external/webrtc.git
2015-01-27 kjellander... Add PRESUBMIT check for GYP files including source... master
2015-01-27 kjellander... Roll chromium_revision 4664fe0..9070a80 (312733:313233)
2015-01-27 asapersson... Update StreamDataCounter with FEC bytes.
2015-01-27 bjornv@webrtc.orgAEC: Implements a new function for calculating delay...
2015-01-27 magjed@webrtc.orgReland of: "Implement elapsed time and capture start...
2015-01-26 glaznev@webrtc.orgSupport VP8 HW decoding on devices with Exynos codec.
2015-01-26 pkasting@chromium.orgFix bug in GetREDStatus(): it doesn't actually return...
2015-01-26 glaznev@webrtc.orgUpdate AppRTCDemo to use renamed GAE messages.
2015-01-26 andrew@webrtc.orgAdd an AudioRingBuffer class wrapper for the ring_buffe...
2015-01-26 pkasting@chromium.orgConsolidate anonymous namespace content and file-static...
2015-01-26 kjellander... Make it easier to use external libyuv + cleanup GYP...
2015-01-26 bjornv@webrtc.orgRefactor common_audio/vad: Removed usage of macro WEBRT...
2015-01-26 tommi@webrtc.orgMove ThreadChecker into rtc_base_approved.
2015-01-26 marpan@webrtc.orgEnable encoder multi-threading for VP9.
2015-01-26 kwiberg@webrtc.orgTemporarily revert r8147 ("Update base/scoped_ptr.h...
2015-01-26 henrik.lundin... Introduce rtc::CheckedDivExact
2015-01-26 kwiberg@webrtc.orgUpdate base/scoped_ptr.h from system_wrappers/interface...
2015-01-25 kjellander... Remove win_asan trybot from PRESUBMIT.py
2015-01-25 kjellander... Roll chromium_revision c086b4e..4664fe0 (312108:312733)
2015-01-23 tkchin@webrtc.orgRevert 8136 "Remove frame copy in ViEExternalRendererIm...
2015-01-23 tkchin@webrtc.orgRevert 8139 "Implement elapsed time and capture start...
2015-01-23 jiayl@webrtc.orgReland r7980:
2015-01-23 fdegans@chromium.orgChange a GYP reference to cpufeatures.gypi
2015-01-23 pbos@webrtc.orgImplement elapsed time and capture start NTP time estim...
2015-01-23 kjellander... Disable DtmfSenderTest.InsertDtmfWithCommaAsDelay due...
2015-01-23 minyue@webrtc.orgRe-allowing RED in voice engine.
2015-01-23 magjed@webrtc.orgRemove frame copy in ViEExternalRendererImpl::RenderFrame
2015-01-23 stefan@webrtc.orgSwitch to use range based loops in the BWE simulation...
2015-01-22 davidben@webrtc.orgLeave BIO_METHOD non-const.
2015-01-22 tommi@webrtc.orgChange GetStreamBySsrc to not copy StreamParams.
2015-01-22 jiayl@webrtc.orgFix a crash in AllocationSequence.
2015-01-22 kjellander... Revert 8125 "Modify some tests to never use DTX disable...
2015-01-22 jlmiller@webrtc.orgChange sprintf use in talk samples to snprintf
2015-01-22 jlmiller@webrtc.orgCorrect GetDriveType error handling.
2015-01-22 henrik.lundin... Modify some tests to never use DTX disable mode
2015-01-22 stefan@webrtc.orgIntegrate send-side BWE into simulation framework.
2015-01-22 asapersson... Split packets/bytes in StreamDataCounter into RtpPacket...
2015-01-22 stefan@webrtc.orgFix bug in thresholds for bitrate probing and adjust...
2015-01-22 henrik.lundin... Make iSAC SWB own its decoder
2015-01-22 jiayl@webrtc.orgFix a use-after-free when sending queued messages is...
2015-01-21 andrew@webrtc.orgFix an unitialized variable warning.
2015-01-21 kjellander... GN: Prepare to remove webrtc_base target
2015-01-21 aluebs@webrtc.orgRe-land "Support 48kHz in AEC"
2015-01-21 aluebs@webrtc.orgFix TransientDetectorTest in modules_unittests on Andro...
2015-01-21 minyue@webrtc.orgDisable AcmSenderBitExactnessOldApi.Opus_stereo_20ms_vo...
2015-01-21 asapersson... Change CreateOrGetReportBlockInformation to have one...
2015-01-21 pbos@webrtc.orgSimplify and guard access to WindowsRealTimeClock.
2015-01-21 tommi@webrtc.orgUpdate StatsReport and by extension StatsCollector...
2015-01-21 kjellander... Remove unnecessary dependencies from webrtc_all target.
