external/webrtc.git
13 hours ago mflodman@webrtc.orgRemoving giles@mozilla.com from WebRTC watchlist. master
18 hours ago pbos@webrtc.orgMake RelayServerTest use VirtualSocketServer.
20 hours ago braveyao@webrtc.orgUse a temporary buffer to scale a screencast in OnFrame...
3 days ago pthatcher@webrtc.orgMove Jingle-specific files from talk/session/media...
3 days ago tkchin@webrtc.orgFix mac video capture leak.
3 days ago tkchin@webrtc.orgAdd initWithCoder to RTCEAGLVideoView.
3 days ago aluebs@webrtc.orgWire up Beamformer in AudioProcessing
3 days ago stefan@webrtc.orgFix the ramp-up-down-up test which was using ts-offset...
3 days ago stefan@webrtc.orgRemove unneccessary lock causing a potential deadlock.
3 days ago jiayl@webrtc.orgAdd a AppRTCDemo setting to change the GAE server.
3 days ago pbos@webrtc.orgRemove the last getters from VideoReceiveStream stats.
3 days ago stefan@webrtc.orgEnable payload-based padding by default and remove...
3 days ago kwiberg@webrtc.orgUnify the two copies of move.h
3 days ago pbos@webrtc.orgRtp-Rtcp sender cleanup.
3 days ago kjellander... GN: Fix build for Mac
3 days ago stefan@webrtc.orgMove updating nack bitrate inside UpdateNACKBitRate.
3 days ago pthatcher@webrtc.orgBreakup Transports and TransportParsers and move Transp...
4 days ago aluebs@webrtc.orgMerge beamformer
4 days ago andrew@webrtc.orgRemove obsolete target_arch == armv7.
4 days ago pthatcher@webrtc.orgSplit up (Jingle)Session from BaseSession. This is...
4 days ago jiayl@webrtc.orgClean up the Channel code in AppRTCDemo and use GAE...
4 days ago pthatcher@webrtc.orgMove session/tunnel to webrtc/libjingle. This is part...
4 days ago asapersson... Store the received report blocks map (mapped per remote...
4 days ago pbos@webrtc.orgRefactor some receive-side stats.
4 days ago pbos@webrtc.orgGet avg_delay_ms from DecoderTiming callback.
4 days ago sprang@webrtc.orgSuppress REMB in bitrate ctrl if it seems lika a short...
4 days ago pbos@webrtc.orgRemove _t from function pointer typedefs.
4 days ago henrik.lundin... Make an AudioEncoder subclass for iSAC redundant encoding
4 days ago pbos@webrtc.orgRename rtpDumpPktHdr_t to RtpDumpPacketHeader.
4 days ago pbos@webrtc.orgRename external_hmac_ctx_t to ExternalHmacContext.
4 days ago pbos@webrtc.orgRename _t struct types in audio_processing.
4 days ago henrik.lundin... Fixing the memory leak in AudioEncoderCopyRedDeathTest...
4 days ago guoweis@webrtc.orgWorkaround for issue 3927 to allow localhost IP even...
4 days ago pthatcher@webrtc.orgRevert "Split up (Jingle)Session from BaseSession....
5 days ago pthatcher@webrtc.orgSplit up (Jingle)Session from BaseSession. This is...
5 days ago guoweis@webrtc.orgFix an assert failure caused by race condition
5 days ago andrew@webrtc.orgMake safe_conversions suitable for rtc_base_approved.
5 days ago pthatcher@webrtc.orgMove jingle examples from talk/ into webrtc/libjingle...
5 days ago guoweis@webrtc.orgMove VirtualSocket into the .h file to allow unit tests...
5 days ago aluebs@webrtc.orgSupport block_size greater than chunk_size in Blocker
5 days ago pbos@webrtc.orgRename _t struct types in audio_coding.
5 days ago tommi@webrtc.orgChange MockStatsObserver to grab values inside of OnCom...
5 days ago pbos@webrtc.orgRemove or rename typedefs with _t prefixes.
5 days ago tommi@webrtc.orgAdd a little utility to capture cpu graphs.
5 days ago sprang@webrtc.orgAdd overshoot of target bitrate for screenshare with...
5 days ago asapersson... Change aggregated fraction loss to be calculated from...
5 days ago kwiberg@webrtc.orgEnable the iSACfix AudioDecoder test (and make it work...
5 days ago braveyao@webrtc.orgIf one of the bundled content is missing in SDP, return...
6 days ago guoweis@webrtc.orgAdd adapter_type into Candidate object.
