Log - b5a191bfe76af4f06eb8abe7d49ebffa9a9ccbdf - platform/external/webrtc

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
eda189b Remove redundant STR_CASE_CMP macro definitions. by andrew@webrtc.org · 12 years ago
a19c9f4 Updated WebRTC version to 3.41 by elham@webrtc.org · 12 years ago
021c42b Lock use of _packetRequestCallback in VCM. by pbos@webrtc.org · 12 years ago
7ebf0e7 Remove include_dirs from video_engine_core.gypi. by pbos@webrtc.org · 12 years ago
59f20bb Break out RTCPSender dependency on ModuleRtpRtcpImpl. by pbos@webrtc.org · 12 years ago
26b0d77 Suppress RTPSender race regardless of codec. by pbos@webrtc.org · 12 years ago
841c8a4 Rename VideoCall to Call. by pbos@webrtc.org · 12 years ago
86136a0 Re-enable tests for Remote Bitrate Estimator by solenberg@webrtc.org · 12 years ago
0181b5f ExternalVideoDecoder for new VideoEngine API. by pbos@webrtc.org · 12 years ago
30e055c Handle empty RTP video packets agnostic to codec. by pbos@webrtc.org · 12 years ago
1b476d9 Disabling channelmanager unittest. This test is causing by mallinath@webrtc.org · 12 years ago
ab5a091 Fixing the build error on Windows. Problem is in coversion from size_t to int. by mallinath@webrtc.org · 12 years ago
1b15f42 Update talk to 51960985. by mallinath@webrtc.org · 12 years ago
b159c2e Reduce cost of PushSincResampler::Resample(). by andrew@webrtc.org · 12 years ago
c7f7086 Clamp camera id to legal values. by fischman@webrtc.org · 12 years ago
b2c8a95 Improving padding rules and breaking out bw allocation to ViEEncoder. by stefan@webrtc.org · 12 years ago
7bb8f02 Adds support for combining RTX and FEC/RED. by stefan@webrtc.org · 12 years ago
5500d93 Add temporal layer factory. by andresp@webrtc.org · 12 years ago
016eec0 Unbreak build by adding new mandatory ICE username param. by fischman@webrtc.org · 12 years ago
f1e807c Removing FrameForStorage by mikhal@webrtc.org · 12 years ago
c31d4d0 AppRTCDemo(iOS): prefer ISAC as audio codec by fischman@webrtc.org · 12 years ago
aa3d1c8 Make unittest log printouts opt-in with a --logs flag. by andrew@webrtc.org · 12 years ago
bebf399 Pre-multiply images for MouseCursorShape. by alexeypa@chromium.org · 12 years ago
31b4a5a Recognize armv7 target_arch for ios support in webrtc common.gyp by fischman@webrtc.org · 12 years ago
be588f9 Apprtc Demo: calling createOffer/Answer without failureCallback is deprecated in FF by braveyao@webrtc.org · 12 years ago
9080518 Restore severity precondition to logging.h. by andrew@webrtc.org · 12 years ago
95e51f5 Remove send and receive streams when destroyed. by pbos@webrtc.org · 12 years ago
164c4f7 Add clockdrift to RtpGenerator by henrik.lundin@webrtc.org · 12 years ago