As AG, set node.rate for output streams that originate from remote
source, so that graph switches rate as needed. This follows what
pipewire-pulse etc. do.
The patch is by Wim.
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because rfcomm is no longer valid.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.
Do not add/connect the profile for unconnected devices.
This fixes the issue where a previously paired ASHA device
would show up in the pavucontrol output device list even
when not connected.
This allows our a2dp-source (sink endpoint) to report delay to the
remote source endpoint. Setting the "Delay" property is already
implemented, but by failing to configure DelayReporting, bluez rejects
the property change.
The resampler delay was off by one sample, so remove the corresponding
fudge factor here. This matters for BAP output synchronization.
The resampler has also some fractional delay, so there can still be
sub-sample offset between the original and resampled timelines. This is
not currently taken into account.
Don't mix endpoint and transport paths, they're generally different.
If ASHA transport already existed, free the old one but not the device,
as the ASHA device existence should track how it appears in DBus.
G722 codec id for ASHA is chosen arbitrarily to be different from
A2DP and BAP codec IDs. ASHA spec does not specify a codec ID like
A2DP/BAP.
In places where codec_id comparisons are done, ensure that the check
is done against the codec of the right type viz. A2DP/BAP/ASHA.
