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gerrit-public.fairphone.software / platform / external / webrtc / c3de0333c0a262c0aacfbb33c286ff5ace781abf
  1. c3de033 Add plot of network delay change computed based on transport feedback. by stefan · 8 years ago
  2. 3ab32dc gn: Do not include "webrtc" in Chromium/iOS builds. by Nico Weber · 8 years ago
  3. d3235f0 Re-order and remove unused Java imports. by Sami Kalliomaki · 8 years ago
  4. 5a74650 Migrated GN target RTPtimeshift. by aleloi · 8 years ago
  5. b0dc30d Improvements to bandwidth plot. See Monorail issue for comparison. by aleloi · 8 years ago
  6. 5556dcb Migrated GN target rtpcat. by aleloi · 8 years ago
  7. 8626eb5 Fix testReturnBufferLate incorrectly calling end to end test. by sakal · 8 years ago
  8. b4ad456 Remove negative include rules from webrtc_nonparallel_tests target by danilchap · 8 years ago
  9. 606daa8 Adding Danil as an owner for the RTP module. by mflodman · 8 years ago
  10. 76cbe19 Migrated RTPchange Gyp target to GN. by aleloi · 8 years ago
  11. 9eb3d5d Add sanity check for arrival timestamps. by sprang · 8 years ago
  12. 8266773 Migrated target RTPencode to GN by aleloi · 8 years ago
  13. 734262c Fixing invalid operator< implementation. by Taylor Brandstetter · 8 years ago
  14. 8853289 Un-flaking TestSrtpError by using a fake clock. by Taylor Brandstetter · 8 years ago
  15. 19ee1e6eb Add cricket::VideoFrame::transport_frame_id() and set it to RTP timestamp. by Sergey Ulanov · 8 years ago
  16. 7fbe2ee Make EGL10 class public. by Alex Glaznev · 8 years ago
  17. b578d6c congestion controller: Fix network route change handling by Irfan Sheriff · 8 years ago
  18. dc35dcd Convenience functions to set axis properties in visualization tool. by terelius · 8 years ago
  19. 8b70faf Send audio packets immediately instead of pacing them. by terelius · 8 years ago
  20. c980062 Use a new method (android_setsocknetwork) to bind socket to network in Android M and later. by Honghai Zhang · 8 years ago
  21. f734c13 Cleanup BWE SendTimeHistory class by danilchap · 8 years ago
  22. 233c4ba New ProbingCalculator class. by philipel · 8 years ago
  23. ddf3e4a GN conversion of audio_decoder_unittests by charujain · 8 years ago
  24. 32cd2c4 Fix issues with RestartingSendStreamPreservesRtpStatesWithRtx by danilchap · 8 years ago
  25. 95e7560 [rtcp] Nack::Parse updated not to use RTCPUtility by danilchap · 8 years ago
  26. 88c1d2b Don't use LOG_END timestamp to compute RtcEventLog duration. by terelius · 8 years ago
  27. 1a518fc Request ownership for visualization tool by terelius · 8 years ago
  28. 8a0b128 Fix CopyOnWriteBuffer::cdata to return const pointer by danilchap · 8 years ago
  29. d8a9c53 Implement CopyOnWriteBuffer move using scoped_refptr move by danilchap · 8 years ago
  30. 76a44d5 Log current time if we stop logging before stop_time_. by terelius · 8 years ago
  31. 7a11ae3 Print entire error object for capture session error handler by haysc · 8 years ago
  32. cd25539 Increase audio buffer duration for iPhone 4s. by tkchin · 8 years ago
  33. 6a850c3 Register header extensions for rtx streams in the event log analyzer. by stefan · 8 years ago
  34. 70cfc14 Adds move assignment operator to scoped_refptr by Danil Chapovalov · 8 years ago
  35. 3d7346f Adds delta-time logging for audio playout by henrika · 8 years ago
  36. b77bd81 Temporarily remove problematic dcheck by Erik Språng · 8 years ago
  37. 1318103 Reland: Add BWE plot to event log analyzer. by Stefan Holmer · 8 years ago
  38. 3d9c71a Add an option to disable proximity sensor to AppRTC Demo Android. by Sami Kalliomaki · 8 years ago
  39. 0e4bffb Add an option to enable level control to Android AppRTC Demo. by Sami Kalliomaki · 8 years ago
  40. 737336d Add NACK rate throttling for audio channels. by Erik Språng · 8 years ago
  41. f388257 Rewrote UpdateToMix in the audio mixer. by aleloi · 8 years ago
  42. ea4c141 Revert of Rewrote UpdateToMix in the audio mixer. (patchset #5 id:120001 of https://codereview.webrtc.org/2132563002/ ) by terelius · 8 years ago
  43. 2942e24 Rewrote UpdateToMix in the audio mixer. by aleloi · 8 years ago
  44. a74363c Remove ports that are not used by any channel after timeout by Honghai Zhang · 8 years ago
  45. c309e0e Don't stop sending media on EWOULDBLOCK by skvlad · 8 years ago
  46. ec4f068 Style cleanups in RtpSender. by Sergey Ulanov · 8 years ago
  47. d762910 Use 15fps for iPhone 4S by tkchin · 8 years ago
  48. b5db1ec Delay destroying a port if new connections are created and destroyed. by Honghai Zhang · 8 years ago
  49. 8eeecab Merge SignalPortPruned and SignalPortsRemoved. by Honghai Zhang · 8 years ago
  50. 8bc7734 Memset svc_params in vp9 wrapper to avoid undefined behavior. by jackychen · 8 years ago
  51. 55b0b37 Use a better method to generate a random ID in IvfFileWriterTest. by deadbeef · 8 years ago
