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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / 80882f3d005f32f5a71bec3a2977cbc8ce1908b7
  1. 80882f3 Replace MapWrapper with std::map<>. by pbos@webrtc.org · 11 years ago
  2. a24f40d Updated WebRTC version to 3.39 by elham@webrtc.org · 11 years ago
  3. f83e3a5 Signal when shutting down DirectTransport. by pbos@webrtc.org · 11 years ago
  4. b0fc85b Avoid acquiring VCM::_receiveCritSect during decode callback. by wuchengli@chromium.org · 11 years ago
  5. 48ac502 Run loopback tests with network thread. by pbos@webrtc.org · 11 years ago
  6. edc86e5 Added Opus stereo support by minyue@webrtc.org · 11 years ago
  7. 6d94c78 Fix crash in screen capturer on Mac by sergeyu@chromium.org · 11 years ago
  8. 0bb1b31 Hand over loopback packets to a network thread. by pbos@webrtc.org · 11 years ago
  9. 16c8462 Don't pace out packets or generate padding when the pacer is disabled. by stefan@webrtc.org · 11 years ago
  10. 4ab008f Remove include_dirs from test/test.gyp. by pbos@webrtc.org · 11 years ago
  11. a0b4f27 Remove unused unreferenced code in webrtc/ by pbos@webrtc.org · 11 years ago
  12. 8a11920 Revert "Avoid acquiring VCM::_receiveCritSect during decode callback." by wuchengli@chromium.org · 11 years ago
  13. 334bf81 Avoid acquiring VCM::_receiveCritSect during decode callback. by wuchengli@chromium.org · 11 years ago
  14. 165febc Allowing decoding with errors, when disabling nack. by mikhal@webrtc.org · 11 years ago
  15. a2505ea Fix duplicate code by niklas.enbom@webrtc.org · 11 years ago
  16. 54e9955 Delete Channels without ChannelManager lock. by pbos@webrtc.org · 11 years ago
  17. be78a05 Adding call to Opus PLC by tina.legrand@webrtc.org · 11 years ago
  18. b263e41 Added logic for kSelectiveErrors to VCMJitterBuffer and corresponding unit tests. by agalusza@google.com · 11 years ago
  19. 9277c94 Ref-counted rewrite of ChannelManager. by pbos@webrtc.org · 11 years ago
  20. 5ce0e78 Code formatting on files touched in r4447. by pbos@webrtc.org · 11 years ago
  21. 7cbdbe6 Added configuration of max delay to ACM and NetEq by pwestin@webrtc.org · 11 years ago
  22. 547e0e3 Added Decoding with errors API to video_coding.h and removed unused DecodeError enum. by agalusza@google.com · 11 years ago
  23. 5c7fa98 Add turaj@webrtc.org to NetEq owners. by turaj@webrtc.org · 11 years ago
  24. babb161 Disabled flaky HardwareTest.BuiltInWasapiAECWorksForAudioWindowsCoreAudioLayer. by phoglund@webrtc.org · 11 years ago
  25. 60ba778 Disabled SsrcPropagatesCorrectly on Linux. by phoglund@webrtc.org · 11 years ago
  26. 79df0bc Better error treatment in NetEqImpl::InsertPacketInternal() by minyue@webrtc.org · 11 years ago
  27. 0c31023 removed NetEq::EnableDtmf() by minyue@webrtc.org · 11 years ago
  28. df8d03f * Update libjingle to 50389769. by wu@webrtc.org · 11 years ago
  29. 0f807b2 Invert dependency between webrtc_utility and media_file targets to reflect reality. by fischman@webrtc.org · 11 years ago
  30. d5fb79c Updated WebRTC version number to 3.38 by elham@webrtc.org · 11 years ago
  31. 50ff6a5 Switch C++-style C headers with their C equivalents. by pbos@webrtc.org · 11 years ago
  32. 30c741a Fix implicit int->bool conversion in VideoSendStream::DeliverRtcp. by pbos@webrtc.org · 11 years ago
  33. 2c00af7 Use RtpHeaderParser in VideoCall implementation. by pbos@webrtc.org · 11 years ago
  34. bf9bc32 Glue code and tests for NACK in new VideoEngine API. by pbos@webrtc.org · 11 years ago
  35. dac40f8 Fix send times in video_full_stack. by pbos@webrtc.org · 11 years ago
  36. 46d2ca1 Add back is.FrameProvider() call lost in r4194. by pbos@webrtc.org · 11 years ago
  37. 0b6e893 Remove redundant conditions key. by andrew@webrtc.org · 11 years ago
  38. 75370f1 Add one API for implementing Initial delay. by turaj@webrtc.org · 11 years ago
  39. 9d939ee Adds all unittests to android NDK-APK framework. by henrike@webrtc.org · 11 years ago
