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gerrit-public.fairphone.software / platform / external / webrtc / ba92c52570d3e803f7e5d0a84295c3de885ad703
  1. ba92c52 Disable GetStats on DrMemory. by pbos@webrtc.org · 11 years ago
  2. 026859b This is related to an earlier CL of enabling Opus 48 kHz. by minyue@webrtc.org · 11 years ago
  3. e6f84ae Initial WebRtcVideoEngine2::GetStats(). by pbos@webrtc.org · 11 years ago
  4. e9e4253 Sleep in ThreadTest thread functions. by pbos@webrtc.org · 11 years ago
  5. d1ea06b Restart VideoReceiveStreams in WebRtcVideoEngine2. by pbos@webrtc.org · 11 years ago
  6. c31651d (Auto)update libjingle 71378257-> 71410012 by buildbot@webrtc.org · 11 years ago
  7. e364ac9 AudioBuffer: Optimize const accesses to arrays that autoconvert int16<->float by kwiberg@webrtc.org · 11 years ago
  8. c145668 Reduce runtime of RingBufferTest by a factor of 100. by andrew@webrtc.org · 11 years ago
  9. 4f5da03 Use _numMixedParticipants instead of audioFrameList->size() to determine if there're more than one participants. by wu@webrtc.org · 11 years ago
  10. aa93611 Connect to the turn server if address cannot be resolved by the browser by using by mallinath@webrtc.org · 11 years ago
  11. e5995aa Assigning a priority to TURN server list passed to PeerConnection. First entry in the TURN server list will get the highest priotity and so forth. by mallinath@webrtc.org · 11 years ago
  12. e10d28c fix by jiayl@webrtc.org · 11 years ago
  13. 8b94e3d Fix issue where padding is sent before media with undefined timestamps if not abs-send-time is enabled. by stefan@webrtc.org · 11 years ago
  14. 4065988 Remove unused ExperimentalNS API in AudioProcessing by aluebs@webrtc.org · 11 years ago
  15. 2b6bc8d AudioBuffer: Eliminate the SplitChannelBuffer class by kwiberg@webrtc.org · 11 years ago
  16. 5301b0f Move additional state into WebRtcVideoSendStream. by pbos@webrtc.org · 11 years ago
  17. 2561d52 Simplify AudioBuffer::mixed_low_pass_data API by aluebs@webrtc.org · 11 years ago
  18. af93fc0 AudioBuffer: Let ChannelBuffer handle bounds checking of channel parameter by kwiberg@webrtc.org · 11 years ago
  19. 2ade42b Add unit test for MediaFile WAV file writing by kwiberg@webrtc.org · 11 years ago
  20. 4a472fb Fixes up rtc so that it compiles on iOS 8 SDK. by tkchin@webrtc.org · 11 years ago
  21. 52eddec Revert 6707 "Add support of multiple STUN servers in UDPPort." by wu@webrtc.org · 11 years ago
  22. c56ae63 r6709 lacks a change in BUILD.gn by minyue@webrtc.org · 11 years ago
  23. 74aaf29 Raw packet loss rate reported by RTP_RTCP module may vary too drastically over time. This CL is to add a filter to the value in VoE before lending it to audio coding module. by minyue@webrtc.org · 11 years ago
  24. 4c3e991 Make sure b lines appear before all the a lines. Per RFC 4566, the order of media description should be: by wu@webrtc.org · 11 years ago
  25. 46fb331 Add support of multiple STUN servers in UDPPort. by jiayl@webrtc.org · 11 years ago
  26. 2e3c97d Compile-time guard for iOS7 specific property. by tkchin@webrtc.org · 11 years ago
  27. a8d8ad2 (Auto)update libjingle 71240799-> 71250251 by buildbot@webrtc.org · 11 years ago
  28. 4070b1d Print an info log instead of return an error if an external encoder is de-registered, but no corresponding internal encoder can be registered automatically. by stefan@webrtc.org · 11 years ago
  29. 63c60ed Remove old padding path in RTPSender. by pbos@webrtc.org · 11 years ago
  30. efb81d8 int16<->float conversions: Use size_t for array length argument, not int by kwiberg@webrtc.org · 11 years ago
  31. 0fa6366 Define convenient FATAL_ERROR() and FATAL_ERROR_IF() macros by kwiberg@webrtc.org · 11 years ago
