external/webrtc.git
51 min ago kjellander... Remove win_asan trybot from PRESUBMIT.py master
60 min ago kjellander... Roll chromium_revision c086b4e..4664fe0 (312108:312733)
46 hours ago tkchin@webrtc.orgRevert 8136 "Remove frame copy in ViEExternalRendererIm...
47 hours ago tkchin@webrtc.orgRevert 8139 "Implement elapsed time and capture start...
2 days ago jiayl@webrtc.orgReland r7980:
2 days ago fdegans@chromium.orgChange a GYP reference to cpufeatures.gypi
2 days ago pbos@webrtc.orgImplement elapsed time and capture start NTP time estim...
2 days ago kjellander... Disable DtmfSenderTest.InsertDtmfWithCommaAsDelay due...
2 days ago minyue@webrtc.orgRe-allowing RED in voice engine.
2 days ago magjed@webrtc.orgRemove frame copy in ViEExternalRendererImpl::RenderFrame
2 days ago stefan@webrtc.orgSwitch to use range based loops in the BWE simulation...
2 days ago davidben@webrtc.orgLeave BIO_METHOD non-const.
2 days ago tommi@webrtc.orgChange GetStreamBySsrc to not copy StreamParams.
2 days ago jiayl@webrtc.orgFix a crash in AllocationSequence.
3 days ago kjellander... Revert 8125 "Modify some tests to never use DTX disable...
3 days ago jlmiller@webrtc.orgChange sprintf use in talk samples to snprintf
3 days ago jlmiller@webrtc.orgCorrect GetDriveType error handling.
3 days ago henrik.lundin... Modify some tests to never use DTX disable mode
3 days ago stefan@webrtc.orgIntegrate send-side BWE into simulation framework.
3 days ago asapersson... Split packets/bytes in StreamDataCounter into RtpPacket...
3 days ago stefan@webrtc.orgFix bug in thresholds for bitrate probing and adjust...
3 days ago henrik.lundin... Make iSAC SWB own its decoder
3 days ago jiayl@webrtc.orgFix a use-after-free when sending queued messages is...
3 days ago andrew@webrtc.orgFix an unitialized variable warning.
3 days ago kjellander... GN: Prepare to remove webrtc_base target
4 days ago aluebs@webrtc.orgRe-land "Support 48kHz in AEC"
4 days ago aluebs@webrtc.orgFix TransientDetectorTest in modules_unittests on Andro...
4 days ago minyue@webrtc.orgDisable AcmSenderBitExactnessOldApi.Opus_stereo_20ms_vo...
4 days ago asapersson... Change CreateOrGetReportBlockInformation to have one...
4 days ago pbos@webrtc.orgSimplify and guard access to WindowsRealTimeClock.
4 days ago tommi@webrtc.orgUpdate StatsReport and by extension StatsCollector...
4 days ago kjellander... Remove unnecessary dependencies from webrtc_all target.
4 days ago asapersson... Only report fraction of lost packets if report_block_st...
4 days ago asapersson... Indentation changes.
4 days ago braveyao@webrtc.orgCorrect the class name in peerconnection_jni.cc.
4 days ago jlmiller@webrtc.orgUpdate libjingle license statements at top of talk...
5 days ago tnakamura@webrtc.orgBump to version 41.
5 days ago minyue@webrtc.orgSetting Opus target application.
5 days ago kjellander... Move internal capture+render to build_with_chromium...
5 days ago kjellander... Roll chromium_revision a6eafec..c086b4e
5 days ago tina.legrand... Revert 8080 "Support 48kHz in AEC"
5 days ago kwiberg@webrtc.orgRemove webrtc/base/compile_assert.h
5 days ago changbin.shao... Cleanup for Rtp Rtcp API test.
5 days ago tommi@webrtc.orgUpdate StatsCollector's interface in preparation of...
6 days ago tommi@webrtc.orgRevert 8095 "Update StatsCollector's interface in prepa...
6 days ago tommi@webrtc.orgUpdate StatsCollector's interface in preparation of...
6 days ago stefan@webrtc.orgAdd UMA stats for tracking the time it takes to reach...
6 days ago phoglund@webrtc.orgFixing LD_LIBRARY_PATH, improving safety for libjingle...
6 days ago kjellander... Adding TRYSERVER_PROJECT to codereview.settings.
6 days ago kjellander... Add /talk/examples/androidtests/{bin,gen} to .gitignore.
