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webrtc_m130/webrtc
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sakal c117e2ed79 Do not classify error after stopping the camera as a startup failure. 
Calling event handlers when the camera is closing is safe because
CameraCapturer checks if the errors are coming from the current session.

Calling onFailure after camera has already started might lead to strange
behavior.

BUG=b/34112992

Review-Url: https://codereview.webrtc.org/2634973002
Cr-Commit-Position: refs/heads/master@{#16101}
2017-01-16 17:40:36 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
RTCTransportStats.dtlsState replaces .activeConnection
2017-01-16 15:38:02 +00:00
audio
Added a new echo likelihood stat that reports the maximum value from a previous time period.
2017-01-15 16:29:46 +00:00
base
Delete unused file faketaskrunner.h.
2017-01-16 08:30:33 +00:00
build
Replace default locale with US locale on Android.
2017-01-16 12:57:32 +00:00
call
Added a new echo likelihood stat that reports the maximum value from a previous time period.
2017-01-15 16:29:46 +00:00
common_audio
Replace RTC_DCHECK(false) with RTC_NOTREACHED().
2017-01-12 10:24:27 +00:00
common_video
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
examples
Espresso test case to control loopback call
2017-01-16 10:23:09 +00:00
logging
Revert of Log audio network adapter decisions in event log. (patchset #14 id:320001 of https://codereview.webrtc.org/2559953002/ )
2017-01-13 14:52:12 +00:00
media
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
modules
Add WebRTC.BWE.MidCallProbing.* metrics.
2017-01-16 17:31:49 +00:00
p2p
RTCTransportStats.dtlsState replaces .activeConnection
2017-01-16 15:38:02 +00:00
pc
Provide better message for when RTCP mux "require" policy is triggered.
2017-01-13 19:34:39 +00:00
sdk
Do not classify error after stopping the camera as a startup failure.
2017-01-16 17:40:36 +00:00
stats
RTCTransportStats.dtlsState replaces .activeConnection
2017-01-16 15:38:02 +00:00
system_wrappers
Reland of Delete unused code from systeminfo. (patchset #1 id:1 of https://codereview.webrtc.org/2584563004/ )
2017-01-09 09:22:16 +00:00
test
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
tools
Revert of Log audio network adapter decisions in event log. (patchset #14 id:320001 of https://codereview.webrtc.org/2559953002/ )
2017-01-13 14:52:12 +00:00
video
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
voice_engine
Revert of Log audio network adapter decisions in event log. (patchset #14 id:320001 of https://codereview.webrtc.org/2559953002/ )
2017-01-13 14:52:12 +00:00
.gitignore
…
BUILD.gn
Rename base/analytics/ to base/numerics/
2017-01-13 17:14:33 +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
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
Remove extra uses of basictypes.h.
2016-12-09 14:54:08 +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
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +00:00
video_send_stream.h
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +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.