external/webrtc.git
6 hours ago kjellander... Revert "This adds an Android apk for running tests... master
11 hours ago perkj@webrtc.orgThis adds an Android apk for running tests on the...
2 days ago thorcarpenter... Remove expensive and unnecessary memory alloc for sendi...
2 days ago andrew@webrtc.orgBuild fix for MIPS Android Webview build.
2 days ago magjed@webrtc.orgcricket::VideoFrame: Refactor ConvertToRgbBuffer into...
2 days ago kjellander... Update mock_frame_dropper.h to use size_t
3 days ago pkasting@chromium.orgUse size_t more consistently for packet/payload lengths.
3 days ago glaznev@webrtc.orgSupport loopback mode and command line execution
3 days ago henrik.lundin... Fix problems if first packet into NetEq is rejected
3 days ago henrik.lundin... Create a NetEq test for when the first incoming payload...
3 days ago asapersson... Change default values for CpuOveruseOptions.
4 days ago magjed@webrtc.orgcricket::VideoAdapter: Drop frames before spending...
4 days ago henrik.lundin... Revert "Add DCHECK to ensure that NetEq's packet buffer...
4 days ago henrik.lundin... Add DCHECK to ensure that NetEq's packet buffer is...
5 days ago henrika@webrtc.orgAppRTCDemoActivity: Add a config CheckBox for enabling...
5 days ago kjellander... Roll chromium_revision 91f1781..d8c9041
6 days ago aluebs@webrtc.orgAdd empty 3 band splitting filter API
6 days ago pkasting@chromium.orgFix ExpectedQueueTimeMs() to avoid truncation or overflow.
6 days ago guoweis@webrtc.orgAdd jmi field for packets discarded due to network...
6 days ago magjed@webrtc.orgAdd preliminary empty file videoframefactory.cc
6 days ago pbos@webrtc.orgAnnotate COMPILE_ASSERT with __attribute__((unused)).
6 days ago minyue@webrtc.orgSetting Opus FEC as default
6 days ago henrik.lundin... Use RtpFileSource in NetEqDecodingTest
7 days ago tommi@webrtc.orgRevert 7707 "cricket::VideoAdapter: Drop frames before...
7 days ago magjed@webrtc.orgcricket::VideoAdapter: Drop frames before spending...
9 days ago henrike@webrtc.orgRevert 7693 "Add jmi field for packets discarded due...
9 days ago aluebs@webrtc.orgWrap the splitting filter in its own class
9 days ago pbos@webrtc.orgDisable EndToEnd.GetStats test.
9 days ago magjed@webrtc.orgRevert 7702 "cricket::VideoAdapter: Drop frames before...
9 days ago magjed@webrtc.orgcricket::VideoAdapter: Drop frames before spending...
9 days ago pbos@webrtc.orgReport total bitrate for all streams in GetStats.
10 days ago magjed@webrtc.orgRevert 7698 "WebRtcVideoMediaChannel::SetSendParams...
10 days ago kjellander... Remove unnecessary copying of libjingle resource files.
10 days ago magjed@webrtc.orgWebRtcVideoMediaChannel::SetSendParams: Don't cap resol...
10 days ago pbos@webrtc.orgMake SetREMBData accept vector of SSRCs.
10 days ago pbos@webrtc.orgFix and enable CanReceiveFec test.
10 days ago bjornv@webrtc.orgSet correct sample rate in far_frame in audioproc tool.
10 days ago kjellander... Update isolate files for Android APK tests.
11 days ago guoweis@webrtc.orgAdd jmi field for packets discarded due to network...
11 days ago jiayl@webrtc.orgFix a platform check to use WEBRTC_WIN instead of OS_WIN.
11 days ago jiayl@webrtc.orgFix a SCTP message reordering issue in datachannel.cc.
11 days ago magjed@webrtc.orgwebrtc::Scaler: Preserve aspect ratio
11 days ago magjed@webrtc.orgVideoSendStreamTest.SwapsI420VideoFrames: Initialize...
11 days ago kjellander... Change the static_library("webrtc") to a source set...
12 days ago andrew@webrtc.orgreplace inline assembly WebRtcAecm_CalcLinearEnergiesNe...
12 days ago andrew@webrtc.orgreplace inline assembly WebRtcAecm_StoreAdaptiveChannel...
12 days ago jiayl@webrtc.orgUse ScreenCapturer to capture the whole and clip to...
12 days ago tnakamura@webrtc.orgBump to version 40
12 days ago magjed@webrtc.orgRevert 7679 "webrtc::Scaler: Preserve aspect ratio"
12 days ago kjellander... Add PROJECT to codereview.settings
12 days ago kjellander... Roll chromium_revision 375f736..91f1781
12 days ago magjed@webrtc.orgwebrtc::Scaler: Preserve aspect ratio
12 days ago asapersson... Add thread annotations to overuse_frame_detector class.
