Roll chromium_revision 91f1781..d8c9041

Relevant changes:
* buildtools: c27f95b..6ea835d
* third_party/icu: d8b2a9d..6242e2f
* tools/gyp: 487c0b6..b13d8f2
* tools/swarming_client: 1f8ba35..5b827c9
Details: https://chromium.googlesource.com/chromium/src/+/91f1781..d8c9041/DEPS

Clang version was not updated in this roll, although the
-Wunused-local-typedef warning was enabled by default.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26189004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7716 4adac7df-926f-26a2-2b94-8c16560cd09d
AppRTCDemoActivity: Add a config CheckBox for enabling/disabling CPU overuse adaptation. (re-land)

This CL was incorrectly reverted in r7647 by the libjingle sync bot.

TBR=kjellander@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/32489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7717 4adac7df-926f-26a2-2b94-8c16560cd09d
Add DCHECK to ensure that NetEq's packet buffer is not empty

This DCHECK ensures that one packet was inserted after the buffer was
flushed.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7719 4adac7df-926f-26a2-2b94-8c16560cd09d
Revert "Add DCHECK to ensure that NetEq's packet buffer is not empty"

This reverts r7719. It failed to compile in Chromium.

TBR=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7720 4adac7df-926f-26a2-2b94-8c16560cd09d
cricket::VideoAdapter: Drop frames before spending time converting/scaling, not after.

In VideoCapturer::OnFrameCaptured, we currently convert cricket::CapturedFrame to cricket::VideoFrame and then send that to VideoAdapter::AdaptFrame. AdaptFrame may then decide to drop the frame. It would be faster to drop the frame before converting to cricket::VideoFrame.

This CL refactors VideoAdapter with a new function AdaptFrameResolution that takes captured resolution as input and output adapted resolution, or 0x0 if the frame should be dropped. Using that function, frames can be dropped before any conversion takes place.

R=fbarchard@google.com, perkj@webrtc.org, tommi@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=7702

Committed: https://code.google.com/p/webrtc/source/detail?r=7707

Review URL: https://webrtc-codereview.appspot.com/29949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7721 4adac7df-926f-26a2-2b94-8c16560cd09d
Change default values for CpuOveruseOptions.
Enabled method based on encode time and modified values for the low (60->55) and high threshold (90->85).

Moved DelayedEncoder to fake_encoder.h and added configuration for the delay.

R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/30969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7722 4adac7df-926f-26a2-2b94-8c16560cd09d
Create a NetEq test for when the first incoming payload type is unknown

This test is currently disabled. It triggers an issue where the NetEq
will trigger asserts on subsequent GetAudio calls if the first inserted
packet is rejected, for instance since the payload type is unknown to
NetEq.

BUG=4021
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7723 4adac7df-926f-26a2-2b94-8c16560cd09d
Fix problems if first packet into NetEq is rejected

This CL fixes the problem described in issue 4021. In summary, of the
very first packet coming in to NetEq gets rejected because the RTP
payload type is unknown, subsequent GetAudio calls would trigger asserts
(in debug builds). The problem was that the first_packet_ variable was
reset and new_codec_ was set, even though the packet was discarded
further down the line. Now, these variables are modified after the
packet has been verified.

BUG=4021
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7724 4adac7df-926f-26a2-2b94-8c16560cd09d
Support loopback mode and command line execution
for Android AppRTCDemo when using WebSocket signaling.

- Add loopback support for new signaling. In loopback mode
only room connection is established, WebSocket connection is
not opened and all candidate/sdp messages are automatically
routed back.
- Fix command line support both for channek and new signaling.
Exit from application when room connection is closed and add
an option to run application for certain time period and exit.
- Plus some fixes for WebSocket signaling - support
POST (not used for now) and DELETE request to WebSocket server
and making sure that all available TURN server are used by
peer connection client.

BUG=3995,3937
R=jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7725 4adac7df-926f-26a2-2b94-8c16560cd09d
Use size_t more consistently for packet/payload lengths.

See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information.

This CL was reviewed and approved in pieces in the following CLs:
https://webrtc-codereview.appspot.com/24209004/
https://webrtc-codereview.appspot.com/24229004/
https://webrtc-codereview.appspot.com/24259004/
https://webrtc-codereview.appspot.com/25109004/
https://webrtc-codereview.appspot.com/26099004/
https://webrtc-codereview.appspot.com/27069004/
https://webrtc-codereview.appspot.com/27969004/
https://webrtc-codereview.appspot.com/27989004/
https://webrtc-codereview.appspot.com/29009004/
https://webrtc-codereview.appspot.com/30929004/
https://webrtc-codereview.appspot.com/30939004/
https://webrtc-codereview.appspot.com/31999004/
Committing as TBR to the original reviewers.

