After AT+CLCC command completion, the calls which has not been listed
should be removed.
This list the calls returned by AT+CLCC to be able to find the ones which
has not been listed but still in the call_list.
Unicast BAP codec switch requires CIG reconfiguration, which cannot be
done if there is an acquired transport.
When doing BAP codec switch, disable nodes of other devices sharing the
same CIG.
To avoid problems with node start/stop, just remove and re-add them.
I'm not aware of any devices that support 48 kHz output in duplex
configuration, so disable that for now.
Doing this properly requires catching errors when on transport Acquire,
and switching to another configuration if the error was due to bad
configuration.
Due to how BAP specification works, it's not necessarily possible to
know whether a configuration was really accepted at earlier stage, and
anyway there's no proper error -> reconfiguration handling currently on
BlueZ side either.
If device supports duplex, show also separate sink-only/source-only
profiles.
Devices don't necessarily support high-quality audio in duplex profile,
so add sink/source only profiles.
This is also a workaround for the current situation that devices may
signal duplex support, but the attempted duplex configuration fails to
work.
Use SelectProperties() DBus API to reconfigure BAP unicast setup.
Add support to spa_bt_ensure_codec() to select whether to configure as
sink/source/duplex.
Simplify codec switching code: determine what switch to perform
immediately in spa_bt_device_ensure_media_codec().
The previous code doing "fallback" switching to various codecs is not
useful, as A2DP generally disconnects on the first failure and all
remote endpoints disappear.
Add iteration over multiple endpoints, for reconfiguring both source and
sink directions at the same time. This is in preparation of supporting
BAP reconfiguration (for A2DP there's usually only one direction
connected at a time).
Use codecs via media_codec in sco-sink instead of implementing the
encoding in-place.
In future, media-sink could replace sco-source to reduce code
duplication.
Use codecs via media_codec in sco-source instead of implementing the
decoding in-place.
Also slightly adjust media-source decode semantics.
In future, media-source could replace sco-source to reduce code
duplication.
This allows to connect the SCO link with old HFP AG devices which
doesn't support the codec negotiation.
The audio connection could be done even without an ongoing call.
When the Bluetooth earphones claim to support AAC `MPEG-2 AAC LC` type only,
PipeWire encodes audio using MPEG-4 FDK-AAC profile which enables unsupported
Perceptual Noise Substitution (PNS) encoder feature.
Use AOT_MP2_AAC_LC pseudo-profile which is AOT_AAC_LC with PNS disabled.
config.h needs to be consistently included before any standard headers
if we ever want to set feature test macros (like _GNU_SOURCE or whatever)
inside. It can lead to hard-to-debug issues without that.
It can also be problematic just for our own HAVE_* that it may define
if it's not consistently made available before our own headers. Just
always include it first, before everything.
We already did this in many files, just not consistently.
When we simply need to change some state for the code executed in the
loop, we can use locked() instead of invoke(). This is more efficient
and avoids some context switches in the normal case.
Including C headers inside of `extern "C"` breaks use from C++. Hoist
the includes of standard C headers above the block so we don't try
to mangle the stdlib.
I initially tried to scope this with a targeted change but it's too
hard to do correctly that way. This way, we avoid whack-a-mole.
Firefox is working around this in their e21461b7b8b39cc31ba53c47d4f6f310c673ff2f
commit.
Bug: https://bugzilla.mozilla.org/1953080
There is no need to encode the potential format in the format.dsp of
control ports, this is just for legacy compatibility with JACK apps. The
actual format can be negotiated with the types field.
Fixes midi port visibility with apps compiled against 1.2, such as JACK
apps in flatpaks.
If kernel socket queues for different streams get out of sync, it will
mess up time alignment of different streams. If that happens, flush to
resync.
If total latency becomes too large, flush queue.
Get accurate queue sizes from tx timestamping.
Use TX timestamping to figure out the accurate amount of unsent data,
including controller buffers. SIOCOUTQ does not report accurate data
size as it includes overheads.
Now that we have ASHA & BAP, this->device_set now refers to the device
set of the active profile. It cannot be used to produce
EnumProfile/EnumRoute.
Fix this by computing the device set for given profile(s) as needed.
3-way incoming calls are created in waiting state. When those calls are
hang-up before being active, the +CIEV: (callsetup = 0) should also be
managed for waiting calls.
One of the ideas behind retrying the sending of a failed packet with the
poll callback was to make sure that we do not end up with missing seqnums
by missing received credit due to some jitter.
However, the rate matching behaviour for ASHA is not clear and we do not
seem to face problems in local testing by just dropping the packet.
The two sides of a ASHA pair rarely if ever start together and the
sequence number was always a bit off due to the stateful nature of
reset_buffer and ASHA needing the sequence number to be matched to
the other side.
Simplify this by setting the sequence number for ASHA just before
flushing.
