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.
The volume synchronization could be done even if there's no audio link
and so no transport opened.
This patch allows to send the Speaker (AT+VGS) and Microphone (AT+VGM)
commands at the end of the SLC. And to exchange volume updates using the
telephony DBus interface, even without a transport.
This allows implementing UI mechanisms to transfer the audio of a call
to the HF (pipewire) only when the user explicitly asks/allows it.
Normally, when a call is connected, the phone initiates a SCO connection
and the HF accepts it, transfering audio automatically. In order to
allow for user interaction, this patch enables the UI to set the RejectSCO
property to 'true' in order to automatically reject the SCO connection.
Later on, at the UI's discression, the audio may be reconnected by calling
the Activate() method, which sends AT+BCC to re-initialize the SCO channel.
A configuration file option is also added to configure the default value
of the RejectSCO property. By setting this to 'true' in the config file,
it is possible to implement rejecting the audio of a call that is already
active at the time the Bluetooth connection to the phone initializes.
This is useful for implementations that do hardware offloading of the
SCO audio channel and need to communicate state information to the
hardware (at least).
Start call id at 1 as for the index calls in HFP, and move this id
to spa_bt_telephony_[ag|call] so they can be used by CLCC to retrieve
the related call.
if enhanced call status is supported, send AT+CLCC on +CIEV events to
get the calls information.
