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Issue
2326513002:
Roll WebRTC 14068:14133 (60 commits) (Closed)
DescriptionRoll WebRTC 14068:14133 (60 commits)
Changes: https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git/+log/6d6af9a..e5402e7
$ git log 6d6af9a..e5402e7 --date=short --no-merges --format=%ad %ae %s
2016-09-08 magjed@webrtc.org Revert of move all reference to carbon api (patchset #2 id:300001 of https://codereview.webrtc.org/2321493002/ )
2016-09-08 magjed@webrtc.org Android ThreadUtils: Propagate exceptions in invoke functions
2016-09-08 henrik.lundin@webrtc.org Revert of Setting up an RTP input fuzzer for NetEq (patchset #2 id:20001 of https://codereview.webrtc.org/2315633002/ )
2016-09-08 ossu@webrtc.org Removed sync packet support from NetEq.
2016-09-08 henrika@webrtc.org Avoids crash in WebRtcAudioTrack.initPlayout
2016-09-08 kwiberg@webrtc.org FilePlayer: Remove backwards compatibility stuff that we no longer need
2016-09-08 ehmaldonado@webrtc.org GN Templates: Introduce rtc_shared_library
2016-09-08 ehmaldonado@webrtc.org MB: Move Linux 32 bots from the WebRTC FYI to the main waterfall.
2016-09-08 kthelgason@webrtc.org Reland of move all reference to carbon api (patchset #1 id:1 of https://codereview.webrtc.org/2316563002/ )
2016-09-08 magjed@webrtc.org Reland of Ignore Camera and Flip bits in CVO when parsing video rotation (patchset #1 id:1 of https://codereview.webrtc.org/2300323002/ )
2016-09-08 henrik.lundin@webrtc.org Fixing NetEqReplacementInput for reordered and missing packets
2016-09-08 aleloi@webrtc.org Python event log analyzer tool: fix of indexing issue.
2016-09-08 aleloi@webrtc.org This CL contains the following small changes:
2016-09-08 asapersson@webrtc.org Use RateCounter for received bitrate stats: "WebRTC.Call.BitrateReceivedInKbps" "WebRTC.Call.VideoBitrateReceivedInKbps" "WebRTC.Call.AudioBitrateReceivedInKbps" "WebRTC.Call.RtcpBitrateReceivedInBps"
2016-09-07 asapersson@webrtc.org Do not report bucket delay for stats when pacer is paused (zero returned).
2016-09-07 VladimirTechMan@gmail.com Copy RTCAudioSource.h and RTCMediaSource.h with other public header files when building the WebRTC framework for iOS / Mac
2016-09-07 deadbeef@webrtc.org If encoding is inactive, don't start sending when stream is reconfigured.
2016-09-07 minyue@webrtc.org Adding AudioNetworkAdaptor interfaces.
2016-09-07 zijiehe@chromium.org Reland of [WebRTC] A real ScreenCapturer test (patchset #1 id:1 of https://codereview.webrtc.org/2310953002/ )
2016-09-07 skvlad@webrtc.org Renamed and restructured the protobuf definitions for the rtc_event_log graphs.
2016-09-07 deadbeef@webrtc.org Fixing stack buffer overflow (read) in SctpDataEngine.
2016-09-07 aleloi@webrtc.org Simplifications of the mixing algorithm.
2016-09-07 solenberg@webrtc.org Moving/renaming webrtc/common.h.
2016-09-07 ehmaldonado@webrtc.org GN: Move variables from //build_overrides/webrtc.gni to //webrtc/build/webrtc.gni
2016-09-07 perkj@webrtc.org Change OverUseFrameDetector to use a task queue instead of ProcessThread to periodically check for overuse. It is made to only operate on a single task queue.
2016-09-07 aleloi@webrtc.org Several lock acquisitions and one of the two lock members are removed. ENSURE_LOCKS_REQUIRED and CalledOnValidThread annotations are added.
2016-09-07 danilchap@webrtc.org Implement PlayoutDelay extension as a trait to be used with rtp::Packet class
2016-09-07 danilchap@webrtc.org Relax expectation in EndToEndTest.CallReportsRttForSender test to reduce flakiness by ignoring potentional rounding errors and minor ntp time adjustments.
2016-09-07 henrik.lundin@webrtc.org Setting up an RTP input fuzzer for NetEq
2016-09-07 danilchap@webrtc.org Merge min_ms and max_ms accessors in PlayoutDelayOracle to reduce CriticalSection enterencies and avoid potentional synchronisation issues.
