GitLab

https://webrtc.googlesource.com/src.git/+log/e769ed90c359..44ca9a392ac6


git log e769ed90c359..44ca9a392ac6 --date=short --no-merges --format='%ad %ae %s'
2018-11-13 mbonadei@webrtc.org Allow usage of stringstream under examples/.
2018-11-13 kwiberg@webrtc.org Remove some unused RentACodec static methods
2018-11-13 peah@webrtc.org AEC3: Corrected erroneous if-statement that always returned true
2018-11-13 nisse@webrtc.org Add missing include of unistd.h
2018-11-13 nisse@webrtc.org Delete deprecated class WrappedI420Buffer
2018-11-13 mbonadei@webrtc.org Configs to run slow_tests.
2018-11-13 nisse@webrtc.org Delete obsolete interface class RtpData
2018-11-13 srte@webrtc.org Adds setup of RTP Extensions in Scenario tests.
2018-11-13 asapersson@webrtc.org Add tests for cpu overuse scaling.
2018-11-12 ouj@fb.com Adding rtcp report interval into RTCConfiguration.
2018-11-12 ouj@fb.com Explicitly retain self in objc blocks to avoid compiler warning.
2018-11-12 srte@webrtc.org Allows change of fake encoder max rate in scenarios tests.
2018-11-12 srte@webrtc.org Add support for screenshare content type in scenario tests.
2018-11-12 srte@webrtc.org Simplifies audio priority rate config in scenario tests.
2018-11-12 eladalon@webrtc.org Remove obsolete comment (WebRtcSessionDescriptionFactory ctor)
2018-11-12 srte@webrtc.org Using early acknowledged rate for safe reset in GoogCC.
2018-11-12 ilnik@webrtc.org In RTP to NTP estimator use linear regression instead of ad hoc filter
2018-11-12 eladalon@webrtc.org Event log - Use ToUnsigned() and ToSigned() on timestamp_ms
2018-11-12 eladalon@webrtc.org Event logs - encode N channels as N-1
2018-11-12 kwiberg@webrtc.org AudioCodingModule: Remove support for creating encoders
2018-11-12 nisse@webrtc.org Tweak ChannelReceive interface, to make it closer to ChannelReceiveProxy
2018-11-12 nisse@webrtc.org Eliminate use of EventWrapper from android audio device tests
2018-11-12 eladalon@webrtc.org Add RtcEvent::timestamp_ms()
2018-11-12 kron@webrtc.org Add offer_extmap_allow_mixed to RTCConfiguration
2018-11-12 danilchap@webrtc.org Revert "Run robolectric tests for Android on several Android API versions"
2018-11-12 aleloi@webrtc.org Fuzzer crash in AGC2.
2018-11-12 jonasolsson@webrtc.org Remove most of api/ortc/.
2018-11-12 kron@webrtc.org Fix overflow for high bitrates in BitrateProber
2018-11-12 yvesg@google.com Revert "Roll "Enable SSE, SSE2, and run-time detected SSE4.1 for libopus.""
2018-11-10 eladalon@webrtc.org Hide RtcEvent members behind accessors
2018-11-10 eladalon@webrtc.org Event logs - separate audio_level and voice_activity
2018-11-09 yvesg@webrtc.org Roll "Enable SSE, SSE2, and run-time detected SSE4.1 for libopus."
2018-11-09 eladalon@webrtc.org Rename fields in rtc_event_log2.proto
2018-11-09 mellem@webrtc.org Fix up an outdated comment in peerconnection_integrationtest.cc.
2018-11-09 Peter) Slatala Signal Network route change in fake ice.
2018-11-09 eladalon@webrtc.org Use delta-encoding in new WebRTC event logs
2018-11-09 phoglund@webrtc.org Clean up root OWNERS.
2018-11-09 artit@webrtc.org Run robolectric tests for Android on several Android API versions
2018-11-09 kron@webrtc.org Pass HdrMetadata between VideoFrame and EncodedImage for VP9
2018-11-09 terelius@webrtc.org Add support for audio in latency visualization.
2018-11-09 jonasolsson@webrtc.org Fix flaky JsepTransportControllerTests.
