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gerrit-public.fairphone.software / platform / external / webrtc / 74767401f2b73648c6b62aa072a2198528c3f9b6
  1. 7476740 Fix a bug preventing FilePlayer from playing encoded wav files by henrik.lundin@webrtc.org · 10 years ago
  2. 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
  3. 727ff69 (Auto)update libjingle 67872893-> 67873348 by buildbot@webrtc.org · 10 years ago
  4. 75cb3dc (Auto)update libjingle 67869540-> 67872893 by buildbot@webrtc.org · 10 years ago
  5. b445f26 Fixing correct UMA metric for PeerConnection enabled with IPv4 Vs IPv6. by mallinath@webrtc.org · 10 years ago
  6. 440e1d1 vie_autotest_android.cc: stop referring to undefined functions. by fischman@webrtc.org · 10 years ago
  7. 4610f1d Roll chromium_revision 266514:272489 by fischman@webrtc.org · 10 years ago
  8. ddc79d0 Rebase webrtc/base with r6232: by henrike@webrtc.org · 10 years ago
  9. 39eccef Disable ChannelManagerTest.StartupShutdownOnUnstartedThread by fischman@webrtc.org · 10 years ago
  10. 7aa1a47 (Auto)update libjingle 67848628-> 67848776 by buildbot@webrtc.org · 10 years ago
  11. e5063b1 Thread: delete racy API (Release()) and fix racy code (started()). by fischman@webrtc.org · 10 years ago
  12. 18f41b8 PRESUBMIT.py: accept variants on the copyright message that are present in the codebase. by fischman@webrtc.org · 10 years ago
  13. 546961a Avoid reading uninitialized values (outside baundary) in DFT arithmatic decoder of iSAC-fix. by turaj@webrtc.org · 10 years ago
  14. 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
  15. 706152d Fix uninitialized reads in IsDefaultBrowserFirefox by pbos@webrtc.org · 10 years ago
  16. 1566ee2 Revert "Revert "Remove VideoSendStreamInput::PutFrame."" by pbos@webrtc.org · 10 years ago
  17. 2cdd433 Revert "Remove VideoSendStreamInput::PutFrame." by pbos@webrtc.org · 10 years ago
  18. f3085e4 Remove VideoSendStreamInput::PutFrame. by pbos@webrtc.org · 10 years ago
  19. 6e98ef4 Fix deadlock in RegisterPreDecodeImageCallback. by pbos@webrtc.org · 10 years ago
  20. bc524ae Added mirror of gtest-parallel. by pbos@webrtc.org · 10 years ago
  21. b60bfe4 Suppress webrtc trace races detected by tsan. by stefan@webrtc.org · 10 years ago
  22. 10f871f Remove the restriction to allow having both webrtc and talk changes in the same cl. by wu@webrtc.org · 10 years ago
  23. 0720758 Bump WebRTC version number to 3.54 TBR=wu@webrtc.org by tnakamura@webrtc.org · 10 years ago
  24. 1bb5da0 Adds missing include of assert header. by henrike@webrtc.org · 10 years ago
  25. 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
  26. 8e755c1 Connect SignalDestroyed in AllocationSequence after TURN ports are destroyed by mallinath@webrtc.org · 10 years ago
  27. 88fbb2d Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h. by henrike@webrtc.org · 10 years ago
  28. 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
  29. f9f1bfb (Auto)update libjingle 67686255-> 67689476 by buildbot@webrtc.org · 10 years ago
  30. 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
  31. ce4201d (Auto)update libjingle 67643194-> 67686255 by buildbot@webrtc.org · 10 years ago
  32. 7ca277b Initializes WINDOWPLACEMENT::length in GetCroppedWindowRect. by jiayl@webrtc.org · 10 years ago
  33. 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
  34. 3b7e282 Disabling systematically failing by mcasas@webrtc.org · 10 years ago
  35. 2fa7f79 Revert 6202 "Switch to using base/constructormagic.h and remove ..." by mcasas@webrtc.org · 10 years ago
  36. c2213b6 Revert 6208 "Patch from henrike@webrtc.org" by mcasas@webrtc.org · 10 years ago
