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Define maximum latency and GST_CLOCK_TIME_NONE to prevent timeouts.
Potential fix for #528.
All gstreamer sources must be synchronized with the gstreamer clock.
Since aravissrc is a live source it can't give reliable increasing timestamps and instead works with durations.
Ideally that would mean first timestamp (0) + duration of frame. New timestamp + duration of frame, etc.
Since we cannot know what the duration of a frame is (some come faster, some slower, thus never being 100% accurate) gstreamer wants to know a latency range.
This patch defined the latency as min=framerate and max=GST_CLOCK_TIME_NONE.
That basically says expect the next buffer between average duration and any given time in the future after that,
users more than enough time to trigger, extended exposure times, etc.