12 days ago henrik.lundin... Follow-up fixes for G722
13 days ago turaj@webrtc.orgRevert 7675 "Make an AudioEncoder subclass for iSAC"
13 days ago kwiberg@webrtc.orgMake an AudioEncoder subclass for iSAC
13 days ago henrike@webrtc.orgChange from talk/p2p (r7664) "(Auto)update libjingle...
13 days ago henrike@webrtc.orgChange from talk/p2p (r7572): "Improve the logging...
13 days ago andrew@webrtc.orgreplace inline assembly WebRtcAecm_ResetAdaptiveChannel...
13 days ago andrew@webrtc.orgclear asm code and unused functions in audio processing...
13 days ago henrike@webrtc.orgRemoves talk/xmllite, talk/xmpp and talk/p2p as they...
13 days ago pbos@webrtc.orgWire up DSCP support in WebRtcVideoEngine2.
13 days ago stefan@webrtc.orgPut send-side bwe probing under finch experiment.
13 days ago pbos@webrtc.orgRefactor SetDefaultEncoderConfig to work on existing...
13 days ago pbos@webrtc.orgAdd unit to dropped frames.
13 days ago kjellander... .gitignore updates
2014-11-07 buildbot@webrtc.org(Auto)update libjingle 79414100-> 79428003
2014-11-07 andresp@webrtc.orgEnable VP9 video codec support on webrtcvideoengine...
2014-11-07 henrik.lundin... Reapply "Advertise G722 as 8 kHz rather than 16 kHz""
2014-11-07 perkj@webrtc.orgChange dummy address to use 0.0.0.0 instead of ::
2014-11-07 pbos@webrtc.orgRemove partially defined WebRtcRTPHeader from Parse().
2014-11-07 pbos@webrtc.orgPrevent a lot of VideoSendStream reconfigures.
2014-11-07 andresp@webrtc.orgRefactor webrtcvideoengines to have the default list...
2014-11-06 henrika@webrtc.orgReland Volume buttons in AppRTCDemo should affect outpu...
2014-11-06 pkasting@chromium.orgUse uint16s for port numbers in webrtc/p2p/base.
2014-11-06 henrike@webrtc.orgFix WebRTC Win64 + BoringSSL build.
2014-11-06 henrika@webrtc.orgVolume buttons in AppRTCDemo should affect output audio...
2014-11-06 henrik.lundin... Revert "Advertise G722 as 8 kHz rather than 16 kHz"
2014-11-06 buildbot@webrtc.org(Auto)update libjingle 79326895-> 79329222
2014-11-06 henrika@webrtc.orgVolume buttons in AppRTCDemo should affect output audio...
2014-11-06 perkj@webrtc.orgRemove deprecated PeerConnection APIs.
2014-11-06 andresp@webrtc.orgRemoving unused method GetDefaultVideoEncoderConfig.
2014-11-06 pbos@webrtc.orgLog formatting fix for VideoEncoderConfig.
2014-11-06 buildbot@webrtc.org(Auto)update libjingle 79285346-> 79320771
2014-11-06 mcasas@webrtc.orgAppRTCDemoActivity: Add a config CheckBox for enabling...
2014-11-06 henrik.lundin... Advertise G722 as 8 kHz rather than 16 kHz
2014-11-06 kwiberg@webrtc.orgRemove the state_ member from AudioDecoder
2014-11-06 kjellander... Revert 7642 "Fix memcheck and dr memory after flakiness...
2014-11-06 kjellander... Fix memcheck and dr memory after flakiness dashboard...
2014-11-06 marpan@webrtc.orgExclude SendsAndReceivesVP9 for linux-memcheck.
2014-11-06 andrew@webrtc.orgChange DrMemory exclusion to match changed test name.
2014-11-06 marpan@webrtc.orgExclude SendsAndReceivesVP9 for WinDrMemory.
2014-11-06 marpan@webrtc.orgAdjust parameter in vp9 rate control test.
2014-11-06 marpan@webrtc.orgIncrease speed setting for VP9 (from 5 to 6) and re...
2014-11-05 tkchin@webrtc.orgThis fixes a small memory leak (found using Xcode/Instr...
2014-11-05 pbos@webrtc.orgRemove uses of build date/time.
2014-11-05 stefan@webrtc.orgWire up bandwidth stats to the new API and webrtcvideoe...
2014-11-05 buildbot@webrtc.org(Auto)update libjingle 79205306-> 79244016
2014-11-05 kjellander... Restore old behavior for Android in fileutils.cc
2014-11-05 kjellander... Roll chromium_revision d3db2ff..375f736
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