3 hours ago mflodman@webrtc.orgMake sure padding is sent on the first sending RTP... master
3 hours ago buildbot@webrtc.org(Auto)update libjingle 71829282-> 71834788
5 hours ago henrike@webrtc.orgRe-revert of 6747 "Refactor StatsCollector and associat...
22 hours ago buildbot@webrtc.org(Auto)update libjingle 71775619-> 71778545
22 hours ago henrike@webrtc.orgRevert 6747 "Refactor StatsCollector and associated...
22 hours ago henrike@webrtc.orgRevert 6766 "Temporarily add a default ctor to StatsRep...
22 hours ago buildbot@webrtc.org(Auto)update libjingle 71766184-> 71775619
24 hours ago buildbot@webrtc.org(Auto)update libjingle 71753329-> 71766184
27 hours ago tommi@webrtc.orgTemporarily add a default ctor to StatsReport and make...
28 hours ago pbos@webrtc.orgEnable SendAndReceive tests.
33 hours ago stefan@webrtc.orgFix flaky ramp-up test.
36 hours ago pbos@webrtc.orgRevert "(Auto)update libjingle 71675033-> 71726409"
36 hours ago buildbot@webrtc.org(Auto)update libjingle 71726409-> 71726772
37 hours ago buildbot@webrtc.org(Auto)update libjingle 71675033-> 71726409
37 hours ago pbos@webrtc.orgImplement suspend-below-min-bitrate option.
37 hours ago pbos@webrtc.orgWire up VideoOptions for payload-based padding.
2 days ago glaznev@webrtc.orgAdd VP8 video decoding hw acceleration support to Java...
2 days ago pbos@webrtc.orgImplement encoder options in WebRtcVideoEngine2.
2 days ago pbos@webrtc.orgRemove unused config.h and math.h includes.
2 days ago minyue@webrtc.orgThe lastest commit on this file was in
2 days ago pbos@webrtc.orgEnable ReceiveStreamReceivingByDefault test.
2 days ago andresp@webrtc.orgRemove no longer used SkipEncodingUnusedStreams.
2 days ago andresp@webrtc.orgRemove remains of WEBRTC_NO_STL.
2 days ago buildbot@webrtc.org(Auto)update libjingle 71599033-> 71605904
2 days ago buildbot@webrtc.org(Auto)update libjingle 71575585-> 71599033
3 days ago andrew@webrtc.orgMIPS optimizations for ISAC (patch #2)
3 days ago tommi@webrtc.orgDisable GetStatsForInvalidTrack while I rewrite it.
3 days ago tommi@webrtc.orgRefactor StatsCollector and associated types.
3 days ago tommi@webrtc.orgRevert 6745 "Refactor StatsCollector and associated...
3 days ago tommi@webrtc.orgRefactor StatsCollector and associated types.
4 days ago pbos@webrtc.orgCheck before send/receive rtp header extensions.
4 days ago pbos@webrtc.orgImplement Base::ConstrainNewCodec2.
5 days ago jiayl@webrtc.orgIgnore empty data in DataChannel::Send to match FF...
5 days ago buildbot@webrtc.org(Auto)update libjingle 71460499-> 71464449
5 days ago jiayl@webrtc.orgRevert "Reland r6707 with the fix for callclient.cc."
5 days ago buildbot@webrtc.org(Auto)update libjingle 71456344-> 71456420
5 days ago buildbot@webrtc.org(Auto)update libjingle 71456173-> 71456344
5 days ago jiayl@webrtc.orgReland r6707 with the fix for callclient.cc.
5 days ago minyue@webrtc.orgThis is to re-open an earlier CL
5 days ago buildbot@webrtc.org(Auto)update libjingle 71452608-> 71453580
5 days ago jiayl@webrtc.orgCreates the default track if the remote media content...
6 days ago tkchin@webrtc.orgRuntime guard for iOS7 property.
6 days ago tkchin@webrtc.orgFix crash in AudioDeviceUtilityIOS::~AudioDeviceUtilityIOS.
6 days ago pbos@webrtc.orgDisable GetStats on DrMemory.
6 days ago minyue@webrtc.orgThis is related to an earlier CL of enabling Opus 48...
6 days ago pbos@webrtc.orgInitial WebRtcVideoEngine2::GetStats().
6 days ago pbos@webrtc.orgSleep in ThreadTest thread functions.
6 days ago pbos@webrtc.orgRestart VideoReceiveStreams in WebRtcVideoEngine2.
6 days ago buildbot@webrtc.org(Auto)update libjingle 71378257-> 71410012
6 days ago kwiberg@webrtc.orgAudioBuffer: Optimize const accesses to arrays that...
