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perkj fa10b557d9 Releand of Let ViEEncoder handle resolution changes. 
The original landed cl is in patchset 1.
The following patchset fix VideoQualityTest as well as fix the case where max_bitrate is set in the SendParams. A unit test is added for that as well.

Original cl description:
Let ViEEncoder handle resolution changes.

This cl move codec reconfiguration due to video frame size changes from WebRtcVideoSendStream to ViEEncoder.

With this change, many variables in WebRtcVideoSendStream no longer need to be locked.

BUG=webrtc:5687, webrtc:6371, webrtc:5332

Review-Url: https://codereview.webrtc.org/2386573002
Cr-Commit-Position: refs/heads/master@{#14467}
2016-10-03 06:45:33 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
Releand of Let ViEEncoder handle resolution changes.
2016-10-03 06:45:33 +00:00
audio
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
base
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
build
Roll chromium_revision f86fb54ec3..dd442d4812 (420104:421425)
2016-09-28 07:45:12 +00:00
call
Releand of Let ViEEncoder handle resolution changes.
2016-10-03 06:45:33 +00:00
common_audio
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
common_video
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
examples
Disable TCPChannelClientTest.testConnectIPv6
2016-09-30 07:20:12 +00:00
libjingle
GN: Change rtc_source_set targets --> rtc_static_library
2016-09-23 07:38:58 +00:00
media
Releand of Let ViEEncoder handle resolution changes.
2016-10-03 06:45:33 +00:00
modules
Creating controller manager from config string in audio network adaptor.
2016-10-02 21:53:45 +00:00
p2p
Emit SignalReadyToSend even for "presumed writable" connections.
2016-09-30 22:16:57 +00:00
pc
Update WebRTC to build against libsrtp 2.0
2016-09-28 21:08:53 +00:00
sdk
Remove Crit::Scope lock by using atomic bool property.
2016-09-30 07:21:28 +00:00
stats
GN: Change rtc_source_set targets --> rtc_static_library
2016-09-23 07:38:58 +00:00
system_wrappers
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
test
Releand of Let ViEEncoder handle resolution changes.
2016-10-03 06:45:33 +00:00
tools
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
video
Releand of Let ViEEncoder handle resolution changes.
2016-10-03 06:45:33 +00:00
voice_engine
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
BUILD.gn
Disable TCPChannelClientTest.testConnectIPv6
2016-09-30 07:20:12 +00:00
call.h
This reverts commit 8eb37a39e79fe1098d3503dcb8c8c2d196203fed. Chrome now have its own implementation of TaskQueues that is based on Chrome threads.
2016-09-01 08:17:43 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
Reland of Ignore Camera and Flip bits in CVO when parsing video rotation (patchset #1 id:1 of https://codereview.webrtc.org/2300323002/ )
2016-09-08 10:25:05 +00:00
common_types.h
Enable the -Wundef warning for clang
2016-09-29 00:42:08 +00:00
common.gyp
Moving/renaming webrtc/common.h.
2016-09-07 14:34:45 +00:00
config.cc
Releand of Let ViEEncoder handle resolution changes.
2016-10-03 06:45:33 +00:00
config.h
Releand of Let ViEEncoder handle resolution changes.
2016-10-03 06:45:33 +00:00
DEPS
Enable the -Wundef warning for clang
2016-09-29 00:42:08 +00:00
engine_configurations.h
Reland of move all reference to carbon api (patchset #1 id:1 of https://codereview.webrtc.org/2317343003/ )
2016-09-09 10:19:54 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete talk directory, and references from build_ios_libs.sh.
2016-09-02 10:07:13 +00:00
OWNERS
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
supplement.gypi
Cleanup MIPS specific link configuration
2016-06-20 05:39:25 +00:00
transport.h
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
2015-12-09 11:13:40 +00:00
typedefs.h
[MIPS] Fix build issue for mips64el
2016-06-23 10:52:35 +00:00
video_decoder.h
Replace scoped_ptr with unique_ptr in webrtc/common_video/
2016-03-02 11:41:40 +00:00
video_encoder.h
Add codec name to CodecSpecificInfo and get the codec name stats from there instead.
2016-09-01 07:21:18 +00:00
video_frame.h
Releand of Let ViEEncoder handle resolution changes.
2016-10-03 06:45:33 +00:00
video_receive_stream.h
Add logging statements to places where the frame might be dropped in WebRTC pipeline.
2016-09-30 13:19:12 +00:00
video_send_stream.h
Add VideoSendStream::Stats::prefered_media_bitrate_bps
2016-09-29 09:49:01 +00:00
webrtc_examples.gyp
Reland of name AppRTCDemo on Android and iOS to AppRTCMobile (patchset #1 id:1 of https://codereview.webrtc.org/2358133003/ )
2016-09-27 08:10:52 +00:00
webrtc_tests.gypi
GYP: Remove targets inside include_tests==1 that are converted to GN.
2016-09-15 11:57:39 +00:00
webrtc.gyp
Move webrtc/audio_*.h to webrtc/api/call
2016-08-31 14:33:14 +00:00

README.chromium 

Name: WebRTC
URL: http://www.webrtc.org
Version: 90
License: BSD
License File: LICENSE

Description:
WebRTC provides real time voice and video processing
functionality to enable the implementation of 
PeerConnection/MediaStream.

Third party code used in this project is described 
in the file LICENSE_THIRD_PARTY.