Chromium Code Reviews Chromium Code Reviews
Issue
2310963002:
Roll WebRTC 14025:14068 (40 commits) (Closed)
DescriptionRoll WebRTC 14025:14068 (40 commits)
Changes: https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/4cb672c..6d6af9a
$ git log 4cb672c..6d6af9a --date=short --no-merges --format=%ad %ae %s
2016-09-05 kwiberg@webrtc.org Restrict the 1-argument ArrayView constructor to types with .size() and .data()
2016-09-05 ehmaldonado@webrtc.org GN Templates: Use the optimize_max compiler config.
2016-09-05 philipel@webrtc.org Plot accumelated packets over time.
2016-09-05 philipel@webrtc.org FrameBuffer::NextFrame now return a ReturnReason and take an additional std::unique_ptr<FrameObject>* as output parameter.
2016-09-05 hbos@webrtc.org RTCStatsCollector collecting stats on multiple threads.
2016-09-05 ehmaldonado@webrtc.org GN Templates: Move common_config to the template.
2016-09-05 nisse@webrtc.org Introduce webrtc::VideoFrame::timestamp_us, and corresponding constructor. Replaces render_time_ms_, but old accessors are kept for compatibility.
2016-09-04 stefan@webrtc.org Only log BWE updates if the actual estimate changed or if we have non-zero loss reports.
2016-09-03 sakal@webrtc.org Remove restart method from VideoTrackSourceInterface.
2016-09-02 zijiehe@chromium.org [WebRTC] A real ScreenCapturer test
2016-09-02 honghaiz@webrtc.org Change a few configurations in AggressiveConfiguration Set bundle policy to max bundle. Set rtcp mux policy to required. Set enable ice renomination to true. This configuration is used by native applications.
2016-09-02 skvlad@webrtc.org Fixed flaky VideoSendStreamTest::SupportsAbsoluteSendTime
2016-09-02 skvlad@webrtc.org EventLogParser: use std::vector to reduce stack usage
2016-09-02 glaznev@google.com Limit initial fps value used in Android HW encoder initialization.
2016-09-02 johan@webrtc.org Provide a default implementation for PeerConnectionInterface::ice_state().
2016-09-02 kjellander@webrtc.org MB: Enable rtc_use_h264 for GYP builds for now.
2016-09-02 danilchap@webrtc.org Introduce helpers to RtpSender to propagate RtpPacketToSend. The helpers intended to replace and deprecate BuildRtpHeader when RtpSenderAudio/RtpSenderVideo will be updated to pass RtpPacket class instead of raw buffer for sending.
2016-09-02 danilchap@webrtc.org Test RtcpParser rewritten to use rtcp packet classes instead of rtcp_utility
2016-09-02 nisse@webrtc.org Delete cricket::VideoFrame::GetCopyWithRotationApplied.
2016-09-02 ehmaldonado@webrtc.org GN Templates: Add //build/config/sanitizers:deps to rtc_executable.
2016-09-02 sakal@webrtc.org Move getSupportedFormats from capturer interface to camera enumerator.
2016-09-02 kjellander@webrtc.org Add kjellander@webrtc.org to more OWNERS for BUILD.gn files.
2016-09-02 palmkvist@webrtc.org Reland of Initial version of file wrapper
2016-09-02 ehmaldonado@webrtc.org GN: Introduce templates.
2016-09-02 stefan@webrtc.org Reland of Add pps id and sps id parsing to the h.264 depacketizer. (patchset #1 id:1 of https://codereview.webrtc.org/2265023002/ )
2016-09-02 stefan@webrtc.org Enable send-side BWE by default.
2016-09-02 nisse@webrtc.org Delete talk directory, and references from build_ios_libs.sh.
2016-09-02 solenberg@webrtc.org - Remove unused unit test webrtc/modules/audio_processing/agc/agc_unittest.cc - Remove webrtc/tools/agc/test_utils.cc/.h - only used from the above test. - Remove webrtc/tools/agc/agc_harness.cc - not used anymore.
2016-09-02 nisse@webrtc.org Revert of Delete cricket::VideoFrame::GetTimeStamp. (patchset #1 id:1 of https://codereview.webrtc.org/2305623002/ )
2016-09-02 nisse@webrtc.org Delete cricket::VideoFrame::GetTimeStamp.
2016-09-02 sakal@webrtc.org Remove VideoSource.stop() and VideoSource.restart() from the Java API.
2016-09-02 magjed@webrtc.org Revert of Ignore Camera and Flip bits in CVO when parsing video rotation (patchset #3 id:80001 of https://codereview.webrtc.org/2280703002/ )
2016-09-02 kwiberg@webrtc.org Add functions to interact with ASan and MSan, and some sample uses
2016-09-01 henrik.lundin@webrtc.org NetEq: Flush and reset if the speech and cng sample rates mismatch
2016-09-01 peah@webrtc.org In order to ensure that the same code is run in the tests as is otherwise run it is important that the same build flags are used in the code being tested. For the debugging functionality inside APM, that was not the case and this is corrected in this CL.
2016-09-01 honghaiz@webrtc.org Use AggressiveConfiguration as the default configuration in IOS
2016-09-01 kwiberg@webrtc.org WebRtcIlbcfix_EnhancerInterface: Let input array be const
2016-09-01 glaznev@webrtc.org Add option to set maximum video encoder bitrate to AppRTCDemo.
2016-09-01 magjed@webrtc.org Ignore Camera and Flip bits in CVO when parsing video rotation
2016-09-01 peah@webrtc.org Moved the place for the aec_debug_dump build flag and changed the name to apm_debug_dump
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/93f28e77a2e7967372647bcff0d58b2639356608
Cr-Commit-Position: refs/heads/master@{#416547}
Patch Set 1 #Messages
Total messages: 5 (2 generated)
|
