GitLab

https://webrtc.googlesource.com/src.git/+log/59e875ce18bb..e9d2b4efdd5d


git log 59e875ce18bb..e9d2b4efdd5d --date=short --no-merges --format='%ad %ae %s'
2019-04-08 jeroendb@webrtc.org Revert "Remove old audio device implementation."
2019-04-08 orphis@webrtc.org Revert "Remove TaskQueue constructor that uses GlobalTaskQueueFactory"
2019-04-08 eladalon@webrtc.org Rename EncoderKeyFrameCallback back to EncoderRtcpFeedback
2019-04-08 eladalon@webrtc.org Give VideoSendStreamImpl access to RTP timestamps
2019-04-08 kthelgason@webrtc.org Remove old audio device implementation.
2019-04-08 sprang@webrtc.org Update VideoStreamEncoder to use new VideoEncoder::SetRates() method.
2019-04-08 chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com Roll chromium_revision baccefbc..7987ebdc (648532:648632)
2019-04-08 eladalon@webrtc.org Add RtpSequenceNumberMap::InsertFrame()
2019-04-08 asapersson@webrtc.org Remove unused members in VCMJitterEstimator.
2019-04-08 nisse@webrtc.org Delete placeholder code for Windows Media Foundation capturer.
2019-04-08 eladalon@webrtc.org Fix timeout in rtcp_receiver_fuzzer - limit input length
2019-04-08 kron@webrtc.org Fix potential bug due to malformed input
2019-04-08 kuddai@google.com Change PlayoutLatency setLatency zero-threshold value.
2019-04-08 titovartem@google.com Add stream labels into PeerConnection level smoke test.
2019-04-08 nisse@webrtc.org Move enum VideoType out of common_types.h
2019-04-08 titovartem@google.com Fix naming in NetworkEmulationManager: endpoint_controller -> endpoint_container
2019-04-08 kron@webrtc.org Make UDP receive buffer size configurable via field trial
2019-04-08 benwright@webrtc.org Added FrameDecryptorInterface::Result constructor and IsOk() member function.
2019-04-07 mbonadei@webrtc.org Implement DefaultAudioQualityAnalyzer.
2019-04-07 chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com Roll chromium_revision 58ec016f..baccefbc (648432:648532)
2019-04-06 chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com Roll chromium_revision 0a611f37..58ec016f (648329:648432)
2019-04-05 chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com Roll chromium_revision 37663bcc..0a611f37 (648215:648329)
2019-04-05 benwright@webrtc.org Add Result FrameDecryptorInterface::Decrypt
2019-04-05 chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com Roll chromium_revision f91f8258..37663bcc (648096:648215)
2019-04-05 srte@webrtc.org Using simulated time for GoogCC tests.
2019-04-05 gustaf@webrtc.org AEC3: Enable usage of external delay estimator
2019-04-05 philipel@webrtc.org Tune VideoCodecTestLibvpx.TemporalLayersVP8 thresholds.
2019-04-05 srte@webrtc.org Running scenario quality unit tests in simulated time.
2019-04-05 danilchap@webrtc.org Remove TaskQueue constructor that uses GlobalTaskQueueFactory


Created with:
  gclient setdep -r src/third_party/webrtc@e9d2b4efdd5d

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.



BUG=chromium:None,chromium:none,chromium:None,chromium:948469,chromium:949020,chromium:939340,chromium:None,chromium:None,chromium:None,chromium:None,chromium:b/130016532,chromium:none
TBR=webrtc-chromium-sheriffs-robots@google.com

Change-Id: I47cf11c55e2695f3d95d4e9109726ab7477a85f0
Reviewed-on: https://chromium-review.googlesource.com/c/chromium/src/+/1557892Reviewed-by: chromium-autoroll <chromium-autoroll@skia-public.iam.gserviceaccount.com>
Commit-Queue: chromium-autoroll <chromium-autoroll@skia-public.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#648722}