Log - 6b6e58d6325067977770ca7dfe4bef8457dd0141 - platform/external/webrtc

6b6e58d Remove unused test_env.py from isolate files + fix nss path. by kjellander@webrtc.org · 10 years ago
85d2794 Adds support for the "apt" format parameter and turns on the RTX feature. by stefan@webrtc.org · 10 years ago
ed7edb8 Enables DelayCorrection tests by bjornv@webrtc.org · 10 years ago
582367f Updated conformance tests and w3c-ified them. by phoglund@webrtc.org · 10 years ago
a1a2c0c Multi-threaded unit test for Audio Coding Module using iSAC by henrik.lundin@webrtc.org · 10 years ago
cb0ea43 audio_processing: Forces extended filter to be used in splitting filter test. by bjornv@webrtc.org · 10 years ago
9c55f0f Rename neteq4 folder to neteq by henrik.lundin@webrtc.org · 10 years ago
31f967c Fix Dr Memory download by kjellander@webrtc.org · 10 years ago
9221ab4 Re-enable AudioCodingModuleMtTest again by henrik.lundin@webrtc.org · 10 years ago
9359eda PRESUBMIT: Add Android ARM64 and remove Linux TSan by kjellander@webrtc.org · 10 years ago
e3cdd99 Revert "Fix the "Failed unprotect audio RTP packet" error when SCTP is bundled with audio." by jiayl@webrtc.org · 10 years ago
013bdf8 APPRTCDemo(objc): Remove regex parsing in favor of JSON struct. by tkchin@webrtc.org · 10 years ago
24c1778 Revert r6358 "AppRTCDemo(Android): only stop the cameraThread's looper after stopping the camera." by fischman@webrtc.org · 10 years ago
c3288c1 Add OpenGL Android video renderer which can display multiple by glaznev@webrtc.org · 10 years ago
b8f5825 Use XErrorTrap in MouseCursorMonitorX11 to catch the error if the shared window has been closed. by jiayl@webrtc.org · 10 years ago
171d941 AppRTCDemo(Android): only stop the cameraThread's looper after stopping the camera. by fischman@webrtc.org · 10 years ago
b464618 Unbreak NDEBUG compile by RTC_UNUSED()ing an assert()d variable. by fischman@webrtc.org · 10 years ago
745a39c Fix the "Failed unprotect audio RTP packet" error when SCTP is bundled with audio. by jiayl@webrtc.org · 10 years ago
b273b60 ViEAutoTestAndroid: Unbreak compile by casting void* to jobject. by fischman@webrtc.org · 10 years ago
9512719 AppRTCDemo(android): support app (UI) & capture rotation. by fischman@webrtc.org · 10 years ago
42694c5 VideoCaptureImpl::IncomingFrame(): avoid deadlock by acquiring _apiCs. by fischman@webrtc.org · 10 years ago
91c9104 (Auto)update libjingle 68701339-> 68703656 by buildbot@webrtc.org · 10 years ago
910473b Fix C++11 -Wnarrowing in channel_unittest.cc. by pbos@webrtc.org · 10 years ago
7b6cbb3 (Auto)update libjingle 68689052-> 68689059 by buildbot@webrtc.org · 10 years ago
6ae48c6 Make VideoSendStream/VideoReceiveStream configs const. by pbos@webrtc.org · 10 years ago
4b83a47 (Auto)update libjingle 68646004-> 68648993 by buildbot@webrtc.org · 10 years ago
4e5f65a Rebase webrtc/base with r6345 version of talk/base: by henrike@webrtc.org · 10 years ago
94454b7 Fix the chain that propagates the audio frame's rtp and ntp timestamp including: by wu@webrtc.org · 10 years ago
130fa64 AppRTCDemo(android): remove HTML/regex hackery in favor of JSON struct. by fischman@webrtc.org · 10 years ago
65d61c3 Opus send rate overflows if over 65 kbps by tina.legrand@webrtc.org · 10 years ago
b51d3ea Revert 6341 "Fixes and enables SystemDelayTests." by bjornv@webrtc.org · 10 years ago
