Log - 6ab45b9dab4611544c7f281da0d9a68f82621745 - platform/external/webrtc

6ab45b9 Implement DesktopRegion subtraction. by sergeyu@chromium.org · 11 years ago
1f09dbe Moving test-only code (stream_generator) out of vcm implemention. by andresp@webrtc.org · 11 years ago
2553450 Fix win trybot errors due to r4729. by andrew@webrtc.org · 11 years ago
6a5cc9d Fix crash in the window capturer on windows by sergeyu@chromium.org · 11 years ago
7959e16 ACM2 integration with NetEq 4. by turaj@webrtc.org · 11 years ago
82a846f Adding Ami to the video renderer and capturer modules. by mallinath@webrtc.org · 11 years ago
36cf4d2 The video render module for iOS. by fischman@webrtc.org · 11 years ago
e509f94 This issue is related to https://chromereviews.googleplex.com/9908014/ by minyue@webrtc.org · 11 years ago
8fa03a1 Make PCM16 available in Chromium builds. by andrew@webrtc.org · 11 years ago
89df092 Make the destructor of AudioCodingModule public. by andrew@webrtc.org · 11 years ago
5eb997a Fix unsigned/signed comparison error due to r4729. by andrew@webrtc.org · 11 years ago
8f94013 Reduce frequency of high audio delay warning logs. by andrew@webrtc.org · 11 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 · 11 years ago
5c678ea Implement 'abs-send-time' extension in VideoSendStream. by pbos@webrtc.org · 11 years ago
6138c5c OpenSl: fixes crashes externally reported in issue 2361 and 2362. by henrike@webrtc.org · 11 years ago
036b743 Adding APIs. These APIs are not implemented yet, they are to help developement of ACM. by turaj@webrtc.org · 11 years ago
a80ee74 AppRTC: using a footer element instead of div#footer in CSS. by braveyao@webrtc.org · 11 years ago
d4d59ac Remove FrameForStorage:Follow up on r4688 by mikhal@webrtc.org · 11 years ago
2902328 Implement 'toffset' extension in VideoSendStream. by pbos@webrtc.org · 11 years ago
554d158 Reset jitter buffer and timing if frames are getting too much delay. by stefan@webrtc.org · 11 years ago
835ef67 Remove repeated conditions key. by andrew@webrtc.org · 11 years ago
82f014a OpenSL (not default): Enables low latency audio on Android. by henrike@webrtc.org · 11 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 · 11 years ago
319c98d Fix format string in video_quality_analysis.cc. by pbos@webrtc.org · 11 years ago
182d025 Remove include_dirs from voice_engine.gyp. by pbos@webrtc.org · 11 years ago
df531a2 Test that VideoSendStream responds to NACK. by pbos@webrtc.org · 11 years ago
f880f86 Convert printing in video quality tests to Chromium's perf format. by kjellander@webrtc.org · 11 years ago
e07049f Lock RTPSender statistics. by pbos@webrtc.org · 11 years ago
744fbc7 Split up EngineTests and RampupTests. by pbos@webrtc.org · 11 years ago
eda189b Remove redundant STR_CASE_CMP macro definitions. by andrew@webrtc.org · 11 years ago
a19c9f4 Updated WebRTC version to 3.41 by elham@webrtc.org · 11 years ago
021c42b Lock use of _packetRequestCallback in VCM. by pbos@webrtc.org · 11 years ago
7ebf0e7 Remove include_dirs from video_engine_core.gypi. by pbos@webrtc.org · 11 years ago
59f20bb Break out RTCPSender dependency on ModuleRtpRtcpImpl. by pbos@webrtc.org · 11 years ago
26b0d77 Suppress RTPSender race regardless of codec. by pbos@webrtc.org · 11 years ago
841c8a4 Rename VideoCall to Call. by pbos@webrtc.org · 11 years ago
86136a0 Re-enable tests for Remote Bitrate Estimator by solenberg@webrtc.org · 11 years ago
0181b5f ExternalVideoDecoder for new VideoEngine API. by pbos@webrtc.org · 11 years ago
30e055c Handle empty RTP video packets agnostic to codec. by pbos@webrtc.org · 11 years ago
1b476d9 Disabling channelmanager unittest. This test is causing by mallinath@webrtc.org · 11 years ago
ab5a091 Fixing the build error on Windows. Problem is in coversion from size_t to int. by mallinath@webrtc.org · 11 years ago
1b15f42 Update talk to 51960985. by mallinath@webrtc.org · 11 years ago
b159c2e Reduce cost of PushSincResampler::Resample(). by andrew@webrtc.org · 11 years ago
c7f7086 Clamp camera id to legal values. by fischman@webrtc.org · 11 years ago
b2c8a95 Improving padding rules and breaking out bw allocation to ViEEncoder. by stefan@webrtc.org · 11 years ago
7bb8f02 Adds support for combining RTX and FEC/RED. by stefan@webrtc.org · 11 years ago
5500d93 Add temporal layer factory. by andresp@webrtc.org · 11 years ago
016eec0 Unbreak build by adding new mandatory ICE username param. by fischman@webrtc.org · 11 years ago
f1e807c Removing FrameForStorage by mikhal@webrtc.org · 11 years ago
c31d4d0 AppRTCDemo(iOS): prefer ISAC as audio codec by fischman@webrtc.org · 11 years ago
aa3d1c8 Make unittest log printouts opt-in with a --logs flag. by andrew@webrtc.org · 11 years ago
