Log - 499392ce105faca0d368e8512e287e81f9392950 - fp2-dev/platform/external/chromium_org/third_party/webrtc

0277aa4 Fix typo in r4765. by pbos@webrtc.org · 11 years ago
54bc776 Fix dangling pointer _encoder in video_sender.cc. by pbos@webrtc.org · 11 years ago
64b5c61 Initialize CodecInst structs in test_api_audio.cc. by pbos@webrtc.org · 11 years ago
79d3355 Dedicated speed test for NetEq4 by henrik.lundin@webrtc.org · 11 years ago
7485573 Add support for multiple report blocks. by stefan@webrtc.org · 11 years ago
e9d2898 This is related to https://code.google.com/p/webrtc/issues/detail?id=1341 by sjlee@webrtc.org · 11 years ago
e3a12da This is related to https://code.google.com/p/webrtc/issues/detail?id=846 by sjlee@webrtc.org · 11 years ago
d8a5b00 To use the channel_transport on the iOS platform, some #if directives are changed. by sjlee@webrtc.org · 11 years ago
b0fb1d6 Call AllowCommandLineReparsing in unit tests. by andrew@webrtc.org · 11 years ago
e8fdc9d Split video coding module unit tests into sender and receiver unit tests. by andresp@webrtc.org · 11 years ago
041d54b Implement NACK over RTX for VideoSendStream. by pbos@webrtc.org · 11 years ago
36c3652 Remove use of vcm->ResetDecoder from modules/utility. by andresp@webrtc.org · 11 years ago
a4bbaa6 Allocate float_buffer_ in the initializer list. by andrew@webrtc.org · 11 years ago
42a65a2 Split VideoCodingModuleImpl into VideoSender and VideoReceiver. by andresp@webrtc.org · 11 years ago
ed0b4fb Prepare to compile ACM1 and ACM2. by turaj@webrtc.org · 11 years ago
26251da Implement DesktopRegion subtraction. by sergeyu@chromium.org · 11 years ago
a26a7f6 Moving test-only code (stream_generator) out of vcm implemention. by andresp@webrtc.org · 11 years ago
388d16c Fix win trybot errors due to r4729. by andrew@webrtc.org · 11 years ago
d0737d9 Fix crash in the window capturer on windows by sergeyu@chromium.org · 11 years ago
3f39c00 ACM2 integration with NetEq 4. by turaj@webrtc.org · 11 years ago
a3351c4 Adding Ami to the video renderer and capturer modules. by mallinath@webrtc.org · 11 years ago
bc375b5 The video render module for iOS. by fischman@webrtc.org · 11 years ago
4489c51 This issue is related to https://chromereviews.googleplex.com/9908014/ by minyue@webrtc.org · 11 years ago
66bfae2 Make PCM16 available in Chromium builds. by andrew@webrtc.org · 11 years ago
5e3379e Make the destructor of AudioCodingModule public. by andrew@webrtc.org · 11 years ago
0fd885e Fix unsigned/signed comparison error due to r4729. by andrew@webrtc.org · 11 years ago
f5556f2 Reduce frequency of high audio delay warning logs. by andrew@webrtc.org · 11 years ago
9fea95a 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
bfad17e Implement 'abs-send-time' extension in VideoSendStream. by pbos@webrtc.org · 11 years ago
8fdce8e OpenSl: fixes crashes externally reported in issue 2361 and 2362. by henrike@webrtc.org · 11 years ago
66dbbd9 Adding APIs. These APIs are not implemented yet, they are to help developement of ACM. by turaj@webrtc.org · 11 years ago
f2982c9 Remove FrameForStorage:Follow up on r4688 by mikhal@webrtc.org · 11 years ago
990c5e3 Implement 'toffset' extension in VideoSendStream. by pbos@webrtc.org · 11 years ago
f0adedc Reset jitter buffer and timing if frames are getting too much delay. by stefan@webrtc.org · 11 years ago
054bc03 Remove repeated conditions key. by andrew@webrtc.org · 11 years ago
f46fff6 OpenSL (not default): Enables low latency audio on Android. by henrike@webrtc.org · 11 years ago
dadb2a1 Fix format string in video_quality_analysis.cc. by pbos@webrtc.org · 11 years ago
7b30ce3 Remove include_dirs from voice_engine.gyp. by pbos@webrtc.org · 11 years ago
eb2d9dd Test that VideoSendStream responds to NACK. by pbos@webrtc.org · 11 years ago
3524ade Convert printing in video quality tests to Chromium's perf format. by kjellander@webrtc.org · 11 years ago
b676ac7 Lock RTPSender statistics. by pbos@webrtc.org · 11 years ago
fa996f2 Split up EngineTests and RampupTests. by pbos@webrtc.org · 11 years ago
5cf83f4 Remove redundant STR_CASE_CMP macro definitions. by andrew@webrtc.org · 11 years ago
0920142 Updated WebRTC version to 3.41 by elham@webrtc.org · 11 years ago
6b4698e Lock use of _packetRequestCallback in VCM. by pbos@webrtc.org · 11 years ago
0245bee Remove include_dirs from video_engine_core.gypi. by pbos@webrtc.org · 11 years ago
4e7777b Break out RTCPSender dependency on ModuleRtpRtcpImpl. by pbos@webrtc.org · 11 years ago
bf6d572 Rename VideoCall to Call. by pbos@webrtc.org · 11 years ago
6a79c9f Re-enable tests for Remote Bitrate Estimator by solenberg@webrtc.org · 11 years ago
618a0ec ExternalVideoDecoder for new VideoEngine API. by pbos@webrtc.org · 11 years ago
