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.
Try to keep half the packet size in the ringbuffer as well. This helps
us adapt to the packet size of the sender.
Drop samples from the ringbuffer for the first packet we read. This
makes us lock onto the stream with the exact requested latency.
Audinate AES67 devices send SAP messages with 30-second interval, so hardcoded timeout has to be bumped. Just bumping it will reduce efficiency of common RTP module use-case, so a config is introduced for this. 70 second will be set as default for AES67 mode.
