Log - 3de1b22f29dc4ae2d0e508d4152c2103c5e8ad1a - fp2-dev/platform/external/chromium_org/third_party/webrtc

3de1b22 Fix include of isolate.gypi by kjellander@webrtc.org · 11 years ago
c73a60e Android OpenSL: Fixes faulty assertion in jni-code. by henrike@webrtc.org · 11 years ago
15eda06 Remove ReturnTrace from DeregisterCallback(). by pbos@webrtc.org · 11 years ago
f9159ec Remove templatization of the AudioVector test by henrik.lundin@webrtc.org · 11 years ago
712d30f Workaround issue with stdin on Windows. by kjellander@webrtc.org · 11 years ago
d6c8fec APK for opensl loopback. by henrike@webrtc.org · 11 years ago
8ee2137 Implement TraceCallbacks in Call. by pbos@webrtc.org · 11 years ago
55312fe Piping AutoMuter interface through to ViE API by henrik.lundin@webrtc.org · 11 years ago
adba142 Added support for sending and receiving RTCP XR packets: by asapersson@webrtc.org · 11 years ago
c6c6c1d Stop timer in ~EventWindows(). by pbos@webrtc.org · 11 years ago
37ae275 Update sampling rate and number of channels of NetEq4 if decoder is changed. by turaj@webrtc.org · 11 years ago
be46543 Test multiple send/receive streams in Call. by pbos@webrtc.org · 11 years ago
7befa0c Remove include_dirs from utility. by pbos@webrtc.org · 11 years ago
d2ca96e PeerConnection(Android): enable tracing to logcat. by fischman@webrtc.org · 11 years ago
cf71152 Reset audio bufer if codec changes, b/10835525. by turaj@webrtc.org · 11 years ago
324dac6 Ensure adjusted "known delay" doesn't drop below zero. by andrew@webrtc.org · 11 years ago
80f29d8 NetEq4: Removing templatization for AudioMultiVector by henrik.lundin@webrtc.org · 11 years ago
d069ddf Support for CELT in NetEq4. by turaj@webrtc.org · 11 years ago
9d8bbb7 Remove include_dirs from video_render. by pbos@webrtc.org · 11 years ago
6ea626e Remove include_dirs from video_capture. by pbos@webrtc.org · 11 years ago
5c9d8ee Revert 4876 "Support for CELT in NetEq4." by tina.legrand@webrtc.org · 11 years ago
38b1789 Propagate AutoMuter interface out to VideoCodingModule by henrik.lundin@webrtc.org · 11 years ago
1364cf1 1. adding request of ACM version in the manual mode of voe_auto_test by minyue@webrtc.org · 11 years ago
10b35b2 Support for CELT in NetEq4. by turaj@webrtc.org · 11 years ago
197a5eb Change the parameters of calculating maximum decode time. by wuchengli@chromium.org · 11 years ago
7540ded Makes OpensSL default audio implementation/device on Android. by henrike@webrtc.org · 11 years ago
aaa1ce6 Add protection to few more methods of AudioDeviceLinuxALSA. Those methods can be called from by wu@webrtc.org · 11 years ago
db10d03 Only use -lm on Linux in ISAC. by andrew@webrtc.org · 11 years ago
45f59dd Remove test parameters from CallTest. by pbos@webrtc.org · 11 years ago
8ac5bf4 With ACM2 and NetEq4, VoE fuzz test very often fails. by minyue@webrtc.org · 11 years ago
1a58624 Remove include_dirs from tools. by pbos@webrtc.org · 11 years ago
109108e Remove include_dirs from test. by pbos@webrtc.org · 11 years ago
1bd61f2 Implemented AutoMuter in MediaOptimization by henrik.lundin@webrtc.org · 11 years ago
cbfa687 Remove include_dirs from pacing. by pbos@webrtc.org · 11 years ago
5cb000f Remove include_dirs from remote_bitrate_estimator. by pbos@webrtc.org · 11 years ago
8895173 Remove include_dirs from bitrate_controller. by pbos@webrtc.org · 11 years ago
a548c20 Remove include_dirs from video_coding. by pbos@webrtc.org · 11 years ago
f8c0ed5 Remove include_dirs from video_processing. by pbos@webrtc.org · 11 years ago
f98e9d8 Remove include_dirs from rtp_rtcp. by pbos@webrtc.org · 11 years ago
749f3e3 Sync-packet insertion into NetEq4. This is related to r3883 & r4052 for NetEq 3. by turaj@webrtc.org · 11 years ago
870404d Move the Config DelayCorrection struct to audio_processing.h. by andrew@webrtc.org · 11 years ago
ad80fde Add an extended filter mode to AEC. by andrew@webrtc.org · 11 years ago
