Add port.ignore-latency prop, which if true causes peer ports to ignore
the latency of the given port.
This is useful for ports that are not intended to affect latency
calculations of other ports, such as ports in monitor streams.
A DBusMessage needs to be unref-ed after sending it regardless
whether or not it was successfully sent. So do that in
`mm_dbus_connection_send_with_reply()` so that the callers
do not need to deal with that.
It is inherently racy, and we have a better way to ensure that
we won't autostart the service:
dbus_message_set_auto_start()
So use that.
This commit also adds a missing call to `dbus_pending_call_unref()`
and indirectly fixes a type mismatch (`dbus_bool_t` vs. `bool`)
that was present in `is_dbus_service_running()`.
The DBusError passed to `dbus_set_error_from_message()` must
be initialized, otherwise libdbus aborts:
dbus[129473]: arguments to dbus_set_error_from_message() were incorrect,
assertion "(error) == NULL || !dbus_error_is_set ((error))"
failed in file dbus-message.c line 4043.
This is normally a bug in some application using the D-Bus library.
The transport set volume call may take a long time or never complete, so
make them async to not block main loop.
Also reduce log level to info for the failed volume setting, as this is
something the user can do nothing about.
When we're using the peaks resampler, allow resampling, even when it is
disabled in the config.
The peaks resampler is just for GUI and would not really change the
signal, so we can allow this.
Do BAP audio location selection properly in SelectProperties, now that
BlueZ provides the supported locations there. Remove a previous
workaround.
The audio location in SelectProperties determines the audio channel
allocation, which determines the channel positions.
Add more properties to sink/source to make them always process and
have the right priorities. Make sure the sink has higher priority than
the source so that we can pull in samples through the graph.
Only trigger the graph cycle when driving the graph.
Make sure to return something from process to signal that the graph can
continue processing.
Use a new boolean to check if the follower is allowed to emit a ready
event. This can be done right after negotiating.
Set the started field to true after we finish setting the state of the
converter and follower. This fields is used to block calling into the
process function before we complete the setup.
This avoid a crash in always-process nodes when the node is scheduled
before the audioconverter completes setup.
The duplex polling issue was due to spa_loop_add_source failing
when source and sink were both using the same fd. We now dup, so the
issue no longer exists.
Remove the now unnecessary workaround, and check the return values from
spa_add_source.
Don't emit node error for A2DP duplex sink channel, or when BAP server.
These can occur under normal conditions (remote side suspends
transport), and are not errors.
When the node receives a set_param Props, it calls the follower
set_param implementation. If that one calls set_param again on the
adapter, discards the original set_param.
This makes it possible for the follower to intercept the Props param and
set a modified version on the converter. This can be used to intercept
volume changes.
When a sink contributing to an ISO CIG does not have data, output
silence for it, as long as at least one sink in the CIG is running.
Only if writes to sockets fail, pause all streams to reset
synchronization.
This way we write exactly the same number of packets for each CIS at the
same time, which probably is the best tested configuration in BT
adapters and devices. We also don't then have to pause output if some
sinks are not running or miss their timing, as we generate silence on
the fly.
When using iso-io, have it initialize the codec instance, and have
media-sink uses that instance, so that silence and actual audio are
encoded with the same codec.
Use the "high-reliability" values for QoS parameters instead of the
low-latency ones. Under some condition BlueZ does not pass on the
endpoint QoS values to us, in which case we may end up selecting bad
latency.
Determine correctly if we are resampling, and have the associated delay.
Add off-by-one sample adjustment to the resampling delay, which seems to
correctly align the resampled audio with non-resampled.
The WebRTC echo canceler can support different rates and channels for
the record, out and playback streams.
Add a new method to pass this config to the echo-canceler.
Resynchronize ISO streams on playback (re)start, so the stream positions
are aligned immediately. This is better than relying on rate matching
to correct any offsets.
Use a separate variable to hold the maximum amount of drift we allow
between driver and follower. Ensure this value is smaller than the max_error
and period size so that we have at most 1 period of drift.
