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webrtc / src.git / d3dcebf6b4e9d057561d94ef36f4aed54b912709
  1. d3dcebf Disable P2PTransportChannelMultihomedTest.TestFailover under Memcheck by henrik.lundin@webrtc.org · 11 years ago
  2. 147f4fe Disables SystemDelayTest.CorrectDelayDuringDrift on Android by bjornv@webrtc.org · 11 years ago
  3. b616e12 Disables some modules_unittests on Android. by bjornv@webrtc.org · 11 years ago
  4. 4436b44 Moved verbose logging in rtcp_receiver.cc to LS_VERBOSE. by andresp@webrtc.org · 11 years ago
  5. 2bdd399 Suppress memcheck error in VideoProcessorIntegrationTest by henrik.lundin@webrtc.org · 11 years ago
  6. 19fc09e Adding missing break in media_file_utility.cc. by mflodman@webrtc.org · 11 years ago
  7. 0cdcd23 (Auto)update libjingle 68501302-> 68506654 by buildbot@webrtc.org · 11 years ago
  8. af81b9b (Auto)update libjingle 68499439-> 68501302 by buildbot@webrtc.org · 11 years ago
  9. 251fdf6 (Auto)update libjingle 68495561-> 68499439 by buildbot@webrtc.org · 11 years ago
  10. 09a71cd talk/ios: Fixes source after corrupt sync in r6305 (which corrupted r6291). by henrike@webrtc.org · 11 years ago
  11. 5321784 (Auto)update libjingle 68465410-> 68487517 by buildbot@webrtc.org · 11 years ago
  12. 4ef254f Enable videoprocessor_integrationtest tests on android. by marpan@webrtc.org · 11 years ago
  13. 83eb7df PeerConnection(java): disable wait for flaky ICEConnection.COMPLETED. by fischman@webrtc.org · 11 years ago
  14. ddc6bc9 Revert 6312 "Re-enable AudioCodingModuleMtTest" by turaj@webrtc.org · 11 years ago
  15. 289a35c Add empty webrtcmediaengine.cc. by pbos@webrtc.org · 11 years ago
  16. 8d13cd1 Re-enable AudioCodingModuleMtTest by henrik.lundin@webrtc.org · 11 years ago
  17. 8e4401b Reformat integer accessors to look like their float counterparts by kwiberg@webrtc.org · 11 years ago
  18. f2e4a99 Add kwiberg@webrtc.org to watchlist for audio_coding and audio_processing by kwiberg@webrtc.org · 11 years ago
  19. b525a9d (Auto)update libjingle 68379861-> 68445177 by buildbot@webrtc.org · 11 years ago
  20. 044bdac Remove kMaxWaitForStatsMs from tsanv2 compilation. by pbos@webrtc.org · 11 years ago
  21. c0035a6 Remove an optimization that's no longer worth the extra complexity it causes by kwiberg@webrtc.org · 11 years ago
  22. 34a08b4 (Auto)update libjingle 68275107-> 68379861 by buildbot@webrtc.org · 11 years ago
  23. a28c697 - Get rid of 'using' from .h by solenberg@webrtc.org · 11 years ago
  24. 2f7c7ce Remove old perf_expectations no longer used. by kjellander@webrtc.org · 11 years ago
  25. 2bd032e Disable MouseCursorMonitorTest by henrik.lundin@webrtc.org · 11 years ago
  26. 4ecae6e Disable MouseCursorMonitorTest.FromScreen by henrik.lundin@webrtc.org · 11 years ago
  27. fe41a8f Adding thread annotations to parts of Audio Coding Module by henrik.lundin@webrtc.org · 11 years ago
  28. 2812b59 Re-enables CommonFormats test for Android. by bjornv@webrtc.org · 11 years ago
  29. 174a674 Enable -Wall, -Wextra and -Wunused-variable for talk/ on clang. by pbos@webrtc.org · 11 years ago
  30. 8a09af3 Fix the build error from OpenSSLStreamAdapter::SSLVerifyCallback by jiayl@webrtc.org · 11 years ago
  31. 360507b VideoCaptureAndroid: don't synchronized on camera thread. by fischman@webrtc.org · 11 years ago
  32. 0163674 Make OpenSSLStreamAdapter verify the leaf certificate digest for chained certificates. by jiayl@webrtc.org · 11 years ago
  33. 222d8d3 Add a TSAN suppression for a benign TRACE_EVENT race. by andrew@webrtc.org · 11 years ago
  34. 56d1146 Fix AppRTC target configuration in libjingle_examples.gyp. by tkchin@webrtc.org · 11 years ago
