Roll src/third_party/webrtc f18b35284288..689b5874d4fb (50 commits)

https://webrtc.googlesource.com/src.git/+log/f18b35284288..689b5874d4fb


git log f18b35284288..689b5874d4fb --date=short --no-merges --format='%ad %ae %s'
2018-09-03 nisse@webrtc.org Use monotonic clock for PhysicalSocketServer timeouts.
2018-09-01 phoglund@webrtc.org Roll chromium_revision c1d47013..bbc67a1b (585833:587546)
2018-09-01 phoglund@webrtc.org Remove MSVC debug bots from CQ.
2018-08-31 julien.isorce@chromium.org ScreenCapturerMac: destroy the streams and remove the DisplayStreamManager
2018-08-31 steveanton@webrtc.org Use AsyncInvoker in DtmfSender instead of MessageHandler
2018-08-31 steveanton@webrtc.org Use AsyncInvoker in DataChannel instead of MessageHandler
2018-08-31 steveanton@webrtc.org Use AsyncInvoker in JsepTransportController instead of MessageHandler
2018-08-31 steveanton@webrtc.org Use AsyncInvoker in PeerConnection instead of MessageHandler
2018-08-31 srte@webrtc.org Removes redundant starting rate.
2018-08-31 devicentepena@webrtc.org AEC3: option for using the stationarity estimator at render from the beginning of the call
2018-08-31 danilchap@webrtc.org Uninline non-trivial AudioOptions functions
2018-08-31 danilchap@webrtc.org Implement periodic cancelable task for task queue
2018-08-31 phoglund@webrtc.org Bump xcode versions for WebRTC bots.
2018-08-31 alessiob@webrtc.org Moving LappedTransform, Blocker and AudioRingBuffer.
2018-08-31 danilchap@webrtc.org Cleanup RtpPacketizerVP8 tests
2018-08-31 nisse@webrtc.org Reland "Add spatial index to EncodedImage."
2018-08-31 peah@webrtc.org AEC3: Parametrize the shadow filter output usage
2018-08-31 qingsi@google.com Add the multicast DNS message format.
2018-08-30 alessiob@webrtc.org Removing the intelligibility enhancer.
2018-08-30 wfh@chromium.org Add no_size_t_to_int_warning suppression to webrtc.
2018-08-30 brandtr@webrtc.org Revert "Reland "Optimize execution time of RTPSender::UpdateDelayStatistics""
2018-08-30 terelius@webrtc.org Reland "Optimize execution time of RTPSender::UpdateDelayStatistics"
2018-08-30 srte@webrtc.org Adds TaskQueue congestion controller tests in VideoSendStreamTest.
2018-08-30 srte@webrtc.org Adds support for frame rate control in FrameGeneratorCapturer.
2018-08-30 srte@webrtc.org Fixes breaking bug in feedback based GoogCC.
2018-08-30 phoglund@webrtc.org Roll chromium_revision 33a17747..c1d47013 (585798:585833)
2018-08-30 nisse@webrtc.org Revert "Refactor TestAudioDeviceModule to not depend on EventTimerWrapper."
2018-08-30 nisse@webrtc.org Use a lock to protect members accessed by RtpVideoStreamReceiver::GetSyncInfo()
2018-08-30 nisse@webrtc.org Delete StreamStatistician::IsRetransmitOfOldPacket
2018-08-30 andersc@webrtc.org Obj-C SDK Cleanup
2018-08-30 nisse@webrtc.org Refactor TestAudioDeviceModule to not depend on EventTimerWrapper.
2018-08-30 danilchap@webrtc.org Cleanup RtpPacketizerVp8
2018-08-30 phoglund@webrtc.org Roll chromium_revision ca3a5e1c..33a17747 (585726:585798)
2018-08-30 benwright@webrtc.org Injects FrameEncryptorInterface into RtpSender.
2018-08-29 benwright@webrtc.org This change integrates the FrameEncryptorInterface and the
2018-08-29 ssilkin@webrtc.org Move VP9 frame rate controller to separate class.
2018-08-29 nisse@webrtc.org Revert "Add spatial index to EncodedImage."
2018-08-29 nisse@webrtc.org Add spatial index to EncodedImage.
2018-08-29 devicentepena@webrtc.org AEC3: Adding a reset of the ERLE estimator after going out from the initial state.
2018-08-29 mbonadei@webrtc.org Remove clang:find_bad_constructs suppression from call:call.
2018-08-29 titovartem@webrtc.org Remove deprecated ctors of DirectTransport and its subclasses and FakeNetworkPipe
2018-08-29 valeriian@webrtc.org Adding CustomAudioAnalyzer interface in APM.
2018-08-29 devicentepena@webrtc.org AEC3: Reset the ERLE estimation after a delay change
2018-08-29 sakal@webrtc.org Use generic video header frame ID as picture ID.
2018-08-29 kthelgason@webrtc.org Add config option to run VideoCodecTest in real time.
2018-08-29 kthelgason@webrtc.org Export constants from RTCAudioSessionConfiguration.
2018-08-29 titovartem@webrtc.org Add support of overriding network simulation in video quality tests.
2018-08-29 kthelgason@webrtc.org Remove kVideoCodecUnknown completely.
2018-08-29 danilchap@webrtc.org Cleanup RtpPacketizer interface
2018-08-29 henrika@webrtc.org Increases max size of webrtc::AudioFrame from 60ms to 120ms @32kHz.


Created with:
  gclient setdep -r src/third_party/webrtc@689b5874d4fb

The AutoRoll server is located here: https://autoroll.skia.org/r/webrtc-chromium-autoroll

Documentation for the AutoRoller is here:
https://skia.googlesource.com/buildbot/+/master/autoroll/README.md

If the roll is causing failures, please contact the current sheriff, who should
be CC'd on the roll, and stop the roller if necessary.

CQ_INCLUDE_TRYBOTS=luci.chromium.try:linux_chromium_archive_rel_ng;luci.chromium.try:mac_chromium_archive_rel_ng

BUG=chromium:None,chromium:851883,chromium:None,chromium:None,chromium:None,chromium:None,chromium:879451,chromium:588506,chromium:None,chromium:878319,chromium:None,chromium:None,chromium:None
TBR=webrtc-chromium-sheriffs-robots@google.com

Change-Id: Ie676e811ca23db391939ff0facf56f6aa46abbcd
Reviewed-on: https://chromium-review.googlesource.com/1201666Reviewed-by: webrtc-chromium-autoroll <webrtc-chromium-autoroll@skia-buildbots.google.com.iam.gserviceaccount.com>
Commit-Queue: webrtc-chromium-autoroll <webrtc-chromium-autoroll@skia-buildbots.google.com.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#588372}

DEPS

@@ -1098,7 +1098,7 @@ deps = {
     Var('chromium_git') + '/external/khronosgroup/webgl.git' + '@' + '7ca87fb1d3da3b3d2060886e8c58e726d74c8219',
 
   'src/third_party/webrtc':
-    Var('webrtc_git') + '/src.git' + '@' + 'f18b35284288ac851b77db88df3e2e8d2273db97',
+    Var('webrtc_git') + '/src.git' + '@' + '689b5874d4fb5ee1767aef0d05e0a929ac2ea46e',
 
   'src/third_party/xdg-utils': {
       'url': Var('chromium_git') + '/chromium/deps/xdg-utils.git' + '@' + 'd80274d5869b17b8c9067a1022e4416ee7ed5e0d',