Log - 6ffc74ee0e28a2e490348dc2d58f4ee927db8ec4 - platform/external/webrtc

6ffc74e Disable flaky RunsRtpRtcpTestWithoutErrors. by andrew@webrtc.org · 12 years ago
eb524d9 Remove deprecated AudioCodingModule::Destroy. by andrew@webrtc.org · 12 years ago
1112c30 Update libjingle to 53057474. by mallinath@webrtc.org · 12 years ago
e2af622 - Reset capture deltas at resolution change. by asapersson@webrtc.org · 12 years ago
bec11ef Reformatting media_optimization.cc and .h by henrik.lundin@webrtc.org · 12 years ago
b533a82 Disabled flaky tests. BUG=2409 R=andrew@webrtc.org, mallinath@webrtc.org by asapersson@webrtc.org · 12 years ago
7a7b929 Updated dc1.html to support SCTP transport. by vikasmarwaha@webrtc.org · 12 years ago
334865e Re-enable VideoCaptureTest.CreateDelete by fischman@webrtc.org · 12 years ago
038e8e6 Updated WebRTC version to 3.42 by elham@webrtc.org · 12 years ago
98fcd2d Adding unit tests for default temporal layer strategy. by andresp@webrtc.org · 12 years ago
cdd3d4d Revert test change in r4808. by stefan@webrtc.org · 12 years ago
269dd42 Reduce flakiness in network down test. by stefan@webrtc.org · 12 years ago
63fe8e1 Enable SetInitialPlayoutDelay on Android. by dwkang@webrtc.org · 12 years ago
2edb642 Fix bugs in DesktopRegion::Subtract(). by sergeyu@chromium.org · 12 years ago
cee0dfb Made sure that DTLS/SRTP is set to false in apprtc demo when testing loopback. See crbug/294881 for details. by vikasmarwaha@webrtc.org · 12 years ago
10e6cc7 VAD changes ported to ACM2. by turaj@webrtc.org · 12 years ago
362a55e Address Windows 64-bits warnings. by turaj@webrtc.org · 12 years ago
0e63e76 Enable FEC for VideoSendStream. by pbos@webrtc.org · 12 years ago
9c74be7 Disable flaky video capture test. by stefan@webrtc.org · 12 years ago
4f3624d Avoid recursively taking critical section. by stefan@webrtc.org · 12 years ago
dd57cd6 Removing the tsan text exclusion since the tests should be passing now. by jiayl@webrtc.org · 12 years ago
d29ab4e Suppress SSL error strings on mac_asan to unbreak that build by fischman@webrtc.org · 12 years ago
76fe930 Use link_settings instead of all_dependent_settings to pacify xcode gyp generator by fischman@webrtc.org · 12 years ago
ccddd0a Roll webrtc's chromium_revision 217707:224141 by fischman@webrtc.org · 12 years ago
6917e19 Rename EngineTest to CallTest. by pbos@webrtc.org · 12 years ago
a03e34e Heap-use-after-free in WebRtcNetEQ_RecInRTPStruct by tina.legrand@webrtc.org · 12 years ago
ab65495 Refactor frame generation code so it can be used by multiple modules. by andresp@webrtc.org · 12 years ago
7a30dfd Disable NACK bandwidth statistics test due to being too flaky. by stefan@webrtc.org · 12 years ago
b5a191b Fixes a flake in network down tests. by stefan@webrtc.org · 12 years ago
d8a9b86 Disable tests for TSan v2 by kjellander@webrtc.org · 12 years ago
967bfff Update talk to 52534915. by wu@webrtc.org · 12 years ago
532f3dc Compile ACM2 and ACM1. by turaj@webrtc.org · 12 years ago
f3930e9 Small refactoring of AudioProcessing use in channel.cc. by andrew@webrtc.org · 12 years ago
0d5da25 NetEq4: Making a few more members scoped_ptrs by henrik.lundin@webrtc.org · 12 years ago
5a43370 Dedicated speed test for NetEq3 by henrik.lundin@webrtc.org · 12 years ago
7a968a8 Add more TSan and Dr Memory suppressions for modules_unittests by kjellander@webrtc.org · 12 years ago
8d1e4d6 Increase the dtmfsender test toleration to 100ms to avoid flaky. by wu@webrtc.org · 12 years ago
8bf755d MIPS optimizations for the functions WebRtcSpl_SqrtFloor, WebRtcSpl_CrossCorrelation, WebRtcSpl_ScaleAndAddVectorsWithRound and the inline functions from signal_processing spl_inl.h file. by andrew@webrtc.org · 12 years ago
5f10516 Fix disabling of tests. by stefan@webrtc.org · 12 years ago
1c77dfd Revert r4772 "Compile ACM1 and ACM2." by stefan@webrtc.org · 12 years ago
40d3fc6 NetEq4: Make some DSP operation classes member variables by henrik.lundin@webrtc.org · 12 years ago
8db81c5 Fix races in vcm::Process(). by stefan@webrtc.org · 12 years ago
e75a1bf Break out glue for old->new Transport. by pbos@webrtc.org · 12 years ago
fe84fda Changing 'frame' method to 'bounds' method. by sjlee@webrtc.org · 12 years ago
367baa6 Compile ACM1 and ACM2. by turaj@webrtc.org · 12 years ago
c8dea6a Use the native sample rate for OpenSL recording. by henrike@webrtc.org · 12 years ago
bf00740 Adds a new voice engine warning for the typing noise off state. by jiayl@webrtc.org · 12 years ago
da79008 Disabling crashing or flaky tests in peerconnection_unittest. by stefan@webrtc.org · 12 years ago
32d640e Fix typo in r4765. by pbos@webrtc.org · 12 years ago
da2c4ce Fix dangling pointer _encoder in video_sender.cc. by pbos@webrtc.org · 12 years ago
be63fd6 Initialize CodecInst structs in test_api_audio.cc. by pbos@webrtc.org · 12 years ago