  52. 59c1ad5 Revert of Add BWE plot to event log analyzer. (patchset #6 id:100001 of https://codereview.webrtc.org/2188033004/ ) by terelius · 8 years ago
  53. 2beea2a Add BWE plot to event log analyzer. by Stefan Holmer · 8 years ago
  54. a23fc62 Fix bug where transport sequence numbers are allocated for packets without the header extension registered. by stefan · 8 years ago
  55. 5deaaaa Fix gitignore after setup_links change. by phoglund · 8 years ago
  56. c62ff86 Adds periodic volume-level logging for Android. by henrika · 8 years ago
  57. a0db81f Removed the memory pool from the mixer. by aleloi · 8 years ago
  58. 1a993d1 aimd: Remove unused variable by isheriff · 8 years ago
  59. 8b2233f Removed Module inheritance and unused methods and members TimeScheduler, Process() and TimeToNextProcess() from mixer. by aleloi · 8 years ago
  60. 178ae90 Fix an UI bug in AppRTC Demo where disable hardware AGC/NS summary was not getting updated. by sakal · 8 years ago
  61. 09f4510 Removed callback in old AudioConferenceMixer. by aleloi · 8 years ago
  62. e6b60a4 Migrated audio_codec_speed_tests target to GN. by aleloi · 8 years ago
  63. 00730c7 Migrated GN target insert_packet_with_timing. by aleloi · 8 years ago
  64. 333f206 Target delay_test migrated to GN. by aleloi · 8 years ago
  65. 9f73f7a Add Robolectric dependencies to setup_link directories. by ehmaldonado · 8 years ago
  66. d356843 Disable EndToEndTest.GetStats on Dr. Memory Full. by deadbeef · 8 years ago
  67. 29ff844 Add PeerConnection IsClosed check. by zhihuang · 8 years ago
  68. 93dd634 Treat foreground event as interruption end. by tkchin · 8 years ago
  69. 43e15bb Android Camera2Enumerator: Remove CameraAccessException to avoid VerifyError by magjed · 8 years ago
  70. 529f83c Add webrtc_perf_tests to BUILD.gn by ehmaldonado · 8 years ago
  71. 81031d6 Cleanup code in Camera2. by sakal · 8 years ago
  72. 85228d6 Regression test for issue where Opus DTX status was being forgotten. by ivoc · 8 years ago
  73. f22d3c4 Filter to formats that match device sensor array aspect ratio on pre-LMR1 devices offering only legacy camera2 support. by sakal · 8 years ago
  74. 4374a09 Only update codec type histogram if lifetime is long enough (10 sec). by asapersson · 8 years ago
  75. bb19a60 Fix a memory leak where EventLogger is created two times in the event tracer. by sakal · 8 years ago
  76. 5f5504f Don't crash if createDataChannel fails. by deadbeef · 8 years ago
  77. 1cd4664 Disable some Opus tests pending an update by flim · 8 years ago
  78. 47bded4 GN migration of target audio_coding/neteq/rtp_analyze. by aleloi · 8 years ago
  79. 2874ed5 [rtcp] App::Parse updated not to use RTCPUtility, by danilchap · 8 years ago
  80. 3022a34 GN migration of target audio_coding/neteq/neteq_test_tools by aleloi · 8 years ago
  81. 2c3f46e Add an option to disable built-in AGC/NS to AppRTC Demo. by sakal · 8 years ago
  82. 63d3f83 Fix a bug where Camera2 tried to start on FPS ranges multiplied by 1000. by sakal · 8 years ago
  83. 2fa1462 Convert Android camera tests to use the new createVideoSource API. by sakal · 8 years ago
  84. 86cc6ff Variable audio bitrate. by mflodman · 8 years ago
  85. 62d695f iOS HW encoder: Increase data rate limit by magjed · 8 years ago
  86. d9c7f8d Use NullSocketServer instead of PhysicalSocketServer in SignalThread by magjed · 8 years ago
  87. aa3520c Roll chromium_revision 0d90f310e3..48e079d573 (407439:407510) by phoglund · 8 years ago
  88. 192717e flaky EndToEndTest.DecodesRetransmittedFrame adjusted by danilchap · 8 years ago
  89. fdd381c Remove unrelated checks from DecodesRetransmittedFrame* tests by danilchap · 8 years ago
  90. 3c87ac4 Roll chromium_revision 32f0eb6bd9..0d90f310e3 (407397:407439) by phoglund · 8 years ago
  91. 8c695b4 Remove redundant UMA stat reporting. Remove logs that are noisy on Windows. by maxmorin · 8 years ago
  92. f89a571 [AFL] Allow webrtc fuzzers to be used with afl-fuzz. by metzman · 8 years ago
  93. 2638c6f Ignore zero bitrate updates in the UMA BWE stats as they represent network being down and would bias the stats. by stefan · 8 years ago
  94. 8058e58 Add loss-based BWE estimate to the outgoing bitrate plot. by terelius · 8 years ago
  95. 1c88e8c Roll chromium_revision 80b5d0ec09..32f0eb6bd9 (407388:407397) by phoglund · 8 years ago
  96. 223df8f Roll chromium_revision fa99476e6e..80b5d0ec09 (407379:407388) by phoglund · 8 years ago
  97. 7626af8 Roll chromium_revision 94de78b48f..fa99476e6e (407377:407379) by phoglund · 8 years ago
  98. 590cbef Roll chromium_revision 4a456989cf..94de78b48f (407367:407377) by phoglund · 8 years ago
  99. 764d83d Roll chromium_revision a042a214c6..4a456989cf (407099:407367) by phoglund · 8 years ago
  100. ff0a96d Fix a bug where SourceState on AndroidVideoTrackSource is set to live even on failure. by Sami Kalliomaki · 8 years ago