- NDI HX Camera Android in the past used 1ns instead of 100ns as unit
for timecodes/timestamps.
- NDI HX Camera iOS uses 0 for all timecodes and the same non-zero
value for all audio timestamps
Detect such situations and try to compensate for them. Also add a new
"auto" timestamping mode that prefers to use timecodes and otherwise
falls back to timestamps or receive times.
Fixes https://github.com/teltek/gst-plugin-ndi/issues/79
Audio/video are in practice not always from the same clock and can have
different behaviours with regards to clock rate and jitter. Handling
them separately generally gives better results for the timestamps output
by the source element.
This is no longer available as this could lead to building a defined
value in Rust which could be interpreted as undefined in C due to
the sentinel `u64::MAX` for `None`.
Use the constants (e.g. `ONE`, `K`, `M`, ...) and operations to build
a value and deref (`*`) to get the quantity as an integer.
Always first try draining queued data in the loop and only start waiting
if there's nothing to drain right now. Otherwise data might have to be
drained right now but we still wait and nothing is ever waking up the
source pad task again.
Also make sure to not wait multiple times on the same gst::ClockId but
instead unset it after waiting on it and no new one was scheduled in the
meantime. Future waits on the same ClockId will immediately return and
instead we should wait on the condvar if no new ClockId is available.
When call_timeout is triggered, request will fail
irrespective of the retry setting. call_timeout define
max time request can take along with retry.
It can be solved by either setting call_timeout to
retry * call_attempt_timeout or not setting the call_timeout.
As per thread call_attempt and rety setting is enough.
https://github.com/awslabs/aws-sdk-rust/issues/558
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1410
Created a new plugin 'webrtchttp' to implement all the
WebRTC HTTP protocols under /net/webrtc-http directory.
WhipSink wraps around 'webrtcbin' with HTTP capabilites
to exchange SDP offer/answer so an ICE/DTLS session can
be established between the encoder/media producer (WHIP client)
and the broadcasting ingestion endpoint (Media Server).
Once the ICE/DTLS session is set up, the media will
flow unidirectionally from the WHIP client to the
broadcasting ingestion endpoint (Media Server).
Spec:
https://www.ietf.org/archive/id/draft-ietf-wish-whip-04.html
