Log - 28631e76cf7bcb40532069eecfe65f6290c05cd4 - fp2-dev/platform/external/chromium_org/third_party/webrtc

28631e7 Refactor frame generation code so it can be used by multiple modules. by andresp@webrtc.org · 11 years ago
a89566f Disable NACK bandwidth statistics test due to being too flaky. by stefan@webrtc.org · 11 years ago
93b9912 Fixes a flake in network down tests. by stefan@webrtc.org · 11 years ago
032f731 Disable tests for TSan v2 by kjellander@webrtc.org · 11 years ago
4d08199 Compile ACM2 and ACM1. by turaj@webrtc.org · 11 years ago
80142aa Small refactoring of AudioProcessing use in channel.cc. by andrew@webrtc.org · 11 years ago
ab34f11 NetEq4: Making a few more members scoped_ptrs by henrik.lundin@webrtc.org · 11 years ago
05dd6c0 Dedicated speed test for NetEq3 by henrik.lundin@webrtc.org · 11 years ago
c61a170 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 · 11 years ago
ec09fcb Revert r4772 "Compile ACM1 and ACM2." by stefan@webrtc.org · 11 years ago
671d90b NetEq4: Make some DSP operation classes member variables by henrik.lundin@webrtc.org · 11 years ago
c2c8e6a Fix races in vcm::Process(). by stefan@webrtc.org · 11 years ago
1ddd57f Break out glue for old->new Transport. by pbos@webrtc.org · 11 years ago
5b7878f Changing 'frame' method to 'bounds' method. by sjlee@webrtc.org · 11 years ago
7556d2d Compile ACM1 and ACM2. by turaj@webrtc.org · 11 years ago
0c57671 Use the native sample rate for OpenSL recording. by henrike@webrtc.org · 11 years ago
39e22a1 Adds a new voice engine warning for the typing noise off state. by jiayl@webrtc.org · 11 years ago
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