- sync mixer logic added
- mixer path creation, empty set in mapping creation, paths added in path creation
- path creation moved inside new port creation as it might be called twice otherwise
- some comments added
This allows us to support the PlaybackVolume and CaptureVolume commands
in UCM, specifying a mixer control to use for hardware volume control.
This only works with ports corresponding to single devices at the
moment, and doesn't support stacking controls for combination ports.
The configuration is intended to provide a control (like Headphone
Playback Volume), but we try to resolve to a simple mixer control
(Headphone) to reuse existing volume paths.
On the UCM side, this also requires that when disabling the device for
the port, the volume should be reset to some default.
When enabling/disabling combination devices, things are a bit iffy since
we have no way to reset the volume before switching to a combination
device. It would be nice to have a combination-transition-sequence
command in UCM to handle this and other similar cases.
PlaybackSwitch and CaptureSwitch are yet to be implemented.
When users select an output device from gnome-control-center UI, the
sink of this output device will be set to the configured_default_sink
and the default_sink, these actions are expected, but after these
actions, the gnome-control-center will call extension_cb() to modify
the entries in the database, let all stream entries to bind the sink
users select, this is not correct since the sink is default_sink now.
This is a temp fix for this issue, after gnome-control-center fixes
this problem, this patch should be reverted.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
When a sink is unlinked, all streams of this sink are moved to
default_sink, this action is implemented in the core rather than
modules now.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
When a new sink appears, all streams that have their preferred_sink
set to the new sink should be moved to the new sink.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
When the default sink changes, the streams from the old default sink
should be moved to the new default sink, unless the preferred_sink
string is set to the old default sink and the active port of the old
default sink is not unavailable
Signed-off-by: Hui Wang <hui.wang@canonical.com>
When the user moves a stream to the current default sink, the
preferred_sink should be set to NULL and module-stream-restore
should clear the routing for that stream in the stream database. From
that point on the stream will be always routed to the default sink.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
If the sink here is NULL, that means users want to clear the
preferred_sink and move the sink-input to the default_sink, otherwise
set the preferred_sink to the sink->name and move the sink-input to
the sink. After that fire the sink_input_change event.
After adding this API, we can use this API to simplify the entry_apply
in the module-stream-restore.c.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
The finial objective is to store the preferred sink name in the
sink-input struct, and use module-stream-restore to save and restore
it.
This patch just replaces the save_sink with preferred_sink, and tries
to keep the original logic.
Signed-off-by: Hui Wang <hui.wang@canonical.com>
pa_modargs_get_value() returns a const string -- instead of discarding
the const qualifier, let's just duplicate the string and free it
explicitly in the failure case.
Add a new module argument, blacklist, which is a regular expression.
If the sink/source name matches the provided blacklist regex, don't
automatically switch to it. By default, no devices are blacklisted.
Add a new function to check whenever a regex pattern is valid, plus
extra NULL asserts in pa_match.
Add an xauthority parameter and use it in the startup script.
Apparently on some systems the X authentication cookie is not stored in
~/.Xauthority but in some dynamic location pointed to by the XAUTHORITY
environment variable. The environment variable therefore needs to be set
in the PulseAudio daemon environment in order to have access to the X
server from the PulseAudio daemon, but the variable is not necessarily
set when starting PulseAudio. For example, systemd starts PulseAudio
outside the X session. The start-pulseaudio-x11 script is run in the
X session, so it has the environment variable available, and can pass it
to the X modules, which then can set the variable in the daemon
environment.
RedHat bug: https://bugzilla.redhat.com/show_bug.cgi?id=1723065
Debian bug: https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=593746
Based on patch by Alexander Kurtz <kurtz.alex@googlemail.com>
This adds a GStreamer-based RTP implementation to replace our own. The
original implementation is retained for cases where it is not possible
to include GStreamer as a dependency.
The idea with this is to be able to start supporting more advanced RTP
features such as RTCP, non-PCM audio, and potentially synchronised
playback.
Signed-off-by: Arun Raghavan <arun@arunraghavan.net>
It is possible that we might want to have a separate userdata to be used
for these callbacks, so let's split them out.
This is particularly needed when using an pa_rtpoll_item around pa_fdsem
since that uses its own before/after callback but will essentially have
whatever is using the fdsem set up the work callback appropriately (and
thus at least the work callback's userdata needs to be separated from
the before/after callback -- we might as well then just separate all
three).
Signed-off-by: Arun Raghavan <arun@arunraghavan.net>
This moves RTP implementation-specific information out of
module-rtp-send/recv. This is basically done by making the
pa_rtp_context structure opaque from the perspective of these modules.
We can then potentially replace the underlying RTP implementation with
something else transparently.
One RTP detail that does "leak" is the RTP timestamp. We provide this to
module-rtp-recv so that it can perform rate adjustments to match the
sender rate.
Signed-off-by: Arun Raghavan <arun@arunraghavan.net>
module-rtp-send itself doesn't really need to handle this, the
implementation can keep track (and make sure sending happens in MTU
sized chunks).
Signed-off-by: Arun Raghavan <arun@arunraghavan.net>
There doesn't seem much value in supporting streaming U8/mulaw/alaw on
the network, and it's unlikely these get any testing. Makes more sense
to drop these formats and just convert to L16 if we're dealing with
source media in that format.
Signed-off-by: Arun Raghavan <arun@arunraghavan.net>
pulseaudio does not link against libbluetooth, as it's only talking to the
bluez daemon over dbus. So the build dependency on libbluetooth is overly
restrictive, as some embedded systems choose to ship without libbluetooth
but still have bluez daemon support.
This syncs the meson to the autotools configuration behavior by changing
the bluez option to a default on boolean.
This was being done automatically by autotools, now we need to manually
specify this for each executable/library with a dependency in a
non-standard directory.
Rename struct rtp_payload to rtp_sbc_payload as it is specific for SBC
codec payload.
Add proper checks for endianity in rtp.h header and use uint8_t type
where appropriated.
Field frame_count is only 4 bit number, so add checks to prevent overflow.
And because is_fragmented field is not parsed by decoder there is no
support for decoding fragmented SBC frames. So throw an error in this case.
Add explanation why minimal bitpool value is used in SBC codec as initial
bitpool value for A2DP source.
Set buffer size for reading/writing from/to A2DP socket to exact link MTU
value. This would ensure that A2DP codec does not produce larger packet as
maximal possible size which can be sent.
Because A2DP socket is of SOCK_SEQPACKET type, it is guaranteed that
we do not read two packets via one read/recvmsg call.
Properly check for all return values of encode/encode methods of A2DP codec
functions. They may fail at different levels. Also encode or decode API
method may return zero length buffer (e.g. because of algorithmic delay of
codec), so do not fail in this case.
This crash occurs when PA is connected to a phone through the oFono
backend.
When disabling the Bluetooth adapter, pa_bluetooth_device is removed before
hf_audio_card. Both keep refs on pa_bluetooth_transport. Those removal will
call pa_bluetooth_transport_free() from device_free() (bluez5-util.c) and
hf_audio_card_free() (backend-ofono.c).
In the end, the call to pa_bluetooth_transport_free() calls
pa_hasmap_remove() through pa_bluetooth_transport_unlink(), but since
memory has already been freed, the second try results in a segfault.
Triggering hf_audio_card removal during pa_bluetooth_device removal allows
hf_audio_card to be freed at the right time.
