Log - e6938185a526c3988057b0a261c7e51562f49f28 - platform/external/webrtc

e693818 Add isolate targets for libjingle by kjellander@webrtc.org · 11 years ago
3f9288f Add APK and isolate target for video_engine_tests by kjellander@webrtc.org · 11 years ago
6c264cc Clean up AudioProcessing defaults and errors. by andrew@webrtc.org · 11 years ago
83b9e5b Add owners to [webrtc,talk]/build and *.isolate by kjellander@webrtc.org · 11 years ago
acb0050 Only declare kDelayDiffOffset when used. by andrew@webrtc.org · 11 years ago
ad2eb6f Unbreaks Android build after r4915. by henrike@webrtc.org · 11 years ago
be9c560 Revert r4913 that reverts r4911. Original CL description: by andresp@webrtc.org · 11 years ago
bab2aa5 Add audio and video parameters for setting media constraints. by andrew@webrtc.org · 11 years ago
4446134 AppRTCDemo(android): support boolean value for MediaStreamConstraints.{audio,video}. by fischman@webrtc.org · 11 years ago
a7266ca Fix clang build break by fischman@webrtc.org · 11 years ago
6c82e04 Android standalone: remove some usages of deprecated APIs and prevent further regressions. by fischman@webrtc.org · 11 years ago
4e65e07 VideoCaptureAndroid: rewrote the (standalone) implementation of video capture on Android. by fischman@webrtc.org · 11 years ago
ddc5a19 AppRTCDemo(android): uncaught exceptions now display a modal dialog box before killing the app. by fischman@webrtc.org · 11 years ago
44db9d1 Revert 4911 "Adding temporal layer strategy that keeps base laye..." by turaj@webrtc.org · 11 years ago
b43d807 Reformatting VPM: First step - No functional changes. by mikhal@webrtc.org · 11 years ago
26f78f7 Adding temporal layer strategy that keeps base layer framerate at an acceptable value. by andresp@webrtc.org · 11 years ago
70df305 Minor fix to avoid breakage by henrik.lundin@webrtc.org · 11 years ago
7ee3efb Disable Receiver unittests on Android. by turaj@webrtc.org · 11 years ago
6ea3d1c ACM test are modified to run with both ACM1 and ACM2. by turaj@webrtc.org · 11 years ago
2a97317 Fix include of isolate.gypi by kjellander@webrtc.org · 11 years ago
f8f78b1 Android OpenSL: Fixes faulty assertion in jni-code. by henrike@webrtc.org · 11 years ago
9b5c807 Remove ReturnTrace from DeregisterCallback(). by pbos@webrtc.org · 11 years ago
4887114 Remove templatization of the AudioVector test by henrik.lundin@webrtc.org · 11 years ago
c0b4c4a Workaround issue with stdin on Windows. by kjellander@webrtc.org · 11 years ago
1fdc51a APK for opensl loopback. by henrike@webrtc.org · 11 years ago
de74b64 Implement TraceCallbacks in Call. by pbos@webrtc.org · 11 years ago
7ea4f24 Piping AutoMuter interface through to ViE API by henrik.lundin@webrtc.org · 11 years ago
8469f7b Added support for sending and receiving RTCP XR packets: by asapersson@webrtc.org · 11 years ago
c016770 Stop timer in ~EventWindows(). by pbos@webrtc.org · 11 years ago
a6101d7 Update sampling rate and number of channels of NetEq4 if decoder is changed. by turaj@webrtc.org · 11 years ago
ee6d0dd Upload Demo page to allow edit offer & Answer sdp in pc1 demo. by vikasmarwaha@webrtc.org · 11 years ago
1d731e4 Roll chromium_revision 224141:226099 to pick up jsoncpp fix for ARM by fischman@webrtc.org · 11 years ago
19134ba Updated device-switch demo page to work with Chrome M30. by vikasmarwaha@webrtc.org · 11 years ago
b74b96f Test multiple send/receive streams in Call. by pbos@webrtc.org · 11 years ago
e546f02 Remove include_dirs from utility. by pbos@webrtc.org · 11 years ago
7e4d0df PeerConnection(Android): enable tracing to logcat. by fischman@webrtc.org · 11 years ago
5222270 Reset audio bufer if codec changes, b/10835525. by turaj@webrtc.org · 11 years ago
8e2f9bc Ensure adjusted "known delay" doesn't drop below zero. by andrew@webrtc.org · 11 years ago
fd11bbf NetEq4: Removing templatization for AudioMultiVector by henrik.lundin@webrtc.org · 11 years ago
6ad6a07 Support for CELT in NetEq4. by turaj@webrtc.org · 11 years ago
7e809c3 Update libjingle to CL 53496343. by mallinath@webrtc.org · 11 years ago
9532fa5 Remove include_dirs from video_render. by pbos@webrtc.org · 11 years ago
1c974ef Remove include_dirs from video_capture. by pbos@webrtc.org · 11 years ago
4cd7622 Revert 4876 "Support for CELT in NetEq4." by tina.legrand@webrtc.org · 11 years ago
572699d Propagate AutoMuter interface out to VideoCodingModule by henrik.lundin@webrtc.org · 11 years ago
cc92e00 1. adding request of ACM version in the manual mode of voe_auto_test by minyue@webrtc.org · 11 years ago
a20a22a Support for CELT in NetEq4. by turaj@webrtc.org · 11 years ago
ad81ab8 Suppress SSL error strings on mac_asan to unbreak that build by mallinath@webrtc.org · 11 years ago
30377c7 Change the parameters of calculating maximum decode time. by wuchengli@chromium.org · 11 years ago
