Never is useful for some RTSP servers that report plain garbage both via
RTCP SR and RTP-Info, for example.
NTP is useful if synchronization should only ever happen based on RTCP
SR or NTP-64 RTP header extension.
Also slightly change the behaviour of always/initial to take RTP-Info
based synchronization into account too. It's supposed to give the same
values as the RTCP SR and is available earlier, so will generally cause
fewer synchronization glitches if it's made use of.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6543>
Both were entangled previously and very hard to follow what happens
under which conditions. Now as a very first step the code decides which
of the two cases it is going to apply, and then proceeds accordingly.
This also avoids calculating completely invalid values along the way and
even printing them int the debug output.
Also improve debug output in various places.
This shouldn't cause any behaviour changes.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6543>
Currently in rtp_session_send_rtp(), the existing ntp-64 RTP header
extension timestamp is updated with the actual NTP time before sending
the packet. However, there are some circumstances where we would like
to preserve the original timestamp obtained from reference timestamp
buffer metadata.
This commit provides the ability to configure whether or not to update
the ntp-64 header extension timestamp with the actual NTP time via the
update-ntp64-header-ext boolean property. The property is also exposed
via rtpbin. Default property value of TRUE will preserve existing
behavior (update ntp-64 header ext with actual NTP time).
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1580
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3451>
When syncing to an RFC7273 clock this will add the original
reconstructed reference clock timestamp to buffers in form
of a GstReferenceTimestampMeta.
This is useful when we want to process or analyse data based
on the original timestamps untainted by any local adjustments,
for example reconstruct AES67 audio streams with sample accuracy.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1964>
Constantly updating the ts_offset results in audiable glitches
when streaming audio using ntp-sync=true. By requiring a minimum
offset before updating ts_offset this can be mitigated. Added a
parameter which can be used to set min_ts_offset in ntp-sync mode.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1409>
