The state_changed callbacks fulfill multiple roles, which is both a problem
regarding separation of concerns and regarding code clarity. De facto,
these callbacks cover error reporting, opening connections, and closing
connection, all in one, depending on a state that is arguably an internal
stream detail. The code in these callbacks tie these internal states to
assumptions that opening/closing callbacks is directly tied to specific
state changes in a common way, which is not always true. For example,
stopping the stream may not _actually_ stop it if a background send timer
is still running.
The notion of a "state_changed" callback is also problematic because the
pw_streams that are used in rtp-sink and rtp-source also have a callback
for state changes, causing confusion.
Solve this by replacing state_changed with three new callbacks:
1. report_error : Used for reporting nonrecoverable errors to the caller.
Note that currently, no one does such error reporting, but the feature
does exist, so this callback is introduced to preserve said feature.
2. open_connection : Used for opening a connection. Its optional return
value informs about success or failure.
3. close_connection : Used for opening a connection. Its optional return
value informs about success or failure.
Importantly, these callbacks do not export any internal stream state. This
improves encapsulation, and also makes it possible to invoke these
callbacks in situations that may not neatly map to a state change. One
example could be to close the connection as part of a stream_start call
to close any connection(s) left over from a previous run. (Followup commits
will in fact introduce such measures.)
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
32 bits are enough, and additionally this also fixes an incorrect
format string, which caused the default `audio.rate` to be
incorrectly set on some platforms, such as 32-bit arm ones.
Fixes#4080
Make a new function to set the rtp stream in the error state.
When we fail to start the stream, set the error state. Otherwise (like
when we try to use an invalid interface name) the socket create will
fail but the stream will still try to send data to the invalid socket.
Use the sess.latency.msec also for the sender and use it to control the
NODE_LATENCY. Make it a float to be in line with the other time values.
Set is to a default of ptime, which was what it used to be.
This makes it possible to set the ptime to a smaller value than the
sess.latency.msec so that we send out multiple packets per quantum.
This will result in some bursty output for now but with a timer that can
be improved later.
Update the docs a little, mention the new rtp.ptime and rtp.frametime.
This method can be used to access the param_changed method of the
underlying pw_stream.
Also adds new public functions rtp_stream_set_param and
rtp_stream_update_params which plum things through to pw_stream_set_param
and pw_stream_update_params respectively.