6 days ago kjellander... Disable tests failing on Android ARM64 (Nexus9).
6 days ago sprang@webrtc.orgDisable WebRtcVideoMediaChannelSimulcastTest::Simulcast...
6 days ago tommi@webrtc.orgRemove unused private data member engine_id_
8 days ago pthatcher@webrtc.orgrelease the turn allocation by sending a refresh reques...
9 days ago decurtis@webrtc.orgRe-enable the messagequeue unittests. These were commen...
9 days ago stefan@webrtc.orgRevert r8076 "Add UMA stats for tracking the time it...
9 days ago andresp@webrtc.orgRemove unnecessary remote bitrate estimator build rule...
9 days ago decurtis@webrtc.orgAdd stats collection for the data channel.
9 days ago decurtis@webrtc.orgFixes reference counting problem when a TransportProxy...
9 days ago tkchin@webrtc.orgUpdate AppRTCDemo UI.
10 days ago aluebs@webrtc.orgSupport 48kHz in AEC
10 days ago guoweis@webrtc.orgFix a case where empty candidate id is used
10 days ago aluebs@webrtc.orgOnly adapt AGC when the desired signal is present
10 days ago stefan@webrtc.orgAdd UMA stats for tracking the time it takes to reach...
10 days ago pbos@webrtc.orgLog configs when creating video streams in Call.
10 days ago henrik.lundin... Remove dual stream functionality in ACM
10 days ago andresp@webrtc.orgClean unnecessary workaround for chromium import.
10 days ago asapersson... Add percentage of fec packets and recovered media packe...
10 days ago guoweis@webrtc.orgFix a case where empty candidate id is used
10 days ago andrew@webrtc.orgAdd WebRtcIsacfix_AllpassFilter2FixDec16Neon()'s intrin...
10 days ago mgraczyk@chromium.orgAdd beamforming to audioproc_float utility.
10 days ago andrew@webrtc.orgMove ring_buffer to common_audio.
10 days ago pthatcher@webrtc.orgAdd BundlePolicy to RTCConfiguration. Don't change...
10 days ago kjellander... Fix searching for DirectX SDK during GN build.
11 days ago pbos@webrtc.orgRemove WebRtcVideoEncoderFactory2.
11 days ago turaj@webrtc.orgRevert removing of compile_assert.h.
11 days ago kjellander... Exclude EndToEndTest.SendsAndReceivesH264 for Dr Memory.
11 days ago stefan@webrtc.orgImproved fairness simulation by starting the flows...
11 days ago pbos@webrtc.orgImplement SimulcastEncoderAdapter support.
11 days ago henrik.lundin... Remove dual stream functionality in VoiceEngine
11 days ago mflodman@webrtc.orgRemove RTX SSRC when deleting the default receive stream.
11 days ago kwiberg@webrtc.orgRemove COMPILE_ASSERT and use static_assert everywhere
11 days ago andresp@webrtc.orgMove system_wrappers.gyp files to the proper directory.
11 days ago kjellander... Add .classpath + talk/app/webrtc/androidtests to .gitignore
11 days ago pbos@webrtc.orgCombine RegKeyTests to prevent parallel execution.
11 days ago phoglund@webrtc.orgNo longer asserting in mocks, split first test case...
11 days ago kjellander... Roll chromium_revision 3dd2edf..a6eafec (310717:311223)
11 days ago mgraczyk@chromium.orgAlways copy processed audio to output buffer in Process...
11 days ago aluebs@webrtc.orgOptimize minimum delay in blocker
11 days ago kwiberg@webrtc.orgUnify the two copies of template_util.h
12 days ago pbos@webrtc.orgOnly return Rtx mode in RTXSendStatus().
12 days ago kwiberg@webrtc.orgUnify the two copies of compile_assert.h
12 days ago kjellander... Roll chromium_revision 271c6cc..3dd2edf (309333:310717)
12 days ago andrew@webrtc.orgRemove useless AudioProcessing::Create() overload.
12 days ago pkasting@chromium.orgUse int64_t more consistently for times, in particular...
12 days ago bjornv@webrtc.orgaudio_processing: Replaced macro WEBRTC_SPL_MUL_16_16...
12 days ago henrik.lundin... Partial revert of r7396
13 days ago glaznev@webrtc.orgAllow 720x1280 frames encoding on Android.
13 days ago pbos@webrtc.orgFix parallelizability in ApmTests.
13 days ago henrika@webrtc.orgUse Java based audio as default for WebRTC.
next