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gerrit-public.fairphone.software / platform / external / webrtc / eae5ab089cfa57ed3ee83ac03e83060ee0725244
  1. eae5ab0 Merge "Update scripts for merging with upstream" am: 081f2a69ce by Treehugger Robot · 4 years ago
  2. 081f2a6 Merge "Update scripts for merging with upstream" by Treehugger Robot · 4 years ago
  3. b045a3f Update scripts for merging with upstream by Yo Chiang · 4 years ago
  4. 2504546 Merge "Disable darwin build of webrtc" am: ff12d79867 by Treehugger Robot · 4 years ago
  5. ff12d79 Merge "Disable darwin build of webrtc" by Treehugger Robot · 4 years ago
  6. 50b1ad4 Disable darwin build of webrtc by Yo Chiang · 4 years ago
  7. b21a524 Merge changes Ie9a90d2f,Ida3bfe62,I2d596942 am: 4ba59966b7 by Jorge Moreira Broche · 4 years ago
  8. 08a471a Merge remote-tracking branch 'upstream-master' am: 8e1bf7eee4 by Jorge E. Moreira · 4 years ago
  9. 1206c45 Generate new Android.bp file and correct build errors am: b6df60492c by Jorge E. Moreira · 4 years ago
  10. 4ba5996 Merge changes Ie9a90d2f,Ida3bfe62,I2d596942 by Jorge Moreira Broche · 4 years ago
  11. 6a057ef Add some scripts useful for merging with upstream-master by Jorge E. Moreira · 4 years ago
  12. 8e1bf7e Merge remote-tracking branch 'upstream-master' by Jorge E. Moreira · 4 years ago
  13. b6df604 Generate new Android.bp file and correct build errors by Jorge E. Moreira · 4 years ago
  14. 1af77b4 [automerger skipped] Merge "Merge branch 'upstream-master'" am: e41ddee6b4 -s ours by Treehugger Robot · 4 years ago
  15. e41ddee Merge "Merge branch 'upstream-master'" by Treehugger Robot · 4 years ago
  16. f8ebb49 Merge branch 'upstream-master' by Jorge E. Moreira · 4 years ago
  17. 708b22c Roll chromium_revision 3a98cccc77..6b2bcf62a8 (790410:790515) by chromium-webrtc-autoroll · 4 years ago
  18. 529c064 Roll chromium_revision abd2dc6530..3a98cccc77 (790286:790410) by chromium-webrtc-autoroll · 4 years ago
  19. 00b8462 Implemented Android Demo Application for VoIP API by Jason Long · 4 years ago
  20. 1062cfe Add list of local_addresses for network stats object by Artem Titov · 4 years ago
  21. 31cb3ab Do not propage RTPFragmentationHeader into rtp_rtcp by Danil Chapovalov · 4 years ago
  22. a5d9c1a In DependencyDescriptor rtp header extension drop partial chain support by Danil Chapovalov · 4 years ago
  23. 0bc68bd Roll chromium_revision be10300048..abd2dc6530 (790129:790286) by chromium-webrtc-autoroll · 4 years ago
  24. c1a0737 Add per source network statistics by Artem Titov · 4 years ago
  25. d1c6472 Merge branch 'upstream-master' by Jorge E. Moreira · 4 years ago
  26. 7a9b96f AV1: set error_resilience to 0. by Jerome Jiang · 4 years ago
  27. 539e878 Roll chromium_revision db5fc682ba..be10300048 (790000:790129) by chromium-webrtc-autoroll · 4 years ago
  28. 8a47847 Roll chromium_revision 9a6944e044..db5fc682ba (789857:790000) by chromium-webrtc-autoroll · 4 years ago
  29. fa504e7 Check that capture time is valid before adjusting it. by Dan Minor · 4 years, 1 month ago
  30. fd5df68 Reduce time that video analyzer holds the frame in pc level framework by Andrey Logvin · 4 years ago
  31. 741a98d Roll chromium_revision 046ea7f66e..9a6944e044 (789730:789857) by chromium-webrtc-autoroll · 4 years ago
  32. d1e8e61 Roll chromium_revision d2f62519e1..046ea7f66e (789613:789730) by chromium-webrtc-autoroll · 4 years ago
