Add jack.connect-audio and jack.connect-midi to specify an array of port
names to link to instead of the default phyisical ports.
Also actually fixes linking the midi ports correctly.
Make the state_changed event and _get_state() function set errno with
the current error value if the state is in error, so that application
can use this to give more detailed error reporting.
Use this in alsa, v4l2 and pulse to give some other error codes than
EIO.
Fixes#4574
Some of the more common errors (caused by packet loss, network jitter, ...)
should be reported with INFO unless there is some indication about how
to fix the problem.
Fixes#4559
We don't initially have the SAP socket open, so we can't generate an SDP
(because we don't have the interface address). So in addition to the
regular flow, also trigger SDP creation after opening the SAP socket, so
we can have a valid SDP for the announcement.
The sending was broken in commit a44afd84. We delay the SAP fd openeing
for reasons explained in commit f2f204d6).
When both node and port are given, check that the port belongs to the
node. If it doesn't, it could be that we found a Port using the
object.id but we should have used the port.id of the node.
Because we don't know the stream state at the start of streaming, if
clients are deciding to connect on the basis of this flag, they will
never connect if we default to true. So let's be optimistic by default
and we'll find out on timeout if there actually isn't data to receive.
Some of the tools would like to connect to the manager socket first
because they are manager style apps. They however completely ignore any
of the configured sockets in the config and assume everything is the
default.
Fix this by adding a remote.intention = "manager" to those apps. This
instructs the protocol to first try to connect to a socket with the
-manager extension before attempting the regular configured socket.
This makes things work when you have sockets configured in /tmp
and have remote.name = /tmp/pipewire-0 in the config.
We use the remote name as a suffix for the default server address and so
it should not contain any slashes. Take everything after the last slash
if there is one.
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 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
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.
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).
