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.
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
If the sender is reset, the RTP stream may return, but may no longer
correspond to the stream for which we loaded this instance of
module-rtp-source. A power cycle or network reset on Dante devices
causes a new SDP and SSRC to be selected, for example.
So let's make sure we don't consider invalid receives while tracking our
"receiving" state. Arguable, we should bail entirely if this happens.
There isn't a good way to surface this information to the module owner
yet, so let's publish the information on the stream so we can try to
manage things in policy.
Expose the acquire_loop/release_loop functions and use them in the
modules.
Make sure the nodes created from the module use the same data loop as
the module. We need to ensure this because otherwise, the nodes might
be scheduled on different data loops and the invoke or timer logic will
fail.
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 `getaddrinfo` in `parse_address` instead of `inet_pton`.
Display Ipv6 addresses with scope identifiers correctly in `get_ip`
functions using `if_indextoname`.
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.
Only listen for packets from source.ip in the unicast (v4) case by
calling connect(). If packets from any source address is wanted, set
source.ip = "0.0.0.0".
Use the timestamps directly as the ringbuffer index. We can save some
conversions to bytes and there is a direct mapping to RTP timestamp,
clock position and ringbuffer index.
Simplify the source a little. Remove the buffering state, we always
start with read and write pointers separted by the target buffering.
Always use the timestamp of the graph clock for RTP packets. Add an
option to apply a random or fixed offset.
Add a ts-refclk option on the sender to specify a reference clock to
use for timestamping. This will activate the direct timestamp mode and
signal this in the SDP.
Parse ts-refclk and ts-offset from the SDP. Make it possible to match
them in rules. Add option to activate the direct-timestamp, where the
rate matching is disabled and the timestamps are used directly for
writing and reading from the ringbuffer using the graph clock.
This makes it possible to set a PTP clock on sink and source and avoid
rate matching by using a shared clock.
