This logic currently prevents loopback calls on Nexus 5X when it's
slightly overloaded to maintain input framerate since encoding at ~25fps
with one framedrop results in >70ms between frames naturally.
With this change applied Nexus 5X can maintain ~25fps both in and out
without building excessive latency (>2 frames) (this is now covered by
CPU adaptation outside the codec wrapper).
BUG=
R=glaznev@webrtc.org
Review URL: https://codereview.webrtc.org/1854413004 .
Cr-Commit-Position: refs/heads/master@{#12317}
Permits going from HD to QVGA in 6 seconds instead of 10. Also adds
windows for going up quickly in the beginning of a call (before any
downscaling happens due to bad quality).
BUG=webrtc:5678
R=glaznev@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1830593003 .
Cr-Commit-Position: refs/heads/master@{#12219}
Boost low QP threashold to 21, otherwise VGA encoding never
scales up even at 2.5 Mbps.
Also reduce high QP threshold to scale down faster.
BUG=b/26504665
R=jackychen@google.com
Review URL: https://codereview.webrtc.org/1717763003 .
Cr-Commit-Position: refs/heads/master@{#11712}
In addition to the code moved from talk/app/webrtc
there were some files in webrtc/api/objctests that still
had the libjingle license header.
BUG=webrtc:5418
TBR=tkchin@webrtc.org
NOTRY=True
Review URL: https://codereview.webrtc.org/1680293005
Cr-Commit-Position: refs/heads/master@{#11552}
The previously disabled warnings that were inherited from
talk/build/common.gypi are now replaced by target-specific disabling
of only the failing warnings. Additional disabling was needed since the stricter
compilation warnings that applies to code in webrtc/.
License headers will be updated in a follow-up CL.
Other modifications:
* Updated the header guards.
* Sorted the includes using chromium/src/tools/sort-headers.py
except for these files:
talk/app/webrtc/peerconnectionendtoend_unittest.cc
talk/app/webrtc/java/jni/androidmediadecoder_jni.cc
talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
webrtc/media/devices/win32devicemanager.cc
The HAVE_SCTP define was added for the peerconnection_unittests target
in api_tests.gyp.
I also checked that none of
SRTP_RELATIVE_PATH
HAVE_SRTP
HAVE_WEBRTC_VIDEO
HAVE_WEBRTC_VOICE
were used by the talk/app/webrtc code.
For Chromium, the following changes will need to be applied to the roll CL that updates the
DEPS for WebRTC and libjingle:
https://codereview.chromium.org/1615433002
BUG=webrtc:5418
NOPRESUBMIT=True
R=deadbeef@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1610243002 .
Cr-Commit-Position: refs/heads/master@{#11545}
