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Issue
2575013002:
Roll WebRTC 15513:15596 (48 commits) (Closed)
DescriptionRoll WebRTC 15513:15596 (48 commits)
Changes: https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/f23e926..2410df4
$ git log f23e926..2410df4 --date=short --no-merges --format=%ad %ae %s
2016-12-14 ivoc@webrtc.org Fix for left shift of negative value in NetEq.
2016-12-14 nisse@webrtc.org Delete method Pathname::url and base/urlencode*
2016-12-14 skvlad@webrtc.org Disable flaky test VideoProcessorIntegrationTest.ProcessNoLossChangeFrameRateFrameDropVP9
2016-12-14 peah@webrtc.org This CL adds the basic framework for AEC3 in the audio processing module. It will be followed by a number of other CLs that extends this framework.
2016-12-14 nisse@webrtc.org Delete unused class rtc::RegKey.
2016-12-14 nisse@webrtc.org Delete stl_util.h. Unused since cl https://codereview.webrtc.org/2447103002
2016-12-14 nisse@webrtc.org Revert of Delete rtc::linked_ptr. Only use, in statstypes.h, replaced bu std::unique_ptr. (patchset #1 id:1 of https://codereview.webrtc.org/2567143003/ )
2016-12-13 nisse@webrtc.org Delete rtc::linked_ptr. Only use, in statstypes.h, replaced with std::unique_ptr.
2016-12-14 kjellander@webrtc.org Disable AudioDeviceTest.StartPlayoutOnTwoInstances on iOS
2016-12-13 deadbeef@webrtc.org Fixing possible crash due to RefCountedChannel assignment operator.
2016-12-13 deadbeef@webrtc.org Fixing integer overflow when parsing bandwidth attribute.
2016-12-13 gyzhou@chromium.org Support external audio mixer in webrtc 2.
2016-12-13 deadbeef@webrtc.org Removing "crypto_required" from MediaContentDescription.
2016-12-13 hnsl@webrtc.org ParseCandidate(): Refactor to fix memcheck false positive. Also make supported protocols explicit in check.
2016-12-13 minyue@webrtc.org Update common_audio/smoothing_filter.
2016-12-13 nisse@webrtc.org Delete VideoFrame default constructor, and IsZeroSize method.
2016-12-13 kthelgason@webrtc.org Disable flaky QualityScaler tests for now.
2016-12-13 hnsl@webrtc.org Refactor "secure bool" into explicit PROTO_TLS.
2016-12-13 thomasanderson@google.com Add a gtk3 port of peerconnection_client on Linux
2016-12-13 palmkvist@webrtc.org Logging basic bad call detection
2016-12-13 hbos@webrtc.org Revert of New PeerConnectionInterface::GetStats: No bogus default implementation. (patchset #1 id:1 of https://codereview.webrtc.org/2566143002/ )
2016-12-13 johan@webrtc.org Rename RtpStreamReceiver::SetCodec() to ::AddCodec().
2016-12-13 hbos@webrtc.org New PeerConnectionInterface::GetStats: No bogus default implementation.
2016-12-13 ivoc@webrtc.org Fix for negative shift value in NetEq.
2016-12-12 nisse@webrtc.org Delete unused class AsyncFile.
2016-12-12 deadbeef@webrtc.org Don't allow changing ICE pool size after SetLocalDescription.
2016-12-12 deadbeef@webrtc.org Implement parsing/serialization of a=bundle-only.
2016-12-12 gyzhou@chromium.org Revert of Support external audio mixer. (patchset #5 id:140001 of https://codereview.webrtc.org/2539213003/ )
2016-12-12 gyzhou@chromium.org Support external audio mixer in webrtc.
2016-12-12 minyue@webrtc.org Making audio network adaptor config proto a JAVA package.
2016-12-12 noahric@chromium.org Fix header guard in thread_annotations.h.
2016-12-12 noahric@chromium.org Destroy encoders that fail to InitEncode.
2016-12-12 deadbeef@webrtc.org Add SSRC to RtpEncodingParameters for audio.
2016-12-12 stefan@webrtc.org Add OWNERS to BWE modules.
2016-12-12 brandtr@webrtc.org Remove sequenced task checker from FlexfecSender.
2016-12-12 henrik.lundin@webrtc.org Delete voice_engine_configurations.h
2016-12-12 philipp.hancke@googlemail.com remove googViewLimitedResolution stat
2016-12-12 peah@webrtc.org Disabling the potentially flaky test VideoProcessorIntegrationTest. ProcessNoLossSpatialResizeFrameDropVP9
2016-12-12 hnsl@webrtc.org Fix out of bound reads in ParseIceServerUrl() for various input.
2016-12-12 brandtr@webrtc.org Try to deflake VideoSendStream tests with ULPFEC.
2016-12-12 hbos@webrtc.org RTCIceCandidatePairStats.consentRequestsSent set by RTCStatsCollector and requestsSent is updated.
2016-12-12 nisse@webrtc.org Replace VideoCaptureDataCallback by VideoSinkInterface.
2016-12-12 kjellander@webrtc.org Change MANUAL -> DISABLED for ScreenCapturerIntegrationTest tests
2016-12-11 ssaroha@gmail.com Set OPENSSL_EC_NAMED_CURVE explicitly on EC key so that certificate has ASN1 OID and NIST curve info. Without this openSSL handshake negotiation fails throwing NO_SHARED_CIPHER error. the change made is along the lines of openssl behavior documented here: https://wiki.openssl.org/index.php/Elliptic_Curve_Diffie_Hellman#ECDH_and_Named_Curves
2016-12-11 minyue@webrtc.org Adding googAudioNetworkAdaptorConfig to MediaConstraintsInterface.
2016-12-10 deadbeef@webrtc.org Implement the "needs-ice-restart" logic for SetConfiguration.
2016-12-10 deadbeef@webrtc.org Adding error enum to be used by PeerConnectionInterface methods.
2016-12-09 zijiehe@chromium.org MANUAL tests of GDI capturers
TBR=
CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.linux:linux_chromium_archive_rel_ng;master.tryserver.chromium.mac:mac_chromium_archive_rel_ng
BUG=
Committed: https://crrev.com/59343ca9b2680dee482a9ac246c944705a0bd3d0
Cr-Commit-Position: refs/heads/master@{#438488}
Patch Set 1 #Messages
Total messages: 7 (4 generated)
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