BUG=chromium:81439
TEST=none
TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom

Review URL: https://webrtc-codereview.appspot.com/23129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
Update mock_frame_dropper.h to use size_t

This mock was missed in the work of
https://webrtc-codereview.appspot.com/23129004 since the file
is not currently used by any test in this repo.

BUG=chromium:81439
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7727 4adac7df-926f-26a2-2b94-8c16560cd09d
cricket::VideoFrame: Refactor ConvertToRgbBuffer into base class

There is also an implementation in Chromium that can be removed if/when this lands:
content/renderer/media/webrtc/webrtc_video_capturer_adapter.cc

R=fbarchard@google.com, pbos@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7728 4adac7df-926f-26a2-2b94-8c16560cd09d
Build fix for MIPS Android Webview build.

Excluding optimizations to support MIPS32R6 platform for Android Webview build (see also https://code.google.com/p/webrtc/source/detail?r=7580).

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7729 4adac7df-926f-26a2-2b94-8c16560cd09d
Remove expensive and unnecessary memory alloc for sending black frames on video
mute.

Remove old crusty is_black_ member var in webrtcvideoengine which was not adding value.

R=henrike@webrtc.org, tpsiaki@google.com

Review URL: https://webrtc-codereview.appspot.com/26229004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7731 4adac7df-926f-26a2-2b94-8c16560cd09d
Merge remote-tracking branch 'upstream/master'

Revision 7731.

No changes at all.

