Audinate AES67 devices send SAP messages with 30-second interval, so hardcoded timeout has to be bumped. Just bumping it will reduce efficiency of common RTP module use-case, so a config is introduced for this. 70 second will be set as default for AES67 mode.
Always first use the env var and then check the properties. So that
PIPEWIRE_CORE=pipewire-1 PIPEWIRE_REMOTE=pipewire-1 make run runs
everything on pipewire-1 sockets regardless of the config files.
Also PIPEWIRE_NODE always needs to be taken into account first.
Implement the combine-sink module with the native module.
Make sure we use the same logic to wait with emitting the module loaded
signal until we have seen all the sink_inputs of our module.
Make sure we also use the timeout to signal module failure when we don't
see the nodes.
The module can:
- Make a sink that sends all or some channels to other sinks.
- Make a source that combines multiple sources into one.
The selection of what streams to combine is implemented with rules so
that the selection is very configurable. By default all Audio/Sink or
Audio/Source nodes are selected.
Actually move the stream properties in stream.props object for the pipe
tunnel module.
Set pipe.filename on the node. Remap this to device.string in pulse.
Add some more default properties on the pipe nodes.
See #2973
Choose the closest match to reduce resampling loss. If match is not equal, resample.
Keep the backwards compatibility retaining `filename` property. When both set, `filenames` takes priority
Make a real debug context with a log function and move it to a new file.
This way we don't need to redefine a macro.
Make a new context for debugging to a log file. Make new functions to
debug to a log file.
Move the stringbuffer to string utils.
Integrate file/line/func and topics into the debug log.
We can remove some more things from the pipewire log_object function and
also add support for topics.
Keep per stream audio info. We copy the global rate and format to
everything but allow for the channel positions to be overwritten
per stream. Invalid channel positions will revert to the default
again.
With a taget.object, this makes it possible to link the echo cancel
stream to specific pro audio sink ports.
Fixes#2939
Refactor get_rt_priority_range().
When we can't set the requested priority, use rlimit to clamp it
and try again. If this clamped value is bigger than RT_PRIO_MIN,
we can allow this.
The result is that the RTPRIO is set to the max of the user rlimit
RTPRIO, as long as it's more than 11. Previously it would fall back
to RTKit.
Add some more formats that can be used in pipewire streams so that
pulseaudio will see them as valid devices/streams.
It is possible that this will result in an invalid format on the client
when there is no pulseaudio format defined, but that is ok.
See !1499
If only one of rates is provided, propagate it. If both are provided and are inequal, warn.
This configuration appears to be broken for obvious reasons
The pulse API uses either the node.name or object.serials so pass this
along in the TARGET_OBJECT instead of the NODE_TARGET now that
wireplumber handles this.
When we start the drain, we unpause the stream. When we conplete the
drain, we unpause again, which does nothing when the stream was already
unpaused. However, this leaves the drained state on the stream and so
the stream will never be able to play new data.
Trigger a new pw_stream_set_active() with the current stream state to
clear the drained state.
Fixes#2928
Incompatible changes between ROC v0.1.X and v0.2.X require
adjusting the ROC modules' code. The largest change is going
from `roc_address` to `roc_endpoint`. There is also a breaking
change, the removal of `local.ip` parameter from module-roc-sink
as `roc_sender_bind()` has been removed.
The API usage was modelled after https://github.com/roc-streaming/roc-pulse
See #1757Fixes#2911
