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sakal 8444405aa4 Try re-enabling two PeerConnectionClientTests: 
- org.appspot.apprtc.test.PeerConnectionClientTest#testCaptureFormatChange
- org.appspot.apprtc.test.PeerConnectionClientTest#testVideoSourceRestart

The tests were previously flaky but significant changes to the test
architecture have been made since. Opportunistically try to re-enable
them.

BUG=webrtc:6478

Review-Url: https://codereview.webrtc.org/2748863003
Cr-Commit-Position: refs/heads/master@{#17245}
2017-03-15 12:50:13 +00:00
..
api
Change the type of session_id() from string to int64_t.
2017-03-13 18:00:54 +00:00
audio
Move delay_based_bwe_ into CongestionController
2017-03-09 14:40:08 +00:00
base
TaskQueue[Win] DOS handling
2017-03-15 11:36:29 +00:00
call
Ignore packets sent on old network route when receiving feedback.
2017-03-15 11:40:25 +00:00
common_audio
iSAC fix entropy coder: Recently added DCHECK could in fact trigger
2017-03-13 12:28:47 +00:00
common_video
Add FullStack test for simulcast screenshare mode.
2017-03-09 17:23:30 +00:00
examples
Try re-enabling two PeerConnectionClientTests:
2017-03-15 12:50:13 +00:00
logging
Enable GN check for webrtc/logging
2017-03-08 10:12:11 +00:00
media
Revert to allowing only 1 unsignaled receive stream for audio.
2017-03-13 12:46:15 +00:00
modules
Fix broken test landed in r17243.
2017-03-15 12:07:24 +00:00
ortc
Add the function ToRtpParameters.
2017-03-09 01:15:06 +00:00
p2p
Making candidate pool size behave as decided in JSEP.
2017-03-10 23:18:00 +00:00
pc
Making candidate pool size behave as decided in JSEP.
2017-03-10 23:18:00 +00:00
sdk
Trigger framelisteners even on frames dropped by the FPS reduction by default.
2017-03-13 12:11:48 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
Delete utf_util_win.h.
2017-03-14 08:32:50 +00:00
test
Add the ability to read/write to WAV files in FakeAudioDevice
2017-03-14 16:01:47 +00:00
tools
Pass label and plot style to the TimeSeries' constructor.
2017-03-15 08:59:12 +00:00
video
Fix crash in vie_encoder when encoder configure fails
2017-03-15 11:24:21 +00:00
voice_engine
Remove VoENetEqStats interface.
2017-03-13 09:36:19 +00:00
.gitignore
…
BUILD.gn
Remove voe_cmd_test.
2017-03-07 09:43:21 +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
Speculative fix for division by zero in Vp8EncoderImpl.
2017-03-10 14:08:26 +00:00
config.cc
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +00:00
config.h
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +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
Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )
2017-02-09 12:53:45 +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
Add |protected_by_flexfec| flag to VideoReceiveStream::Config.
2017-03-09 14:13:20 +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
Reland of Setting is_component_build to false by default (patchset #1 id:1 of https://codereview.webrtc.org/2731703004/ )
2017-03-04 03:41:59 +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.