Conflicts:
	DEPS
	talk/app/webrtc/java/jni/peerconnection_jni.cc
	talk/examples/android/res/values/arrays.xml
	talk/examples/android/res/values/strings.xml
	talk/examples/android/res/xml/preferences.xml
	talk/examples/android/src/org/appspot/apprtc/AppRTCClient.java
	talk/examples/android/src/org/appspot/apprtc/AppRTCDemoActivity.java
	talk/examples/android/src/org/appspot/apprtc/ConnectActivity.java
	talk/examples/android/src/org/appspot/apprtc/GAEChannelClient.java
	talk/examples/android/src/org/appspot/apprtc/GAERTCClient.java
	talk/examples/android/src/org/appspot/apprtc/PeerConnectionClient.java
	talk/examples/android/src/org/appspot/apprtc/SettingsActivity.java
	talk/libjingle.gyp
	talk/libjingle_examples.gyp
	talk/media/base/videoadapter.cc
	talk/media/base/videoadapter.h
	talk/media/base/videoadapter_unittest.cc
	talk/media/base/videocapturer.cc
	talk/media/base/videoframe.cc
	talk/media/base/videoframe.h
	talk/media/base/videoframefactory.cc
	talk/media/base/videoframefactory.h
	talk/media/webrtc/fakewebrtcvideoengine.h
	talk/media/webrtc/fakewebrtcvoiceengine.h
	talk/media/webrtc/webrtcvideocapturer.cc
	talk/media/webrtc/webrtcvideocapturer.h
	talk/media/webrtc/webrtcvideoengine.cc
	talk/media/webrtc/webrtcvideoengine.h
	talk/media/webrtc/webrtcvideoengine2.cc
	talk/media/webrtc/webrtcvideoframe.cc
	talk/media/webrtc/webrtcvideoframe.h
	talk/media/webrtc/webrtcvoiceengine.cc
	talk/media/webrtc/webrtcvoiceengine.h
	webrtc/base/BUILD.gn
	webrtc/base/base.gyp
	webrtc/build/common.gypi
	webrtc/common_audio/common_audio.gyp
	webrtc/common_types.h
	webrtc/common_video/libyuv/include/webrtc_libyuv.h
	webrtc/common_video/libyuv/libyuv_unittest.cc
	webrtc/common_video/libyuv/scaler_unittest.cc
	webrtc/common_video/libyuv/webrtc_libyuv.cc
	webrtc/modules/audio_coding/codecs/cng/include/webrtc_cng.h
	webrtc/modules/audio_coding/codecs/cng/webrtc_cng.c
	webrtc/modules/audio_coding/codecs/isac/fix/source/isacfix.gypi
	webrtc/modules/audio_coding/main/acm2/acm_receive_test.cc
	webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
	webrtc/modules/audio_coding/main/acm2/acm_receiver.h
	webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc
	webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc
	webrtc/modules/audio_coding/main/acm2/acm_send_test.cc
	webrtc/modules/audio_coding/main/acm2/acm_send_test.h
	webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc
	webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h
	webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
	webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
	webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc
	webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
	webrtc/modules/audio_coding/main/interface/audio_coding_module.h
	webrtc/modules/audio_coding/main/test/Channel.cc
	webrtc/modules/audio_coding/main/test/Channel.h
	webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
	webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h
	webrtc/modules/audio_coding/main/test/RTPFile.cc
	webrtc/modules/audio_coding/main/test/RTPFile.h
	webrtc/modules/audio_coding/main/test/TestAllCodecs.cc
	webrtc/modules/audio_coding/main/test/TestAllCodecs.h
	webrtc/modules/audio_coding/main/test/TestStereo.cc
	webrtc/modules/audio_coding/main/test/TestStereo.h
	webrtc/modules/audio_coding/main/test/dual_stream_unittest.cc
	webrtc/modules/audio_coding/main/test/target_delay_unittest.cc
	webrtc/modules/audio_coding/neteq/dtmf_buffer.cc
	webrtc/modules/audio_coding/neteq/dtmf_buffer.h
	webrtc/modules/audio_coding/neteq/interface/neteq.h
	webrtc/modules/audio_coding/neteq/mock/mock_payload_splitter.h
	webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
	webrtc/modules/audio_coding/neteq/neteq_impl.cc
	webrtc/modules/audio_coding/neteq/neteq_impl.h
	webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
	webrtc/modules/audio_coding/neteq/neteq_unittest.cc
	webrtc/modules/audio_coding/neteq/packet.h
	webrtc/modules/audio_coding/neteq/payload_splitter.cc
	webrtc/modules/audio_coding/neteq/payload_splitter.h
	webrtc/modules/audio_coding/neteq/payload_splitter_unittest.cc
	webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.cc
	webrtc/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h
	webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
	webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h
	webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
	webrtc/modules/audio_processing/audio_processing.gypi
	webrtc/modules/interface/module_common_types.h
	webrtc/modules/media_file/interface/media_file.h
	webrtc/modules/media_file/source/avi_file.cc
	webrtc/modules/media_file/source/avi_file.h
	webrtc/modules/media_file/source/media_file_impl.cc
	webrtc/modules/media_file/source/media_file_impl.h
	webrtc/modules/media_file/source/media_file_unittest.cc
	webrtc/modules/media_file/source/media_file_utility.cc
	webrtc/modules/media_file/source/media_file_utility.h
