external/webrtc.git
9 hours ago henrike@webrtc.orglibjingle: stop building files in talk/base as they... master
12 hours ago fbarchard@google.comDisable warning 4702 which affects map, xlist and other...
19 hours ago fbarchard@google.comroll libyuv to r1038 from r1035 to add gyp define that...
19 hours ago buildbot@webrtc.org(Auto)update libjingle 72097588-> 72159069
21 hours ago solenberg@webrtc.orgRemove dependency on openssl for android, add dependenc...
22 hours ago andrew@webrtc.orgUse C functions in aec for MIPS
28 hours ago asapersson... Integrate rtcp packet class to rtcp receiver tests.
32 hours ago henrike@webrtc.orgmerge_libs.py: fixes Windows breakage: there should...
38 hours ago buildbot@webrtc.org(Auto)update libjingle 72016417-> 72097588
4 days ago pbos@webrtc.orgRemove a disabled test.
4 days ago pbos@webrtc.orgRemove clang-format rm_binaries.py DEPS entry.
4 days ago henrike@webrtc.orgwebrtc/base: FileModifyTime -> OlderThan as that's...
4 days ago sergeyu@chromium.orgFix compilation on windows with clang, indentation...
4 days ago pbos@webrtc.orgSet NACK/REMB when setting receive codecs.
4 days ago fgalligan@google.comRoll chromium 282879:285412.
4 days ago henrike@webrtc.orgRevert of 6778 "Refactor StatsCollector and associated...
4 days ago henrike@webrtc.orgFixes "argument list too long" problem on Linux by...
4 days ago turaj@webrtc.orgRemove timestamp retreival warning/error.
4 days ago sergeyu@chromium.orgRevert "Fix compilation on windows with clang, indentat...
4 days ago sergeyu@chromium.orgFix compilation on windows with clang, indentation...
5 days ago tommi@webrtc.orgRefactor StatsCollector and associated types.
5 days ago jiayl@webrtc.orgFix a crash in statscollector.cc caused by invoking...
5 days ago mflodman@webrtc.orgMake sure padding is sent on the first sending RTP...
5 days ago buildbot@webrtc.org(Auto)update libjingle 71829282-> 71834788
5 days ago henrike@webrtc.orgRe-revert of 6747 "Refactor StatsCollector and associat...
6 days ago buildbot@webrtc.org(Auto)update libjingle 71775619-> 71778545
6 days ago henrike@webrtc.orgRevert 6747 "Refactor StatsCollector and associated...
6 days ago henrike@webrtc.orgRevert 6766 "Temporarily add a default ctor to StatsRep...
6 days ago buildbot@webrtc.org(Auto)update libjingle 71766184-> 71775619
6 days ago buildbot@webrtc.org(Auto)update libjingle 71753329-> 71766184
6 days ago tommi@webrtc.orgTemporarily add a default ctor to StatsReport and make...
6 days ago pbos@webrtc.orgEnable SendAndReceive tests.
7 days ago stefan@webrtc.orgFix flaky ramp-up test.
7 days ago pbos@webrtc.orgRevert "(Auto)update libjingle 71675033-> 71726409"
7 days ago buildbot@webrtc.org(Auto)update libjingle 71726409-> 71726772
7 days ago buildbot@webrtc.org(Auto)update libjingle 71675033-> 71726409
7 days ago pbos@webrtc.orgImplement suspend-below-min-bitrate option.
7 days ago pbos@webrtc.orgWire up VideoOptions for payload-based padding.
7 days ago glaznev@webrtc.orgAdd VP8 video decoding hw acceleration support to Java...
7 days ago pbos@webrtc.orgImplement encoder options in WebRtcVideoEngine2.
7 days ago pbos@webrtc.orgRemove unused config.h and math.h includes.
8 days ago minyue@webrtc.orgThe lastest commit on this file was in
8 days ago pbos@webrtc.orgEnable ReceiveStreamReceivingByDefault test.
8 days ago andresp@webrtc.orgRemove no longer used SkipEncodingUnusedStreams.
8 days ago andresp@webrtc.orgRemove remains of WEBRTC_NO_STL.
8 days ago buildbot@webrtc.org(Auto)update libjingle 71599033-> 71605904
8 days ago buildbot@webrtc.org(Auto)update libjingle 71575585-> 71599033
8 days ago andrew@webrtc.orgMIPS optimizations for ISAC (patch #2)
9 days ago tommi@webrtc.orgDisable GetStatsForInvalidTrack while I rewrite it.
9 days ago tommi@webrtc.orgRefactor StatsCollector and associated types.
9 days ago tommi@webrtc.orgRevert 6745 "Refactor StatsCollector and associated...