681aaae Remove remaining samples (AppRTC) since moved to Github by kjellander@webrtc.org · 10 years ago
1f971b5 Fixes and enables SystemDelayTests. by bjornv@webrtc.org · 10 years ago
2f816bb NetEq: Add thread annotation to const scoped_ptrs by henrik.lundin@webrtc.org · 10 years ago
eae7924 Adding back platform specific renderer to video loopback test. by mflodman@webrtc.org · 10 years ago
0d523ee Remove static initializer from WebRtcVideoEngine2. by pbos@webrtc.org · 10 years ago
aafd7a8 The correct fix of workaround in r6261. by bjornv@webrtc.org · 10 years ago
edbe886 common_audio/signal_processing: Removed macro WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND by bjornv@webrtc.org · 10 years ago
ef92755 Have RTX be enabled by setting an RTX payload type instead of by setting an RTX SSRC. by stefan@webrtc.org · 10 years ago
c578962 Disable a test in libjingle_peerconnection_unittest for DrMemory by henrik.lundin@webrtc.org · 10 years ago
f1adbee (Auto)update libjingle 68562943-> 68571194 by buildbot@webrtc.org · 10 years ago
e6e1391 Android: cleanup gtest_target_type conditions. by henrike@webrtc.org · 10 years ago
738df89 Fix retain cycle in RTCEAGLVideoView. by tkchin@webrtc.org · 10 years ago
c6db88b Make it possible to build webrtc for arm64. by solenberg@webrtc.org · 10 years ago
6f23776 (Auto)update libjingle 68507189-> 68543735 by buildbot@webrtc.org · 10 years ago
40b45fc (Auto)update libjingle 68506654-> 68507189 by buildbot@webrtc.org · 10 years ago
d3dcebf Disable P2PTransportChannelMultihomedTest.TestFailover under Memcheck by henrik.lundin@webrtc.org · 10 years ago
147f4fe Disables SystemDelayTest.CorrectDelayDuringDrift on Android by bjornv@webrtc.org · 10 years ago
b616e12 Disables some modules_unittests on Android. by bjornv@webrtc.org · 10 years ago
4436b44 Moved verbose logging in rtcp_receiver.cc to LS_VERBOSE. by andresp@webrtc.org · 10 years ago
2bdd399 Suppress memcheck error in VideoProcessorIntegrationTest by henrik.lundin@webrtc.org · 10 years ago
19fc09e Adding missing break in media_file_utility.cc. by mflodman@webrtc.org · 10 years ago
0cdcd23 (Auto)update libjingle 68501302-> 68506654 by buildbot@webrtc.org · 10 years ago
af81b9b (Auto)update libjingle 68499439-> 68501302 by buildbot@webrtc.org · 10 years ago
251fdf6 (Auto)update libjingle 68495561-> 68499439 by buildbot@webrtc.org · 10 years ago
09a71cd talk/ios: Fixes source after corrupt sync in r6305 (which corrupted r6291). by henrike@webrtc.org · 10 years ago
5321784 (Auto)update libjingle 68465410-> 68487517 by buildbot@webrtc.org · 10 years ago
4ef254f Enable videoprocessor_integrationtest tests on android. by marpan@webrtc.org · 10 years ago
83eb7df PeerConnection(java): disable wait for flaky ICEConnection.COMPLETED. by fischman@webrtc.org · 10 years ago
ddc6bc9 Revert 6312 "Re-enable AudioCodingModuleMtTest" by turaj@webrtc.org · 10 years ago
289a35c Add empty webrtcmediaengine.cc. by pbos@webrtc.org · 10 years ago
8d13cd1 Re-enable AudioCodingModuleMtTest by henrik.lundin@webrtc.org · 10 years ago
8e4401b Reformat integer accessors to look like their float counterparts by kwiberg@webrtc.org · 10 years ago
f2e4a99 Add kwiberg@webrtc.org to watchlist for audio_coding and audio_processing by kwiberg@webrtc.org · 10 years ago
b525a9d (Auto)update libjingle 68379861-> 68445177 by buildbot@webrtc.org · 10 years ago
044bdac Remove kMaxWaitForStatsMs from tsanv2 compilation. by pbos@webrtc.org · 10 years ago
