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gerrit-public.fairphone.software / platform / external / webrtc / c7a6569713699df07e3a717778014bcc8eff97b3
  1. c7a6569 Revert of Disable failing modules_unittests for UBSan. (patchset #1 id:40001 of https://codereview.webrtc.org/1915813002/ ) by pbos · 8 years ago
  2. 82d7862 Change default timestamp to 64 bits in all webrtc directories. by Honghai Zhang · 8 years ago
  3. e76db89 Fix BoringSSL license path. by tkchin · 8 years ago
  4. dd32486 Bitrate prober now keep track of probing cluster id. by philipel · 8 years ago
  5. f2eae33 Corrected bug in checking the third number and added extra checks by dkirovbroadsoft · 8 years ago
  6. dc7d0d2 Move, almost, all receive side references to RTP to RtpStreamReceiver. by mflodman · 8 years ago
  7. b56069e Enable NACK for audio even if there are no send streams. by deadbeef · 8 years ago
  8. 31fec40 Set rtcp_send_transport for AudioReceiveStreams. This was forgotten in https://codereview.webrtc.org/1909333002/. by solenberg · 8 years ago
  9. 3a33465 Fix the flaky WebRtcSessionTest.TestRtxRemovedByCreateAnswer. by zhihuang · 8 years ago
  10. 44c8a37 Removed the file echo_cancellation_internal.h and moved by peah · 8 years ago
  11. cf5b37c Accept all the media profiles required by JSEP. by zhihuang · 8 years ago
  12. 39a3670 Rename OpenH264 frame-type conversion function. by pbos · 8 years ago
  13. 3f08dc6 Introduced the new APM data logging functionality into the AEC echo_cancellation.* API layer. by peah · 8 years ago
  14. e84cd2e Cache a ClientHello received before the DTLS handshake has started. by deadbeef · 8 years ago
  15. fac23f0 Tune QP threshold for HW codecs. by Alex Glaznev · 8 years ago
  16. 600246e Removed SSRC knowledge from ViEEncoder. by perkj · 8 years ago
  17. ef00ec1 Update CPU monitor to use moving averages. by Alex Glaznev · 8 years ago
  18. 28a4456 Revert "Revert of Remove SendPacer from ViEEncoder (patchset #13 id:240001 of https://codereview.webrtc.org/1917793002/ )" by Per · 8 years ago
  19. b49ac78 Revert of Use RC_TIMESTAMP_MODE for OpenH264. (patchset #1 id:1 of https://codereview.webrtc.org/1945763002/ ) by pbos · 8 years ago
  20. 1aa435c Reland of Android GlDrawer: Add frame size as argument to draw functions (patchset #1 id:1 of https://codereview.webrtc.org/1950953002/ ) by ivoc · 8 years ago
  21. 1726831 Revert of Android GlDrawer: Add frame size as argument to draw functions (patchset #2 id:20001 of https://codereview.webrtc.org/1948473002/ ) by ivoc · 8 years ago
  22. 274c1dc Added flag for FEC for video_loopback. by philipel · 8 years ago
  23. 73987c9 Run "git cl format --full" on a pair of files with ancient formatting by kwiberg · 8 years ago
  24. 053f917 Partial revert of Enable -Winconsistent-missing-override flag. (patchset #5 id:80001 of https://codereview.webrtc.org/1921653002/ ) by ivoc · 8 years ago
  25. 71af75d Android GlDrawer: Add frame size as argument to draw functions by magjed · 8 years ago
  26. c6c00b3 Revert of Remove the rtc_relative_path GYP variable and similar defines (patchset #1 id:1 of https://codereview.webrtc.org/1925733002/ ) by phoglund · 8 years ago
  27. 825eb58 Revert of Remove SendPacer from ViEEncoder (patchset #13 id:240001 of https://codereview.webrtc.org/1917793002/ ) by perkj · 8 years ago
  28. 857c5cc Remove SendPacer from ViEEncoder by perkj · 8 years ago
  29. cfc8e3b Removed all RTP dependencies from ViEChannel and renamed class. by mflodman · 8 years ago
  30. fe4b216 Roll chromium_revision 0b4adfd25e..58963e5878 (390907:391406) by buildbot · 8 years ago
