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gerrit-public.fairphone.software / platform / external / webrtc / e2a017725570ead5946a4ca8235af27470ca0df9
  1. e2a0177 Style cleanups in rtp header extension traits: by danilchap · 8 years ago
  2. af27ed0 Add algorithm for Residual Echo Detector. by ivoc · 8 years ago
  3. 5f1b051 Reland Change ViEEncoder to not reconfigure the encoder until the video resolution is known. by perkj · 8 years ago
  4. c4b9b94 Revert of Replace FileWrapper with File (in audio_device) (patchset #3 id:40001 of https://codereview.webrtc.org/2386963003/ ) by terelius · 8 years ago
  5. 6ceab08 GN: New conventions, default target and refactorings by kjellander · 8 years ago
  6. 9f4a4a0 Add empty residual echo detector. by ivoc · 8 years ago
  7. a9a1ac2 Moved rtc::Base64 to base approved. by philipel · 8 years ago
  8. 02ba211 Move RTCStatsCollector helper functions to anonymous namespace. by hbos · 8 years ago
  9. f005a00 Added calling of the stream_analog_level api in audioproc_f by peah · 8 years ago
  10. 6d6762c Add UINavigationController and settings bar button to AppRTCMobile. by denicija · 8 years ago
  11. 5da65f2 Revert of Change ViEEncoder to not reconfigure the encoder until the video resolution is known. (patchset #4 id:60001 of https://codereview.webrtc.org/2455063002/ ) by perkj · 8 years ago
  12. 48dfab5 Revert of New statistics interface for APM (patchset #11 id:200001 of https://codereview.webrtc.org/2433153003/ ) by ivoc · 8 years ago
  13. 461c29e Change ViEEncoder to not reconfigure the encoder until the video resolution is known. by perkj · 8 years ago
  14. 135259a In order to be able to analyze the AGC behavior on by peah · 8 years ago
  15. 04055e9 Removes all uses of FileWrapper in audio_device. by palmkvist · 8 years ago
  16. 8b8d3e4 New statistics interface for APM by ivoc · 8 years ago
  17. 37abf53 Delete FrameObject::size member. by nisse · 8 years ago
  18. 11f72b1 Fix compile error for non Intel platforms by Gordana.Cmiljanovic · 8 years ago
  19. 9922016 Fix "IsLoopbackIp" to cover all loopback addresses; not just 127.0.0.1. by deadbeef · 8 years ago
  20. 6be0a65 Move ScreenCapturer 'real' tests out of screen_capturer_unittest.cc. by zijiehe · 8 years ago
  21. 32bcaf6 Improve RTC_DCHECK_op so that it won't trigger useless compiler warnings by kwiberg · 8 years ago
  22. a73df55 Do not rely on specific ordering on generated candidates in TestGetAllPortsPortRange by honghaiz · 8 years ago
  23. 45d18eb Re-enable the PostDelayed TaskQueue test on all platforms except Windows. by tommi · 8 years ago
  24. 492ee28 Use bayesian estimate of acked bitrate. by Stefan Holmer · 8 years ago
  25. 9890a58 Testing of FileVideoCapturer. by mandermo · 8 years ago
  26. da35f3e Delete unused features of rtc::FilesystemInterface and related classes. by nisse · 8 years ago
  27. a101e56 Remove LOGGING=1 define. by kjellander · 8 years ago
  28. fe90b41 Improves audio logs of native audio layers on Android by henrika · 8 years ago
  29. 68e6bdd Remove use of VoECodec in video/call tests. by solenberg · 8 years ago
  30. 5e49c2f Restore symbol level for Android builds. by kjellander · 8 years ago
  31. bc59b06 Add gn_isolate_map.pyl file for WebRTC stand-alone tests by kjellander · 8 years ago
  32. b112568 Roll chromium_revision 9b5bb47fa0..04e7c673d9 (426837:427632) by kjellander · 8 years ago
  33. e183121 Enable clang style plugin in webrtc/modules/desktop_capture by sergeyu · 8 years ago