6 days ago andrew@webrtc.orgReduce runtime of RingBufferTest by a factor of 100.
6 days ago wu@webrtc.orgUse _numMixedParticipants instead of audioFrameList...
6 days ago mallinath@webrtc.orgConnect to the turn server if address cannot be resolve...
7 days ago mallinath@webrtc.orgAssigning a priority to TURN server list passed to...
7 days ago jiayl@webrtc.orgfix
7 days ago stefan@webrtc.orgFix issue where padding is sent before media with undef...
7 days ago aluebs@webrtc.orgRemove unused ExperimentalNS API in AudioProcessing
7 days ago kwiberg@webrtc.orgAudioBuffer: Eliminate the SplitChannelBuffer class
7 days ago pbos@webrtc.orgMove additional state into WebRtcVideoSendStream.
7 days ago aluebs@webrtc.orgSimplify AudioBuffer::mixed_low_pass_data API
7 days ago kwiberg@webrtc.orgAudioBuffer: Let ChannelBuffer handle bounds checking...
7 days ago kwiberg@webrtc.orgAdd unit test for MediaFile WAV file writing
7 days ago tkchin@webrtc.orgFixes up rtc so that it compiles on iOS 8 SDK.
7 days ago wu@webrtc.orgRevert 6707 "Add support of multiple STUN servers in...
7 days ago minyue@webrtc.orgr6709 lacks a change in BUILD.gn
7 days ago minyue@webrtc.orgRaw packet loss rate reported by RTP_RTCP module may...
7 days ago wu@webrtc.orgMake sure b lines appear before all the a lines. Per...
7 days ago jiayl@webrtc.orgAdd support of multiple STUN servers in UDPPort.
8 days ago tkchin@webrtc.orgCompile-time guard for iOS7 specific property.
8 days ago buildbot@webrtc.org(Auto)update libjingle 71240799-> 71250251
8 days ago stefan@webrtc.orgPrint an info log instead of return an error if an...
8 days ago pbos@webrtc.orgRemove old padding path in RTPSender.
8 days ago kwiberg@webrtc.orgint16<->float conversions: Use size_t for array length...
8 days ago kwiberg@webrtc.orgDefine convenient FATAL_ERROR() and FATAL_ERROR_IF...
8 days ago kwiberg@webrtc.orgnrsh1 is written before tmp321 is read, so needs to...
8 days ago pbos@webrtc.orgImplement unittest for SetSendCodecsChangesExistingStreams.
8 days ago jiayl@webrtc.orgFix an invalid memory access due to typo in win/cursor.cc.
8 days ago tkchin@webrtc.orgAfter an audio interruption the audio unit no longer...
9 days ago tommi@webrtc.orgMinor refactoring of StatsCollector.
9 days ago tkchin@webrtc.orgRemove Thread::RunningForChannelManager().
9 days ago stefan@webrtc.orgImprovements to the pacer where it lost some budget...
9 days ago pbos@webrtc.orgFix breakage introduced by r6691.
9 days ago pbos@webrtc.orgMake RTCP sender report send media bytes.
9 days ago kwiberg@webrtc.orgEliminate unnecessary #include
9 days ago kwiberg@webrtc.orgrtc::Fatal output: Print space between # and message
9 days ago pbos@webrtc.orgRemove the VPM denoiser.
9 days ago tommi@webrtc.orgHandle the case if an unusually long peer name is provi...
9 days ago pbos@webrtc.orgReplace strcpy with talk_base::strcpyn.
9 days ago fbarchard@google.comRoll libyuv from 1033 to 1035 to get cpuid fix for...
9 days ago fgalligan@google.comRoll chromium 282462:282879.
9 days ago henrike@webrtc.orgRebase webrtc/base with r6682 version of talk/base:
9 days ago henrike@webrtc.orgAdd a facility to the Thread class to catch blocking...
9 days ago tkchin@webrtc.orgEnable SCTP compile for iOS.
9 days ago buildbot@webrtc.org(Auto)update libjingle 71116846-> 71117224
9 days ago tommi@webrtc.orgAdd a facility to the Thread class to catch blocking...
9 days ago tommi@webrtc.orgA step towards changing StatsReport::Value::name to...
9 days ago tommi@webrtc.orgMake StatsCollector depend on always having a valid...
9 days ago tommi@webrtc.orgMinor refactoring of the session classes.
9 days ago buildbot@webrtc.org(Auto)update libjingle 71107853-> 71115715
10 days ago buildbot@webrtc.org(Auto)update libjingle 71099685-> 71107853