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.
Currently, the PipeWire daemon registers BlueZ LE Media Endpoints
with audio capabilities covering all settings defined in the BAP spec.
However, some scenarios might require the capabilities to be restricted
to specific configurations.
This adds a method to read LC3 codec specific capabilities from the
Wireplumber config file, and provide those settings when registering
Media Endpoint objects with BlueZ. If the values are not present in
the config file, all settings will be used by default.
Below is an example of how to set the LC3 capabilities in the config
file, to support the 16_2 setting from the BAP spec:
bluez5.bap-server-capabilities.rates = [16000]
bluez5.bap-server-capabilities.durations = [10]
bluez5.bap-server-capabilities.channels = [1, 2]
bluez5.bap-server-capabilities.framelen_min = 40
bluez5.bap-server-capabilities.framelen_max = 40
bluez5.bap-server-capabilities.max_frames = 2
For a BAP Broadcast Source endpoint, the QoS sync_factor enables the user
to adjust the Periodic Advertising interval based on the ISO interval
configured for the stream:
PA_Interval = sync_factor * ISO_Interval
Currently, this value is hardcoded to 2. This commit makes the sync_factor
configurable in the Wireplumber config file, along with the other config
parameters for BIGs.
Encoders and some decoders have additional internal latency that needs
to be accounted for.
This mostly matters for AAC (~40ms), as the other BT codecs have much
lower delays (~5ms).
Airpods don't follow the specification and set multiple bits in AAC
object type, including the ELD bit, but actually want AAC-LC. So check
the AOT in the right order.
When in A2DP sink role and remote end switches codec, BlueZ nowadays
appears sometimes emit first SetConfiguration (creating new transport),
and then ClearConfiguration (removing old transport).
Handle this case: emit profiles_changed event always when transports
come/go.
Redefine profiles_changed() to take bitmask of profiles whose connection
status changed, so we don't need to emit two remove+add events.
Interpolate buffer level to current playback position, and change its
definition so it directly corresponds to the total buffer latency. This
is also a bit simpler.
Now that BlueZ supports delay reporting in A2DP sink role, implement
that.
Report value that gives the total latency between packet reception and
audio rendering.
Also make Latency parameter in media-source to be not just a dummy
value.