6 days ago andrew@webrtc.orgFix path to mock_agc.h
6 days ago pthatcher@webrtc.orgRevert "Split up (Jingle)Session from BaseSession....
6 days ago pthatcher@webrtc.orgSplit up (Jingle)Session from BaseSession. This is...
6 days ago pthatcher@webrtc.orgMove ViewRequest and MediaStreams to streamparams.h...
6 days ago henrik.lundin... Disable AudioEncoderCopyRedDeathTest.NullSpeechEncoder
6 days ago pthatcher@webrtc.orgMake one OWNERS files for all of webrtc/libjingle so...
6 days ago andrew@webrtc.orgAdd a manageable command-line tool for AudioProcessing.
6 days ago aluebs@webrtc.orgAdd 48kHz support to AGC
6 days ago andrew@webrtc.orgAdd (safe) uint32_t cast to fix Win64 build.
6 days ago andrew@webrtc.orgHandle all permissible PCM fields with WavReader.
6 days ago pbos@webrtc.orgAdd AGC manager tests.
6 days ago henrik.lundin... Make an AudioEncoder subclass for RED
6 days ago kwiberg@webrtc.orgAudioEncoder subclass for iSACfix
6 days ago kjellander... Cleanup: Remove 'const' qualifier from OnReceivedEstima...
6 days ago asapersson... Add field to counters for when first rtp/rtcp packet...
6 days ago bjornv@webrtc.orgaudio_processing: Moved legacy AGC code to webrtc/modul...
6 days ago guoweis@webrtc.orgRevert "Add adapter_type into Candidate object."
7 days ago marpan@webrtc.orgFix vp9 setting in vie loopback test.
7 days ago guoweis@webrtc.orgAdd adapter_type into Candidate object.
7 days ago pkasting@chromium.orgUse int64_t for milliseconds more often, primarily...
7 days ago aluebs@webrtc.orgRemove 20ms support in AGC
7 days ago guoweis@webrtc.orgReenable test case P2PTransportChannelTest.TestIPv6Conn...
7 days ago pbos@webrtc.orgMerge in AGC manager and AGC tools.
7 days ago bjornv@webrtc.orgRemoves unused test files by audio_processing/transient
7 days ago bjornv@webrtc.orgresources/audio_processing: Removed unused test files
7 days ago minyue@webrtc.orgSuppressing warnings in GetRTT() in VoE.
7 days ago tommi@webrtc.orgClean up StatsObserver's OnComplete methods (address...
7 days ago pbos@webrtc.orgUse webrtc_root instead of DEPTH for iSAC.
7 days ago buildbot@webrtc.org(Auto)update libjingle 82121498-> 82126219
7 days ago tommi@webrtc.orgRemove unneeded ctor and add a more practical one
7 days ago tommi@webrtc.orgAdd thread asserts to StatsCollector.
7 days ago pbos@webrtc.orgMerge audio_processing changes.
7 days ago pbos@webrtc.orgRevert r7885.
7 days ago andrew@webrtc.orgAdd WebRtcIsacfix_FilterMaLoopNeon's intrinsics version.
7 days ago pbos@webrtc.orgRevert r7886:7887.
7 days ago andrew@webrtc.orgAdd NEON intrinsics version for min_max_operations_neon.c
8 days ago magjed@webrtc.orgMove WebRtcVideoRenderFrame from webrtcvideoengine2...
10 days ago pthatcher@webrtc.orgPut pseudotcp back because remoting uses it.
10 days ago pthatcher@webrtc.orgMove the obvious/easy Jingle-specific code into webrtc...
10 days ago guoweis@webrtc.orgAdd adapter_type into Candidate object.
10 days ago tommi@webrtc.orgSwitch kStatsValueName* constants to be enums instead...
10 days ago henrik.lundin... Moving encoded_bytes into EncodedInfo
10 days ago kjellander... Fix webrtc gn windows build.
10 days ago jansson@webrtc.orgRemoving manual test pages because they have been moved...
10 days ago pthatcher@webrtc.orgCleanup little things found when refactoring.
11 days ago aluebs@webrtc.orgMove the downmixing out of AudioBuffer
11 days ago minyue@webrtc.orgAdding DTX to WebRTC Opus wrapper (relanding).
11 days ago pbos@webrtc.orgMerge AEC changes.
11 days ago pbos@webrtc.orgWire up RTT statistics to webrtc::Call.
11 days ago pbos@webrtc.orgRemove old_factory from WebRtcVideoEngine.
11 days ago perkj@webrtc.orgRevert "Revert 7826 "Change Android PeerConnectionUnitt...
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