Currently there is a hard limit for the estimated captured frame
interval of 45ms. As the encoder utilization is calculated as
(input frame interval)/(encode time), overuse signals can be triggered
even though there is plenty of time to go around if the fps is low.
However, in order to avoid falsly estimating low encode usage in case
the capturer has a dynamic frame rate, set the frame interval based on
the actual current max framerate.
BUG=webrtc:4172
Review-Url: https://codereview.webrtc.org/2918143003
Cr-Commit-Position: refs/heads/master@{#18610}
The only implementation which used a nullptr was a mock used in tests,
so add a dummy instance there instead.
Remove tests for stats_proxy_ in vie_encoder and just dcheck in the
constructor instead.
BUG=None
Review-Url: https://codereview.webrtc.org/2695643002
Cr-Commit-Position: refs/heads/master@{#16577}
This removes the VideoSendStream::LoadObserver interface and the implementation in WebrtcVideoSendStream and replace it with VideoSinkWants through the VideoSourceInterface.
To do that that, some stats for CPU adaptation is moved into VideoSendStream. Also handling of the CVO rtp header extension is moved to VideoSendStreamImpl.
BUG=webrtc:5687
TBR=mflodman@webrtc.org
Review-Url: https://codereview.webrtc.org/2304363002
Cr-Commit-Position: refs/heads/master@{#14877}
The renaming is to reflect this class is only used for RTCP interaction
and not for other transports.
This Cl will be followed by multiple CLs moving all send-side RTP
functionality to a separate class, rtp module ownership away from
VideoSendStream and use TaskQueue instead of ProcessThread for RTP.
BUG=webrtc:6456
Review-Url: https://codereview.webrtc.org/2390463002
Cr-Commit-Position: refs/heads/master@{#14556}
