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philipel 46948c17fd Revert of Propagate probing cluster id to SendTimeHistory. (patchset #5 id:80001 of https://codereview.webrtc.org/2005313003/ ) 
Reason for revert:
Breaks google3 buildbot:  http://webrtc-buildbot-master.mtv.corp.google.com:21000/builders/WebRTC%20google3%20Importer/builds/8640

Original issue's description:
> Propagate probing cluster id to SendTimeHistory, both for packets and padding.
>
> BUG=webrtc:5859
>
> Committed: https://crrev.com/5be28c848b91bc6e4800eac07a3f5ac09a32ad70
> Cr-Commit-Position: refs/heads/master@{#12985}

TBR=danilchap@webrtc.org,stefan@webrtc.org,mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5859

Review-Url: https://codereview.webrtc.org/2032463003
Cr-Commit-Position: refs/heads/master@{#12987}
2016-06-01 11:04:49 +00:00
..
androidjunit
Reland of Android: Change camera fps range selection (patchset #1 id:1 of https://codereview.webrtc.org/2021233002/ )
2016-06-01 07:44:07 +00:00
api
peerconnection: remove unused include
2016-06-01 10:42:42 +00:00
audio
Remove use of RtpHeaderExtension and clean up
2016-05-26 18:25:04 +00:00
base
Use scoped_refptr for On(Add|Remove)Stream and OnDataChannel.
2016-05-31 20:02:30 +00:00
build
Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc}
2016-06-01 09:17:56 +00:00
call
Take ownership and close the platform file even if we fail to start logging.
2016-05-27 09:22:54 +00:00
common_audio
Rename APK tests workaround to make it more generic.
2016-05-31 11:01:47 +00:00
common_video
Rename APK tests workaround to make it more generic.
2016-05-31 11:01:47 +00:00
examples
Use scoped_refptr for On(Add|Remove)Stream and OnDataChannel.
2016-05-31 20:02:30 +00:00
libjingle
Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc}
2016-06-01 09:17:56 +00:00
media
Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc}
2016-06-01 09:17:56 +00:00
modules
Revert of Propagate probing cluster id to SendTimeHistory. (patchset #5 id:80001 of https://codereview.webrtc.org/2005313003/ )
2016-06-01 11:04:49 +00:00
p2p
Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc}
2016-06-01 09:17:56 +00:00
pc
Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc}
2016-06-01 09:17:56 +00:00
sdk
Revert of Replacing DtlsIdentityStoreInterface with RTCCertificateGeneratorInterface. (patchset #2 id:20001 of https://codereview.webrtc.org/2013523002/ )
2016-05-27 13:08:58 +00:00
system_wrappers
Rename APK tests workaround to make it more generic.
2016-05-31 11:01:47 +00:00
test
Rename APK tests workaround to make it more generic.
2016-05-31 11:01:47 +00:00
tools
Rename APK tests workaround to make it more generic.
2016-05-31 11:01:47 +00:00
video
Disabled RestartingSendStreamPreservesRtpStatesWithRtx on Mac due to flakiness.
2016-05-31 12:17:41 +00:00
voice_engine
Revert of Propagate probing cluster id to SendTimeHistory. (patchset #5 id:80001 of https://codereview.webrtc.org/2005313003/ )
2016-06-01 11:04:49 +00:00
.gitignore
…
audio_receive_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
audio_send_stream.h
Remove webrtc/base/scoped_ptr.h
2016-05-15 02:44:18 +00:00
audio_sink.h
…
audio_state.h
…
BUILD.gn
Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc}
2016-06-01 09:17:56 +00:00
call.h
Update the call when the network route changes
2016-04-19 22:41:53 +00:00
codereview.settings
…
common_types.cc
…
common_types.h
Add histogram stats for average send delay of sent packets for a sent video stream. The delay is measured from a packet is sent to the transport until leaving the socket.
2016-05-03 06:44:11 +00:00
common.gyp
Revert of Add missing headers and fix some missing dependencies (patchset #1 id:20001 of https://codereview.webrtc.org/1990593002/ )
2016-05-18 12:00:55 +00:00
common.h
#include "webrtc/base/constructormagic.h" where appropriate
2016-04-26 15:14:48 +00:00
config.cc
Remove use of RtpHeaderExtension and clean up
2016-05-26 18:25:04 +00:00
config.h
Remove use of RtpHeaderExtension and clean up
2016-05-26 18:25:04 +00:00
DEPS
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
engine_configurations.h
…
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
Add root owners to webrtc/OWNERS
2016-05-11 12:47:43 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
rtc_unittests.isolate
…
supplement.gypi
Disable libyuv jpeg support on Android
2016-05-17 15:39:15 +00:00
transport.h
…
typedefs.h
Remove runtime NEON detection
2016-05-17 17:56:48 +00:00
video_decoder.h
…
video_encoder.h
Remove QualityScaler framerate reduction.
2016-04-26 11:37:20 +00:00
video_engine_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
video_engine_tests.isolate
…
video_frame.h
Reland of Delete webrtc::VideoFrame methods buffer and stride. (patchset #1 id:1 of https://codereview.webrtc.org/1983583002/ )
2016-05-17 11:05:51 +00:00
video_receive_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
video_send_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
webrtc_examples.gyp
UI for AppRTC Android demo that doesn't require Design Support Library
2016-05-24 07:18:18 +00:00
webrtc_nonparallel_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_nonparallel_tests.isolate
…
webrtc_perf_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_perf_tests.isolate
…
webrtc_tests.gypi
Reland of Android: Change camera fps range selection (patchset #1 id:1 of https://codereview.webrtc.org/2021233002/ )
2016-06-01 07:44:07 +00:00
webrtc.gyp
Revert of Add missing headers and fix some missing dependencies (patchset #1 id:20001 of https://codereview.webrtc.org/1990593002/ )
2016-05-18 12:00:55 +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.