Log - 4bb33620d4ec46400b0ed0a2eac5810ab16775e3 - fp2-dev/platform/external/chromium_org/third_party/webrtc

4bb3362 Disable tests for TSan v2 by kjellander@webrtc.org · 11 years ago
edf08ee Compile ACM2 and ACM1. by turaj@webrtc.org · 11 years ago
bee99b1 Small refactoring of AudioProcessing use in channel.cc. by andrew@webrtc.org · 11 years ago
c12119c NetEq4: Making a few more members scoped_ptrs by henrik.lundin@webrtc.org · 11 years ago
b00b61d Dedicated speed test for NetEq3 by henrik.lundin@webrtc.org · 11 years ago
fccf64c 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
65a237a Revert r4772 "Compile ACM1 and ACM2." by stefan@webrtc.org · 11 years ago
7c41c3b NetEq4: Make some DSP operation classes member variables by henrik.lundin@webrtc.org · 11 years ago
3f9ebdb Fix races in vcm::Process(). by stefan@webrtc.org · 11 years ago
26d75f3 Break out glue for old->new Transport. by pbos@webrtc.org · 11 years ago
cfdf698 Changing 'frame' method to 'bounds' method. by sjlee@webrtc.org · 11 years ago
19c663b Compile ACM1 and ACM2. by turaj@webrtc.org · 11 years ago
0ae4638 Use the native sample rate for OpenSL recording. by henrike@webrtc.org · 11 years ago
e30fde1 Adds a new voice engine warning for the typing noise off state. by jiayl@webrtc.org · 11 years ago
7901868 Fix typo in r4765. by pbos@webrtc.org · 11 years ago
5777a0a Fix dangling pointer _encoder in video_sender.cc. by pbos@webrtc.org · 11 years ago
c9b400c Initialize CodecInst structs in test_api_audio.cc. by pbos@webrtc.org · 11 years ago
ed8ce36 Dedicated speed test for NetEq4 by henrik.lundin@webrtc.org · 11 years ago
0c0f882 Add support for multiple report blocks. by stefan@webrtc.org · 11 years ago
bc90ee3 This is related to https://code.google.com/p/webrtc/issues/detail?id=1341 by sjlee@webrtc.org · 11 years ago
1a8c9b3 This is related to https://code.google.com/p/webrtc/issues/detail?id=846 by sjlee@webrtc.org · 11 years ago
29fce82 To use the channel_transport on the iOS platform, some #if directives are changed. by sjlee@webrtc.org · 11 years ago
e8eaed8 Call AllowCommandLineReparsing in unit tests. by andrew@webrtc.org · 11 years ago
0180fc4 Split video coding module unit tests into sender and receiver unit tests. by andresp@webrtc.org · 11 years ago
f952fce Implement NACK over RTX for VideoSendStream. by pbos@webrtc.org · 11 years ago
3b6ab4a Remove use of vcm->ResetDecoder from modules/utility. by andresp@webrtc.org · 11 years ago
91b0d23 Allocate float_buffer_ in the initializer list. by andrew@webrtc.org · 11 years ago
f458c43 Split VideoCodingModuleImpl into VideoSender and VideoReceiver. by andresp@webrtc.org · 11 years ago
e125ca7 Prepare to compile ACM1 and ACM2. by turaj@webrtc.org · 11 years ago
cda8e61 Implement DesktopRegion subtraction. by sergeyu@chromium.org · 11 years ago
564ba1e Moving test-only code (stream_generator) out of vcm implemention. by andresp@webrtc.org · 11 years ago
1d8ceab Fix win trybot errors due to r4729. by andrew@webrtc.org · 11 years ago
985848d Fix crash in the window capturer on windows by sergeyu@chromium.org · 11 years ago
242b8a5 ACM2 integration with NetEq 4. by turaj@webrtc.org · 11 years ago
54164d5 Adding Ami to the video renderer and capturer modules. by mallinath@webrtc.org · 11 years ago
4d57e48 The video render module for iOS. by fischman@webrtc.org · 11 years ago
1963a68 This issue is related to https://chromereviews.googleplex.com/9908014/ by minyue@webrtc.org · 11 years ago
a5b7b8c Make PCM16 available in Chromium builds. by andrew@webrtc.org · 11 years ago
40bd492 Make the destructor of AudioCodingModule public. by andrew@webrtc.org · 11 years ago
93da8cb Fix unsigned/signed comparison error due to r4729. by andrew@webrtc.org · 11 years ago
e45a8a8 Reduce frequency of high audio delay warning logs. by andrew@webrtc.org · 11 years ago
9d775a6 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
e22b761 Implement 'abs-send-time' extension in VideoSendStream. by pbos@webrtc.org · 11 years ago
75e7cff OpenSl: fixes crashes externally reported in issue 2361 and 2362. by henrike@webrtc.org · 11 years ago
811e4c9 Adding APIs. These APIs are not implemented yet, they are to help developement of ACM. by turaj@webrtc.org · 11 years ago
7efd262 Remove FrameForStorage:Follow up on r4688 by mikhal@webrtc.org · 11 years ago
905cebd Implement 'toffset' extension in VideoSendStream. by pbos@webrtc.org · 11 years ago
910520a Reset jitter buffer and timing if frames are getting too much delay. by stefan@webrtc.org · 11 years ago
96da891 Remove repeated conditions key. by andrew@webrtc.org · 11 years ago