  33. 20b701f Revert "sdp: parse and serialize b=TIAS" by Taylor · 4 years ago
  34. de90862 Roll chromium_revision 3cd7ca077b..d2f62519e1 (789472:789613) by chromium-webrtc-autoroll · 4 years ago
  35. 8fddf1f Delete callbacks from RtpDemuxer on ssrc binding by Danil Chapovalov · 4 years ago
  36. c359a17 Roll chromium_revision b8a4405eb1..3cd7ca077b (789356:789472) by chromium-webrtc-autoroll · 4 years ago
  37. e6542f2 Removed unused include from encoded_image.h. by philipel · 4 years ago
  38. fc4668d configure target bitrate in opus dtx tests by Philipp Hancke · 4 years ago
  39. b9d3809 Allows bitrate prober to discard delayed probes, unit type refactorings by Erik Språng · 4 years ago
  40. d74c0e6 Add presubmit test to use RTC_OBJC_TYPE on RTC_OBJC_EXPORT types. by Mirko Bonadei · 4 years ago
  41. 3cb525b Rename CriticalSection to RecursiveCriticalSection. by Markus Handell · 4 years ago
  42. 1a09fae Delete SignalDataChannelTransportNegotiated by Niels Möller · 4 years ago
  43. 6b82716 Delete unused enum values for DataChannelType by Niels Möller · 4 years ago
  44. 13fbc08 Explicitly set presentation style to full screen, because, by dharmesh · 4 years ago
  45. b8c03e4 Roll chromium_revision d850211387..b8a4405eb1 (789218:789356) by chromium-webrtc-autoroll · 4 years ago
  46. 0c5f244 Roll chromium_revision 46d8bf63b3..d850211387 (789115:789218) by chromium-webrtc-autoroll · 4 years ago
  47. 1ad8944 Implement Source enumeration and selection for WGC capturer by Austin Orion · 4 years ago
  48. f678870 Roll chromium_revision ef63f40c0d..46d8bf63b3 (789011:789115) by chromium-webrtc-autoroll · 4 years ago
  49. 820021d Ignore fragmentation header when packetizing H264 by Danil Chapovalov · 4 years ago
  50. 3d22108 Remove unused critical section includes. by Markus Handell · 4 years ago
  51. 11bfeb3 Roll chromium_revision f831fc29d7..ef63f40c0d (788907:789011) by chromium-webrtc-autoroll · 4 years ago
  52. 686a370 opus: take SILK vad result into account for voice detection by Philipp Hancke · 4 years ago
  53. 3592839 Add default values for VideoEncoderFactory::CodecInfo by Niels Möller · 4 years ago
  54. 007271f Delete obsolete TODO item by Niels Möller · 4 years ago
  55. fca3087 Do not use internal source in video send stream tests. by Mirta Dvornicic · 4 years ago
  56. e51d6ac Fix override declarations and delete related TODOs by Niels Möller · 4 years ago
  57. e43648a Add constrained high profile level for h264 codec to media_constants by Andrey Logvin · 4 years ago
  58. c8c01f3 Roll chromium_revision 4b7890cdbe..f831fc29d7 (788759:788907) by chromium-webrtc-autoroll · 4 years ago
  59. 4c7bb27 Remove rtc::GlobalLock. by Markus Handell · 4 years ago
  60. e7e17d3 Roll chromium_revision bc87af3aed..4b7890cdbe (788656:788759) by chromium-webrtc-autoroll · 4 years ago
  61. c2f113b Trigger CI bots. by Mirko Bonadei · 4 years ago
  62. c179575 Roll chromium_revision a29ceb7864..bc87af3aed (788510:788656) by chromium-webrtc-autoroll · 4 years ago
  63. 0dd35d3 Migrate to webrtc::GlobalMutex. by Markus Handell · 4 years ago
  64. a13e7a1 Add factory to create scalability structures by name by Danil Chapovalov · 4 years ago
  65. 97910da Do not use internal source in H.264 bitstream rewriting tests. by Mirta Dvornicic · 4 years ago
  66. c6801d4 sdp: parse and serialize b=TIAS by Philipp Hancke · 4 years ago