  32. e8ea33c nrsh1 is written before tmp321 is read, so needs to be earlyclobber by kwiberg@webrtc.org · 11 years ago
  33. 38ce7d0 Implement unittest for SetSendCodecsChangesExistingStreams. by pbos@webrtc.org · 11 years ago
  34. bac5f0f Fix an invalid memory access due to typo in win/cursor.cc. by jiayl@webrtc.org · 11 years ago
  35. 122caa5 After an audio interruption the audio unit no longer invokes its render callback, which results in a loss of audio. Restarting the audio unit post interruption fixes the issue. by tkchin@webrtc.org · 11 years ago
  36. 4721895 Minor refactoring of StatsCollector. by tommi@webrtc.org · 11 years ago
  37. 42fe435 Remove Thread::RunningForChannelManager(). by tkchin@webrtc.org · 11 years ago
  38. 89fd1e8 Improvements to the pacer where it lost some budget due to truncation errors. by stefan@webrtc.org · 11 years ago
  39. 376b4ea Fix breakage introduced by r6691. by pbos@webrtc.org · 11 years ago
  40. 2f4b14e Make RTCP sender report send media bytes. by pbos@webrtc.org · 11 years ago
  41. ffa8dca Eliminate unnecessary #include by kwiberg@webrtc.org · 11 years ago
  42. 324f63c rtc::Fatal output: Print space between # and message by kwiberg@webrtc.org · 11 years ago
  43. bc73871 Remove the VPM denoiser. by pbos@webrtc.org · 11 years ago
  44. 2adc51c Handle the case if an unusually long peer name is provided in the peerconnection example. by tommi@webrtc.org · 11 years ago
  45. cb859ec Replace strcpy with talk_base::strcpyn. by pbos@webrtc.org · 11 years ago
  46. 6823479 Roll libyuv from 1033 to 1035 to get cpuid fix for AVX2 that avoids misdetect causing a crash in AVX2 code on cpus that do not have AVX2. by fbarchard@google.com · 11 years ago
  47. d873540 Roll chromium 282462:282879. by fgalligan@google.com · 11 years ago
  48. 92a9bac Rebase webrtc/base with r6682 version of talk/base: by henrike@webrtc.org · 11 years ago
  49. 1b84116 Add a facility to the Thread class to catch blocking regressions. by henrike@webrtc.org · 11 years ago
  50. b038c72 Enable SCTP compile for iOS. by tkchin@webrtc.org · 11 years ago
  51. aac1497 (Auto)update libjingle 71116846-> 71117224 by buildbot@webrtc.org · 11 years ago
  52. 5be649f Add a facility to the Thread class to catch blocking regressions. by tommi@webrtc.org · 11 years ago
  53. 242068d A step towards changing StatsReport::Value::name to an enum. by tommi@webrtc.org · 11 years ago
  54. 03505bc Make StatsCollector depend on always having a valid session pointer. by tommi@webrtc.org · 11 years ago
  55. b5348c6 Minor refactoring of the session classes. by tommi@webrtc.org · 11 years ago
  56. d852434 (Auto)update libjingle 71107853-> 71115715 by buildbot@webrtc.org · 11 years ago
  57. b92f6f9 (Auto)update libjingle 71099685-> 71107853 by buildbot@webrtc.org · 11 years ago
  58. a4da771 Fix deadlock in Android stopCapture() call. by glaznev@webrtc.org · 11 years ago
  59. 5f43ce6 Fix a type cast issue for compiling webrtc with BoringSSL. by jiayl@webrtc.org · 11 years ago
  60. e04cb0e (Auto)update libjingle 70948025-> 70959275 by buildbot@webrtc.org · 11 years ago
  61. 9bef551 GN: Fix include paths for WebRTC in Chromium build. by kjellander@webrtc.org · 11 years ago
  62. 9e1acc8 Fix bugs introduced by https://code.google.com/p/webrtc/source/detail?r=6667 . by tommi@webrtc.org · 11 years ago
  63. dd6780d Remove always-true expression. by tommi@webrtc.org · 11 years ago
  64. eec6ecd Landing pkasting's webrtc fixes for MSVC level 4 warnings in WebRTC. --- by tommi@webrtc.org · 11 years ago
  65. 180e516 Thread annotate RTCPSender. by pbos@webrtc.org · 11 years ago
