RTCVideoEncoder should use webrtc timestamp for |capture_time_ms|.

Previously we were using the time delta between base::Time::Now() at
BitstreamBufferReady() and a "time base", but this does not match
the internal frame of timeticks used by the webrtc RTP sender. As a
result the sender is comparing apples to oranges, and frequently
pauses HW encoder when it incorrectly believes that sender queue is
too large.

The solution is to use the same webrtc timestamp routine to set the
|capture_time_ms|. It is nearly impossible to replicate the exact
timestamp using chromium's base::Time routines.

TODO(hshi): we should ideally obtain the actual timestamp when capture
occurs, and not in BitstreamBufferReady. This may introduce jitter
in the stream.

BUG=284783
TEST=observe that EncoderPaused events are eliminated in traces.
R=fischman@chromium.org, juberti@chromium.org

Review URL: https://codereview.chromium.org/24623002

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@225590 0039d316-1c4b-4281-b951-d872f2087c98
Removed unused feedback_ui.cc

R=xiyuan@chromium.org
BUG=None.

Review URL: https://codereview.chromium.org/24435003

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@225591 0039d316-1c4b-4281-b951-d872f2087c98
SVN changes up to revision 225591