Log - ce3ac5375775c25ed5d78bee2f8904d3cc21950b - platform/external/webrtc

ce3ac53 Adding TRYSERVER_PROJECT to codereview.settings. by kjellander@webrtc.org · 9 years ago
018c087 Add /talk/examples/androidtests/{bin,gen} to .gitignore. by kjellander@webrtc.org · 9 years ago
a32d154 Disable tests failing on Android ARM64 (Nexus9). by kjellander@webrtc.org · 9 years ago
ff9462e Disable WebRtcVideoMediaChannelSimulcastTest::SimulcastSend_* on tsan. by sprang@webrtc.org · 9 years ago
2624b1e Remove unused private data member engine_id_ by tommi@webrtc.org · 9 years ago
fe672e3 release the turn allocation by sending a refresh request with lifetime 0 by pthatcher@webrtc.org · 9 years ago
d7de120 Re-enable the messagequeue unittests. These were commented out at one point but never reenabled. by decurtis@webrtc.org · 9 years ago
a1aea10 Revert r8076 "Add UMA stats for tracking the time it takes to reach a BWE of 500, 1000 and 2000 kbps." by stefan@webrtc.org · 9 years ago
4ba1e44 Remove unnecessary remote bitrate estimator build rule which serves no purpose. by andresp@webrtc.org · 9 years ago
487a444 Add stats collection for the data channel. by decurtis@webrtc.org · 9 years ago
357469d Fixes reference counting problem when a TransportProxy points to a Transport prior to creating channels. by decurtis@webrtc.org · 9 years ago
ef2a5dd Update AppRTCDemo UI. by tkchin@webrtc.org · 9 years ago
64d3c4b Support 48kHz in AEC by aluebs@webrtc.org · 9 years ago
89aa276 Fix a case where empty candidate id is used by guoweis@webrtc.org · 9 years ago
d82f55d Only adapt AGC when the desired signal is present by aluebs@webrtc.org · 9 years ago
3e42a8a Add UMA stats for tracking the time it takes to reach a BWE of 500, 1000 and 2000 kbps. by stefan@webrtc.org · 9 years ago
32e8528 Log configs when creating video streams in Call. by pbos@webrtc.org · 9 years ago
1f67b53 Remove dual stream functionality in ACM by henrik.lundin@webrtc.org · 9 years ago
9ce01e6 Clean unnecessary workaround for chromium import. by andresp@webrtc.org · 9 years ago
0800db7 Add percentage of fec packets and recovered media packets to histogram stats: by asapersson@webrtc.org · 9 years ago
61c1247 Fix a case where empty candidate id is used by guoweis@webrtc.org · 9 years ago
6c38552 Add WebRtcIsacfix_AllpassFilter2FixDec16Neon()'s intrinsics version. by andrew@webrtc.org · 9 years ago
5a92b78 Add beamforming to audioproc_float utility. by mgraczyk@chromium.org · 9 years ago
6b63015 Move ring_buffer to common_audio. by andrew@webrtc.org · 9 years ago
fd630a5 Add BundlePolicy to RTCConfiguration. Don't change any behavior. Just make it possible to make progress in Chromium while we work on the behavior. by pthatcher@webrtc.org · 9 years ago
693e01c Fix searching for DirectX SDK during GN build. by kjellander@webrtc.org · 9 years ago
f1c8b90 Remove WebRtcVideoEncoderFactory2. by pbos@webrtc.org · 9 years ago
e5a31e1 Revert removing of compile_assert.h. by turaj@webrtc.org · 9 years ago
85fa94d Exclude EndToEndTest.SendsAndReceivesH264 for Dr Memory. by kjellander@webrtc.org · 9 years ago
387841a Improved fairness simulation by starting the flows 20 seconds apart. by stefan@webrtc.org · 9 years ago
f18fba2 Implement SimulcastEncoderAdapter support. by pbos@webrtc.org · 9 years ago
8315d7d Remove dual stream functionality in VoiceEngine by henrik.lundin@webrtc.org · 9 years ago
