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gerrit-public.fairphone.software / platform / external / webrtc / 1d66be22c8f929e1170f288472aac9d4b44b7a05
  1. 1d66be2 (Auto)update libjingle 68203780-> 68206793 by buildbot@webrtc.org · 10 years ago
  2. 8dcd43c Make MediaSessionDescriptionFactory accept offers with UDP/TLS/RTP/SAVPF. by jiayl@webrtc.org · 10 years ago
  3. abe01dd AppRTCDemo(android): run in full-screen & immersive mode. by fischman@webrtc.org · 10 years ago
  4. 21a5d44 Increase VPMVideoDecimator's initial max_frame_rate_ to 60, which allow us potentially do 60fps. by wu@webrtc.org · 10 years ago
  5. 7a9a3b7 * Revert clock.cc changes made in 6178, but keep the changes to the test. by wu@webrtc.org · 10 years ago
  6. 2a8efa8 Update generated asm offsets scripts. by fgalligan@google.com · 10 years ago
  7. caa01b1 Rebase webrtc/base with r6250: by henrike@webrtc.org · 10 years ago
  8. 5dc51fb Closes the DataChannel when the send buffer is full or on transport errors. by jiayl@webrtc.org · 10 years ago
  9. 001fd2d Fire OnRenegotiationNeeded only for the first SCTP DataChannel. by jiayl@webrtc.org · 10 years ago
  10. 9aa7d8d Increase the threshold for CallPerfTest.CaptureNtpTimeWithNetworkDelay to avoid flaky. by wu@webrtc.org · 10 years ago
  11. d6a0efd VideoCaptureAndroid: quit & join the camera thread on stopCapture. by fischman@webrtc.org · 10 years ago
  12. 43a1395 AppRTCDemo(android): README updates for a shrinking envsetup.sh world. by fischman@webrtc.org · 10 years ago
  13. b364016 Revert r6161 "Drop the DataChannel message if it's received when the channel is not open." by jiayl@webrtc.org · 10 years ago
  14. f15c14b Echo canceler: Saturate output to guarantee it'll be in the allowed range by kwiberg@webrtc.org · 10 years ago
  15. 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 · 10 years ago
  16. aca5939 common_audio/signal_processing: Fixes arm compilation issues with gcc 4.8 by bjornv@webrtc.org · 10 years ago
  17. 0aa3ee6 Better buffer size estimation in NetEq for redundant packets by minyue@webrtc.org · 10 years ago
  18. 1b9df05 Revert 6257 "Rename neteq4 folder to neteq" by henrik.lundin@webrtc.org · 10 years ago
  19. 637c55f Add support of texture frames for video capturer. by wuchengli@chromium.org · 10 years ago
  20. a90f6d6 Rename neteq4 folder to neteq by henrik.lundin@webrtc.org · 10 years ago
  21. 27e884c Disable MouseCursorMonitorTest due to flake on Windows. by andrew@webrtc.org · 10 years ago
  22. 0ef565e Roll libvpx 267596:269083 by marpan@webrtc.org · 10 years ago
  23. 033aa22 video_engine_tests_apk: enable running by adding nativeRunTests dependency. by fischman@webrtc.org · 10 years ago
  24. 89e8ffb Revert "Add support of texture frames for video capturer." by wuchengli@chromium.org · 10 years ago
  25. efe1535 Add support of texture frames for video capturer. by wuchengli@chromium.org · 10 years ago
  26. 59336e8 Adding R/W lock to SimulatedClock by henrik.lundin@webrtc.org · 10 years ago
  27. f666ecc Disabling flaky libjingle tests after fixit week. by phoglund@webrtc.org · 10 years ago
  28. ab6bf4f Added api for getting cpu measures using a struct. by asapersson@webrtc.org · 10 years ago
  29. 7476740 Fix a bug preventing FilePlayer from playing encoded wav files by henrik.lundin@webrtc.org · 10 years ago
  30. 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 · 10 years ago
  31. 727ff69 (Auto)update libjingle 67872893-> 67873348 by buildbot@webrtc.org · 10 years ago
  32. 75cb3dc (Auto)update libjingle 67869540-> 67872893 by buildbot@webrtc.org · 10 years ago
  33. b445f26 Fixing correct UMA metric for PeerConnection enabled with IPv4 Vs IPv6. by mallinath@webrtc.org · 10 years ago
  34. 440e1d1 vie_autotest_android.cc: stop referring to undefined functions. by fischman@webrtc.org · 10 years ago