  40. aa0dac5 Add some virtual and OVERRIDEs in webrtc/common_audio/ by pbos@webrtc.org · 11 years ago
  41. 0bf6b98 Fix some chromium-style warnings in webrtc/modules/audio_processing/ by pbos@webrtc.org · 11 years ago
  42. f72eb49 Fix crash in DesktopRegion::Intersect(). by sergeyu@chromium.org · 11 years ago
  43. 42ef0f5 Fix some chromium-style warnings in webrtc/system_wrappers/ by pbos@webrtc.org · 11 years ago
  44. 28dda63 Removed lines preventing simultaneous kHardNack and decoding with errors. Also made changes recommended by gcl lint (with the exception of changing non-const references to pointers). by agalusza@google.com · 11 years ago
  45. 26a30e6 Unbreak clang/android build of webrtc. by fischman@webrtc.org · 11 years ago
  46. 53d1ade Adding possibility to use encoding time when trigger underuse for frame based overuse detection. by mflodman@webrtc.org · 11 years ago
  47. 9b748e5 Merge r4374 from stable to trunk. by xians@webrtc.org · 11 years ago
  48. 2d4c1a1 Merge r4394 from stable to trunk. by xians@webrtc.org · 11 years ago
  49. f686778 Merge r4326 from stable to trunk. by xians@webrtc.org · 11 years ago
  50. dadf0f7 Handel zero correlation if at the same time distortion is also zero. by turaj@webrtc.org · 11 years ago
  51. e2df770 Add some virtual and OVERRIDEs in webrtc/modules/audio_coding/ by pbos@webrtc.org · 11 years ago
  52. 7df7f61 Fix some chromium-style warnings in webrtc/modules/desktop_capture/ by pbos@webrtc.org · 11 years ago
  53. 463eb03 Fix some chromium-style warnings in webrtc/modules/pacing/ by pbos@webrtc.org · 11 years ago
  54. d0557b5 Fix some chromium-style warnings in webrtc/modules/rtp_rtcp/ by pbos@webrtc.org · 11 years ago
  55. ff3f7f6 Fix some chromium-style warnings in webrtc/modules/remote_bitrate_estimator/ by pbos@webrtc.org · 11 years ago
  56. ee34820 Fix some chromium-style warnings in webrtc/modules/bitrate_controller/ by pbos@webrtc.org · 11 years ago
  57. 81e21c6 Added libjingle_peerconnection_java_unittest to buildbot_tests.py by phoglund@webrtc.org · 11 years ago
  58. fe8ba4d Move internal aec_core defines out of header. by andrew@webrtc.org · 11 years ago
  59. 3ea4830 Add svn:ignore properties for all spuriously-removed dirs on Linux64 Release (internal). by fischman@webrtc.org · 11 years ago
  60. b36d452 Correcting Turaj's email. by turaj@webrtc.org · 11 years ago
  61. 80df10d Fix some chromium-style warnings in webrtc/modules/video_coding/ by pbos@webrtc.org · 11 years ago
  62. ae6d494 Fix some chromium-style warnings in webrtc/test/ by pbos@webrtc.org · 11 years ago
  63. 9bf2b46 Fix some chromium-style warnings in webrtc/tools/ by pbos@webrtc.org · 11 years ago
  64. e142b98 Fix some chromium-style warnings in webrtc/modules/audio_device/ by pbos@webrtc.org · 11 years ago
  65. 46688dd Sets up framework for decoding with errors: collects frame sizes (in number of packets) in JB and passes this information to VCMSessionInfo with rtt_ms as FrameData. by agalusza@google.com · 11 years ago
  66. 03bfae8 PeerConnectionTest.java: make the test work for the bots' v4l2loopback. by fischman@webrtc.org · 11 years ago
  67. 2a61170 Land http://webrtc-codereview.appspot.com/1632005/ by niklas.enbom@webrtc.org · 11 years ago
  68. 0b8a595 Updated WebRTC version to 3.37 TBR=tnakamura@webrtc.org by elham@webrtc.org · 11 years ago
  69. 263411c Improved error messages when binaries are missing. Also stderr = stdout now. by phoglund@webrtc.org · 11 years ago
  70. 6429cdb To fix a bug in InverseFFTAndWindow() function in AECM. by kma@webrtc.org · 11 years ago
  71. 7b97b16 Updated WebRtcNsx_PrepareSpectrumNeon() in accordance with the new real FFT interface in APM. For reference, you can check https://webrtc-codereview.appspot.com/1830004/diff/92001/webrtc/modules/audio_processing/ns/nsx_core.c, line 594 "static void PrepareSpectrumC()". by kma@webrtc.org · 11 years ago