b4e5d1b Remove RTX SSRC when deleting the default receive stream. by mflodman@webrtc.org · 9 years ago
2ebfac5 Remove COMPILE_ASSERT and use static_assert everywhere by kwiberg@webrtc.org · 9 years ago
86e1e48 Move system_wrappers.gyp files to the proper directory. by andresp@webrtc.org · 9 years ago
a35f741 Add .classpath + talk/app/webrtc/androidtests to .gitignore by kjellander@webrtc.org · 9 years ago
f7a5893 Combine RegKeyTests to prevent parallel execution. by pbos@webrtc.org · 9 years ago
ef09092 No longer asserting in mocks, split first test case in two methods. by phoglund@webrtc.org · 9 years ago
69f4738 Roll chromium_revision 3dd2edf..a6eafec (310717:311223) by kjellander@webrtc.org · 9 years ago
d6e84d9 Always copy processed audio to output buffer in ProcessStream. by mgraczyk@chromium.org · 9 years ago
c0da63c Optimize minimum delay in blocker by aluebs@webrtc.org · 9 years ago
af9d56f Unify the two copies of template_util.h by kwiberg@webrtc.org · 9 years ago
0b0c241 Only return Rtx mode in RTXSendStatus(). by pbos@webrtc.org · 9 years ago
3df38b4 Unify the two copies of compile_assert.h by kwiberg@webrtc.org · 9 years ago
58a1ba6 Roll chromium_revision 271c6cc..3dd2edf (309333:310717) by kjellander@webrtc.org · 9 years ago
46323b3 Remove useless AudioProcessing::Create() overload. by andrew@webrtc.org · 9 years ago
16825b1 Use int64_t more consistently for times, in particular for RTT values. by pkasting@chromium.org · 9 years ago
a7add19 audio_processing: Replaced macro WEBRTC_SPL_MUL_16_16 with * in high_pass_filter by bjornv@webrtc.org · 9 years ago
2a26734 Partial revert of r7396 by henrik.lundin@webrtc.org · 9 years ago
be40eb0 Allow 720x1280 frames encoding on Android. by glaznev@webrtc.org · 9 years ago
a525c98 Fix parallelizability in ApmTests. by pbos@webrtc.org · 9 years ago
45db7ee Use Java based audio as default for WebRTC. by henrika@webrtc.org · 9 years ago
81134d0 Use proxy macro for PeerConnectionFactory instead of sending messages internally in PeerConnectionFactory. by perkj@webrtc.org · 9 years ago
88a4298 common_audio: Made input vector const in WebRtcSpl_LevinsonDurbin() by bjornv@webrtc.org · 9 years ago
c14e357 common_audio: Made input signal const in WebRtcSplFilterMAFastQ12() by bjornv@webrtc.org · 9 years ago
19e4e8d Add support for trying alternate server (STUN 300 error message) on TCP by guoweis@webrtc.org · 9 years ago
0ba1533 Added support for an Origin header in STUN messages. by pthatcher@webrtc.org · 9 years ago
2693a54 Add WEBRTC_BEAMFORMER define to BUILD.gn by aluebs@webrtc.org · 9 years ago
8f27fcc Revert 8028 "Support associated payload type when registering Rt..." by andrew@webrtc.org · 9 years ago
80452d7 Sync Android AppRTCDemo with internal repo. by glaznev@webrtc.org · 9 years ago
9657265 Revert "Accept incoming pings before remote answer is set to reduce connection latency." by pthatcher@webrtc.org · 9 years ago
f3fd8e7 Add NEON intrinsics version for transform_neon by andrew@webrtc.org · 9 years ago
1592df7 PRESUBMIT: Add GN trybots for Windows and Mac. by kjellander@webrtc.org · 9 years ago
2a16964 Support associated payload type when registering Rtx payload type. by pbos@webrtc.org · 9 years ago
8649fed GN: Fix Windows build. by kjellander@webrtc.org · 9 years ago
2ead571 Hard define the GUID for AudioEndpoint to avoid conflicts during compile. by decurtis@webrtc.org · 9 years ago
