Set different icons for A2DP & HFP output routes, so that they look
different (in Gnome).
Don't call the non-HFP output route as "headset" or "handsfree" in this
case, to be less ambiguous about microphone availability.
Also set device.icon-name for the device too.
While ASHA does not really use the D-Bus device set interface
but since ASHA has a device-wide HiSyncId and needs to handle
left and right side via a combine sink, use the device set
notion for ASHA as well.
Let's just directly use the next timer value from the other side
directly, and the reference time as well to calculate the expected next
sample position we want to send from on this side.
For ASHA stereo, the timer for flushing packets on both sides of a
pair should be as closed to each other as possible.
Also set seqnum before first flush so other side sends the correct
packets at the start.
In ASHA, we might need to sync sequence numbers between
left and right side media sink. If the sequence number
is added in start encode, it becomes difficult to set
the sequence number again when a call to reset_buffer
in media sink might have happened already.
This effectively reverts commit ab5f81b9a.
ASHA devices with the same HiSyncId are a pair and this
will be used later on to set them up together with a
combine sink like device set for BAP. Side information
is required for channel information when setting up the
combine sink.
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.
Use kernel-provided packet reception timestamps to get less jitter in
packet timings. Mostly matters for ISO/SCO which have regular schedule.
A2DP (L2CAP) doesn't currently do RX timestamps in kernel, but we can as
well use the same mechanism for it.
Add configuration options for the BAP/PACS sink and source endpoint
location (= channel positions) and context settings.
Although BlueZ associates these with individual endpoints, in the PACS
spec they are actually device-global, so configure directly in monitor
settings.
Parse BAP settings in a single place, and simplify QoS customization a
bit.
Ensure the selected preset gets selected.
For all the BAP codec settings, use device settings instead of global
monitor ones.
The current BAP QoS configuration allows to register a sampling
frequency based on the configuration done using wireplumber configuration.
However, for a scenario were the user need to use a specific SDU framelength
it cannot be done as the select_bap_qos function selects the QOS based on
priority and hence it will use the best possible config rather than the user
configured.
This PR adds option to select the QoS set based on user configured value. If
the remote device doesn't have the user configured capabilities it will always
use the best priority config.
Further, this change also allows the user to set RTN, Latency, Delay QoS config
for certain use case to have controller use optimum bandwidth usage.
Below are the example configuration on setting LC3 capabilities in config file:
bluez5.bap.set_name = "48_2_1"
bluez5.bap.rtn = 5
bluez5.bap.latency = 20
bluez5.bap.delay = 40000
bluez5.framing = false
The current BAP unicast QoS configuration allows to register a sampling
frequency based on the configuration done using wireplumber configuration.
However, for a scenario were the user need to use a specific SDU framelength
it cannot be done as the select_bap_qos function selects the QoS based on
priority and hence it will use the best possible config rather than the user
configured.
This PR adds option to select the QoS set based on user configured value. If
the remote device doesn't have the user configured capabilities it will always
use the best priority config.
Further, this change also allows the user to set RTN, Latency, Delay QoS config
for certain use case to have controller use optimum bandwidth usage.
Below is the example for the options that can be configured & selected
in config file:
bluez5.bap.set_name = "48_2_1"
bluez5.bap.rtn = 5
bluez5.bap.latency = 20
bluez5.bap.delay = 40000
bluez5.framing = false
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.
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).