  35. acca675 Implement mac version of AppRTCDemo. by tkchin@webrtc.org · 11 years ago
  36. 9f8164c Fix two bugs in DataChannel state transition. by jiayl@webrtc.org · 11 years ago
  37. 1fddd61 Add a Reset() method to AudioFrame. by andrew@webrtc.org · 11 years ago
  38. af48aaa Disable AudioCodingModuleMtTest due to memcheck and tsan failures. by andrew@webrtc.org · 11 years ago
  39. 1678db9 (Auto)update libjingle 68230113-> 68244456 by buildbot@webrtc.org · 11 years ago
  40. 288bd15 Multi-threaded test for Audio Coding Module by henrik.lundin@webrtc.org · 11 years ago
  41. b4e3c25 Add native_test dependency to webrtc_perf_tests. by pbos@webrtc.org · 11 years ago
  42. 420b256 Fix bug where RTP headers in the packet history were replaced with the RTX wrapped headers. by stefan@webrtc.org · 11 years ago
  43. a816180 Fixing a bug regarding VOE packet loss rate feedback to ACM by minyue@webrtc.org · 11 years ago
  44. 6e732c6 Revert 6272 "Update generated asm offsets scripts." by sprang@webrtc.org · 11 years ago
  45. 540a225 (Auto)update libjingle 68230011-> 68230113 by buildbot@webrtc.org · 11 years ago
  46. 35efb83 Implement new-API test RecvStreamWithoutRtx. by pbos@webrtc.org · 11 years ago
  47. c34bb3a Log default receive stream creation. by pbos@webrtc.org · 11 years ago
  48. 1986474 Implement and fix new-API NackIsEnabled test. by pbos@webrtc.org · 11 years ago
  49. 1d66be2 (Auto)update libjingle 68203780-> 68206793 by buildbot@webrtc.org · 11 years ago
  50. 8dcd43c Make MediaSessionDescriptionFactory accept offers with UDP/TLS/RTP/SAVPF. by jiayl@webrtc.org · 11 years ago
  51. abe01dd AppRTCDemo(android): run in full-screen & immersive mode. by fischman@webrtc.org · 11 years ago
  52. 21a5d44 Increase VPMVideoDecimator's initial max_frame_rate_ to 60, which allow us potentially do 60fps. by wu@webrtc.org · 11 years ago
  53. 7a9a3b7 * Revert clock.cc changes made in 6178, but keep the changes to the test. by wu@webrtc.org · 11 years ago
  54. 2a8efa8 Update generated asm offsets scripts. by fgalligan@google.com · 11 years ago
  55. caa01b1 Rebase webrtc/base with r6250: by henrike@webrtc.org · 11 years ago
  56. 5dc51fb Closes the DataChannel when the send buffer is full or on transport errors. by jiayl@webrtc.org · 11 years ago
  57. 001fd2d Fire OnRenegotiationNeeded only for the first SCTP DataChannel. by jiayl@webrtc.org · 11 years ago
  58. 9aa7d8d Increase the threshold for CallPerfTest.CaptureNtpTimeWithNetworkDelay to avoid flaky. by wu@webrtc.org · 11 years ago
  59. d6a0efd VideoCaptureAndroid: quit & join the camera thread on stopCapture. by fischman@webrtc.org · 11 years ago
  60. 43a1395 AppRTCDemo(android): README updates for a shrinking envsetup.sh world. by fischman@webrtc.org · 11 years ago
  61. b364016 Revert r6161 "Drop the DataChannel message if it's received when the channel is not open." by jiayl@webrtc.org · 11 years ago
  62. f15c14b Echo canceler: Saturate output to guarantee it'll be in the allowed range by kwiberg@webrtc.org · 11 years ago
  63. c1a40a7 This CL is to adding feedback of packet loss rate to encoder in voice engine. A direct reason for doing it is to make use of Opus FEC, which can adapt itself to changes in the packet loss rate. by minyue@webrtc.org · 11 years ago
  64. aca5939 common_audio/signal_processing: Fixes arm compilation issues with gcc 4.8 by bjornv@webrtc.org · 11 years ago
  65. 0aa3ee6 Better buffer size estimation in NetEq for redundant packets by minyue@webrtc.org · 11 years ago
  66. 1b9df05 Revert 6257 "Rename neteq4 folder to neteq" by henrik.lundin@webrtc.org · 11 years ago