	webrtc/modules/pacing/include/mock/mock_paced_sender.h
	webrtc/modules/pacing/include/paced_sender.h
	webrtc/modules/pacing/paced_sender.cc
	webrtc/modules/pacing/paced_sender_unittest.cc
	webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
	webrtc/modules/remote_bitrate_estimator/overuse_detector.cc
	webrtc/modules/remote_bitrate_estimator/overuse_detector.h
	webrtc/modules/remote_bitrate_estimator/rate_statistics.cc
	webrtc/modules/remote_bitrate_estimator/rate_statistics.h
	webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc
	webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc
	webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h
	webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.cc
	webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h
	webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp_play.cc
	webrtc/modules/rtp_rtcp/interface/fec_receiver.h
	webrtc/modules/rtp_rtcp/interface/receive_statistics.h
	webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h
	webrtc/modules/rtp_rtcp/interface/rtp_receiver.h
	webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h
	webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
	webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
	webrtc/modules/rtp_rtcp/source/bitrate.cc
	webrtc/modules/rtp_rtcp/source/bitrate.h
	webrtc/modules/rtp_rtcp/source/fec_receiver_impl.cc
	webrtc/modules/rtp_rtcp/source/fec_receiver_impl.h
	webrtc/modules/rtp_rtcp/source/fec_test_helper.cc
	webrtc/modules/rtp_rtcp/source/fec_test_helper.h
	webrtc/modules/rtp_rtcp/source/forward_error_correction.cc
	webrtc/modules/rtp_rtcp/source/forward_error_correction.h
	webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
	webrtc/modules/rtp_rtcp/source/producer_fec.cc
	webrtc/modules/rtp_rtcp/source/producer_fec.h
	webrtc/modules/rtp_rtcp/source/producer_fec_unittest.cc
	webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
	webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
	webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
	webrtc/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
	webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
	webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
	webrtc/modules/rtp_rtcp/source/rtcp_sender.h
	webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
	webrtc/modules/rtp_rtcp/source/rtp_fec_unittest.cc
	webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc
	webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h
	webrtc/modules/rtp_rtcp/source/rtp_format_vp8.cc
	webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h
	webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.cc
	webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h
	webrtc/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc
	webrtc/modules/rtp_rtcp/source/rtp_header_extension.cc
	webrtc/modules/rtp_rtcp/source/rtp_header_extension.h
	webrtc/modules/rtp_rtcp/source/rtp_header_extension_unittest.cc
	webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc
	webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
	webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
	webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
	webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
	webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
	webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
	webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
	webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
	webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
	webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h
	webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
	webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
	webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
	webrtc/modules/rtp_rtcp/source/rtp_sender.cc
	webrtc/modules/rtp_rtcp/source/rtp_sender.h
	webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
	webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
	webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
	webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
	webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
	webrtc/modules/rtp_rtcp/source/rtp_utility.cc
	webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.cc
	webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator.h
	webrtc/modules/rtp_rtcp/source/vp8_partition_aggregator_unittest.cc
	webrtc/modules/rtp_rtcp/test/BWEStandAlone/BWEStandAlone.cc
	webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.cc
	webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h
	webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.cc
	webrtc/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h
	webrtc/modules/rtp_rtcp/test/testAPI/test_api.h
	webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
	webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc
	webrtc/modules/rtp_rtcp/test/testFec/test_fec.cc