  67. 9ad1f6f Reland "Delete PeerConnectionInterface::BitrateParameters" by Niels Möller · 4 years ago
  68. c888ffa Delete CompositeDataChannelTransport by Niels Möller · 4 years ago
  69. 121f32c Roll chromium_revision 812a4946f7..a29ceb7864 (788405:788510) by chromium-webrtc-autoroll · 4 years ago
  70. 77ca290 Roll chromium_revision 20d0aa1d03..812a4946f7 (788294:788405) by chromium-webrtc-autoroll · 4 years ago
  71. 26d41f2 Roll chromium_revision 3e578a0ddf..20d0aa1d03 (787863:788294) by chromium-webrtc-autoroll · 4 years ago
  72. 80d2159 Use Android Q API to test if MediaCodecInfo is HW Accelerated by Byoungchan Lee · 4 years ago
  73. 84bb634 Delete legacy cricket::RtpHeaderExtension struct as unused by Danil Chapovalov · 4 years ago
  74. 47f87ab Roll chromium_revision 432e660d6d..3e578a0ddf (787714:787863) by chromium-webrtc-autoroll · 4 years ago
  75. c7c412a Check for null before accessing SctpTransport map. by Taylor Brandstetter · 4 years ago
  76. f60d4c2 Revert "Delete PeerConnectionInterface::BitrateParameters" by Artem Titov · 4 years ago
  77. 6e4416c Roll chromium_revision f629a87d19..432e660d6d (787611:787714) by chromium-webrtc-autoroll · 4 years ago
  78. 4ab7dde Migrate stray leftovers from rtc_base/ and test/ to webrtc::Mutex. by Markus Handell · 4 years ago
  79. 4379a7d Reland "Migrate modules/desktop_capture and modules/video_capture to webrtc::Mutex." by Markus Handell · 4 years ago
  80. 30c8eaa Add CPEPrefix. by Adrian Taylor · 4 years ago
  81. e2dfe74 Delete PeerConnectionInterface::BitrateParameters by Niels Möller · 4 years ago
  82. 4819098 Auto roller: send trooper notifications only when Commit-Queue+2. by Jeremy Leconte · 4 years ago
  83. 21621e9 Delete obsolete method JsepTransport::NegotiateDatagramTransport by Niels Möller · 4 years ago
  84. acc1123 Roll chromium_revision 598b095453..f629a87d19 (787511:787611) by chromium-webrtc-autoroll · 4 years ago
  85. 59f3c68 Inclusive language in DEPS. by Mirko Bonadei · 4 years ago
  86. 35926d9 Revert "Test luci.notifier." by Mirko Bonadei · 4 years ago
  87. 662918e Roll chromium_revision f0f2118569..598b095453 (787142:787511) by chromium-webrtc-autoroll · 4 years ago
  88. d3ec791 Reland "Test luci.notifier." by Mirko Bonadei · 4 years ago
  89. 2e978e3 Revert "Test luci.notifier." by Mirko Bonadei · 4 years ago
  90. bd393b2 Add target_sdk_version to rtc_test. by Mirko Bonadei · 4 years ago
  91. 84a812e Check old_vector_size prior to copying in RTPFragmentationHeader::Resize by Dan Minor · 4 years ago
  92. 8df59bc Test luci.notifier. by Mirko Bonadei · 4 years ago
  93. 1added5 Revert "Migrate modules/desktop_capture and modules/video_capture to webrtc::Mutex." by Markus Handell · 4 years ago
  94. 0800010 peerconnection: remove old helper function by Philipp Hancke · 4 years ago
  95. edacbd5 Reland "Implement packets_(sent | received) for RTCTransportStats" by Artem Titov · 4 years ago
  96. 76ad2e0 Add jitter buffer delay metric to pc level tests by Andrey Logvin · 4 years ago
  97. 265931e Migrate VideoStreamDecoderImpl to webrtc::Mutex. by Markus Handell · 4 years ago
  98. bc633d7 Roll chromium_revision a69b9c614c..f0f2118569 (786202:787142) by chromium-webrtc-autoroll · 4 years ago
  99. 4e5bc9f Reland "Complete migration from "track" to "inbound-rtp" stats" by Eldar Rello · 4 years ago
  100. 5e98742 Add missing dependencies to java targets. by Mirko Bonadei · 4 years ago