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.
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.
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.
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.
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.
The +BCS event may interrupt any of the initialization commands after
SLC is established and by changing the state here we may lose track
of the initialization sequence.
There is no reason to have the hfp_hf_bcs state anyway. If the
initialization sequence is over, we can remain in the hfp_hf_vgm state.
Print the error in td->err (the SO_ERROR) or else let the user know
this is a regular hangup. The previous printout was always reporting
EAGAIN (remainder in errno from the event loop code, I suppose)
as an error, without any real data.
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.
HFP/HF/TWC/BV-03-C test, which setup an active and a held calls,
expects to receive AT+CHLD=1 (release and swap calls) instead of
AT+CHUP on active call hang up request.
As this changes the active call to disconnected and held call to
being active, the call states should be managed in hfp_hf_hangup
instead of waiting for +CIEV (callheld=0) event which will drop
the previously held call before AT+CLCC reply can inform this call
is now active.
HFP/HF/TWC/BV-01-C test creates an incoming call as soon as the SLC is
completed, i.e. a +CIEV: <callsetup>,1 event just after AT+CHLD=? reply
has been received. This try to parse the rfcomm->telephony_ag->call_list
which has not yet been created.
This commit move the telephony_ag creation to the SLC completed event.
In this error condition, execution is supposed to return immediately
because rfcomm is no longer valid. However, the code was incorrectly
changed to jump to the done label, which would try to use rfcomm
again to process pending commands.
To be sure that the AG reply correspond to the command sent, this
postpone the new command if previous reply (OK, ERROR or +CME) has
not yet been received.
The postponed command is sent on reception of the reply.
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.
The HF should declare the features supported before trying to send
related AT commands.
It should only send AT+CHLD related commands if the remote AG supports
them.
On some phones (at least), sending AT+CHLD=0 while there is no held
call returns an error. Make sure we don't do this to avoid returning
an error on D-Bus when in fact hangup_all actually succeeds.
This commit implements the dial, answer and hangup callbacks to manage
call when connected to HFP AG device.
This adds +CLIP support to be able to get remote party phone number.