a27be8e Update libjingle to CL 53398036. by mallinath@webrtc.org · 11 years ago
34c50c1 Makes OpensSL default audio implementation/device on Android. by henrike@webrtc.org · 11 years ago
a39b323 Add tools/sharding_supervisor to .gitignore by kjellander@webrtc.org · 11 years ago
8b7ec82 Exclude P2PTransportChannelSameNatTest.TestConesBehindSameCone for TSan Linux by kjellander@webrtc.org · 11 years ago
4475905 Disable flaky RapidSpeakerChange test. by andrew@webrtc.org · 11 years ago
6049787 Add protection to few more methods of AudioDeviceLinuxALSA. Those methods can be called from by wu@webrtc.org · 11 years ago
137b379 Only use -lm on Linux in ISAC. by andrew@webrtc.org · 11 years ago
287f07b Add sharding_supervisor to DEPS to prepare for swarm/isolated testing. by kjellander@webrtc.org · 11 years ago
2e246b4 Remove test parameters from CallTest. by pbos@webrtc.org · 11 years ago
3223a3d Roll libvpx 212975:225010 to pick up iOS Release fixes by fischman@webrtc.org · 11 years ago
663da0a With ACM2 and NetEq4, VoE fuzz test very often fails. by minyue@webrtc.org · 11 years ago
f26e8f6 Remove include_dirs from tools. by pbos@webrtc.org · 11 years ago
f8b2966 Remove include_dirs from test. by pbos@webrtc.org · 11 years ago
544b17c Implemented AutoMuter in MediaOptimization by henrik.lundin@webrtc.org · 11 years ago
04b6179 Remove include_dirs from pacing. by pbos@webrtc.org · 11 years ago
97eefb7 Remove include_dirs from remote_bitrate_estimator. by pbos@webrtc.org · 11 years ago
339fe12 Remove include_dirs from bitrate_controller. by pbos@webrtc.org · 11 years ago
054ccd2 Remove include_dirs from video_coding. by pbos@webrtc.org · 11 years ago
73f2076 Remove include_dirs from video_processing. by pbos@webrtc.org · 11 years ago
dc3fa08 Remove include_dirs from rtp_rtcp. by pbos@webrtc.org · 11 years ago
7b75ac6 Sync-packet insertion into NetEq4. This is related to r3883 & r4052 for NetEq 3. by turaj@webrtc.org · 11 years ago
6b1e219 Move the Config DelayCorrection struct to audio_processing.h. by andrew@webrtc.org · 11 years ago
1760a17 Add an extended filter mode to AEC. by andrew@webrtc.org · 11 years ago
becbefa Fix WindowCapturerWin to capture window decorations after window size changes. by sergeyu@chromium.org · 11 years ago
3fdeddb 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
3e77036 Remove unused constants, so chrome can enable a warning for that. Patch from thakis@ by niklas.enbom@webrtc.org · 11 years ago
cecaae2 Updated WebRTC version to 3.43 TBR=mallinath@webrtc.org by elham@webrtc.org · 11 years ago
0c0fae8 Re-enable verbose logging in NetEq4. by turaj@webrtc.org · 11 years ago
69fc315 Convert DeviceInfoImpl::_captureCapabilities from a map to a vector. by fischman@webrtc.org · 11 years ago
ce014d9 Revert 4837 "Add an extended filter mode to AEC." by asapersson@webrtc.org · 11 years ago
2f240b4 Disable some flaky libjingle base tests. by andrew@webrtc.org · 11 years ago
26e02f0 Add an extended filter mode to AEC. by andrew@webrtc.org · 11 years ago
d6a7a5f Small fixes to run ACM2 tests. by turaj@webrtc.org · 11 years ago
ff43c85 API add to set background noise mode. by turaj@webrtc.org · 11 years ago
8d757ac Fix window capturer not to leak HDC. by sergeyu@chromium.org · 11 years ago
958cdf6 Fix window capturer to stop capturing when the target is minimized. by sergeyu@chromium.org · 11 years ago
f832a55 Disable flaky TestPartialFrameHeader. by andrew@webrtc.org · 11 years ago
c2ac5ed Add more TSAN suppressions for libjingle_media_unittest. by andrew@webrtc.org · 11 years ago
641587f Disable some VP8 tests on Android. by andrew@webrtc.org · 11 years ago
f0f92fa Disable flaky SendDataMultipleClocks. by andrew@webrtc.org · 11 years ago
9b6eefc Fix for Heap-use-after-free in webrtc::voe::Channel::SendRTCPPacket by henrika@webrtc.org · 11 years ago
ebd3ab0 Add libjingle_peerconnection_objc_test to buildbot_tests.py by kjellander@webrtc.org · 11 years ago
6e86349 Disable tests that crash the OS X kernel when run under memcheck. by kjellander@webrtc.org · 11 years ago
b0e6eb5 Revert r4823 "Reenable test and remove flaky expects." by stefan@webrtc.org · 11 years ago
01aad09 Reenable test and remove flaky expects. by stefan@webrtc.org · 11 years ago
b426c46 MediaOptimization: Converting a few members to scoped_ptrs by henrik.lundin@webrtc.org · 11 years ago
6ffc74e Disable flaky RunsRtpRtcpTestWithoutErrors. by andrew@webrtc.org · 11 years ago
eb524d9 Remove deprecated AudioCodingModule::Destroy. by andrew@webrtc.org · 11 years ago
1112c30 Update libjingle to 53057474. by mallinath@webrtc.org · 11 years ago
e2af622 - Reset capture deltas at resolution change. by asapersson@webrtc.org · 11 years ago
bec11ef Reformatting media_optimization.cc and .h by henrik.lundin@webrtc.org · 11 years ago