  66. 336e8e8 Fixing memcheck leak suppressions for XMPPClient tests. by pbos@webrtc.org · 11 years ago
  67. 168f23f Move pacer to fully use webrtc::Clock instead of webrtc::TickTime. by stefan@webrtc.org · 11 years ago
  68. ccbed3b Implement unittest SetRecvCodecsAcceptDefaultCodecs. by pbos@webrtc.org · 11 years ago
  69. a1bfcad Cast payload types to int for logging. by pbos@webrtc.org · 11 years ago
  70. fb2e7c2 Document that channels are stored contiguously in AudioBuffer by aluebs@webrtc.org · 11 years ago
  71. d212ffc Remove unnecessary build message. by tommi@webrtc.org · 11 years ago
  72. 4ef438e Remove the send-side cname getter APIs from voice and video engine. by stefan@webrtc.org · 11 years ago
  73. 0f42668 Roll chromium_revision 280876:282462 by henrikg@webrtc.org · 11 years ago
  74. cb97368 roll libyuv to r1033 for clang-cl support on windows. by fbarchard@google.com · 11 years ago
  75. b614d06 Rebase webrtc/base with r6655 version of talk/base: by henrike@webrtc.org · 11 years ago
  76. 72491b9 Count total bytes sent in RTPSender::Bytes(). by pbos@webrtc.org · 11 years ago
  77. 0422100 Fix data race in VCMTiming::ResetDecodeTime. by pbos@webrtc.org · 11 years ago
  78. bd9c092 Skip encoding in fake VP8 encoder. by pbos@webrtc.org · 11 years ago
  79. 7ae9108 Remove more unused tsan suppressions and fix call test passing the same decoder to multiple received streams. by andresp@webrtc.org · 11 years ago
  80. 91f1752 Support VP8 encoder settings in VideoSendStream. by pbos@webrtc.org · 11 years ago
  81. 8f15121 Refactor registerable callbacks for FrameCountObserver from rtp_rtcp module into vie_channel. by andresp@webrtc.org · 11 years ago
  82. 5bde66e audio_processing: Updates aec_core_sse2.c with changes made to aec_common.h by bjornv@webrtc.org · 11 years ago
  83. 555fc78 Neon version of SubbandCoherence() by bjornv@webrtc.org · 11 years ago
  84. ac800c8 Neon version of rftbsub_128() by bjornv@webrtc.org · 11 years ago
  85. 5ac876b Revert "Remove remains of WEBRTC_NO_STL." (rev 6641). by andresp@webrtc.org · 11 years ago
  86. e91ba26 Revert 6643 "Revert 6637 "Revert 6636 "Roll chromium_revision 28..." by henrikg@webrtc.org · 11 years ago
  87. 02dce51 Revert 6637 "Revert 6636 "Roll chromium_revision 280876:281479"" by henrikg@webrtc.org · 11 years ago
  88. 7267020 (Auto)update libjingle 70813271-> 70818369 by buildbot@webrtc.org · 11 years ago
  89. 47d1c98 Remove remains of WEBRTC_NO_STL. by andresp@webrtc.org · 11 years ago
  90. 10ef8fe Create FullScreenChromeWindowDetector in DesktopConfigurationOptions::CreateDefault. by jiayl@webrtc.org · 11 years ago
  91. 4b1f330 Fix a bug in SocketAddress where "a.b.c.d:1" and "b.b.c.d:1" are incorrectly considered equal. by jiayl@webrtc.org · 11 years ago
  92. 7af12be Thread annotations for vie_encoder.cc/.h by stefan@webrtc.org · 11 years ago
  93. e7771d0 Revert 6636 "Roll chromium_revision 280876:281479" by henrikg@webrtc.org · 11 years ago
  94. 543da99 Roll chromium_revision 280876:281479 by henrikg@webrtc.org · 11 years ago
  95. 045a9b1 Remove unnecessary race suppressions copied from chromium. by andresp@webrtc.org · 11 years ago
  96. b8e9e44 Add full stack test cases with a fake network pipe. by stefan@webrtc.org · 11 years ago
  97. e9cefde Improve libjingle's ASSERT and VERIFY macros on Windows. by tommi@webrtc.org · 11 years ago
  98. 01bda20 Fixed the stats problem when new track is using the same ssrc as the previous track. by xians@webrtc.org · 11 years ago
  99. b753762 delay_estimator: Increases test coverage and makes input spectrum const by bjornv@webrtc.org · 11 years ago
  100. 12b4efe Implement a work around for Chrome full-screen tab switch on Mac. by jiayl@webrtc.org · 11 years ago