9 days ago tommi@webrtc.orgRefactor StatsCollector and associated types.
9 days ago pbos@webrtc.orgCheck before send/receive rtp header extensions.
9 days ago pbos@webrtc.orgImplement Base::ConstrainNewCodec2.
11 days ago jiayl@webrtc.orgIgnore empty data in DataChannel::Send to match FF...
11 days ago buildbot@webrtc.org(Auto)update libjingle 71460499-> 71464449
11 days ago jiayl@webrtc.orgRevert "Reland r6707 with the fix for callclient.cc."
11 days ago buildbot@webrtc.org(Auto)update libjingle 71456344-> 71456420
11 days ago buildbot@webrtc.org(Auto)update libjingle 71456173-> 71456344
11 days ago jiayl@webrtc.orgReland r6707 with the fix for callclient.cc.
11 days ago minyue@webrtc.orgThis is to re-open an earlier CL
11 days ago buildbot@webrtc.org(Auto)update libjingle 71452608-> 71453580
11 days ago jiayl@webrtc.orgCreates the default track if the remote media content...
11 days ago tkchin@webrtc.orgRuntime guard for iOS7 property.
11 days ago tkchin@webrtc.orgFix crash in AudioDeviceUtilityIOS::~AudioDeviceUtilityIOS.
11 days ago pbos@webrtc.orgDisable GetStats on DrMemory.
12 days ago minyue@webrtc.orgThis is related to an earlier CL of enabling Opus 48...
12 days ago pbos@webrtc.orgInitial WebRtcVideoEngine2::GetStats().
12 days ago pbos@webrtc.orgSleep in ThreadTest thread functions.
12 days ago pbos@webrtc.orgRestart VideoReceiveStreams in WebRtcVideoEngine2.
12 days ago buildbot@webrtc.org(Auto)update libjingle 71378257-> 71410012
12 days ago kwiberg@webrtc.orgAudioBuffer: Optimize const accesses to arrays that...
12 days ago andrew@webrtc.orgReduce runtime of RingBufferTest by a factor of 100.
12 days ago wu@webrtc.orgUse _numMixedParticipants instead of audioFrameList...
12 days ago mallinath@webrtc.orgConnect to the turn server if address cannot be resolve...
12 days ago mallinath@webrtc.orgAssigning a priority to TURN server list passed to...
12 days ago jiayl@webrtc.orgfix
12 days ago stefan@webrtc.orgFix issue where padding is sent before media with undef...
13 days ago aluebs@webrtc.orgRemove unused ExperimentalNS API in AudioProcessing
13 days ago kwiberg@webrtc.orgAudioBuffer: Eliminate the SplitChannelBuffer class
13 days ago pbos@webrtc.orgMove additional state into WebRtcVideoSendStream.
13 days ago aluebs@webrtc.orgSimplify AudioBuffer::mixed_low_pass_data API
13 days ago kwiberg@webrtc.orgAudioBuffer: Let ChannelBuffer handle bounds checking...
13 days ago kwiberg@webrtc.orgAdd unit test for MediaFile WAV file writing
13 days ago tkchin@webrtc.orgFixes up rtc so that it compiles on iOS 8 SDK.
13 days ago wu@webrtc.orgRevert 6707 "Add support of multiple STUN servers in...
13 days ago minyue@webrtc.orgr6709 lacks a change in BUILD.gn
13 days ago minyue@webrtc.orgRaw packet loss rate reported by RTP_RTCP module may...
13 days ago wu@webrtc.orgMake sure b lines appear before all the a lines. Per...
13 days ago jiayl@webrtc.orgAdd support of multiple STUN servers in UDPPort.
13 days ago tkchin@webrtc.orgCompile-time guard for iOS7 specific property.
13 days ago buildbot@webrtc.org(Auto)update libjingle 71240799-> 71250251
2014-07-16 stefan@webrtc.orgPrint an info log instead of return an error if an...
2014-07-16 pbos@webrtc.orgRemove old padding path in RTPSender.
2014-07-16 kwiberg@webrtc.orgint16<->float conversions: Use size_t for array length...
2014-07-16 kwiberg@webrtc.orgDefine convenient FATAL_ERROR() and FATAL_ERROR_IF...
2014-07-16 kwiberg@webrtc.orgnrsh1 is written before tmp321 is read, so needs to...
2014-07-16 pbos@webrtc.orgImplement unittest for SetSendCodecsChangesExistingStreams.
2014-07-15 jiayl@webrtc.orgFix an invalid memory access due to typo in win/cursor.cc.
2014-07-15 tkchin@webrtc.orgAfter an audio interruption the audio unit no longer...
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