When the invoke ringbuffer is full, sleep a little and try again.
Add an option to set the retty timeout, setting this to 0 restores
the old behaviour of returning -EPIPE.
Most callers don't check the return values and might assume the invoke
call is queued or executed, which could cause crashes or leaks.
When the queue overruns, it's better to log a warning and hope that the
problem is resolved soon. We might abort or return the error to the
caller later if we want to break the retry loop.
See !1887
The file is moved into a new "include-private" directory. This is done
because otherwise adjustments would have to be made to the list of installed
headers, the way include tests currently work and which files are
used for generating documentation.
Fix some sanity checks and add mtu check. Don't use
spa_return_val_if_fail here as it can spam stderr.
The buffer size check in codec_encode can't be hit.
Don't show a "codecless" profile for HFP, similarly as we do for A2DP.
Simplify codec handling: for HFP/A2DP there's at most one transport for
each profile, so no need to check it has right codec. There's also no
need for "fallback profile", we always just emit nodes for the transport
we find.
Always reevaluate the rate matching even when we did not change the
follower state.
It is possible that we were a follower from some node with the same
clock and now become a follower of a node with a different clock. The
follower state doesn't change but we need to activate the rate matching
logic in that case.
Fixes rate matching in pro audio (playback) when capture and playback
are moved to another driver.
The AAC-ELD support was not properly tested on devices. In theory it
should be OK, but it's untested.
Bump it down in priority so it won't be selected by default.
Also log info on FDK-AAC AAC-ELD support status.
After we set the format, probe if we can do EXPBUF and enable/disable
the ALLOC_BUFFERS flag on the port.
This should gracefully handle the case where EXPBUF is not available.
Fixes#3821
When we try to alloc buffers but EXPBUF is not supported, make sure to
clear the alloc_buffers flag so that the caller can try again with
allocated buffers instead.
See #3821
Reduce fallback delay values used when BT device doesn't provide the
information itself.
It may be better to have audio late than early, so use values that are
probably close to or below the delays of majority of headsets.
Don't include the quantum in latency: the latency relative to graph
cycle start doesn't depend on the quantum. Instead, the audio packet
size determines it.
Enable the /Internal media class hack also for SCO.
Session manager can use this to adjust SCO sink/source media.class when
it is going to emit front-end nodes hiding the hardware ones.
The rfcomm list may contain various AG & HF ones, so the profile must be
checked everywhere they are looked up.
Fix the rfcomm lookups everywhere to do it.
Fixes Pipewire<->Pipewire HFP connections, and sending HFP HF commands
to HSP or AG.
HFP 1.9 adds LC3 as a possible codec in addition to CVSD & mSBC.
E.g. Pixel Buds Pro latest firmware supports it.
Add the RFCOMM side and codec selection for it.
Move some of the tracking code for the DLL to where it is used.
Add resync.ms (default 10) option at which we give up rate adjusting
and instead do a hard resync. This results in a jump in the position
of the graph clock.
Devices may advertise other values, but not certain they will work well
in duplex configuration.
E.g. my Samsung Galaxy Buds2 Pro emits buzzing sound with 48kHz duplex
input.
Don't believe QoS values recommended by the device, which may be
suboptimal. Instead, pick the values from the BAP v1.0.1 Table 5.2.
Link: https://github.com/bluez/bluez/issues/713
The PAC profile UUIDs do not appear in the UUID list, but are still
useful to know before SelectProperties.
Set them ahead of time based on the visible remote endpoints.
The "default" codec is the one with fill_caps != NULL, and should be
picked if we don't know which one we are using.
Fixes showing AAC-ELD as supported when it's not, which happened because
it's ordered before the default AAC in the codec list unlike the other
"shared endpoint" codecs.
