in PA (for
things like AC3/DTS encoding for multichannel output over S/PDIF).
This patch was already added earlier with commit ID 2f86ba4f, but the
changes got reverted by commit 3adc43b ("win32: Make once-test work").
However, this still doesn't work on OSX as here, pthread is in general
available, but the barrier APIs aren't.
The purpose of this patch is to make it possible to configure stream volumes
before pulseaudio is run for the first time. This is useful, for example, in
embedded products where the default volumes have to be sensible already in
the first boot.
These are not used for anything at this point, but this
makes it easy to add ad-hoc debug prints that show the
memblockq name and to convert between bytes and usecs.
