Log - 5a92b78e868983096a883e7016727981b375582d - platform/external/webrtc

5a92b78 Add beamforming to audioproc_float utility. by mgraczyk@chromium.org · 10 years ago
6b63015 Move ring_buffer to common_audio. by andrew@webrtc.org · 10 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 · 10 years ago
693e01c Fix searching for DirectX SDK during GN build. by kjellander@webrtc.org · 10 years ago
f1c8b90 Remove WebRtcVideoEncoderFactory2. by pbos@webrtc.org · 10 years ago
e5a31e1 Revert removing of compile_assert.h. by turaj@webrtc.org · 10 years ago
85fa94d Exclude EndToEndTest.SendsAndReceivesH264 for Dr Memory. by kjellander@webrtc.org · 10 years ago
387841a Improved fairness simulation by starting the flows 20 seconds apart. by stefan@webrtc.org · 10 years ago
f18fba2 Implement SimulcastEncoderAdapter support. by pbos@webrtc.org · 10 years ago
8315d7d Remove dual stream functionality in VoiceEngine by henrik.lundin@webrtc.org · 10 years ago
b4e5d1b Remove RTX SSRC when deleting the default receive stream. by mflodman@webrtc.org · 10 years ago
2ebfac5 Remove COMPILE_ASSERT and use static_assert everywhere by kwiberg@webrtc.org · 10 years ago
86e1e48 Move system_wrappers.gyp files to the proper directory. by andresp@webrtc.org · 10 years ago
a35f741 Add .classpath + talk/app/webrtc/androidtests to .gitignore by kjellander@webrtc.org · 10 years ago
f7a5893 Combine RegKeyTests to prevent parallel execution. by pbos@webrtc.org · 10 years ago
ef09092 No longer asserting in mocks, split first test case in two methods. by phoglund@webrtc.org · 10 years ago
69f4738 Roll chromium_revision 3dd2edf..a6eafec (310717:311223) by kjellander@webrtc.org · 10 years ago
d6e84d9 Always copy processed audio to output buffer in ProcessStream. by mgraczyk@chromium.org · 10 years ago
c0da63c Optimize minimum delay in blocker by aluebs@webrtc.org · 10 years ago
af9d56f Unify the two copies of template_util.h by kwiberg@webrtc.org · 10 years ago
0b0c241 Only return Rtx mode in RTXSendStatus(). by pbos@webrtc.org · 10 years ago
3df38b4 Unify the two copies of compile_assert.h by kwiberg@webrtc.org · 10 years ago
58a1ba6 Roll chromium_revision 271c6cc..3dd2edf (309333:310717) by kjellander@webrtc.org · 10 years ago
46323b3 Remove useless AudioProcessing::Create() overload. by andrew@webrtc.org · 10 years ago
16825b1 Use int64_t more consistently for times, in particular for RTT values. by pkasting@chromium.org · 10 years ago
a7add19 audio_processing: Replaced macro WEBRTC_SPL_MUL_16_16 with * in high_pass_filter by bjornv@webrtc.org · 10 years ago
2a26734 Partial revert of r7396 by henrik.lundin@webrtc.org · 10 years ago
be40eb0 Allow 720x1280 frames encoding on Android. by glaznev@webrtc.org · 10 years ago
a525c98 Fix parallelizability in ApmTests. by pbos@webrtc.org · 10 years ago
45db7ee Use Java based audio as default for WebRTC. by henrika@webrtc.org · 10 years ago
81134d0 Use proxy macro for PeerConnectionFactory instead of sending messages internally in PeerConnectionFactory. by perkj@webrtc.org · 10 years ago
88a4298 common_audio: Made input vector const in WebRtcSpl_LevinsonDurbin() by bjornv@webrtc.org · 10 years ago
c14e357 common_audio: Made input signal const in WebRtcSplFilterMAFastQ12() by bjornv@webrtc.org · 10 years ago