  35. 4610f1d Roll chromium_revision 266514:272489 by fischman@webrtc.org · 10 years ago
  36. ddc79d0 Rebase webrtc/base with r6232: by henrike@webrtc.org · 10 years ago
  37. 39eccef Disable ChannelManagerTest.StartupShutdownOnUnstartedThread by fischman@webrtc.org · 10 years ago
  38. 7aa1a47 (Auto)update libjingle 67848628-> 67848776 by buildbot@webrtc.org · 10 years ago
  39. e5063b1 Thread: delete racy API (Release()) and fix racy code (started()). by fischman@webrtc.org · 10 years ago
  40. 18f41b8 PRESUBMIT.py: accept variants on the copyright message that are present in the codebase. by fischman@webrtc.org · 10 years ago
  41. 546961a Avoid reading uninitialized values (outside baundary) in DFT arithmatic decoder of iSAC-fix. by turaj@webrtc.org · 10 years ago
  42. 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 · 10 years ago
  43. 706152d Fix uninitialized reads in IsDefaultBrowserFirefox by pbos@webrtc.org · 10 years ago
  44. 1566ee2 Revert "Revert "Remove VideoSendStreamInput::PutFrame."" by pbos@webrtc.org · 10 years ago
  45. 2cdd433 Revert "Remove VideoSendStreamInput::PutFrame." by pbos@webrtc.org · 10 years ago
  46. f3085e4 Remove VideoSendStreamInput::PutFrame. by pbos@webrtc.org · 10 years ago
  47. 6e98ef4 Fix deadlock in RegisterPreDecodeImageCallback. by pbos@webrtc.org · 10 years ago
  48. bc524ae Added mirror of gtest-parallel. by pbos@webrtc.org · 10 years ago
  49. b60bfe4 Suppress webrtc trace races detected by tsan. by stefan@webrtc.org · 10 years ago
  50. 10f871f Remove the restriction to allow having both webrtc and talk changes in the same cl. by wu@webrtc.org · 10 years ago
  51. 0720758 Bump WebRTC version number to 3.54 TBR=wu@webrtc.org by tnakamura@webrtc.org · 10 years ago
  52. 1bb5da0 Adds missing include of assert header. by henrike@webrtc.org · 10 years ago
  53. 21f7d6d WebRTCDemo: move the deletion of CritSect to end of the dtor to fix a crash in Android video renderer. by braveyao@webrtc.org · 10 years ago
  54. 8e755c1 Connect SignalDestroyed in AllocationSequence after TURN ports are destroyed by mallinath@webrtc.org · 10 years ago
  55. 88fbb2d Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h. by henrike@webrtc.org · 10 years ago
  56. 99b4162 Rebase webrtc/base 6163:6216 (svn diff -r 6163:6216 http://webrtc.googlecode.com/svn/trunk/talk/base, apply diff manually) by henrike@webrtc.org · 10 years ago
  57. f9f1bfb (Auto)update libjingle 67686255-> 67689476 by buildbot@webrtc.org · 10 years ago
  58. a148704 Rename webrtc/base's IS_ALIGNED macro to RTC_IS_ALIGNED to avoid conflict between webrtc/base/basictypes.h and third_party/.../vpx_codec.h. by henrike@webrtc.org · 10 years ago
  59. ce4201d (Auto)update libjingle 67643194-> 67686255 by buildbot@webrtc.org · 10 years ago
  60. 7ca277b Initializes WINDOWPLACEMENT::length in GetCroppedWindowRect. by jiayl@webrtc.org · 10 years ago
  61. 000658a Revert of 6211 as it was committed despite of PRESUBMIT.py warning. The commit breaks the sync bot. by henrike@webrtc.org · 10 years ago
  62. 3b7e282 Disabling systematically failing by mcasas@webrtc.org · 10 years ago
  63. 2fa7f79 Revert 6202 "Switch to using base/constructormagic.h and remove ..." by mcasas@webrtc.org · 10 years ago
  64. c2213b6 Revert 6208 "Patch from henrike@webrtc.org" by mcasas@webrtc.org · 10 years ago
  65. 86df8ac Patch from henrike@webrtc.org by mcasas@webrtc.org · 10 years ago
  66. 1a79bb8 WebRTCDemo: clean the error message due to API clean up and add ability to route the audio through all three outputs, headset/earpiece/loudspeaker by braveyao@webrtc.org · 10 years ago
  67. 49a6a27 (Auto)update libjingle 67555838-> 67643194 by buildbot@webrtc.org · 10 years ago
  68. 82c4b85 Calculate capture ntp timestamp in local timebase for decoded audio frame. by wu@webrtc.org · 10 years ago
  69. 48438c2 Enabling NetEq bit-exactness test for Win x64 by henrik.lundin@webrtc.org · 10 years ago