bebf399 Pre-multiply images for MouseCursorShape. by alexeypa@chromium.org · 11 years ago
31b4a5a Recognize armv7 target_arch for ios support in webrtc common.gyp by fischman@webrtc.org · 11 years ago
be588f9 Apprtc Demo: calling createOffer/Answer without failureCallback is deprecated in FF by braveyao@webrtc.org · 11 years ago
9080518 Restore severity precondition to logging.h. by andrew@webrtc.org · 11 years ago
95e51f5 Remove send and receive streams when destroyed. by pbos@webrtc.org · 11 years ago
164c4f7 Add clockdrift to RtpGenerator by henrik.lundin@webrtc.org · 11 years ago
7e1bf31 Allow unknown flags in test_main.cc. by pbos@webrtc.org · 11 years ago
36439bf NetEq4: Small change to reduce allocs in AudioMultiVector by henrik.lundin@webrtc.org · 11 years ago
e2d4da6 Enable EngineTest.ReceivesPliAndRecoversWithNack and fix memcheck suppression filter. by mflodman@webrtc.org · 11 years ago
be23b32 Adding tsan suppression for BUG 2349. by mflodman@webrtc.org · 11 years ago
77bf5c2 Clean capture timestamp code. by andresp@webrtc.org · 11 years ago
06f1f74 Disable EngineTest.ReceivesPliAndRecoversWithNack. by mflodman@webrtc.org · 11 years ago
b21e528 Protecting Bitrate to avoid data race found by tsan. by mflodman@webrtc.org · 11 years ago
65abb6b Revert 4671 "Enable SetInitialPlayoutDelay on Android." by mflodman@webrtc.org · 11 years ago
310ac91 Enable SetInitialPlayoutDelay on Android. by dwkang@webrtc.org · 11 years ago
3abb82d Suppress video engine test by mikhal@webrtc.org · 11 years ago
3c5a924 Don't force cont' when enabling kWithErrors by mikhal@webrtc.org · 11 years ago
635b2b8 Removing some TODO's from libyuv by mikhal@webrtc.org · 11 years ago
2b810bf Removing non decodable count from session info: This value isn't used, and therfore can be removed. This is a step towards the refactor of the session info to use maps. by mikhal@webrtc.org · 11 years ago
ccf8b56 AppRTCDemo(android): prefer ISAC for audio codec. by fischman@webrtc.org · 11 years ago
8788167 PeerConnection Java: explicitly cast DataChannel* to jlong for Java. by fischman@webrtc.org · 11 years ago
c8c3263 Remove JpegEncoder suppression as jpeg is now removed. by kjellander@webrtc.org · 11 years ago
f5f5da0 Adding TSAN suppression for test posix udp transport. by mflodman@webrtc.org · 11 years ago
3a6ff41 Document the source of test scenarios for Dummynet wrapper script. by kjellander@webrtc.org · 11 years ago
cac7325 Adding critsect for child_modules_ in ModuleRtpRtcpImpl::Process() to avoid race with ModuleRtpRtcp::RegisterChildModule. by mflodman@webrtc.org · 11 years ago
cb5118c Add FakeEncoder to VideoSendStream tests. by pbos@webrtc.org · 11 years ago
8fb8953 Correcting two nits in InputAudioFile by henrik.lundin@webrtc.org · 11 years ago
8d32066 Changed method name. by mflodman@webrtc.org · 11 years ago
814d5e9 Renamed method. by mflodman@webrtc.org · 11 years ago
d51bcff Function name change. by mflodman@webrtc.org · 11 years ago
dfbf52b Fixing capture frame race in ViECapturer. by mflodman@webrtc.org · 11 years ago
5aedb29 Add TSan and Dr Memory suppressions for Windows by kjellander@webrtc.org · 11 years ago
b3e905c Disable all LS_VERBOSE logging in NetEq4 by henrik.lundin@webrtc.org · 11 years ago
c487c6a NetEq4: Make the algorithm buffer a member variable by henrik.lundin@webrtc.org · 11 years ago
cadf904 Update talk to 51664136. by wu@webrtc.org · 11 years ago
a957570 Overuse detection based on capture-input jitter. by pbos@webrtc.org · 11 years ago
0b960cf Libjpeg is needed for Libyuv by mikhal@webrtc.org · 11 years ago
cf61bee Removing JPEG as it is not used. by mikhal@webrtc.org · 11 years ago
45d2840 Zero comfort noise for stereo insted of assertion. by turaj@webrtc.org · 11 years ago
3170b57 Reorder and add critical section to the public method NetEqImpl::PacketBufferStatistics(). by turaj@webrtc.org · 11 years ago
9ded07e Fix typo in InvertedDesktopFrame by sergeyu@chromium.org · 11 years ago
bfde359 Revert accidental checkin of DEPS by kjellander@webrtc.org · 11 years ago
c520fc9 Add svn:ignore on dirs that shouldn't be wiped during gclient revert by kjellander@webrtc.org · 11 years ago
de49966 Fix fileutils.cc for tests running under Win memory tools. by kjellander@webrtc.org · 11 years ago
f8c16b8 Disabling CondVarTest for TSan v2 (take 2) by kjellander@webrtc.org · 11 years ago
b295a3f Update SSRC in RtpRtcp for audio channel so that it can have NTP values for further AV sync. by dwkang@webrtc.org · 11 years ago
d730177 update neteq 4 to facilitate NACK by minyue@webrtc.org · 11 years ago
8ae641e Add suppressions file for Leak Sanitizer. by kjellander@webrtc.org · 11 years ago
5f8d05a Disabling CondVarTest for TSan v2. by kjellander@webrtc.org · 11 years ago