  72. 1095587 Access receiving_ under receive_cs critical section by braveyao@webrtc.org · 11 years ago
  73. dbdcf16 Don't set clang_use_chrome_plugins in common.gypi by sergeyu@chromium.org · 11 years ago
  74. 0bd4d89 Fixes resources and data path in modules_unittests.isolate. by henrike@webrtc.org · 11 years ago
  75. fd87865 Downstream latest Chromium SincResampler changes. by andrew@webrtc.org · 11 years ago
  76. b31f64f Update include paths in device_info_external.cc by sergeyu@chromium.org · 11 years ago
  77. d13f24b Add a Config class interface to AudioProcessing for passing options. by andrew@webrtc.org · 11 years ago
  78. 2cf4d85 Fix include path in video_capture_external.cc by niklas.enbom@webrtc.org · 11 years ago
  79. a3c7fa2 Formalized Real 16-bit FFT for APM. by kma@webrtc.org · 11 years ago
  80. 51f7c7e Fix ScreenCapturerLinux not to use XDamage when requested. by sergeyu@chromium.org · 11 years ago
  81. 530f40f webrtc/common_types.h: Document bitrate fields' units. by fischman@webrtc.org · 11 years ago
  82. 7b87e6b Makes webrtc and libjingle build from the same gyp-file. Also, the libjingle and webrtc DEPS revisions were mismatching. This cl takes the most recent revision of mismatches. Also disables 64 bit Mac builds for libjingle by henrike@webrtc.org · 11 years ago
  83. ecbeb2b Hooking up first simple CPU adaptation version. by mflodman@webrtc.org · 11 years ago
  84. cb2fb3f Revert 4382 "Makes webrtc and libjingle build from the same gyp-..." by henrike@webrtc.org · 11 years ago
  85. 4432261 Makes webrtc and libjingle build from the same gyp-file. Also, the libjingle and webrtc DEPS revisions were mismatching. This cl takes the most recent revision of mismatches. Also disables 64 bit Mac builds for libjingle by henrike@webrtc.org · 11 years ago
  86. 675eead Modified the presubmit checks such that difference license templates are checked for in webrtc and talk folder. by henrike@webrtc.org · 11 years ago
  87. 95a4477 Correctly rebuild WebRTCDemo after jni/ source file changes by yujie.mao@webrtc.org · 11 years ago
  88. 331a9b2 Revert 4372 "Makes webrtc and libjingle build from the same gyp-..." by henrike@webrtc.org · 11 years ago
  89. 30a83c1 Makes webrtc and libjingle build from the same gyp-file. Also, the libjingle and webrtc DEPS revisions were mismatching. This cl takes the most recent revision of mismatches. by henrike@webrtc.org · 11 years ago
  90. 7c4152b AppRTCDemo: build fixes for iOS build in webrtc by fischman@webrtc.org · 11 years ago
  91. 382ef1e Undo libvpx include changes in r4348 to fix build. by tnakamura@webrtc.org · 11 years ago
  92. a491674 Default constructor for RtcpAppHandler. by pbos@webrtc.org · 11 years ago
  93. 62365d0 clean up incomplete revert in r4357 Also revert r4319, will follow up with pbos by tnakamura@webrtc.org · 11 years ago
  94. 1628267 Revert r4301 by tnakamura@webrtc.org · 11 years ago
  95. 8eea45a Fixes: Resolves conflict that will happen when merging libjingle's and WebRTC's supplemental.gyp. By separating build_with_chromium and build_with_libjingle one can now just define build_with_libjingle in libjingle's supplemental.gyp. Once that is done it will be possible to merge the two supplemental.gyp-files. I.e. in WebRTC the supplemental.gyp would only set build_with_chromium to 0 since there is no longer any reason to disable logging and tests as they will be accessible in the same repository as libjingle. by henrike@webrtc.org · 11 years ago
  96. c64c74c Include files from webrtc/.. paths in signal_processing/. by pbos@webrtc.org · 11 years ago
  97. 93c9ccb Include files from webrtc/.. paths in media_file/. by pbos@webrtc.org · 11 years ago
  98. d92e5df Make sure first RTP packet counts as in-order. by pbos@webrtc.org · 11 years ago
  99. 3fa41a6 Include files from webrtc/.. paths in bitrate_controller/. by pbos@webrtc.org · 11 years ago
  100. 2a2e7ff Include files from webrtc/.. paths in video_coding/. by pbos@webrtc.org · 11 years ago