  70. aed31fe Modifying WATCHLISTS by henrik.lundin@webrtc.org · 10 years ago
  71. 125ffd7 Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h. by henrike@webrtc.org · 10 years ago
  72. 4059c2f Disable CallPerfTest.CaptureNtpTimeWithNetworkDelay due to being flaky. by stefan@webrtc.org · 10 years ago
  73. 70bb2d5 Revert r6198 "Expose the original packet length in in the RTP play tools." by stefan@webrtc.org · 10 years ago
  74. 83599cb Reenable WebRtcVideoEngineTestFake.SendReceiveBitratesStats under DrMemory. by stefan@webrtc.org · 10 years ago
  75. e208458 Expose the original packet length in in the RTP play tools. by stefan@webrtc.org · 10 years ago
  76. be4ab99 Disabling RealFFTTest.RealAndComplexMatch and AudioProcessingTest.Formats as they currently are broken with gcc 4.8. by stefan@webrtc.org · 10 years ago
  77. a36db97 Suppress GMOCK printouts from TestVideoSenderWithVp8 by henrik.lundin@webrtc.org · 10 years ago
  78. f3e1341 VoEVolumeTest: Enabled Linux flaky tests by bjornv@webrtc.org · 10 years ago
  79. a826006 Add NACK and RPSI packet types to RTCP packet builder. by asapersson@webrtc.org · 10 years ago
  80. 2db9f45 Reduce flakiness of voe_auto_test MixingTest by checking dumped audio size by minyue@webrtc.org · 10 years ago
  81. 1732a59 Add a UIView for rendering a video track. by tkchin@webrtc.org · 10 years ago
  82. 7ca1edb Remove IOKit linkage from iOS builds. by tkchin@webrtc.org · 10 years ago
  83. 40bc777 talk_base: remove lock inversion between MessageQueue and MessageQueueManager. by fischman@webrtc.org · 10 years ago
  84. cb711f7 Add interface to propagate audio capture timestamp to the renderer. by wu@webrtc.org · 10 years ago
  85. ebb467f Avoid NACK-list flush error on keyframe packets. by pbos@webrtc.org · 10 years ago
  86. 64339a7 Don't crash if a frame returned from the decoder is too old. by stefan@webrtc.org · 10 years ago
  87. 725e582 Use the new gyp_var_prefix local variable set by gyp instead of the by michaelbai@google.com · 10 years ago
  88. 14abcc7 libvpx's UNUSED macro conflicts with webrtc/base's. Added missing include of assert.h. Globally defined function "Unused" in talk/base and its copy (webrtc/base) is causing a conflict. by henrike@webrtc.org · 10 years ago
  89. a3b5673 common_audio/signal_processing: Removes macro WEBRTC_SPL_UMUL_RSFT16 by bjornv@webrtc.org · 10 years ago
  90. 1e019d1 Fix delivery error-checking missed in r6151. by pbos@webrtc.org · 10 years ago
  91. 57e0602 Fix flaky test SendRtpRtcpHeaderExtensionsTest.SentPackets*. by solenberg@webrtc.org · 10 years ago
  92. 60015d2 Wire up --force_fieldtrials for vie_auto_test and for test targets linking with test/test.gyp:{test_main|test_support_main} by andresp@webrtc.org · 10 years ago
  93. 1b21a57 common_audio/signal_processing: Removed macro WEBRTC_SPL_SUB_SAT_W16 by bjornv@webrtc.org · 10 years ago
  94. d83d607 common_audio/signal_processing: Removed macro WEBRTC_SPL_MAX_SEED_USED by bjornv@webrtc.org · 10 years ago
  95. 75718cf * Implement WindowsRealTimeClock::CurrentTimeVal with GetSystemTimeAsFileTime as it supposes to return a POSIX gettimeofday, so that later it can be converted to NTP timee correctly. by wu@webrtc.org · 10 years ago
  96. bf58a75 removed webrtc_base_tests_utils from merge libs as it was breaking some builds. by henrike@webrtc.org · 10 years ago
  97. 508795f Made the presubmit script accept license headers back to 2003 by henrike@webrtc.org · 10 years ago
  98. cfdf420 Rebase webrtc/base 6129:6163 (svn diff -r 6129:6163 http://webrtc.googlecode.com/svn/trunk/talk/base apply diff manually) by henrike@webrtc.org · 10 years ago
  99. 6bfd619 (Auto)update libjingle 67052073-> 67134648 by buildbot@webrtc.org · 10 years ago
  100. 6aeeac9 Fix Windows debug compile of overrides/ logging. by pbos@webrtc.org · 10 years ago