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Issue
2293433002:
Roll WebRTC 13869:13951 (74 commits) (Closed)
DescriptionRoll WebRTC 13869:13951 (74 commits)
Changes: https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/446b8bf..f80f5b3
$ git log 446b8bf..f80f5b3 --date=short --no-merges --format=%ad %ae %s
2016-08-29 ehmaldonado@webrtc.org Remove Chromium Clang warnings only on Windows.
2016-08-29 sakal@webrtc.org Remove the old AndroidVideoCapturer stack code.
2016-08-29 nisse@webrtc.org Reland of Delete method cricket::VideoFrame::Copy. (patchset #1 id:1 of https://codereview.webrtc.org/2275313003/ )
2016-08-29 nisse@webrtc.org Delete deprecated and unused method VideoFrame::SetTimeStamp.
2016-08-28 kjellander@webrtc.org MB: Flip Windows bots to GN by default
2016-08-26 deadbeef@webrtc.org Combining "SetTransportChannel" and "SetRtcpTransportChannel".
2016-08-26 deadbeef@webrtc.org Add parameter to TransportController to not change ICE role on restart.
2016-08-26 kwiberg@webrtc.org Fix Chromium clang plugin warnings
2016-08-26 ehmaldonado@webrtc.org GN: Fix windows clang errors. Attempt 2.
2016-08-26 kjellander@webrtc.org Adding GYP/GN owners to stats/ aligning with all other dirs.
2016-08-26 zhihuang@webrtc.org Log how often DTLS negotiation failed because of incompatible ciphersuites.
2016-08-26 peah@webrtc.org Added logging of the level controller metrics.
2016-08-26 danilchap@webrtc.org Change RtpSender::OnReceiveNACK name and signature Name changed to follow style. list replaced with vector to decrease number of included headers.
2016-08-26 peah@webrtc.org Deactivated the intelligibility enhancement functionality by default
2016-08-26 peah@webrtc.org Revert of Added functionality for specifying the initial signal level to use for the gain estimation in the l… (patchset #8 id:160001 of https://codereview.webrtc.org/2254973003/ )
2016-08-26 kjellander@webrtc.org MB: Flip Linux bots to GN by default.
2016-08-26 danilchap@webrtc.org AbsoluteSendTime rtp header extension publish MsTo24Bit conversion Since this conversion is used in multiple place and extension seems right place to keep it in.
2016-08-26 ossu@webrtc.org Removed virtual from several methods in DecoderDatabase to minimize the number of points that need to be mocked for testing.
2016-08-26 peah@webrtc.org This CL adds functionality in the level controller to receive a signal level to use initially, instead of the default initial signal level.
2016-08-26 kwiberg@webrtc.org Replace calls to assert() with RTC_DCHECK_*() in .c code
2016-08-26 johan@webrtc.org Skip unit test if GYP_DEFINES="rtc_use_h264=1" is not set.
2016-08-25 henrik.lundin@webrtc.org NetEq: Update CNG code to accommodate 48 kHz sample rate
2016-08-25 peah@webrtc.org Adding AecDump functionality to AppRTCDemo for iOS
2016-08-25 deadbeef@webrtc.org Renaming BaseChannel methods and adding comments for added clarity.
2016-08-25 philipel@webrtc.org Revert of Delete method cricket::VideoFrame::Copy. (patchset #3 id:210001 of https://codereview.webrtc.org/2275243002/ )
2016-08-25 nisse@webrtc.org Reland of Delete method cricket::VideoFrame::Copy. (patchset #1 id:1 of https://codereview.webrtc.org/2087923004/ )
2016-08-25 henrika@webrtc.org Avoids java.lang.NullPointerException in WebRtcAudioRecord
2016-08-25 ivoc@webrtc.org Moved format_macros.h from rtc_base to rtc_base_approved.
2016-08-25 ehmaldonado@webrtc.org GN: Fix Windows Clang errors
2016-08-25 maxmorin@webrtc.org Fix error when accumulating floats in an int.
2016-08-25 hbos@webrtc.org Refactor certificate stats collection, added SSLCertificateStats.
2016-08-25 magjed@webrtc.org Implement CVO for iOS capturer
2016-08-25 ehmaldonado@webrtc.org Add missing "//build/config/sanitizers:deps" to executable targets.
2016-08-25 brandtr@webrtc.org Move InsertZeroColumns and CopyColumn to ::internal.