  67. 637c55f Add support of texture frames for video capturer. by wuchengli@chromium.org · 11 years ago
  68. a90f6d6 Rename neteq4 folder to neteq by henrik.lundin@webrtc.org · 11 years ago
  69. 27e884c Disable MouseCursorMonitorTest due to flake on Windows. by andrew@webrtc.org · 11 years ago
  70. 0ef565e Roll libvpx 267596:269083 by marpan@webrtc.org · 11 years ago
  71. 033aa22 video_engine_tests_apk: enable running by adding nativeRunTests dependency. by fischman@webrtc.org · 11 years ago
  72. 89e8ffb Revert "Add support of texture frames for video capturer." by wuchengli@chromium.org · 11 years ago
  73. efe1535 Add support of texture frames for video capturer. by wuchengli@chromium.org · 11 years ago
  74. 59336e8 Adding R/W lock to SimulatedClock by henrik.lundin@webrtc.org · 11 years ago
  75. f666ecc Disabling flaky libjingle tests after fixit week. by phoglund@webrtc.org · 11 years ago
  76. ab6bf4f Added api for getting cpu measures using a struct. by asapersson@webrtc.org · 11 years ago
  77. 7476740 Fix a bug preventing FilePlayer from playing encoded wav files by henrik.lundin@webrtc.org · 11 years ago
  78. 1457b47 First incoming packet was not accounted for in receive stats. Changed call order for incoming packet to receive statistics class. by asapersson@webrtc.org · 11 years ago
  79. 727ff69 (Auto)update libjingle 67872893-> 67873348 by buildbot@webrtc.org · 11 years ago
  80. 75cb3dc (Auto)update libjingle 67869540-> 67872893 by buildbot@webrtc.org · 11 years ago
  81. b445f26 Fixing correct UMA metric for PeerConnection enabled with IPv4 Vs IPv6. by mallinath@webrtc.org · 11 years ago
  82. 440e1d1 vie_autotest_android.cc: stop referring to undefined functions. by fischman@webrtc.org · 11 years ago
  83. 4610f1d Roll chromium_revision 266514:272489 by fischman@webrtc.org · 11 years ago
  84. ddc79d0 Rebase webrtc/base with r6232: by henrike@webrtc.org · 11 years ago
  85. 39eccef Disable ChannelManagerTest.StartupShutdownOnUnstartedThread by fischman@webrtc.org · 11 years ago
  86. 7aa1a47 (Auto)update libjingle 67848628-> 67848776 by buildbot@webrtc.org · 11 years ago
  87. e5063b1 Thread: delete racy API (Release()) and fix racy code (started()). by fischman@webrtc.org · 11 years ago
  88. 18f41b8 PRESUBMIT.py: accept variants on the copyright message that are present in the codebase. by fischman@webrtc.org · 11 years ago
  89. 546961a Avoid reading uninitialized values (outside baundary) in DFT arithmatic decoder of iSAC-fix. by turaj@webrtc.org · 11 years ago
  90. aa5ea1c 1. Make a clear distinction between codec internal FEC and RED, confusing mentioning of FEC in the old codes is replaced by RED by minyue@webrtc.org · 11 years ago
  91. 706152d Fix uninitialized reads in IsDefaultBrowserFirefox by pbos@webrtc.org · 11 years ago
  92. 1566ee2 Revert "Revert "Remove VideoSendStreamInput::PutFrame."" by pbos@webrtc.org · 11 years ago
  93. 2cdd433 Revert "Remove VideoSendStreamInput::PutFrame." by pbos@webrtc.org · 11 years ago
  94. f3085e4 Remove VideoSendStreamInput::PutFrame. by pbos@webrtc.org · 11 years ago
  95. 6e98ef4 Fix deadlock in RegisterPreDecodeImageCallback. by pbos@webrtc.org · 11 years ago
  96. bc524ae Added mirror of gtest-parallel. by pbos@webrtc.org · 11 years ago
  97. b60bfe4 Suppress webrtc trace races detected by tsan. by stefan@webrtc.org · 11 years ago
  98. 10f871f Remove the restriction to allow having both webrtc and talk changes in the same cl. by wu@webrtc.org · 11 years ago
  99. 0720758 Bump WebRTC version number to 3.54 TBR=wu@webrtc.org by tnakamura@webrtc.org · 11 years ago
  100. 1bb5da0 Adds missing include of assert header. by henrike@webrtc.org · 11 years ago