19e4e8d Add support for trying alternate server (STUN 300 error message) on TCP by guoweis@webrtc.org · 10 years ago
0ba1533 Added support for an Origin header in STUN messages. by pthatcher@webrtc.org · 10 years ago
2693a54 Add WEBRTC_BEAMFORMER define to BUILD.gn by aluebs@webrtc.org · 10 years ago
8f27fcc Revert 8028 "Support associated payload type when registering Rt..." by andrew@webrtc.org · 10 years ago
80452d7 Sync Android AppRTCDemo with internal repo. by glaznev@webrtc.org · 10 years ago
9657265 Revert "Accept incoming pings before remote answer is set to reduce connection latency." by pthatcher@webrtc.org · 10 years ago
f3fd8e7 Add NEON intrinsics version for transform_neon by andrew@webrtc.org · 10 years ago
1592df7 PRESUBMIT: Add GN trybots for Windows and Mac. by kjellander@webrtc.org · 10 years ago
2a16964 Support associated payload type when registering Rtx payload type. by pbos@webrtc.org · 10 years ago
8649fed GN: Fix Windows build. by kjellander@webrtc.org · 10 years ago
2ead571 Hard define the GUID for AudioEndpoint to avoid conflicts during compile. by decurtis@webrtc.org · 10 years ago
758d6d4 audio_processing/aecm: Removed usage of macro WEBRTC_SPL_MUL_16_16 by bjornv@webrtc.org · 10 years ago
dec649c audio_processing/ns: Replaced WEBRTC_SPL_MUL_16_16 macro with * by bjornv@webrtc.org · 10 years ago
5e5b327 audio_processing/agc: Removed usage of macro WEBRTC_SPL_MUL_16_16 in legacy/agc by bjornv@webrtc.org · 10 years ago
124b9c7 Suppress races in event tracing code. by pbos@webrtc.org · 10 years ago
5f09564 Suppress AsyncHttpRequestTest.TestCancel leak for LSan by kjellander@webrtc.org · 10 years ago
823c9b8 Add histograms stats for sent/received fraction loss for a stream: by asapersson@webrtc.org · 10 years ago
d730b28 Remove WebRtcSpl_ScaleAndAddVectorsWithRoundNeon by andrew@webrtc.org · 10 years ago
59062d5 Rename SendAndReceiveH264SvcQqvga to VP8 instead. by pbos@webrtc.org · 10 years ago
8af1104 Avoid reading past end of string in GetLine. by decurtis@webrtc.org · 10 years ago
3663fb0 Reenable dlclose() for InternalUnloadDll on TSan. by pbos@webrtc.org · 10 years ago
bab7995 Convert FileMediaEngineTest to use more expects. by pbos@webrtc.org · 10 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 · 10 years ago
c10ecea Always tag SRTP_PROTECTION_PROFILE and BIO_METHOD as const. by henrike@webrtc.org · 10 years ago
dfef028 Ignore virtual box interfaces. by pthatcher@webrtc.org · 10 years ago
25dd754 Excluding a flaky test from DrMemory by tina.legrand@webrtc.org · 10 years ago
7fbf278 Suppress memcheck error in video_engine_tests by kjellander@webrtc.org · 10 years ago
1777880 Roll gtest-parallel. by pbos@webrtc.org · 10 years ago
07c83a1 Disable WebRtcVideoMediaChannelSimulcastTest.SimulcastSend tests on Win (take 2) by kjellander@webrtc.org · 10 years ago
4e5115a RTCPeerConnectionFactory: Explicitly create new worker and signaling threads. by tkchin@webrtc.org · 10 years ago
f6a9714 Remove peer connection and signaling calls from UI thread. by glaznev@webrtc.org · 10 years ago
2ec50f2 Memcheck suppression for uninitalized memory in WebRtcIsac_Decode by kjellander@webrtc.org · 10 years ago
d95435c Disable WebRtcVideoMediaChannelSimulcastTest.SimulcastSend tests on Win by kjellander@webrtc.org · 10 years ago