  34. 54b0acb Change destruction order to fix potential invalid pointer dereference. by erikchen · 8 years ago
  35. 489c0d4 Decrease threshold for key frame generation. by glaznev · 8 years ago
  36. 91c2d43 Disable TaskQueueTest.PostDelayed because of flakiness by terelius · 8 years ago
  37. e5ba44e Implement framesDecoded stat in video receive ssrc stats. by sakal · 8 years ago
  38. 784a831 Check that stats_proxy_ is non-NULL before use. by nisse · 8 years ago
  39. 5819660 MB: Add Linux swarming bots with memory sanitizers on the FYI waterfall. by ehmaldonado · 8 years ago
  40. 059fb44 - Replace FakeAudioProcessing in WVoE unittest with MockAudioProcessing. by solenberg · 8 years ago
  41. 16b6d6d Reland of "Separating video settings in VideoQualityTest". by minyue · 8 years ago
  42. c1600c5 Follow standard sending CVO rtp header extension by danilchap · 8 years ago
  43. b906172 Reland of Move bitstream parser to more appropriate directory. (patchset #1 id:1 of https://codereview.webrtc.org/2430353004/ ) by kthelgason · 8 years ago
  44. 12ba186 Move parsing from tests to Transport helper in RTPSenderTests by danilchap · 8 years ago
  45. a8bec8d Testing of VideoFileRenderer with byte frames by mandermo · 8 years ago
  46. 940b6d6 Clean up logging in AudioSendStream::SetupSendCodec(). by solenberg · 8 years ago
  47. da389e3 PrintTo functions for RTCStats added in rtcstatscollector_unittest.cc by hbos · 8 years ago
  48. d89ab14 Introduce rtc::PacketTransportInterface and let cricket::TransportChannel inherit. by johan · 8 years ago
  49. 57e13de Minor cleanup of rtc::BasicPacketSocketFactory implementation. by johan · 8 years ago
  50. 0d8ade5 Remove remnants of libsrtp1 by mattdr · 8 years ago
  51. 257dc39 Refactoring: Hide VideoCodec.codecSpecific as "private" by hta · 8 years ago
  52. 189f9b1 Revert of Clean up logging in AudioSendStream::SetupSendCodec(). (patchset #3 id:40001 of https://codereview.webrtc.org/2446963003/ ) by terelius · 8 years ago
  53. d0af5c6 Fix a deadlock in EglRenderer.releaseEglSurface. by sakal · 8 years ago
  54. 2d81eb3 Fix BWE simulations so that it uses the delay based BWE. by terelius · 8 years ago
  55. 1836fd6 Clean up logging in AudioSendStream::SetupSendCodec(). by solenberg · 8 years ago
  56. 701d628 Moved the AGC render sample queue into the audio processing module by peah · 8 years ago
  57. d8872c5 Removed the file resources/audioproc.aecdump.sha1 file which is no longer used. by peah · 8 years ago
  58. a062460 Several subcomponents inside APM copy render audio from by peah · 8 years ago
  59. 67c8bc4 RTCStats equality operator added. by hbos · 8 years ago
  60. 01bbc3c Reland of Deflake ChangingNetworkRoute test. by stefan · 8 years ago
  61. 77f5953 Revert of Deflake ChangingNetworkRoute test. (patchset #1 id:1 of https://codereview.webrtc.org/2426073002/ ) by ehmaldonado · 8 years ago
  62. 6711820 Deflake ChangingNetworkRoute test. by stefan · 8 years ago
  63. cc34833 Remove now unused code in RtpHeaderExtensionMap by danilchap · 8 years ago
  64. 611f267 Make WebRTC compatible with OpenH264 v1.6. by sprang · 8 years ago
  65. af1ae31 Remove dead dependencies on xmllite and xmpp. by kjellander · 8 years ago
  66. dbf6705 codec_unittest.cc: Fix TEST vs TEST_F mismatch by Magnus Jedvert · 8 years ago
  67. 06c8e1e Revert of H264 codec: Check profile-level-id when matching (patchset #2 id:60001 of https://codereview.webrtc.org/2347863003/ ) by Magnus Jedvert · 8 years ago