	webrtc/modules/utility/interface/rtp_dump.h
	webrtc/modules/utility/source/coder.cc
	webrtc/modules/utility/source/coder.h
	webrtc/modules/utility/source/file_player_impl.cc
	webrtc/modules/utility/source/file_recorder_impl.cc
	webrtc/modules/utility/source/file_recorder_impl.h
	webrtc/modules/utility/source/rtp_dump_impl.cc
	webrtc/modules/utility/source/rtp_dump_impl.h
	webrtc/modules/utility/source/video_coder.cc
	webrtc/modules/utility/source/video_coder.h
	webrtc/modules/video_capture/android/video_capture_android.cc
	webrtc/modules/video_capture/android/video_capture_android.h
	webrtc/modules/video_capture/include/video_capture_defines.h
	webrtc/modules/video_capture/ios/rtc_video_capture_ios_objc.mm
	webrtc/modules/video_capture/mac/qtkit/video_capture_qtkit_objc.mm
	webrtc/modules/video_capture/test/video_capture_unittest.cc
	webrtc/modules/video_capture/video_capture_impl.cc
	webrtc/modules/video_capture/video_capture_impl.h
	webrtc/modules/video_capture/windows/sink_filter_ds.cc
	webrtc/modules/video_capture/windows/sink_filter_ds.h
	webrtc/modules/video_coding/codecs/i420/main/interface/i420.h
	webrtc/modules/video_coding/codecs/i420/main/source/i420.cc
	webrtc/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h
	webrtc/modules/video_coding/codecs/test/packet_manipulator.cc
	webrtc/modules/video_coding/codecs/test/packet_manipulator.h
	webrtc/modules/video_coding/codecs/test/packet_manipulator_unittest.cc
	webrtc/modules/video_coding/codecs/test/stats.cc
	webrtc/modules/video_coding/codecs/test/stats.h
	webrtc/modules/video_coding/codecs/test/videoprocessor.cc
	webrtc/modules/video_coding/codecs/test/videoprocessor.h
	webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc
	webrtc/modules/video_coding/codecs/test_framework/normal_async_test.cc
	webrtc/modules/video_coding/codecs/test_framework/normal_async_test.h
	webrtc/modules/video_coding/codecs/test_framework/packet_loss_test.cc
	webrtc/modules/video_coding/codecs/test_framework/packet_loss_test.h
	webrtc/modules/video_coding/codecs/test_framework/test.h
	webrtc/modules/video_coding/codecs/test_framework/unit_test.cc
	webrtc/modules/video_coding/codecs/test_framework/unit_test.h
	webrtc/modules/video_coding/codecs/test_framework/video_source.cc
	webrtc/modules/video_coding/codecs/test_framework/video_source.h
	webrtc/modules/video_coding/codecs/tools/video_quality_measurement.cc
	webrtc/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
	webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
	webrtc/modules/video_coding/codecs/vp8/vp8_impl.h
	webrtc/modules/video_coding/codecs/vp8/vp8_sequence_coder.cc
	webrtc/modules/video_coding/codecs/vp9/vp9_impl.cc
	webrtc/modules/video_coding/codecs/vp9/vp9_impl.h
	webrtc/modules/video_coding/main/interface/video_coding.h
	webrtc/modules/video_coding/main/interface/video_coding_defines.h
	webrtc/modules/video_coding/main/source/codec_database.cc
	webrtc/modules/video_coding/main/source/codec_database.h
	webrtc/modules/video_coding/main/source/encoded_frame.h
	webrtc/modules/video_coding/main/source/frame_buffer.cc
	webrtc/modules/video_coding/main/source/generic_encoder.cc
	webrtc/modules/video_coding/main/source/generic_encoder.h
	webrtc/modules/video_coding/main/source/media_optimization.cc
	webrtc/modules/video_coding/main/source/media_optimization.h
	webrtc/modules/video_coding/main/source/media_optimization_unittest.cc
	webrtc/modules/video_coding/main/source/packet.cc
	webrtc/modules/video_coding/main/source/packet.h
	webrtc/modules/video_coding/main/source/qm_select.cc
	webrtc/modules/video_coding/main/source/qm_select.h
	webrtc/modules/video_coding/main/source/qm_select_unittest.cc
	webrtc/modules/video_coding/main/source/session_info.cc
	webrtc/modules/video_coding/main/source/session_info.h
	webrtc/modules/video_coding/main/source/session_info_unittest.cc
	webrtc/modules/video_coding/main/source/video_coding_impl.cc
	webrtc/modules/video_coding/main/source/video_coding_impl.h
	webrtc/modules/video_coding/main/source/video_receiver.cc
	webrtc/modules/video_coding/main/source/video_receiver_unittest.cc
	webrtc/modules/video_coding/main/source/video_sender_unittest.cc
	webrtc/modules/video_coding/main/test/generic_codec_test.cc
	webrtc/modules/video_coding/main/test/generic_codec_test.h
	webrtc/modules/video_coding/main/test/media_opt_test.cc
	webrtc/modules/video_coding/main/test/media_opt_test.h
	webrtc/modules/video_coding/main/test/mt_test_common.cc
	webrtc/modules/video_coding/main/test/mt_test_common.h
	webrtc/modules/video_coding/main/test/normal_test.cc
	webrtc/modules/video_coding/main/test/normal_test.h
	webrtc/modules/video_coding/main/test/quality_modes_test.cc
	webrtc/modules/video_coding/main/test/quality_modes_test.h
	webrtc/modules/video_coding/main/test/receiver_tests.h
	webrtc/modules/video_coding/main/test/rtp_player.cc
	webrtc/modules/video_coding/main/test/test_callbacks.cc