2016-09-07 stefan@webrtc.org Only parse PPS up to PPS and SPS ids in the depacketizater.
2016-09-07 kjellander@webrtc.org Disable -Wsentinel warning for Linux 32-bit builds.
2016-09-07 stefan@webrtc.org Add time line for acked bitrate.
2016-09-06 skvlad@webrtc.org Fixed flaky StunRequestTests which depended on the wall clock
2016-09-06 skvlad@webrtc.org Increase timeout for flaky tests for ProcessThreadImpl
2016-09-06 glaznev@google.com Add dynamic bitrate tracker and adjustment for Exynos VP8 HW encoder.
2016-09-06 aluebs@webrtc.org Fix chromium-style errors in IntelligibilityEnhancer
2016-09-06 nisse@webrtc.org Reland of Delete cricket::VideoFrame::GetTimeStamp. (patchset #1 id:1 of https://codereview.webrtc.org/2315703002/ )
2016-09-06 aleloi@webrtc.org Improvements to UI to python event log analyzer tool.
2016-09-06 kwiberg@webrtc.org iSAC float: Handle errors in upper band decoding
2016-09-06 nisse@webrtc.org Revert of Delete cricket::VideoFrame::GetTimeStamp. (patchset #2 id:150001 of https://codereview.webrtc.org/2310043002/ )
2016-09-06 nisse@webrtc.org Reland of Delete cricket::VideoFrame::GetTimeStamp. (patchset #1 id:1 of https://codereview.webrtc.org/2306953002/ )
2016-09-06 danilchap@webrtc.org Remove dedicated unittest file for remb format RembStatus moved to RtcpSender unittest where it fits better Creating remb in Compound/ReducedSize modes already covered by RtcpSender unittests. Parsing remb already covered by RtcpReceiverTest.ReceivesRemb
2016-09-06 kjellander@webrtc.org MB: Cleanup no-longer-used GN configurations
2016-09-06 ehmaldonado@webrtc.org GN: Set WEBRTC_RESTRICT_LOGGING as is set in GYP.
2016-09-06 sakal@webrtc.org Remove stop method from VideoTrackSourceInterface.
2016-09-06 hbos@webrtc.org Significantly increased max_num_buffers_ of Vp9FrameBufferPool.
2016-09-06 henrik.lundin@webrtc.org neteq_rtpplay: Add an error message for unmatched SSRC
2016-09-06 kjellander@webrtc.org MB: Add Linux 32-bit Debug and Release trybots
2016-09-06 kthelgason@webrtc.org Revert of Remove all reference to carbon api (patchset #2 id:20001 of https://codereview.webrtc.org/2299633002/ )
2016-09-06 kthelgason@webrtc.org Remove all reference to carbon api
2016-09-05 zijiehe@chromium.org [WebRTC] Two DirectX capturers cannot work concurrently
2016-09-05 zijiehe@chromium.org An early analysis shows in DirectX based capturer, Windows API returns larger dirty region than the real screen change. A similar behavior may happen on other platforms with damage notification support. So it's better to have an individual layer to handle the Differ logic, and remove capturing independent logic out of each ScreenCapturer* implementation.
2016-09-05 ehmaldonado@webrtc.org GN: Apply optimize_max only on windows
2016-09-05 danilchap@webrtc.org RtcpReceiverTest rewritten using public available interface IncomingPacket(const uint8_t*, size_t) is used as entry point instead of IncomingRTCPPacket(PacketInformation* out, RtcpParser* in); Result is validated by checking which callbacks were called instead of checking intermediate structure PacketInformaion.
2016-09-05 kwiberg@webrtc.org rtc::Buffer: Let SetData and AppendData accept anything with .data() and .size()
2016-09-05 danilchap@webrtc.org Use RtpPacketToSend in RtpSenderAudio. this eliminates reparsing of rtp packet on send audio path
2016-09-05 asapersson@webrtc.org Revert of [WebRTC] A real ScreenCapturer test (patchset #8 id:240001 of https://codereview.webrtc.org/2268093002/ )
2016-09-05 ehmaldonado@webrtc.org MB: Add WebRTC FYI bots to mb_config.pyl.
2016-09-05 ehmaldonado@webrtc.org GN Templates: Move common_inherited_config to the template.
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/97647b7f0ade60db7dca0d6f2d57e2f8dd79cc20
Cr-Commit-Position: refs/heads/master@{#417276}
Patch Set 1 #Messages
Total messages: 5 (2 generated)
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