e5f36e0 Fix WindowCapturerWin to capture window decorations after window size changes. by sergeyu@chromium.org · 11 years ago
e45a98b Disable a NetEq unittest on Android. The test tries to register iSAC-swb as send codec and fails. by turaj@webrtc.org · 11 years ago
d49bc92 Remove unused constants, so chrome can enable a warning for that. Patch from thakis@ by niklas.enbom@webrtc.org · 11 years ago
6b42cc0 Updated WebRTC version to 3.43 TBR=mallinath@webrtc.org by elham@webrtc.org · 11 years ago
38cef0a Re-enable verbose logging in NetEq4. by turaj@webrtc.org · 11 years ago
17583a4 Convert DeviceInfoImpl::_captureCapabilities from a map to a vector. by fischman@webrtc.org · 11 years ago
85c5e8f Revert 4837 "Add an extended filter mode to AEC." by asapersson@webrtc.org · 11 years ago
000fecc Add an extended filter mode to AEC. by andrew@webrtc.org · 11 years ago
3d9ce0c Small fixes to run ACM2 tests. by turaj@webrtc.org · 11 years ago
d4b2d0c API add to set background noise mode. by turaj@webrtc.org · 11 years ago
2b7e421 Fix window capturer not to leak HDC. by sergeyu@chromium.org · 11 years ago
f6795cd Fix window capturer to stop capturing when the target is minimized. by sergeyu@chromium.org · 11 years ago
0ab1945 Disable some VP8 tests on Android. by andrew@webrtc.org · 11 years ago
a064105 Fix for Heap-use-after-free in webrtc::voe::Channel::SendRTCPPacket by henrika@webrtc.org · 11 years ago
88bcc98 Add libjingle_peerconnection_objc_test to buildbot_tests.py by kjellander@webrtc.org · 11 years ago
957be53 Revert r4823 "Reenable test and remove flaky expects." by stefan@webrtc.org · 11 years ago
7e07f16 Reenable test and remove flaky expects. by stefan@webrtc.org · 11 years ago
48b1173 MediaOptimization: Converting a few members to scoped_ptrs by henrik.lundin@webrtc.org · 11 years ago
653dfe1 Disable flaky RunsRtpRtcpTestWithoutErrors. by andrew@webrtc.org · 11 years ago
d24ce00 Remove deprecated AudioCodingModule::Destroy. by andrew@webrtc.org · 11 years ago
5cca7ed - Reset capture deltas at resolution change. by asapersson@webrtc.org · 11 years ago
72fde7b Reformatting media_optimization.cc and .h by henrik.lundin@webrtc.org · 11 years ago
584890b Re-enable VideoCaptureTest.CreateDelete by fischman@webrtc.org · 11 years ago
0715778 Updated WebRTC version to 3.42 by elham@webrtc.org · 11 years ago
697b7f3 Adding unit tests for default temporal layer strategy. by andresp@webrtc.org · 11 years ago
21318a9 Revert test change in r4808. by stefan@webrtc.org · 11 years ago
dd3f2e4 Reduce flakiness in network down test. by stefan@webrtc.org · 11 years ago
0580c2c Enable SetInitialPlayoutDelay on Android. by dwkang@webrtc.org · 11 years ago
26caab2 Fix bugs in DesktopRegion::Subtract(). by sergeyu@chromium.org · 11 years ago
69dfcb4 VAD changes ported to ACM2. by turaj@webrtc.org · 11 years ago
344a2d7 Address Windows 64-bits warnings. by turaj@webrtc.org · 11 years ago
aa693dd Enable FEC for VideoSendStream. by pbos@webrtc.org · 11 years ago
8797e63 Disable flaky video capture test. by stefan@webrtc.org · 11 years ago
0d416cf Avoid recursively taking critical section. by stefan@webrtc.org · 11 years ago
0c6a78c Use link_settings instead of all_dependent_settings to pacify xcode gyp generator by fischman@webrtc.org · 11 years ago
927bbc2 Roll webrtc's chromium_revision 217707:224141 by fischman@webrtc.org · 11 years ago
60fa827 Rename EngineTest to CallTest. by pbos@webrtc.org · 11 years ago
14f44ba Heap-use-after-free in WebRtcNetEQ_RecInRTPStruct by tina.legrand@webrtc.org · 11 years ago
9612f5a Refactor frame generation code so it can be used by multiple modules. by andresp@webrtc.org · 11 years ago
97e2f4e Disable NACK bandwidth statistics test due to being too flaky. by stefan@webrtc.org · 11 years ago
71c8df6 Fixes a flake in network down tests. by stefan@webrtc.org · 11 years ago
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