	webrtc/modules/video_coding/main/test/test_callbacks.h
	webrtc/modules/video_coding/main/test/test_util.h
	webrtc/modules/video_coding/main/test/vcm_payload_sink_factory.cc
	webrtc/modules/video_coding/main/test/video_source.cc
	webrtc/modules/video_coding/main/test/video_source.h
	webrtc/modules/video_coding/utility/frame_dropper.cc
	webrtc/modules/video_coding/utility/include/frame_dropper.h
	webrtc/modules/video_coding/utility/include/mock/mock_frame_dropper.h
	webrtc/modules/video_processing/main/test/unit_test/video_processing_unittest.h
	webrtc/modules/video_render/mac/video_render_agl.cc
	webrtc/modules/video_render/mac/video_render_agl.h
	webrtc/modules/video_render/mac/video_render_nsopengl.h
	webrtc/modules/video_render/mac/video_render_nsopengl.mm
	webrtc/system_wrappers/interface/file_wrapper.h
	webrtc/system_wrappers/source/file_impl.cc
	webrtc/system_wrappers/source/file_impl.h
	webrtc/test/channel_transport/channel_transport.cc
	webrtc/test/channel_transport/include/channel_transport.h
	webrtc/test/channel_transport/udp_socket2_manager_win.h
	webrtc/test/channel_transport/udp_socket2_win.cc
	webrtc/test/channel_transport/udp_socket2_win.h
	webrtc/test/channel_transport/udp_socket_posix.cc
	webrtc/test/channel_transport/udp_socket_posix.h
	webrtc/test/channel_transport/udp_socket_wrapper.h
	webrtc/test/channel_transport/udp_transport.h
	webrtc/test/channel_transport/udp_transport_impl.cc
	webrtc/test/channel_transport/udp_transport_impl.h
	webrtc/test/channel_transport/udp_transport_unittest.cc
	webrtc/test/configurable_frame_size_encoder.cc
	webrtc/test/configurable_frame_size_encoder.h
	webrtc/test/fake_encoder.cc
	webrtc/test/fake_encoder.h
	webrtc/test/mock_transport.h
	webrtc/test/rtcp_packet_parser.cc
	webrtc/test/rtcp_packet_parser.h
	webrtc/test/testsupport/packet_reader.cc
	webrtc/test/testsupport/packet_reader.h
	webrtc/test/testsupport/packet_reader_unittest.cc
	webrtc/test/testsupport/unittest_utils.h
	webrtc/tools/frame_editing/frame_editing_lib.cc
	webrtc/tools/frame_editing/frame_editing_unittest.cc
	webrtc/video/call_perf_tests.cc
	webrtc/video/end_to_end_tests.cc
	webrtc/video/rampup_tests.cc
	webrtc/video/send_statistics_proxy_unittest.cc
	webrtc/video/transport_adapter.cc
	webrtc/video/transport_adapter.h
	webrtc/video/video_receive_stream.cc
	webrtc/video/video_send_stream.cc
	webrtc/video/video_send_stream_tests.cc
	webrtc/video_encoder.h
	webrtc/video_engine/include/vie_base.h
	webrtc/video_engine/include/vie_capture.h
	webrtc/video_engine/include/vie_image_process.h
	webrtc/video_engine/include/vie_network.h
	webrtc/video_engine/include/vie_render.h
	webrtc/video_engine/include/vie_rtp_rtcp.h
	webrtc/video_engine/overuse_frame_detector_unittest.cc
	webrtc/video_engine/test/auto_test/automated/vie_network_test.cc
	webrtc/video_engine/test/auto_test/primitives/framedrop_primitives.cc
	webrtc/video_engine/test/auto_test/primitives/framedrop_primitives.h
	webrtc/video_engine/test/auto_test/source/vie_autotest_capture.cc
	webrtc/video_engine/test/auto_test/source/vie_autotest_codec.cc
	webrtc/video_engine/test/auto_test/source/vie_autotest_image_process.cc
	webrtc/video_engine/test/auto_test/source/vie_autotest_render.cc
	webrtc/video_engine/test/libvietest/helpers/vie_file_capture_device.cc
	webrtc/video_engine/test/libvietest/helpers/vie_to_file_renderer.cc
	webrtc/video_engine/test/libvietest/include/tb_I420_codec.h
	webrtc/video_engine/test/libvietest/include/tb_external_transport.h
	webrtc/video_engine/test/libvietest/include/vie_external_render_filter.h
	webrtc/video_engine/test/libvietest/include/vie_to_file_renderer.h
	webrtc/video_engine/test/libvietest/testbed/tb_I420_codec.cc
	webrtc/video_engine/test/libvietest/testbed/tb_external_transport.cc
	webrtc/video_engine/vie_capturer.cc
	webrtc/video_engine/vie_capturer.h
	webrtc/video_engine/vie_channel.cc
	webrtc/video_engine/vie_channel.h
	webrtc/video_engine/vie_channel_group.cc
	webrtc/video_engine/vie_encoder.cc
	webrtc/video_engine/vie_encoder.h
	webrtc/video_engine/vie_network_impl.cc
	webrtc/video_engine/vie_network_impl.h
	webrtc/video_engine/vie_receiver.cc
	webrtc/video_engine/vie_receiver.h
	webrtc/video_engine/vie_renderer.cc
	webrtc/video_engine/vie_sender.cc
	webrtc/video_engine/vie_sender.h
	webrtc/video_frame.h
	webrtc/voice_engine/channel.cc
	webrtc/voice_engine/channel.h
	webrtc/voice_engine/include/voe_network.h
	webrtc/voice_engine/include/voe_rtp_rtcp.h
	webrtc/voice_engine/test/android/android_test/jni/android_test.cc
	webrtc/voice_engine/test/auto_test/fakes/fake_external_transport.cc
	webrtc/voice_engine/test/auto_test/fakes/fake_external_transport.h
	webrtc/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h
	webrtc/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc
	webrtc/voice_engine/test/win_test/WinTestDlg.cc
	webrtc/voice_engine/voe_network_impl.cc
	webrtc/voice_engine/voe_network_impl.h