d1fc5d4 Dedicated speed test for NetEq4 by henrik.lundin@webrtc.org · 12 years ago
28a331e Add support for multiple report blocks. by stefan@webrtc.org · 12 years ago
fc10c5c This is related to https://code.google.com/p/webrtc/issues/detail?id=1341 by sjlee@webrtc.org · 12 years ago
e6ac163 This is related to https://code.google.com/p/webrtc/issues/detail?id=846 by sjlee@webrtc.org · 12 years ago
c3e51ac To use the channel_transport on the iOS platform, some #if directives are changed. by sjlee@webrtc.org · 12 years ago
15e979b Call AllowCommandLineReparsing in unit tests. by andrew@webrtc.org · 12 years ago
b3af8ae Verify local and remote transport description before negotiation. by mallinath@webrtc.org · 12 years ago
f6ae62f Add Win TSan exclude and Dr Memory suppressions by kjellander@webrtc.org · 12 years ago
eddbfb8 Add more Dr Memory suppressions for common_audio_unittests by kjellander@webrtc.org · 12 years ago
e401c2e Split video coding module unit tests into sender and receiver unit tests. by andresp@webrtc.org · 12 years ago
ab800f6 Disable flaky libjingle tests under tsan and memcheck. by stefan@webrtc.org · 12 years ago
5860de0 Implement NACK over RTX for VideoSendStream. by pbos@webrtc.org · 12 years ago
8fa436b Remove use of vcm->ResetDecoder from modules/utility. by andresp@webrtc.org · 12 years ago
62b816a Fixed pylint warnings. by phoglund@webrtc.org · 12 years ago
15b8871 Allocate float_buffer_ in the initializer list. by andrew@webrtc.org · 12 years ago
8a14489 Disable WebRtcSessionTest.TestCreateOfferWithSctpEnabledWithoutStreams by sergeyu@chromium.org · 12 years ago
f7eb75b Split VideoCodingModuleImpl into VideoSender and VideoReceiver. by andresp@webrtc.org · 12 years ago
a59696b Update libjingle to 52300956 by sergeyu@chromium.org · 12 years ago
48af652 Prepare to compile ACM1 and ACM2. by turaj@webrtc.org · 12 years ago
bc189fb * Prefer to send ISAC on clank. by wu@webrtc.org · 12 years ago
6ab45b9 Implement DesktopRegion subtraction. by sergeyu@chromium.org · 12 years ago
1f09dbe Moving test-only code (stream_generator) out of vcm implemention. by andresp@webrtc.org · 12 years ago
2553450 Fix win trybot errors due to r4729. by andrew@webrtc.org · 12 years ago
6a5cc9d Fix crash in the window capturer on windows by sergeyu@chromium.org · 12 years ago
7959e16 ACM2 integration with NetEq 4. by turaj@webrtc.org · 12 years ago
82a846f Adding Ami to the video renderer and capturer modules. by mallinath@webrtc.org · 12 years ago
36cf4d2 The video render module for iOS. by fischman@webrtc.org · 12 years ago
e509f94 This issue is related to https://chromereviews.googleplex.com/9908014/ by minyue@webrtc.org · 12 years ago
8fa03a1 Make PCM16 available in Chromium builds. by andrew@webrtc.org · 12 years ago
89df092 Make the destructor of AudioCodingModule public. by andrew@webrtc.org · 12 years ago
5eb997a Fix unsigned/signed comparison error due to r4729. by andrew@webrtc.org · 12 years ago
8f94013 Reduce frequency of high audio delay warning logs. by andrew@webrtc.org · 12 years ago
256b831 Removes function that is not used anywhere but somehow still causing library load issues on Android Release build. by henrike@webrtc.org · 12 years ago
5c678ea Implement 'abs-send-time' extension in VideoSendStream. by pbos@webrtc.org · 12 years ago
6138c5c OpenSl: fixes crashes externally reported in issue 2361 and 2362. by henrike@webrtc.org · 12 years ago
036b743 Adding APIs. These APIs are not implemented yet, they are to help developement of ACM. by turaj@webrtc.org · 12 years ago
a80ee74 AppRTC: using a footer element instead of div#footer in CSS. by braveyao@webrtc.org · 12 years ago
d4d59ac Remove FrameForStorage:Follow up on r4688 by mikhal@webrtc.org · 12 years ago
2902328 Implement 'toffset' extension in VideoSendStream. by pbos@webrtc.org · 12 years ago
554d158 Reset jitter buffer and timing if frames are getting too much delay. by stefan@webrtc.org · 12 years ago
835ef67 Remove repeated conditions key. by andrew@webrtc.org · 12 years ago
82f014a OpenSL (not default): Enables low latency audio on Android. by henrike@webrtc.org · 12 years ago
6413409 Show the signaling state and ice connection state in AppRTC by hooking up the peerconnections .onsignalingstatechange and .oniceconnectionstatechange events. by braveyao@webrtc.org · 12 years ago
319c98d Fix format string in video_quality_analysis.cc. by pbos@webrtc.org · 12 years ago
182d025 Remove include_dirs from voice_engine.gyp. by pbos@webrtc.org · 12 years ago
df531a2 Test that VideoSendStream responds to NACK. by pbos@webrtc.org · 12 years ago
f880f86 Convert printing in video quality tests to Chromium's perf format. by kjellander@webrtc.org · 12 years ago
e07049f Lock RTPSender statistics. by pbos@webrtc.org · 12 years ago
744fbc7 Split up EngineTests and RampupTests. by pbos@webrtc.org · 12 years ago