  37. 86df8ac Patch from henrike@webrtc.org by mcasas@webrtc.org · 10 years ago
  38. 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
  39. 49a6a27 (Auto)update libjingle 67555838-> 67643194 by buildbot@webrtc.org · 10 years ago
  40. 82c4b85 Calculate capture ntp timestamp in local timebase for decoded audio frame. by wu@webrtc.org · 10 years ago
  41. 48438c2 Enabling NetEq bit-exactness test for Win x64 by henrik.lundin@webrtc.org · 10 years ago
  42. aed31fe Modifying WATCHLISTS by henrik.lundin@webrtc.org · 10 years ago
  43. 125ffd7 Switch to using base/constructormagic.h and remove system_wrappers/interface/constructor_magic.h. by henrike@webrtc.org · 10 years ago
  44. 4059c2f Disable CallPerfTest.CaptureNtpTimeWithNetworkDelay due to being flaky. by stefan@webrtc.org · 10 years ago
  45. 70bb2d5 Revert r6198 "Expose the original packet length in in the RTP play tools." by stefan@webrtc.org · 10 years ago
  46. 83599cb Reenable WebRtcVideoEngineTestFake.SendReceiveBitratesStats under DrMemory. by stefan@webrtc.org · 10 years ago
  47. e208458 Expose the original packet length in in the RTP play tools. by stefan@webrtc.org · 10 years ago
  48. be4ab99 Disabling RealFFTTest.RealAndComplexMatch and AudioProcessingTest.Formats as they currently are broken with gcc 4.8. by stefan@webrtc.org · 10 years ago
  49. a36db97 Suppress GMOCK printouts from TestVideoSenderWithVp8 by henrik.lundin@webrtc.org · 10 years ago
  50. f3e1341 VoEVolumeTest: Enabled Linux flaky tests by bjornv@webrtc.org · 10 years ago
  51. a826006 Add NACK and RPSI packet types to RTCP packet builder. by asapersson@webrtc.org · 10 years ago
  52. 2db9f45 Reduce flakiness of voe_auto_test MixingTest by checking dumped audio size by minyue@webrtc.org · 10 years ago
  53. 1732a59 Add a UIView for rendering a video track. by tkchin@webrtc.org · 10 years ago
  54. 7ca1edb Remove IOKit linkage from iOS builds. by tkchin@webrtc.org · 10 years ago
  55. 40bc777 talk_base: remove lock inversion between MessageQueue and MessageQueueManager. by fischman@webrtc.org · 10 years ago
  56. cb711f7 Add interface to propagate audio capture timestamp to the renderer. by wu@webrtc.org · 10 years ago
  57. ebb467f Avoid NACK-list flush error on keyframe packets. by pbos@webrtc.org · 10 years ago
  58. 64339a7 Don't crash if a frame returned from the decoder is too old. by stefan@webrtc.org · 10 years ago
  59. 725e582 Use the new gyp_var_prefix local variable set by gyp instead of the by michaelbai@google.com · 10 years ago
  60. 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
  61. a3b5673 common_audio/signal_processing: Removes macro WEBRTC_SPL_UMUL_RSFT16 by bjornv@webrtc.org · 10 years ago
  62. 1e019d1 Fix delivery error-checking missed in r6151. by pbos@webrtc.org · 10 years ago
  63. 57e0602 Fix flaky test SendRtpRtcpHeaderExtensionsTest.SentPackets*. by solenberg@webrtc.org · 10 years ago
  64. 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
  65. 1b21a57 common_audio/signal_processing: Removed macro WEBRTC_SPL_SUB_SAT_W16 by bjornv@webrtc.org · 10 years ago
  66. d83d607 common_audio/signal_processing: Removed macro WEBRTC_SPL_MAX_SEED_USED by bjornv@webrtc.org · 10 years ago
  67. 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
  68. bf58a75 removed webrtc_base_tests_utils from merge libs as it was breaking some builds. by henrike@webrtc.org · 10 years ago
  69. 508795f Made the presubmit script accept license headers back to 2003 by henrike@webrtc.org · 10 years ago
  70. 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
  71. 6bfd619 (Auto)update libjingle 67052073-> 67134648 by buildbot@webrtc.org · 10 years ago