3965d1f OpenSL (not default): Enables low latency audio on Android. by henrike@webrtc.org · 11 years ago
4c94668 Fix format string in video_quality_analysis.cc. by pbos@webrtc.org · 11 years ago
d1deeb6 Remove include_dirs from voice_engine.gyp. by pbos@webrtc.org · 11 years ago
af73083 Test that VideoSendStream responds to NACK. by pbos@webrtc.org · 11 years ago
b1b278e Convert printing in video quality tests to Chromium's perf format. by kjellander@webrtc.org · 11 years ago
0a477d1 Lock RTPSender statistics. by pbos@webrtc.org · 11 years ago
4d1cb14 Split up EngineTests and RampupTests. by pbos@webrtc.org · 11 years ago
81c4d24 Remove redundant STR_CASE_CMP macro definitions. by andrew@webrtc.org · 11 years ago
98691c2 Updated WebRTC version to 3.41 by elham@webrtc.org · 11 years ago
0313e5b Lock use of _packetRequestCallback in VCM. by pbos@webrtc.org · 11 years ago
462460f Remove include_dirs from video_engine_core.gypi. by pbos@webrtc.org · 11 years ago
38ba534 Break out RTCPSender dependency on ModuleRtpRtcpImpl. by pbos@webrtc.org · 11 years ago
fdc4352 Rename VideoCall to Call. by pbos@webrtc.org · 11 years ago
9c843fd Re-enable tests for Remote Bitrate Estimator by solenberg@webrtc.org · 11 years ago
0ee03f9 ExternalVideoDecoder for new VideoEngine API. by pbos@webrtc.org · 11 years ago
b8aa042 Handle empty RTP video packets agnostic to codec. by pbos@webrtc.org · 11 years ago
d44ec1c Reduce cost of PushSincResampler::Resample(). by andrew@webrtc.org · 11 years ago
252b16f Clamp camera id to legal values. by fischman@webrtc.org · 11 years ago
5ee7139 Improving padding rules and breaking out bw allocation to ViEEncoder. by stefan@webrtc.org · 11 years ago
5632a64 Adds support for combining RTX and FEC/RED. by stefan@webrtc.org · 11 years ago
b49897c Add temporal layer factory. by andresp@webrtc.org · 11 years ago
4a4d15b Removing FrameForStorage by mikhal@webrtc.org · 11 years ago
1e88712 Make unittest log printouts opt-in with a --logs flag. by andrew@webrtc.org · 11 years ago
5a196e6 Pre-multiply images for MouseCursorShape. by alexeypa@chromium.org · 11 years ago
7ac916b Recognize armv7 target_arch for ios support in webrtc common.gyp by fischman@webrtc.org · 11 years ago
744235e Restore severity precondition to logging.h. by andrew@webrtc.org · 11 years ago
6da93db Remove send and receive streams when destroyed. by pbos@webrtc.org · 11 years ago
cade3c3 Add clockdrift to RtpGenerator by henrik.lundin@webrtc.org · 11 years ago
882b499 Allow unknown flags in test_main.cc. by pbos@webrtc.org · 11 years ago
5b5cf3c NetEq4: Small change to reduce allocs in AudioMultiVector by henrik.lundin@webrtc.org · 11 years ago
934ddca Enable EngineTest.ReceivesPliAndRecoversWithNack and fix memcheck suppression filter. by mflodman@webrtc.org · 11 years ago
9d1d4b1 Clean capture timestamp code. by andresp@webrtc.org · 11 years ago
9333ee7 Disable EngineTest.ReceivesPliAndRecoversWithNack. by mflodman@webrtc.org · 11 years ago
d7e6388 Protecting Bitrate to avoid data race found by tsan. by mflodman@webrtc.org · 11 years ago
b1af9a8 Revert 4671 "Enable SetInitialPlayoutDelay on Android." by mflodman@webrtc.org · 11 years ago
8f34f73 Enable SetInitialPlayoutDelay on Android. by dwkang@webrtc.org · 11 years ago
179fc03 Don't force cont' when enabling kWithErrors by mikhal@webrtc.org · 11 years ago
1a4a552 Removing some TODO's from libyuv by mikhal@webrtc.org · 11 years ago
29befe4 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
0e84525 Adding critsect for child_modules_ in ModuleRtpRtcpImpl::Process() to avoid race with ModuleRtpRtcp::RegisterChildModule. by mflodman@webrtc.org · 11 years ago
43ec357 Add FakeEncoder to VideoSendStream tests. by pbos@webrtc.org · 11 years ago
777e192 Correcting two nits in InputAudioFile by henrik.lundin@webrtc.org · 11 years ago
c88d905 Changed method name. by mflodman@webrtc.org · 11 years ago
81ef3b8 Renamed method. by mflodman@webrtc.org · 11 years ago
a911872 Function name change. by mflodman@webrtc.org · 11 years ago
b7f97fc Fixing capture frame race in ViECapturer. by mflodman@webrtc.org · 11 years ago
efdafa9 Disable all LS_VERBOSE logging in NetEq4 by henrik.lundin@webrtc.org · 11 years ago
0cb9df1 NetEq4: Make the algorithm buffer a member variable by henrik.lundin@webrtc.org · 11 years ago
ee42c34 Overuse detection based on capture-input jitter. by pbos@webrtc.org · 11 years ago
fd74f30 Removing JPEG as it is not used. by mikhal@webrtc.org · 11 years ago
7a776d2 Zero comfort noise for stereo insted of assertion. by turaj@webrtc.org · 11 years ago