I used a command like this to update the paths:
perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"`
The only manual edit is to add an include of webrtc/rtc_base/checks.h in
webrtc/modules/audio_device/android/opensles_common.h, which likely
was needed due to changed include paths due to 'git cl format'.
BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2969653002
Cr-Commit-Position: refs/heads/master@{#18871}
1. Fix the case of key frame accumulation being incorrect due to the chunk
size being computed at the time of leak based on input frame rate. The issue
is that the count is computed based on key frame ratio and the actual chunk
size computed from current input frame rate. These can be wildly different
especially at the beginning of the stream (key frame ratio defaults based
on 30 fps) resulting in incorrect key frame accumulation causing large frame
drops when the input frame rate is low.
2. Add large delta frame compensation. The current code accounts for key frames
but not large delta frames. This is a common occurence in some application
(remote desktop as an example)
3. Fixes an issue identified by the unit tests. The accumulation of
key frames had an issue in the scenario of a high key frame ratio where
the full key frame was not being accounted for.
3. Removes fast mode and other methods that are mostly dead code.
4. Cleans up variable names as per chromium style.
Review URL: https://codereview.webrtc.org/1750493002
Cr-Commit-Position: refs/heads/master@{#11884}
