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Stefan Holmer f9b6e5e4b5 Fix KeepsHighBitrateWhenReconfiguringSender to avoid flakiness if probing succeeds in between encoder reconfigurations. 
BUG=webrtc:7111
R=brandtr@webrtc.org

Review-Url: https://codereview.webrtc.org/2676973004 .
Cr-Commit-Position: refs/heads/master@{#16454}
2017-02-06 16:17:57 +00:00
..
api
Adding C++ versions of currently spec'd "RtpParameters" structs.
2017-02-04 20:09:01 +00:00
audio
Reland of Always call RemoteBitrateEstimator::IncomingPacket from Call. (patchset #1 id:1 of https://codereview.webrtc.org/2668973003/ )
2017-02-06 10:23:00 +00:00
base
Pick the DTLS handshake timeout based on the ICE RTT estimate
2017-02-03 01:18:37 +00:00
call
Fix KeepsHighBitrateWhenReconfiguringSender to avoid flakiness if probing succeeds in between encoder reconfigurations.
2017-02-06 16:17:57 +00:00
common_audio
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ )
2017-01-24 14:58:22 +00:00
common_video
Move frame_generator_capture.{cc, h} and video_capturer.h to video_test_common.
2017-02-06 10:21:11 +00:00
examples
Remove NewApi lint suppression.
2017-02-06 13:34:26 +00:00
logging
Reland of Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters. (patchset #1 id:1 of https://codereview.webrtc.org/2649323010/ )
2017-01-27 12:53:07 +00:00
media
Adding C++ versions of currently spec'd "RtpParameters" structs.
2017-02-04 20:09:01 +00:00
modules
Enable send-side BWE by default for video in call tests.
2017-02-06 14:29:38 +00:00
p2p
Fix Chromium FYI bot
2017-02-03 02:22:46 +00:00
pc
Adding C++ versions of currently spec'd "RtpParameters" structs.
2017-02-04 20:09:01 +00:00
sdk
Remove NewApi lint suppression.
2017-02-06 13:34:26 +00:00
stats
RTCRTPStreamStats.mediaTrackId renamed to trackId.
2017-01-27 14:35:16 +00:00
system_wrappers
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ )
2017-01-24 14:58:22 +00:00
test
Enable send-side BWE by default for video in call tests.
2017-02-06 14:29:38 +00:00
tools
Remove all occurrences of "using std::string".
2017-02-02 16:10:00 +00:00
video
Reland of Improve and re-enable FEC end-to-end tests. (patchset #1 id:1 of https://codereview.webrtc.org/2672373002/ )
2017-02-06 14:35:47 +00:00
voice_engine
Move frame_generator_capture.{cc, h} and video_capturer.h to video_test_common.
2017-02-06 10:21:11 +00:00
.gitignore
…
BUILD.gn
Move frame_generator_capture.{cc, h} and video_capturer.h to video_test_common.
2017-02-06 10:21:11 +00:00
call.h
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
codereview.settings
…
common_types.cc
Reject XR TargetBitrate items with unsupported layer indices
2016-12-06 14:09:00 +00:00
common_types.h
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
config.cc
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
config.h
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
DEPS
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
no_op_function.cc
Moving no_op_function.cc out of webrtc/build
2017-01-24 13:25:12 +00:00
OWNERS
Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it
2016-11-29 10:52:22 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
typedefs.h
Only define NO_RETURN if undefined
2017-01-30 08:54:19 +00:00
video_decoder.h
Move VideoDecoder::Create() logic to separate internal video decoder factory
2016-12-01 08:27:35 +00:00
video_encoder.h
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +00:00
video_frame.h
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
video_receive_stream.h
Remove temporary AddRtxInfo member function.
2017-02-06 10:34:25 +00:00
video_send_stream.h
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
webrtc.gni
Adding build switch for Opus that supports 120ms ptime.
2017-02-02 01:31:11 +00:00
whitespace.txt
Moving whitespace file up by one folder
2017-01-24 13:19:42 +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.