758d6d4 audio_processing/aecm: Removed usage of macro WEBRTC_SPL_MUL_16_16 by bjornv@webrtc.org · 9 years ago
dec649c audio_processing/ns: Replaced WEBRTC_SPL_MUL_16_16 macro with * by bjornv@webrtc.org · 9 years ago
5e5b327 audio_processing/agc: Removed usage of macro WEBRTC_SPL_MUL_16_16 in legacy/agc by bjornv@webrtc.org · 9 years ago
124b9c7 Suppress races in event tracing code. by pbos@webrtc.org · 9 years ago
5f09564 Suppress AsyncHttpRequestTest.TestCancel leak for LSan by kjellander@webrtc.org · 9 years ago
823c9b8 Add histograms stats for sent/received fraction loss for a stream: by asapersson@webrtc.org · 9 years ago
d730b28 Remove WebRtcSpl_ScaleAndAddVectorsWithRoundNeon by andrew@webrtc.org · 9 years ago
59062d5 Rename SendAndReceiveH264SvcQqvga to VP8 instead. by pbos@webrtc.org · 9 years ago
8af1104 Avoid reading past end of string in GetLine. by decurtis@webrtc.org · 9 years ago
3663fb0 Reenable dlclose() for InternalUnloadDll on TSan. by pbos@webrtc.org · 9 years ago
bab7995 Convert FileMediaEngineTest to use more expects. by pbos@webrtc.org · 9 years ago
69472e7 Add a dummy implemenation of SChannelAdapter::SetMode that makes sure that StartSSL fails if the mode is set to DTLS. by pthatcher@webrtc.org · 9 years ago
c10ecea Always tag SRTP_PROTECTION_PROFILE and BIO_METHOD as const. by henrike@webrtc.org · 9 years ago
dfef028 Ignore virtual box interfaces. by pthatcher@webrtc.org · 9 years ago
25dd754 Excluding a flaky test from DrMemory by tina.legrand@webrtc.org · 9 years ago
7fbf278 Suppress memcheck error in video_engine_tests by kjellander@webrtc.org · 9 years ago
1777880 Roll gtest-parallel. by pbos@webrtc.org · 9 years ago
07c83a1 Disable WebRtcVideoMediaChannelSimulcastTest.SimulcastSend tests on Win (take 2) by kjellander@webrtc.org · 9 years ago
4e5115a RTCPeerConnectionFactory: Explicitly create new worker and signaling threads. by tkchin@webrtc.org · 9 years ago
f6a9714 Remove peer connection and signaling calls from UI thread. by glaznev@webrtc.org · 9 years ago
2ec50f2 Memcheck suppression for uninitalized memory in WebRtcIsac_Decode by kjellander@webrtc.org · 9 years ago
d95435c Disable WebRtcVideoMediaChannelSimulcastTest.SimulcastSend tests on Win by kjellander@webrtc.org · 9 years ago
cbe7ca8 Roll chromium_revision 8e72e1d..271c6cc (307131:309333) by kjellander@webrtc.org · 9 years ago
3a63a3c iOS AppRTC: First unit test. by tkchin@webrtc.org · 9 years ago
4796cb9 Disable flaky RelayServerTest.TestExpiration on all platforms. by andrew@webrtc.org · 9 years ago
fb7a039 Use array geometry in Beamformer by aluebs@webrtc.org · 9 years ago
a37bf2c Hack clock_unittest fix for parallel execution. by andrew@webrtc.org · 9 years ago
c37e72e Make setting identical RTP extensions a no-op. by pbos@webrtc.org · 9 years ago
e5a921a Use tmp files in file_utils_unittests by aluebs@webrtc.org · 9 years ago
76bc981 Use a temp file in FileLockTest. by pbos@webrtc.org · 9 years ago
433006a Fixed style issues from lint and got rid of unused fields. by wzh@webrtc.org · 9 years ago
c4ad157 Re-enable test: VideoProcessorIntegrationTest.ProcessNoLossChangeFrameRateFrameDropVP9. by marpan@webrtc.org · 9 years ago
215bbbd Fix for log typo in ViEExternalCodecImpl::RegisterExternalReceiveCodec. by mflodman@webrtc.org · 9 years ago
aeb0dd3 Disable RelayServerTest.TestExpiration on Mac. by kjellander@webrtc.org · 9 years ago