  31. 3815655 Change aggregation window of aecDivergentFilterFraction to 1 second. by minyue · 8 years ago
  32. 7dd7ab5 Changed the name of the variable overdriveSm and removed the by peah · 8 years ago
  33. 55dd708 Support RtpEncodingParameters::active in voice engine. by Taylor Brandstetter · 8 years ago
  34. 1746179 Reducing neteq sync buffer size. by minyue · 8 years ago
  35. 4adbbcf Move ADM Create() method to public interface. by Peter Boström · 8 years ago
  36. 9bfa106 Change the threshold for external VNR. by jackychen · 8 years ago
  37. c4deee4 Use RC_TIMESTAMP_MODE for OpenH264. by Peter Boström · 8 years ago
  38. c8fe991 Removing SpatialAudio test code by henrik.lundin · 8 years ago
  39. 87f8c0d Adding in objc vars for WebRTC GN config. by Patrik Höglund · 8 years ago
  40. b1fb72b NetEq: Move counting of generated CNG samples from DecisionLogic by henrik.lundin · 8 years ago
  41. b46083e This CL introduces a new data logging functionality by peah · 8 years ago
  42. 696a802 Re-enable Vp9FlexModeRefCount by philipel · 8 years ago
  43. 35fdb2a Log WebRTC.Video.AVSyncOffsetInMs. by pbos · 8 years ago
  44. 5178ee8 NetEq: Use a BuiltinAudioDecoderFactory to create decoders by kwiberg · 8 years ago
  45. ddf1653 Android EGL: Synchronize calls to eglCreateContext by magjed · 8 years ago
  46. 30f118e This cl deletes the class webrtc::VideoRendererCallback. by nisse · 8 years ago
  47. fc88ffe Fix allocation size in CricketToJavaI420Frame, taking stride into account. by nisse · 8 years ago
  48. 35151f3 Add histogram stats for average send delay of sent packets for a sent video stream. The delay is measured from a packet is sent to the transport until leaving the socket. by asapersson · 8 years ago
  49. 5a24637 Do not stop a session unless the candidate of a writable connection belongs to the by Honghai Zhang · 8 years ago
  50. 5dd42fd Fixing a segfault that can occur when changing the track of an RtpSender. by deadbeef · 8 years ago
  51. acf1431 Removing unused resources from building files. by minyue · 8 years ago
  52. 376b192 Remove VideoCodingModule::VCMPacketizationCallback by perkj · 8 years ago
  53. 16ac328 Remove VCMRenderBufferSizeCallback. by Peter Boström · 8 years ago
  54. 1bffc1d Rename rtc::Time64 --> rtc::TimeMillis. by nisse · 8 years ago
  55. bc75d97 Remove PayloadRouter dependency from ViEEncoder. by perkj · 8 years ago
  56. 5bd3397 Adding 120 ms frame length support in NetEq. by minyue · 8 years ago
  57. e4246b6 Roll chromium_revision 8e44a16c27..0b4adfd25e (390884:390907) by buildbot · 8 years ago
  58. 7d4a6c3 Adds timeout for audio record thread in Java layer by henrika · 8 years ago
  59. 53ff70f Reland "Avoiding overflow in cross correlation in NetEq." by minyue · 8 years ago
  60. a017b8e Remove asapersson from webrtc/modules/utility/OWNERS. by asapersson · 8 years ago
  61. d1d96b2 VideoCapturerAndroid: Remove deprecated create function with egl context argument by Magnus Jedvert · 8 years ago
  62. 1ba8d39 Remove webrtc/stream.h and unutilized inheritance. by pbos · 8 years ago
  63. 60d7e7c Roll chromium_revision 927a9e3510..8e44a16c27 (390871:390884) by buildbot · 8 years ago
  64. 7b94c69 Remove DropOldEvents from list of excluded tests. As of last week, the test no longer exists. (https://codereview.webrtc.org/1687703002/) by terelius · 8 years ago
  65. c043052 Reland of move VCMQmRobustness. (patchset #1 id:1 of https://codereview.webrtc.org/1935753002/ ) by pbos · 8 years ago
  66. 0a1f14e Roll chromium_revision 9cd13b2cd1..927a9e3510 (390867:390871) by buildbot · 8 years ago
  67. bfefb03 Replace scoped_ptr with unique_ptr everywhere by kwiberg · 8 years ago