cbe7ca8 Roll chromium_revision 8e72e1d..271c6cc (307131:309333) by kjellander@webrtc.org · 10 years ago
3a63a3c iOS AppRTC: First unit test. by tkchin@webrtc.org · 10 years ago
4796cb9 Disable flaky RelayServerTest.TestExpiration on all platforms. by andrew@webrtc.org · 10 years ago
fb7a039 Use array geometry in Beamformer by aluebs@webrtc.org · 10 years ago
a37bf2c Hack clock_unittest fix for parallel execution. by andrew@webrtc.org · 10 years ago
c37e72e Make setting identical RTP extensions a no-op. by pbos@webrtc.org · 10 years ago
e5a921a Use tmp files in file_utils_unittests by aluebs@webrtc.org · 10 years ago
76bc981 Use a temp file in FileLockTest. by pbos@webrtc.org · 10 years ago
433006a Fixed style issues from lint and got rid of unused fields. by wzh@webrtc.org · 10 years ago
c4ad157 Re-enable test: VideoProcessorIntegrationTest.ProcessNoLossChangeFrameRateFrameDropVP9. by marpan@webrtc.org · 10 years ago
215bbbd Fix for log typo in ViEExternalCodecImpl::RegisterExternalReceiveCodec. by mflodman@webrtc.org · 10 years ago
aeb0dd3 Disable RelayServerTest.TestExpiration on Mac. by kjellander@webrtc.org · 10 years ago
8390c27 Add two unit tests for Android AppRTCDemo. by glaznev@webrtc.org · 10 years ago
896888b Remove min bitrate from simulcast streams. by pbos@webrtc.org · 10 years ago
bac0012 Extend delay estimation window in AEC to 500 ms on all platforms by bjornv@webrtc.org · 10 years ago
9eacb8c Make P2PTestConductor use VirtualSocketServer. by pbos@webrtc.org · 10 years ago
c62749f Parallelize MediaRecorder unittests. by pbos@webrtc.org · 10 years ago
3a70625 audio_processing: Added back ATTRIBUTE_UNUSED lost in r7877 by bjornv@webrtc.org · 10 years ago
27f5317 Use the prod GAE server in AppRTCDemo for iOS. by jiayl@webrtc.org · 10 years ago
5eb71eb Fix style issues from lint. by jiayl@webrtc.org · 10 years ago
34ac956 Do not use openmax_dl for MIPS64 platform. by andrew@webrtc.org · 10 years ago
b2bda67 Removing old channel code from a few more places. by glaznev@webrtc.org · 10 years ago
a9b1ec0 Support for DTLS in OpenSSLAdapter by pthatcher@webrtc.org · 10 years ago
c5fd66d Accept incoming pings before remote answer is set to reduce connection latency. by jiayl@webrtc.org · 10 years ago
84d8447 Minor fixes regarding accumulator usage on MIPS platforms. by andrew@webrtc.org · 10 years ago
b024da3 Add support for audio device selection in AppRTCDemo. by henrika@webrtc.org · 10 years ago
5ad4178 Move the Jingle-specific network code into webrtc/libjingle. by pthatcher@webrtc.org · 10 years ago
46d4d29 Add field trial for screenshare bitrates when using temporal layers. by sprang@webrtc.org · 10 years ago
1be0a78 Removing giles@mozilla.com from WebRTC watchlist. by mflodman@webrtc.org · 10 years ago
53cb741 Make RelayServerTest use VirtualSocketServer. by pbos@webrtc.org · 10 years ago
086c8d5 Use a temporary buffer to scale a screencast in OnFrameCaptured by braveyao@webrtc.org · 10 years ago
4c0544a Move Jingle-specific files from talk/session/media to webrtc/libjingle/session/media. This is part of an ongoing effort to remove Jingle-specific files from the WebRTC repository. by pthatcher@webrtc.org · 10 years ago
ed1a48b Fix mac video capture leak. by tkchin@webrtc.org · 10 years ago
7ce4a58 Add initWithCoder to RTCEAGLVideoView. by tkchin@webrtc.org · 10 years ago