Improve the spa_ump_to_midi function so that it can consume multiple UMP
messages and produce multiple midi messages.
Some UMP messages (like program changes) need to be translated into up
to 3 midi messages. Do this byt adding a state to the function and by
making it consume the input bytes, just like the spa_ump_from_midi
function.
Adapt code to this new world. This is a little API break..
Use kernel BT_PKT_SEQNUM (likely in Linux v6.17) to provide the ISO
packets sequence numbers. Fall back to counting packets if kernel is too
old to support the feature.
When we skip the CIEV event in order to update the call list with CLCC,
we may receive multiple +CLCC events or even none, if the calls are
disconnected. To avoid any mistakes, update the hfp_hf_in_progress flag
after the CLCC update is entirely done.
This removes the need to call poll() on the rfcomm socket in order
to wait for replies from the AG.
Use a queue to buffer all the commands that are to be sent to the AG
and match them to replies when they are received. Optionally associate
each command with a DBusMessage that is assumed to be a method call
from the telephony interface, which is then replied to when the rfcomm
command reply is received. Also associate each command with a state,
so that it is always deterministic what gets executed after the reply
is received.
On the telephony module, pass on the DBusMessage on the callbacks and
add a method to allow the receiver to send a reply. Only send FAILED
directly when the callback is not handled. Also, remove the return value
from the Dial() command (it was not advertised on the introspection
anyway) to make things easier.
Pass zero-length packets to the codec. BAP/ISO may use these to indicate
missing data.
Fix A2DP codecs to not parse input with spa_return_val_if_fail, that's
meant for assertions. Just return -EINVAL directly, it's normal that
input data may contain garbage.
If packet sequence number jumps ahead, or we would underflow, use
codec-provided packet loss concealment to produce some audio data.
When we produce it during underflow, skip the corresponding number of
sequence numbers of future packets.
If codec doesn't have PLC, keep the previous behavior (pad with zeros,
buffering pauses to wait for data).
LC3 and Opus have built-in support for packet loss concealment.
Add codec interface for that, and implement for LC3.
Extend media_codec interface so that packets not aligned with socket
reads can be handled, as in HFP. This is required for correct sequence
number counting, and for being able to run codec PLC *before* decoding
the next correctly received packet.
Update rate matching only once per process(). This ensures all nodes in
the group update their rate matching in the same way.
Also account for audio data in ISO output buffer in the reference time.
The calculations is in system clock domain, so when converting from
samples/duration to time rate difference should be accounted.
This does not have much effect in practice.
The rate matching calculations are done in the system clock domain. If
the driver ticks at a different rate, the correction factor needs to be
adjusted by the rate_diff.
This fixes ISO streams getting out of sync with each other when target
delay changes. This happens because typically one of them is the driver
and the other follower. Driver adjust clock rate, and follower does its
own adjustment *on top of that* so it rate matches more or less at
double speed. (The DLL of the follower to some degree corrects for
this, but can't do that when hitting RATE_CTL_DIFF_MAX and moreover it
acts with a delay.)
A2DP v1.4 uses the rfa bits for adding 5.1 and 7.1 configurations.
Clear those bits properly when sending configuration, in case remote
device sets them.
When dialing an incorrect phone number some phones (e.g. iOS 18.5)
replies with OK but never send +CIEV updates, so there's no way to
know that the dial is not in progress and the call object should be
removed.
This change waits for +CIEV event to create the call object.
Change media-source to use sco-io for HFP codecs.
Replace sco-source with media-source.
sco-source is mostly copypaste from media-source, only differed in the
IO handling.
Change media-sink to use sco-io for HFP codecs.
Move SCO fragmentation to sco-io side.
Replace sco-sink with media-sink.
sco-sink is mostly copypaste from media-sink, and only differed in the
fragmentation detail, which can as well be handled on sco-io side.
Don't try to write data in too large blocks.
This controls the maximum amount of data to send at once. sco-io will
buffer and fragment packets to the right size.
Previously in sco-sink, SO_SNDBUF was not set, so there could be a
longer queue in the socket.
E.g. LDAC doesn't have that. Add the missing guard that was accidentally
dropped in the rewrite.
Also explicitly filter out non-A2DP/BAP codecs that can't be used in
ensure_media_codec
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).