2016-08-25 kwiberg@webrtc.org GN build rules for four audio processing test executables
2016-08-25 henrik.lundin@webrtc.org Make neteq_rtpplay parse RTP header extensions
2016-08-25 aleloi@webrtc.org Removed inline definitions and added destructors to fix chromium-style.
2016-08-25 henrik.lundin@webrtc.org NetEq: only update current_rtp_payload_type_ when validated
2016-08-24 deadbeef@webrtc.org Fix setting the MTU for SCTP.
2016-08-24 deadbeef@webrtc.org Fixing inconsistency with behavior of `ClearGettingPorts`.
2016-08-24 deadbeef@webrtc.org Fixing segfault caused by TurnServer.
2016-08-24 deadbeef@webrtc.org Fixing off-by-one error with max SCTP id.
2016-08-24 deadbeef@webrtc.org Fixing timestamp comparison assert.
2016-08-24 glaznev@webrtc.org Increase QP threshold for H.264 encoder QP based scaling.
2016-08-24 tkchin@webrtc.org Restart capture session if needed on active.
2016-08-24 henrik.lundin@webrtc.org NetEq: Don't check sample rate and frame size upon error
2016-08-24 henrik.lundin@webrtc.org Make FakeDecodeFromFile handle codec-internal CNG
2016-08-24 kjellander@webrtc.org MB: Flip Mac bots to GN by default.
2016-08-24 ehmaldonado@webrtc.org Roll chromium_revision e3860bd297..938114be1e (412289:414059)
2016-08-24 kjellander@webrtc.org GN: Synchronize resources between Android and iOS.
2016-08-24 maxmorin@webrtc.org Make dependency of audio_device of ApplicationServices explicit. Tested in https://codereview.webrtc.org/2276903002.
2016-08-24 philipel@webrtc.org Now probe for x3 and x6 of the initial start bitrate and allow for faster receive bitrates when calculating probing estimates.
2016-08-24 ivoc@webrtc.org Added GN target for webrtc_opus_fec_test.
2016-08-24 ehmaldonado@webrtc.org Disable examples for GYP Android bots.
2016-08-24 sakal@webrtc.org Revert of GN build rules for four audio processing test executables (patchset #3 id:40001 of https://codereview.webrtc.org/2267403003/ )
2016-08-24 kwiberg@webrtc.org GN build rules for four audio processing test executables
2016-08-24 philipel@webrtc.org Only use payload size within the know send/receive interval for probing calculations.
2016-08-24 kwiberg@webrtc.org iLBC: Handle a case of bad input data
2016-08-24 philipel@webrtc.org Set send side bitrate estimate on successful probing attempt.
2016-08-24 kjellander@webrtc.org GN: Add resources for webrtc_perf_tests on Android
2016-08-24 ivoc@webrtc.org Added GN target for isac_test.
2016-08-24 aleloi@webrtc.org Removals and renamings in the new audio mixer.
2016-08-24 nisse@webrtc.org Add ThreadChecker to the TimestampAligner class.
2016-08-24 aleloi@webrtc.org Increased column width for python tool rtp_analyzer.py.
2016-08-24 aleloi@webrtc.org Updated mixer unittests and fixed a related bug in the new mixer.
2016-08-24 hbos@webrtc.org RTCStats and RTCStatsReport added (webrtc/stats).
2016-08-24 aleloi@webrtc.org Added a level indicator to new mixer.
2016-08-24 kthelgason@webrtc.org Remove outdated symlink
2016-08-24 sakal@webrtc.org Fix AppRTC Android Demo GN build when is_component_build=true.
2016-08-24 kjellander@webrtc.org MB: Flip Android bots to GN by default.
2016-08-23 terelius@webrtc.org Define a protobuf format for representing plots. Add code to convert the C-representation generated by the RtcEventLog analysis tool, to the new protobuf format.
2016-08-23 terelius@webrtc.org Adds function for computing moving average to visualization tool.
2016-08-23 honghaiz@webrtc.org Add logs and small change in BasicPortAllocator.
2016-08-23 isheriff@google.com ProbingEstimator: Erase history based on time threshold
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://chromium.googlesource.com/chromium/src/+/55b05938527117dfcb5db481cd8bd8590c65f0a4
Patch Set 1 #Messages
Total messages: 12 (6 generated)
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