Let codec decides when rtp packet need to be sent (terminated by MTU size in most case).
LDAC encoding loop can now be terminated by reading if frame_num is written, no 'frame_count' updating is needed.
RTP payload fragmentation can now be implemented more easily based on this.
When we set the volume on a port, make sure we also set the save flag
so that the flag is put into the volume changed event and the session
manager can save it.
Fixes#995
Allocate up to the first 16 ports, use the last 2 ports and free the
first 14 ports.
This ensure our ports are not among the first ports so that port 128
and following are for normal apps, what is usually expected when
PipeWire is not running.
Fixes#951
The start-delay adds extra silence to the buffer before starting the
playback. The idea is to have more time to adapt to the device
startup and set the timer more accurately.
See #983, #431
Auto-connect all paired & trusted devices on startup.
Since devices that already connected or powering off would reject the connecting requests, it should be fine with this behavior.
Also reconnect remaining profiles if only partial profiles are connected.
If oFono or hsphfpd are not installed, just log an info message
that they are not used. The native backend will work fine and we
don't want to log warnings.
See #971
When we are following the resampler requested size in capture, never
keep queued buffers around or we might get out of sync with the
requested size and cause cracks and pops in the resampler.
See #805
So that we can know if this node has routes or not.
Nodes without routes might need their volumes restored directly
with the node properties, like streams. Nodes with a route need their
volumes set on the device managing the node.
Just leave them be. Something else is supposed to changed the volumes
if needed. In the usual case where nothing changes, we don't want to
override the volumes.
When I enforing HFP codec switching between HFP profiles, it has a chance of being aborted by ECONNABORTED. Seems a retry is sufficient for work around it.
Remove the soft mute/volume events, add a new method to get the soft
volume and use the volume_changed event to emit the changed soft and
monitor (real) volumes event for the node.
Make sure the monitor ports always uses the monitor volume, which is the
real volume unaffected by the mixer volumes.
This configures the soft and real volume on the sink/source in all
cases and makes the monitor port follow the real volume of the sink.
See #897
Implement switching HFP codecs in bluez5-devices, currently only for
backend-native.
Codecs are exposed via profiles similarly as for A2DP.
Some hardware appears to not properly reply to the +BCS message. Catch
these cases with a timeout, in which case we fall back to previously
existing transports.
Initial Props value are parsed from device settings, further changes are triggered by 'set_param' on a2dp node.
Codec can then use props to tweak its transcoder.
Add media.name so that restore-stream can store this stream.
Remove node.driver because the stream is slaved to the connected ALSA device.
Remove pause-on-idle in all cases, as this is now configurable from
the config file.
This profile is meant to be used with audio gateways, such as mobile
phones, making pipewire act as a headset. It activates all 3 "dynamic"
nodes (all of which are "Stream/*/Audio"), allowing both A2DP source
and HSP/HFP AG to be available at the same time. Ultimately, the remote
device (the AG), is the one that decides which profile to use and pipewire
just creates/destroys the appropriate stream nodes dynamically.
To make things less confusing, the HFP/HSP profile is now only available
if the remote device is a Head Unit and the A2DP profile is only available
if the remote device has an A2DP Sink.
If the device has both A2DP Source & A2DP Sink (not sure if this is a real world
possibility, but just in case...), the A2DP profile allows using them both,
while the AG profile will only allow the source.
In addition, to keep things less complex, the routes are now only used for
device nodes (the "Audio/*" ones). A2DP source and HSP/HFP AG never have a route.
Restoring their props should be possible to be handled by the restore-stream
module.