2018-11-09 kron@webrtc.org Add RTP header extension for HDR metadata
2018-11-09 ilnik@webrtc.org In RTP to NTP estimator do not allow huge jumps in NTP timestamps
2018-11-09 yvesg@webrtc.org Reintroduce missing dependencies in libwebrtc.a library.
2018-11-09 mellem@webrtc.org Implement data channels over media transport.
2018-11-08 ouj@fb.com Reland "Use the factory instead of using the builtin code path in `VideoCodecInitializer`"
2018-11-08 yvesg@webrtc.org [Win/boringSSL] Add nasm as part of required dependencies.
2018-11-08 Peter) Slatala Callback changes to media transport interface:
2018-11-08 Peter) Slatala Add owners for media_transport_interface
2018-11-08 sprang@webrtc.org Add ability to specify if rate controller of video encoder is trusted.
2018-11-08 sprang@webrtc.org In Android encoders, cache EncoderInfo in InitEncode.
2018-11-08 nisse@webrtc.org Delete rtc::Filesystem. Move needed functions to filerotatingstream.cc.
2018-11-08 nisse@webrtc.org Eliminate use of EventWrapper from mac audio device
2018-11-08 sprang@webrtc.org Add magjed/nisse/sprang/brandtr as api/video_codecs owners
2018-11-08 danilchap@webrtc.org Introduce RtpPacket::GetExtension accessor that return result
2018-11-08 yujo@chromium.org Split a separate codecs target off of :video_jni
2018-11-08 nisse@webrtc.org Eliminate use of EventWrapper from ios audio device tests
2018-11-08 alessiob@webrtc.org Tolerate optional chunks in WAV files
2018-11-08 saza@webrtc.org Add flag for fast jitter buffer playout in neteq simulation
2018-11-08 alessiob@webrtc.org MsanUninitialized: restric type check to msan case.
2018-11-08 nisse@webrtc.org Delete classes EventFactory and EventFactoryImpl.
2018-11-08 oprypin@webrtc.org Make the bitrate_allocator param optional to prepare for its removal
2018-11-08 nisse@webrtc.org Reenable test RampUpTest.AudioTransportSequenceNumber
2018-11-08 kwiberg@webrtc.org Add a style rule about not using const optional<T>& arguments
2018-11-08 saza@webrtc.org Add missing conditional defines to neteq test and tools targets
2018-11-08 nisse@webrtc.org Deprecate EventFactory and delete all usage.
2018-11-07 sprang@webrtc.org Update H264 encoder to use GetEncoderInfo
2018-11-07 sprang@webrtc.org Update LibVpxVp8Encoder to use GetEncoderInfo
2018-11-07 sprang@webrtc.org Update VP9 encoder to use GetEncoderInfo
2018-11-07 orphis@webrtc.org Remove multiple RTX codec entries in GetRtpReceiver/SenderCapabilities
2018-11-07 sprang@webrtc.org Update SimulcastEncoderAdapter merging of EncoderInfo
2018-11-07 ilnik@webrtc.org Clear FrameBuffer if there were no frames received for 10 minutes
2018-11-07 alessiob@webrtc.org Reland "Isolating APM API build target: making :api an actual target."
2018-11-07 brandtr@webrtc.org Add field trial for target bitrate RTCP XR message.
2018-11-07 nisse@webrtc.org Delete NullEventFactory


Created with:
  gclient setdep -r src/third_party/webrtc@44ca9a392ac6

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:none,chromium:None,chromium:901661,chromium:None,chromium:None,chromium:None,chromium:766721,chromium:None,chromium:None,chromium:None,chromium:none,chromium:None
TBR=webrtc-chromium-sheriffs-robots@google.com

Change-Id: I80b2d4e7908e09e4b4b99e592eca5879ce252ca2
Reviewed-on: https://chromium-review.googlesource.com/c/1333849Reviewed-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@{#607647}