e97b69f Handle empty RTP video packets agnostic to codec. by pbos@webrtc.org · 11 years ago
11a8868 Reduce cost of PushSincResampler::Resample(). by andrew@webrtc.org · 11 years ago
ca20f3d Clamp camera id to legal values. by fischman@webrtc.org · 11 years ago
7dc1790 Improving padding rules and breaking out bw allocation to ViEEncoder. by stefan@webrtc.org · 11 years ago
db74c61 Adds support for combining RTX and FEC/RED. by stefan@webrtc.org · 11 years ago
4014302 Add temporal layer factory. by andresp@webrtc.org · 11 years ago
31a8ce7 Removing FrameForStorage by mikhal@webrtc.org · 11 years ago
e41c6b2 Make unittest log printouts opt-in with a --logs flag. by andrew@webrtc.org · 11 years ago
f2ef20c Pre-multiply images for MouseCursorShape. by alexeypa@chromium.org · 11 years ago
6f458ed Recognize armv7 target_arch for ios support in webrtc common.gyp by fischman@webrtc.org · 11 years ago
06eaa54 Restore severity precondition to logging.h. by andrew@webrtc.org · 11 years ago
0020858 Remove send and receive streams when destroyed. by pbos@webrtc.org · 11 years ago
0e2cb29 Add clockdrift to RtpGenerator by henrik.lundin@webrtc.org · 11 years ago
4998966 Allow unknown flags in test_main.cc. by pbos@webrtc.org · 11 years ago
787364c NetEq4: Small change to reduce allocs in AudioMultiVector by henrik.lundin@webrtc.org · 11 years ago
c77dcb0 Enable EngineTest.ReceivesPliAndRecoversWithNack and fix memcheck suppression filter. by mflodman@webrtc.org · 11 years ago
7a21a64 Clean capture timestamp code. by andresp@webrtc.org · 11 years ago
1cd055c Disable EngineTest.ReceivesPliAndRecoversWithNack. by mflodman@webrtc.org · 11 years ago
00c95bf Protecting Bitrate to avoid data race found by tsan. by mflodman@webrtc.org · 11 years ago
0f62690 Revert 4671 "Enable SetInitialPlayoutDelay on Android." by mflodman@webrtc.org · 11 years ago
0fe8944 Enable SetInitialPlayoutDelay on Android. by dwkang@webrtc.org · 11 years ago
0987043 Don't force cont' when enabling kWithErrors by mikhal@webrtc.org · 11 years ago
9787291 Removing some TODO's from libyuv by mikhal@webrtc.org · 11 years ago
4dae3c6 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
1cc93a2 Adding critsect for child_modules_ in ModuleRtpRtcpImpl::Process() to avoid race with ModuleRtpRtcp::RegisterChildModule. by mflodman@webrtc.org · 11 years ago
9e70940 Add FakeEncoder to VideoSendStream tests. by pbos@webrtc.org · 11 years ago
1c9d3fe Correcting two nits in InputAudioFile by henrik.lundin@webrtc.org · 11 years ago
324a016 Changed method name. by mflodman@webrtc.org · 11 years ago
94ef274 Renamed method. by mflodman@webrtc.org · 11 years ago
710d2e1 Function name change. by mflodman@webrtc.org · 11 years ago
a594db2 Fixing capture frame race in ViECapturer. by mflodman@webrtc.org · 11 years ago
88a2327 Disable all LS_VERBOSE logging in NetEq4 by henrik.lundin@webrtc.org · 11 years ago
797eb64 NetEq4: Make the algorithm buffer a member variable by henrik.lundin@webrtc.org · 11 years ago
ce9de71 Overuse detection based on capture-input jitter. by pbos@webrtc.org · 11 years ago
42baf15 Removing JPEG as it is not used. by mikhal@webrtc.org · 11 years ago
1aa0938 Zero comfort noise for stereo insted of assertion. by turaj@webrtc.org · 11 years ago
b22fe00 Reorder and add critical section to the public method NetEqImpl::PacketBufferStatistics(). by turaj@webrtc.org · 11 years ago
d808778 Fix typo in InvertedDesktopFrame by sergeyu@chromium.org · 11 years ago
27103e5 Fix fileutils.cc for tests running under Win memory tools. by kjellander@webrtc.org · 11 years ago
e0e5a6a Disabling CondVarTest for TSan v2 (take 2) by kjellander@webrtc.org · 11 years ago
c766a74 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
42758b3 update neteq 4 to facilitate NACK by minyue@webrtc.org · 11 years ago
e879919 Fix metrics_unittests on Android. by kjellander@webrtc.org · 11 years ago
8c6633c Add isolate configuration for Android for all tests. by kjellander@webrtc.org · 11 years ago
584b688 Re-organizing ACM tests by tina.legrand@webrtc.org · 11 years ago
9b7bdee Revert r4562 by elham@webrtc.org · 11 years ago
fd8cc12 Fix image flipping for OpenGL-based screen capturer on Mac. by sergeyu@chromium.org · 11 years ago
e8acc7c Enable ObjC build by default and reenable 64-bit mac libjingle build by fischman@webrtc.org · 11 years ago
6203090 Updated WebRTC version to 3.40 by elham@webrtc.org · 11 years ago
d9416e6 VCM:Accounting for bounds when inserting packets. We currently receive indicators to the first and last packets of the frame, but not have any sanity to verify that all packets are indeed within the bounds (when available). This cl attempts to fix that, by mikhal@webrtc.org · 11 years ago