2015-01-21 asapersson... Only report fraction of lost packets if report_block_st...
2015-01-21 asapersson... Indentation changes.
2015-01-21 braveyao@webrtc.orgCorrect the class name in peerconnection_jni.cc.
2015-01-20 jlmiller@webrtc.orgUpdate libjingle license statements at top of talk...
2015-01-20 tnakamura@webrtc.orgBump to version 41.
2015-01-20 minyue@webrtc.orgSetting Opus target application.
2015-01-20 kjellander... Move internal capture+render to build_with_chromium...
2015-01-20 kjellander... Roll chromium_revision a6eafec..c086b4e
2015-01-20 tina.legrand... Revert 8080 "Support 48kHz in AEC"
2015-01-20 kwiberg@webrtc.orgRemove webrtc/base/compile_assert.h
2015-01-20 changbin.shao... Cleanup for Rtp Rtcp API test.
2015-01-19 tommi@webrtc.orgUpdate StatsCollector's interface in preparation of...
2015-01-19 tommi@webrtc.orgRevert 8095 "Update StatsCollector's interface in prepa...
2015-01-19 tommi@webrtc.orgUpdate StatsCollector's interface in preparation of...
2015-01-19 stefan@webrtc.orgAdd UMA stats for tracking the time it takes to reach...
2015-01-19 phoglund@webrtc.orgFixing LD_LIBRARY_PATH, improving safety for libjingle...
2015-01-19 kjellander... Adding TRYSERVER_PROJECT to codereview.settings.
2015-01-19 kjellander... Add /talk/examples/androidtests/{bin,gen} to .gitignore.
2015-01-19 kjellander... Disable tests failing on Android ARM64 (Nexus9).
2015-01-19 sprang@webrtc.orgDisable WebRtcVideoMediaChannelSimulcastTest::Simulcast...
2015-01-19 tommi@webrtc.orgRemove unused private data member engine_id_
2015-01-17 pthatcher@webrtc.orgrelease the turn allocation by sending a refresh reques...
2015-01-16 decurtis@webrtc.orgRe-enable the messagequeue unittests. These were commen...
2015-01-16 stefan@webrtc.orgRevert r8076 "Add UMA stats for tracking the time it...
2015-01-16 andresp@webrtc.orgRemove unnecessary remote bitrate estimator build rule...
2015-01-15 decurtis@webrtc.orgAdd stats collection for the data channel.
2015-01-15 decurtis@webrtc.orgFixes reference counting problem when a TransportProxy...
2015-01-15 tkchin@webrtc.orgUpdate AppRTCDemo UI.
2015-01-15 aluebs@webrtc.orgSupport 48kHz in AEC
2015-01-15 guoweis@webrtc.orgFix a case where empty candidate id is used
2015-01-15 aluebs@webrtc.orgOnly adapt AGC when the desired signal is present
2015-01-15 stefan@webrtc.orgAdd UMA stats for tracking the time it takes to reach...
2015-01-15 pbos@webrtc.orgLog configs when creating video streams in Call.
2015-01-15 henrik.lundin... Remove dual stream functionality in ACM
2015-01-15 andresp@webrtc.orgClean unnecessary workaround for chromium import.
2015-01-15 asapersson... Add percentage of fec packets and recovered media packe...
2015-01-15 guoweis@webrtc.orgFix a case where empty candidate id is used
2015-01-15 andrew@webrtc.orgAdd WebRtcIsacfix_AllpassFilter2FixDec16Neon()'s intrin...
2015-01-15 mgraczyk@chromium.orgAdd beamforming to audioproc_float utility.
2015-01-15 andrew@webrtc.orgMove ring_buffer to common_audio.
2015-01-14 pthatcher@webrtc.orgAdd BundlePolicy to RTCConfiguration. Don't change...
2015-01-14 kjellander... Fix searching for DirectX SDK during GN build.
2015-01-14 pbos@webrtc.orgRemove WebRtcVideoEncoderFactory2.
2015-01-14 turaj@webrtc.orgRevert removing of compile_assert.h.
2015-01-14 kjellander... Exclude EndToEndTest.SendsAndReceivesH264 for Dr Memory.
2015-01-14 stefan@webrtc.orgImproved fairness simulation by starting the flows...
2015-01-14 pbos@webrtc.orgImplement SimulcastEncoderAdapter support.
2015-01-14 henrik.lundin... Remove dual stream functionality in VoiceEngine
2015-01-14 mflodman@webrtc.orgRemove RTX SSRC when deleting the default receive stream.
2015-01-14 kwiberg@webrtc.orgRemove COMPILE_ASSERT and use static_assert everywhere
2015-01-14 andresp@webrtc.orgMove system_wrappers.gyp files to the proper directory.
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