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.