  72. 6aeeac9 Fix Windows debug compile of overrides/ logging. by pbos@webrtc.org · 10 years ago
  73. d5da250 Revert "Revert "Audio processing: Feed each processing step its choice by mflodman@webrtc.org · 10 years ago
  74. 024e4d5 Fix Win VideoSendStream::...::ToString() compiles. by pbos@webrtc.org · 10 years ago
  75. 1e92b0a Add ToString() to VideoSendStream::Config. by pbos@webrtc.org · 10 years ago
  76. 1aae6bf common_audio: Removes unused macros by bjornv@webrtc.org · 10 years ago
  77. b4e80e0 Re-enable almost all NetEqDecodingTests for Android by henrik.lundin@webrtc.org · 10 years ago
  78. 7cb4752 WebRTCDemo: couldn't run a second time. The reason is voe could register/unregister for each run, but vie would expect initialization only once per process. by braveyao@webrtc.org · 10 years ago
  79. 54231f0 Ignore the return value of UpdateRtcpTimestamp instead of printing warning. Because UpdateRtcpTimestamp may fail when there's no valid RTCP SR, which can happen in the first couple seconds or when the channel is a send only channel. Either case we don't want the warning log. by wu@webrtc.org · 10 years ago
  80. bb6201a TCP remote socket address should have both server hostname and IP address. by mallinath@webrtc.org · 10 years ago
  81. a150bc9 PeerConnection(android): allow initializing either (or neither) of {Voice,Video}Engine. by fischman@webrtc.org · 10 years ago
  82. ef5a752 (Auto)update libjingle 67043374-> 67044055 by buildbot@webrtc.org · 10 years ago
  83. 3e92468 (Auto)update libjingle 67037200-> 67043374 by buildbot@webrtc.org · 10 years ago
  84. 4f58014 Drop the DataChannel message if it's received when the channel is not open. by jiayl@webrtc.org · 10 years ago
  85. 372701a (Auto)update libjingle 67023528-> 67036361 by buildbot@webrtc.org · 10 years ago
  86. 21299d4 Remove the use of AudioFrame::energy_ from AudioProcessing and VoE. by andrew@webrtc.org · 10 years ago
  87. 688ed69 (Auto)update libjingle 67017551-> 67023528 by buildbot@webrtc.org · 10 years ago
  88. c50bf7c Added namespace rtc to some base classes and functions. It was causing linker error in the FYI bots: http://chromegw.corp.google.com/i/internal.chromium.webrtc.fyi/builders/Android%20Builder%20%28dbg%29/builds/1808/steps/compile/logs/stdio but also, not doing it pollutes the global namespace. by henrike@webrtc.org · 10 years ago
  89. 3147b97 LSan suppressions for libjingle tests (fix) by kjellander@webrtc.org · 10 years ago
  90. 7c0f6e1 LSan suppressions for libjingle tests (more) by kjellander@webrtc.org · 10 years ago
  91. 2c98af7 PeerConnection(Java): auto-WrapCurrentThread() when creating PeerConnectionFactory. by fischman@webrtc.org · 10 years ago
  92. a70dff4 LSan suppressions for libjingle tests. by kjellander@webrtc.org · 10 years ago
  93. 88abf11 Move the capture ntp computing code to ntp_calculator so that later it can be shared with voe. by wu@webrtc.org · 10 years ago
  94. 4e545cc Update webrtcvideoengine2.cc to use DeliveryStatus. by pbos@webrtc.org · 10 years ago
  95. caba2d2 Add DeliveryStatus enum to DeliverPacket(). by pbos@webrtc.org · 10 years ago
  96. 581e217 Fix libjingle to provide a field_trial implementation. by andresp@webrtc.org · 10 years ago
  97. 01edf2e Updating LSan third party suppressions. by kjellander@webrtc.org · 10 years ago
  98. a36ad69 Add webrtc field trials API. by andresp@webrtc.org · 10 years ago
  99. 9f27735 Removes parts of the webrtc::VoEDtmf sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 10 years ago
  100. f383a1b Removes parts of the webrtc::VoEVolumeControl sub API as part of a clean-up operation where the goal is to remove unused APIs. by henrika@webrtc.org · 10 years ago