Use the "high-reliability" values for QoS parameters instead of the
low-latency ones. Under some condition BlueZ does not pass on the
endpoint QoS values to us, in which case we may end up selecting bad
latency.
Determine correctly if we are resampling, and have the associated delay.
Add off-by-one sample adjustment to the resampling delay, which seems to
correctly align the resampled audio with non-resampled.
Resynchronize ISO streams on playback (re)start, so the stream positions
are aligned immediately. This is better than relying on rate matching
to correct any offsets.
Emit BAP device set nodes, which the session manager can use to combine
the sinks/sources of a device set to a single sink/source.
Emit the actual sinks/sources with media.class=.../Internal to hide them
from pipewire-pulse.
Add separate device set routes to the set leader device. Other routes
of the set members will be marked as unavailable when the set is active.
Accordingly, return failure for attempts to set these unavailable
routes, so that volumes etc. of the "internal" nodes are only controlled
via the device set route.
For ISO server sockets, the QOS struct from getsockopt contains values
with different meaning from ISO client socket. Get the values via DBus
instead, which is right in both cases.
If BAP codec configuration is mono channel with unspecified location,
set the channel position from transport location.
This in principle should be set in SelectProperties, but currently BlueZ
doesn't tell us that yet there, so we hack it up later on.
The number of channels is determined by Audio_Channel_Allocation.
One frame block contains all channels. (BAP v1.0.1 Sec. 4.2)
Fix the handling of frame blocks and counting of numbers of channels. We
support and configure only one frame block per packet.
Consider omitted Audio_Channel_Allocation to indicate MONO stream (see
BAP v1.0.1 Sec 4.3.2).
Use the ISO IO helpers to get synchronized BAP output, and rate match to
the ISO schedule.
The rate matching is necessary, since the driver may be ticking at a
corrected rate, different from the ISO interval rate.
Add factored out helper for ISO socket I/O.
ISO sockets need synchronization of writes and audio position for
different stream fds in the same isochronous group, and it's easier to
separate out the part that coordinates it.
Avoiding unnecessary release + reacquire when nodes restart makes sense
for all transport types. Do timed releases for all transport types, not
only SCO.
Do transport release synchronously for simplicity. BlueZ handles
releasing while acquire is pending, but acquire while release is pending
would fail the acquire.
Otherwise we need to maintain an operation chain to handle trying to
acquire/release while the other operation is pending. This makes things
complex with little gained, as releases generally don't block for a long
time.
Drivers should only read the target_ values in the timeout, update the
timeout with the new duration and then update the position.
For the position we simply need to add the previous duration to the
position and then set the new duration + rate.
Otherwise, everything else should read the duration/rate and not use
the target_ values.
Make BAP nodes align the first sample of their packets at multiples of
the ISO interval, counted in the shared graph sample position. This
skips a few samples (< 10ms) at the start of playback to ensure the
alignment.
Since the sinks align their flush timing to the graph time, this also
results to them sending packets corresponding to the same graph time at
the same real time instants.
Due to packet queues in kernel/controller, the playback may still be off
by multiples of packets. Kernel changes are needed to address that part.
This works towards making BAP left and right channels to be
synchronized in TWS headsets, where the two earpieces currently appear
as different devices.
If transport goes into error state too often, fail instead of trying to
acquire it again.
This avoids getting into a tight acquire->fail->reacquire loop.
We need to acquire and release all transports in the same CIG at the
same time.
Due to current kernel ISO socket limitations, this cannot be done one by
one.
Now that sinks/sources can do transport acquire asynchronously, remove
the workaround that made it synchronous. Do release still synchronously
however.
Change A2DP/BAP transport acquire and release to be async.
Since BlueZ acquiring ISO sockets blocks until all sockets in same CIG
are acquired, BAP transports must be acquired asynchronously.
Allow asynchronous changes in transport state in the sinks/sources.
Also allow transport acquire to be actually synchronous, in this case it
must set transport state during acquire call.
Separate driver start/stop from transport start/stop.
Emit any remove node events before resetting initial profile. It
indicates to the session manager that nodes if any went away before
device disconnected.
Usually the profile is removed first which removes the nodes. This
depends on ordering of events from bluez, which apparently can be
different depending on how remote device disconnects.
Add some guards against doing processing when there has been an error or
the node is not started. Set error status to IO. Continue driving on IO
errors.
In media-sink, there's no need to set RCVBUF.
In media-source, we don't need to set NONBLOCK, as reads are done with
DONTWAIT. Don't set SNDBUF as it's not needed there. Don't set RCVBUF,
but use the (big) kernel default value: decode-buffer will handle any
overruns. Small values of RCVBUF might cause problems if kernel is
sending packets in a burst faster than we wake up.
On underflow in sources, pad with explicit silence. This avoids the
audioadapter from getting off sync from the cycle. That causes problems
as driver when we want to produce a buffer only a the start of the
cycle.
In some cases, it's also possible that the io already has buffer at the
start of the cycle when rate matching as driver. Currently, we don't
produce buffer in this case, but we should. Fix that by doing things in
the exact same way as ALSA sources do.
Delay output by one packet, so that we never need to wait for
node_process to supply more data when a packet is due out, and can write
audio packets at exactly equal intervals (up to timer/io accuracy).
In principle, this should not be necessary. However, enable it for now,
in case this improves the various stutter/etc. bug reports.
After flushing a packet, encode the next one immediately if we already
have the data. This makes the flush timing more accurate (std ~4x
smaller) as we don't need to wait for the encode.
The maximum receive buffer target of 6 packets may be too small when
there's huge jitter in reception. Increase it so that we may use all
buffer available if needed (2*quantum_limit = 370 ms @ 44100).
For SCO, explicitly set maximum buffer to 40 ms, so that latency cannot
grow too large there. For A2DP duplex, set it to 80 ms for same reason.
These are close to the old 6*packet limit.
For BAP server audio sink, set buffering target so that we try to match
the target presentation delay. Also adjust requested node latency to be
smaller than the delay.
Also fix BAP transport presentation delay value parsing, and parse also
the other BAP transport properties. Of these, transport latency value
needs to be taken into account in the total sink latency.