  68. fcba8fe Delete left-over file profiler_unittest.cc. by nisse · 8 years ago
  69. 74097fd Delete unused file screencastid.h. by nisse · 8 years ago
  70. e58d73d Fix more swarming test failures by using the fake clock or longer timeout. by honghaiz · 8 years ago
  71. a6b8298 Use relative names in GN to make Chromium happy by kwiberg · 8 years ago
  72. 4a18f16 Update XServerPixelBuffer to handle errors returned from XGetImage(). by sergeyu · 8 years ago
  73. da2bf4e Stop using old AudioCodingModule::RegisterReceiveCodec overloads by kwiberg · 8 years ago
  74. 88b7074 Remove unused function implementations from FakeWebRtcVoiceEngine. by solenberg · 8 years ago
  75. fb70b45 Preventing TURN redirects to loopback addresses. by deadbeef · 8 years ago
  76. 838cdb3 Revert of Fix chromium-style warnings. (patchset #1 id:1 of https://codereview.webrtc.org/2400993002/ ) by terelius · 8 years ago
  77. 5d79a7c rtcstats_objects.h updated with TODOs about stats not being collected by hbos · 8 years ago
  78. a6f495c Simplifying audio network adaptor by moving receiver frame length range to ctor. by minyue · 8 years ago
  79. a73f6c9 NetEq now works with packets as values, rather than pointers. by ossu · 8 years ago
  80. d312713 Roll chromium_revision 1362287708..9b5bb47fa0 (426760:426837) + roll Android SDK to N by ehmaldonado · 8 years ago
  81. 86b92e0 Drop VP8 frames older than the last sync frame in the RtpFrameReferenceFinder. by philipel · 8 years ago
  82. 1655e45 Elimiteted race condition in the AudioMixer. by aleloi · 8 years ago
  83. 2206c95 Revert of Fix some chromium style warnings in remote_bitrate_estimator.h (patchset #1 id:1 of https://codereview.webrtc.org/2387113008/ ) by terelius · 8 years ago
  84. 0140408 Add tests and fix thread annotations by danilchap · 8 years ago
  85. b60d196 Eliminate left shift of negative value by using multiplication instead by kwiberg · 8 years ago
  86. 2fa7c67 RTCTransportStats[1] added, supporting all members. by hbos · 8 years ago
  87. 5de3a7e Remove unused variable from delay based BWE. by terelius · 8 years ago
  88. 509eadd Fix chromium-style warnings. by terelius · 8 years ago
  89. c22bcf4 Fix some chromium style warnings in remote_bitrate_estimator.h by terelius · 8 years ago
  90. d7ce668 Roll chromium_revision f9e01d4887..1362287708 (426685:426760) by buildbot · 8 years ago
  91. a3cac05 GN: move webrtc/video/ targets from webrtc/BUILD.gn into webrtc/video/BUILD.gn by kjellander · 8 years ago
  92. 43536c3 Implement framesEncoded stat in video send ssrc stats. by sakal · 8 years ago
  93. 2675274 Remove cricket::VideoCodec with, height and framerate properties by perkj · 8 years ago
  94. d3c4008 Delete always-zero ByteBufferWriter::start_. by nisse · 8 years ago
  95. 61c053e Reland of Delete webrtc::VideoFrame::CopyFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2397943003/ ) by nisse · 8 years ago
  96. ebf5240 Allow using Java classes that don't require JNI in Chromium. by sakal · 8 years ago
  97. 66712b0 Revert of Add method cricket::VideoCapturer::NeedsDenoising, use in VideoCapturerTrackSource. (patchset #5 id:80001 of https://codereview.webrtc.org/2334683002/ ) by nisse · 8 years ago
  98. 151572b Delete unused class AudioSourceWithMixStatus. by nisse · 8 years ago
  99. 25445d3 Integrate FlexfecReceiveStream with Call. by brandtr · 8 years ago
  100. 764e364 Several subcomponents inside APM copy render audio from by peah · 8 years ago