c0035a6 Remove an optimization that's no longer worth the extra complexity it causes by kwiberg@webrtc.org · 10 years ago
34a08b4 (Auto)update libjingle 68275107-> 68379861 by buildbot@webrtc.org · 10 years ago
a28c697 - Get rid of 'using' from .h by solenberg@webrtc.org · 10 years ago
2f7c7ce Remove old perf_expectations no longer used. by kjellander@webrtc.org · 10 years ago
2bd032e Disable MouseCursorMonitorTest by henrik.lundin@webrtc.org · 10 years ago
4ecae6e Disable MouseCursorMonitorTest.FromScreen by henrik.lundin@webrtc.org · 10 years ago
fe41a8f Adding thread annotations to parts of Audio Coding Module by henrik.lundin@webrtc.org · 10 years ago
2812b59 Re-enables CommonFormats test for Android. by bjornv@webrtc.org · 10 years ago
174a674 Enable -Wall, -Wextra and -Wunused-variable for talk/ on clang. by pbos@webrtc.org · 10 years ago
8a09af3 Fix the build error from OpenSSLStreamAdapter::SSLVerifyCallback by jiayl@webrtc.org · 10 years ago
360507b VideoCaptureAndroid: don't synchronized on camera thread. by fischman@webrtc.org · 10 years ago
0163674 Make OpenSSLStreamAdapter verify the leaf certificate digest for chained certificates. by jiayl@webrtc.org · 10 years ago
222d8d3 Add a TSAN suppression for a benign TRACE_EVENT race. by andrew@webrtc.org · 10 years ago
56d1146 Fix AppRTC target configuration in libjingle_examples.gyp. by tkchin@webrtc.org · 10 years ago
acca675 Implement mac version of AppRTCDemo. by tkchin@webrtc.org · 10 years ago
9f8164c Fix two bugs in DataChannel state transition. by jiayl@webrtc.org · 10 years ago
1fddd61 Add a Reset() method to AudioFrame. by andrew@webrtc.org · 10 years ago
af48aaa Disable AudioCodingModuleMtTest due to memcheck and tsan failures. by andrew@webrtc.org · 10 years ago
1678db9 (Auto)update libjingle 68230113-> 68244456 by buildbot@webrtc.org · 10 years ago
288bd15 Multi-threaded test for Audio Coding Module by henrik.lundin@webrtc.org · 10 years ago
b4e3c25 Add native_test dependency to webrtc_perf_tests. by pbos@webrtc.org · 10 years ago
420b256 Fix bug where RTP headers in the packet history were replaced with the RTX wrapped headers. by stefan@webrtc.org · 10 years ago
a816180 Fixing a bug regarding VOE packet loss rate feedback to ACM by minyue@webrtc.org · 10 years ago
6e732c6 Revert 6272 "Update generated asm offsets scripts." by sprang@webrtc.org · 10 years ago
540a225 (Auto)update libjingle 68230011-> 68230113 by buildbot@webrtc.org · 10 years ago
35efb83 Implement new-API test RecvStreamWithoutRtx. by pbos@webrtc.org · 10 years ago
c34bb3a Log default receive stream creation. by pbos@webrtc.org · 10 years ago
1986474 Implement and fix new-API NackIsEnabled test. by pbos@webrtc.org · 10 years ago
1d66be2 (Auto)update libjingle 68203780-> 68206793 by buildbot@webrtc.org · 10 years ago
8dcd43c Make MediaSessionDescriptionFactory accept offers with UDP/TLS/RTP/SAVPF. by jiayl@webrtc.org · 10 years ago
abe01dd AppRTCDemo(android): run in full-screen & immersive mode. by fischman@webrtc.org · 10 years ago
21a5d44 Increase VPMVideoDecimator's initial max_frame_rate_ to 60, which allow us potentially do 60fps. by wu@webrtc.org · 10 years ago
7a9a3b7 * Revert clock.cc changes made in 6178, but keep the changes to the test. by wu@webrtc.org · 10 years ago
2a8efa8 Update generated asm offsets scripts. by fgalligan@google.com · 10 years ago