Use a wildcard rate for DSP ports.
Handle wildcards for rate and channels.
Calculate required in/out samples using quantum
Limit monitor and output number of samples.
Assume that capture and playback nodes from a device have different
clocks. This enables the adative resampler to match them. A lot of devices
actually have slightly different rates and would work out of the box
with this fix.
Make an exception when the card is configured in the pro audio profile.
Then we force the same clock on all device nodes and avoid resampling
and rate matching. This can still be changed with a session manager
override.
It causes some headsets behave strangely. See pipewire#2391,
pipewire#1853.
The BlueZ issue of AVRCP volume sometimes missing that this worked
around was fixed in recent versions. The issue of some headsets not
sending volume without AVRCP player remains, but it appears this breaks
more headsets than fixes.
Don't try to reconnect or wait for profiles, which cannot be connected
because the adapter doesn't have the counterpart sink/source profile.
E.g. we should not reconnect/wait for HFP HF on remote device, if
we don't have the corresponding HFP AG.
If A2DP remote does not acquire its pending transport within a timeout,
we won't get a write error in a2dp-sink, but instead the transport
becomes idle. Currently, we continue writing to the socket as if
everything was fine, even though the data won't be processed at the
remote end.
Handle this by stopping the node and emitting a node error event.
Pipewire may then restart the node to retry.
If the follower of the adapter emits an error event, the adapter needs
to forward it to upper levels so that they know the node has errored,
and handle the situation.
Endpoints without decode/encode capability are skipped in the object
manager, but we should also skip them in the registration calls (even
though in practice this doesn't appear to matter).
When filling up the channels, either fill up the positions with one
of the know layouts or use AUX channels, never try to mix them.
This avoid cards with a large number channels to show a strange mix
of surround and AUX channels.
Register a pthread cleanup handler to guarantee
that `spa_source::{priv, rmask}` are cleared even
if the thread is cancelled while the loop is dispatching.
This is necessary, otherwise `spa_source::priv` could point
to the stack of the cancelled thread, which will lead to
problems like this later:
Program terminated with signal SIGSEGV, Segmentation fault.
#0 0x00007f846b025be2 in detach_source (source=0x7f845f435f60) at ../spa/plugins/support/loop.c:144
144 e->data = NULL;
[Current thread is 1 (LWP 5274)]
(gdb) p e
$1 = (struct spa_poll_event *) 0x7f845e297820
(gdb) bt
#0 0x00007f846b025be2 in detach_source (source=0x7f845f435f60) at ../spa/plugins/support/loop.c:144
#1 0x00007f846b0276ad in free_source (s=0x7f845f435f60) at ../spa/plugins/support/loop.c:359
#2 0x00007f846b02a453 in loop_destroy_source (object=0x7f845f3af478, source=0x7f845f435f60) at ../spa/plugins/support/loop.c:786
#3 0x00007f846b02a886 in impl_clear (handle=0x7f845f3af478) at ../spa/plugins/support/loop.c:859
#4 0x00007f846b172f40 in unref_handle (handle=0x7f845f3af450) at ../src/pipewire/pipewire.c:211
#5 0x00007f846b173579 in pw_unload_spa_handle (handle=0x7f845f3af478) at ../src/pipewire/pipewire.c:346
#6 0x00007f846b15a761 in pw_loop_destroy (loop=0x7f845f434e30) at ../src/pipewire/loop.c:159
#7 0x00007f846b135d8e in pw_data_loop_destroy (loop=0x7f845f434cb0) at ../src/pipewire/data-loop.c:166
#8 0x00007f846b12c31c in pw_context_destroy (context=0x7f845f41c690) at ../src/pipewire/context.c:485
#9 0x00007f846b3ddf9e in jack_client_close (client=0x7f845f3c1030) at ../pipewire-jack/src/pipewire-jack.c:3481
...