  68. e319059 Roll chromium_revision c6dd4461f5..9cd13b2cd1 (390857:390867) by buildbot · 8 years ago
  69. a1aad8f Roll chromium_revision 551bb334c0..c6dd4461f5 (390850:390857) by buildbot · 8 years ago
  70. d130ecc Roll chromium_revision b7ebd042d1..551bb334c0 (390693:390850) by buildbot · 8 years ago
  71. 322c4a0 Replace scoped_ptr with unique_ptr in webrtc/libjingle/ by kwiberg · 8 years ago
  72. a97611a Stop QuicDataChannel and QuicDataTransport unit tests from segfaulting by mikescarlett · 8 years ago
  73. e774867 Allow TransportController to create a QuicTransportChannel by mikescarlett · 8 years ago
  74. 9bc517f Add QuicDataChannel and QuicDataTransport classes by mikescarlett · 8 years ago
  75. 70035ca Fix QuicSession to unbuffer data when the QuicTransportChannel reconnects by mikescarlett · 8 years ago
  76. 602316c Revert of Remove VCMQmRobustness. (patchset #1 id:1 of https://codereview.webrtc.org/1917083003/ ) by pbos · 8 years ago
  77. 8d37d29 Update QuicTransportChannel to latest version of libquic (Chromium: f03d2c62) by mikescarlett · 8 years ago
  78. f3569c8 Added the API to create an RTCRtpSender to the Objective C wrapper. by skvlad · 8 years ago
  79. b3bedca Roll chromium_revision d908daba0f..b7ebd042d1 (390420:390693) by buildbot · 8 years ago
  80. 0bdebd4 Re-add a (dummy) webrtc/base/buffer.cc to hopefully unbreak the Chromium build by Karl Wiberg · 8 years ago
  81. 4f90677 Making NetEq bitexactness test independent on reference files. by minyue · 8 years ago
  82. 05e61ed Remove usage of VoENetwork from VoEWrapper and FakeWebRtcVoiceEngine. by solenberg · 8 years ago
  83. 79e2842 Add tracing to MessageQueue::Dispatch. by pbos · 8 years ago
  84. a4ac478 Define rtc::BufferT, like rtc::Buffer but for any trivial type by kwiberg · 8 years ago
  85. ef8b61e Enable -Winconsistent-missing-override flag. by nisse · 8 years ago
  86. b296d05 Revert of New task queueing primitive for async tasks: TaskQueue. (patchset #5 id:80001 of https://codereview.webrtc.org/1919733002/ ) by tommi · 8 years ago
  87. 0c9df5e New task queueing primitive for async tasks: TaskQueue. by tommi · 8 years ago
  88. 4edf93b Remove deprecated functions in rtp_rtcp module by danilchap · 8 years ago
  89. 0565451 Reland of Delete cricket::VideoFrame methods GetYPlane and GetYPitch. (patchset #1 id:1 of https://codereview.webrtc.org/1921493004/ ) by nisse · 8 years ago
  90. 5b3c443 Revert of Delete webrtc::VideoFrame methods buffer and stride. (patchset #14 id:250001 of https://codereview.webrtc.org/1900673002/ ) by nisse · 8 years ago
  91. a0591b5 Delete webrtc::VideoFrame methods buffer and stride. by nisse · 8 years ago
  92. b99395a Reland of Delete video_render module. (patchset #1 id:1 of https://codereview.webrtc.org/1923613003/ ) by nisse · 8 years ago
  93. 3d7db26 Switch voice transport to use Call and Stream instead of VoENetwork. by mflodman · 8 years ago
  94. 8f8c96d NetEq: Use TickTimer in DelayManager by henrik.lundin · 8 years ago
  95. fa66659 Rename ViEReceiver and move ownership to VideoReceiveStream. by mflodman · 8 years ago
  96. acbc8d0 Roll chromium_revision bd6e76400e..d908daba0f (390344:390420) by kjellander · 8 years ago
  97. 0090b68 Script for concatenating licenses. by tkchin · 8 years ago
  98. 4896aaa Limit the maximum and minimum gain the IntelligibilityEnhancer can apply by aluebs · 8 years ago
  99. 11d4a42 Compensate for the compression gain in the IntelligibilityEnhancer by aluebs · 8 years ago
  100. c01c6a4 New interface (AudioDecoderFactory), with an implementation by kwiberg · 8 years ago