If the rate is 0/0, converting nsec to ticks doesn't work and will
result in 0 ticks, and it is not possible to convert ticks back to a
timestamp.
This can be reproduced by connecting a GStreamer pipewiresrc to a
libcamera node. The libcamera-utils has a rate of 0/0 and the
pipewireclock won't be able to determine a correct time with that. This
error was caused by Commit 89993a3cc6 ("gst: enable the pipewire ticks
as a clock source").
Fix this by using the nsec as ticks and setting the appropriate rate.
Idle the source when no packets are received and resume when new packets
arrive.
Add a stream.may-pause property to pause the stream when no packets are
received during the timeout window.
Make sure the rtp.streaming property is updated correctly and as soon as
we get the first packet.
Fixes#4456
Reorganize some code to separate the creation and sending of the SAP
message.
Check if when the node changed, we have an actual change in the SDP
before we send BYE and the new SAP message. It's possible that nothing
changed, for example when the node simply changed state or an unrelated
property.
When we do any other blocking in the render function, we should unblock
and call _wait_preroll() when we go to PAUSED.
We can have this situation when all the buffers are queued in the
pw_stream and we get a new _render() call. We can't get more buffers
from the pool and so we must block and wait. When we go to PAUSED we
need to unlock and go to _wait_preroll(). Implement this by setting a
pool paused flag that is set when the sink goes to paused, we can then
return a special value that does the wait_preroll().
See !2248
Setting bufferpool to flushing state in PAUSED state is preventing the
buffer flow if there is a seek/flush event.
Instead, set the bufferpool to flushing during the `flush-start` event
and clear it during the `flush-stop`
Deactivate/activate the stream during flush event only if the sink is
in the PLAYING state. In the PAUSED or READY state, the stream would be
inactive and we do not want to alter that
When we write samples, check if we make a jump in the ringbuffer and
clear the samples we jumped over.
If we don't do this, the reader side might pick up old samples that we
didn't write or clear but that are now available for reading after we
made a jump in the ringbuffer.
This migh not be exactly what pulseaudio does but it is good for now.
Fixes#4464
flush the pw buffers to the stream's queue during a FLUSH_START event
and return the unqueued pw buffers, if they are dropped/released without
being rendered, so they can be available to be dequeued for the subsequent
`acquire` calls
We now automatically move non-rt clients into non-rt threads so the
client-rt.conf is obsolete.
Move the module-rt in client.conf and add conditions to disable modules.
Transparently load client.conf in case applications still specify
client-rt.conf.
Custon configuration in the client-rt.conf.d/ should be moved to
client.conf.d/
Wireplumber and other system services use local real time timestamps in
logging, so it's more convenient if also PW uses them.
Add env var for selecting the timestamp type, default to "local".
Copy the server value to the context so that the locally allocated
buffers match the server quantum-limit and we don't cause xruns because
of too small buffers.
See #4490
Remove the chunk and add separate arrays with data and n_samples. This aligns
better with other methods and makes it possible to more easily reuse
arrays of pointers as input and output.
For some streams, the buffer size is changed and may exceed
the acquired buffer size which is acquired from the pool of
pipewiresink. Need split buffer and send them in turn for
this case.
We want to track the difference between the PTP timestamp (now) and the
last RTP send, not the synthesized next RTP timestamp (which will always
be smoothly incrementing).
Add support for latencyOffsetNsec Prop, which just controls the nsec
part of the ProcessLatency.
This is needed to support latency offset in Pulseaudio apps when using
loopbacks as front-end nodes to underlying sinks.
