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gerrit-public.fairphone.software / platform / external / webrtc / 37cf2455a420124b341ad06ac27fa3c4dbd29d3c
  1. 37cf245 Revert "Propagate media transport to media channel." by Oleh Prypin · 6 years ago
  2. f409246 Roll chromium_revision 3b54b6aa8b..03013c95df (599343:599460) by chromium-webrtc-autoroll · 6 years ago
  3. 8c16f74 Propagate media transport to media channel. by Anton Sukhanov · 6 years ago
  4. dbc2ea7 Roll chromium_revision c12ec9eedc..3b54b6aa8b (599188:599343) by chromium-webrtc-autoroll · 6 years ago
  5. 55cd3ac Modernize rtc::SSLCertificate by Steve Anton · 6 years ago
  6. 47f3240 Reland: Use unique_ptr and ArrayView in SSLFingerprint by Steve Anton · 6 years ago
  7. 5e23a41 Removes backwards compatability CryptoOptions support. by Benjamin Wright · 6 years ago
  8. 23e48fb Move expectations from eventlog unittests to helper functions. by Bjorn Terelius · 6 years ago
  9. f7fee39 Remove rtc_base:rtc_base_generic. by Mirko Bonadei · 6 years ago
  10. b354f74 Roll chromium_revision d47784f23e..c12ec9eedc (599082:599188) by chromium-webrtc-autoroll · 6 years ago
  11. 6af1c92 Add mock_video_encoder.h to api/test by Erik Språng · 6 years ago
  12. 3b4b4f5 Mitigate miscalculation of rtp packet size by Danil Chapovalov · 6 years ago
  13. 781b2bd Restore "device type" for iOS internal.client.webrtc by Artem Titarenko · 6 years ago
  14. 62b1345 Get rid of thread_darwin file. by Kári Tristan Helgason · 6 years ago
  15. c34cf71 Revert "Remove old video_bitrate_allocator.h" by Oleh Prypin · 6 years ago
  16. 93428bf Move SdpType from/to string definition close to declaration. by Mirko Bonadei · 6 years ago
  17. 55d1af1 Remove support for microsecond resolution in RtcEventLogs. by Bjorn Terelius · 6 years ago
  18. 4529fbc Move TemporalLayers to api/video_codecs. by Erik Språng · 6 years ago
  19. 28d200c Roll chromium_revision 37b6d53f02..d47784f23e (598967:599082) by chromium-webrtc-autoroll · 6 years ago
  20. a54daf1 Reland "Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h" by Benjamin Wright · 6 years ago
  21. edd204e Roll chromium_revision 9d052f4b6f..37b6d53f02 (598839:598967) by chromium-webrtc-autoroll · 6 years ago
  22. 8f4bc41 Revert "Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h" by Oleh Prypin · 6 years ago
  23. 1cd39fa make CreateOffer/CreateAnswer use ice credentials of pooled sessions. by Jonas Oreland · 6 years ago
  24. df1bf00 Headers shouldn't include themselves. by Yves Gerey · 6 years ago
  25. ac2f3d1 Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h by Benjamin Wright · 6 years ago
  26. 8285841 Adds handling of untracked data to congestion controller. by Sebastian Jansson · 6 years ago
  27. ca51189 Roll chromium_revision f34485ffde..9d052f4b6f (598711:598839) by chromium-webrtc-autoroll · 6 years ago
  28. 0d399a8 Removes socket addresses from PacketInfo struct. by Sebastian Jansson · 6 years ago
  29. 20ad254 Adds tracking of allocated but unacknowledged bitrate. by Sebastian Jansson · 6 years ago
  30. 26968ba Delete unused utf8 conversion utilities by Niels Möller · 6 years ago
  31. e8038e9 Adds IP overhead info to PacketInfo. by Sebastian Jansson · 6 years ago
  32. 74cd1ef AEC3: Enabling by default the use of the stationarity properties at render at init by Jesús de Vicente Peña · 6 years ago
  33. 5350d1c RtcEventLogSource no longer uses deprecated parsing functions. by Bjorn Terelius · 6 years ago
  34. 499bc6c Fix race conditions for ReofferDoesNotCallOnTrack test. by Yves Gerey · 6 years ago
  35. 53e2211 AEC3: Kill kill-switches by Gustaf Ullberg · 6 years ago
  36. 8b3cc49 Adds default values for feedback/allocation indicators. by Sebastian Jansson · 6 years ago
  37. fb226af Remove some old logging in goog_cc for congestion window. by Ying Wang · 6 years ago
  38. a1d9ca4 Revert "Add ability to specify if rate controller of video encoder is trusted." by Oleh Prypin · 6 years ago
  39. cdc959f Compute video freeze metrics on rendered frames instead of on decoded by Ilya Nikolaevskiy · 6 years ago
  40. 3bdbc84 Moves pushback controller to GoogCC by Sebastian Jansson · 6 years ago
  41. f81170b Add error logs to RtpPacketHistory::GetBestFittingPacket when no packet is found. by Per Kjellander · 6 years ago
  42. ade98c9 Adds srte to WATCHLISTS. by Sebastian Jansson · 6 years ago
  43. 2b15626 Revert "Use unique_ptr and ArrayView in SSLFingerprint" by Henrik Grunell · 6 years ago
  44. 703259c Don't CHECK when parsing AEC3 parameters from json by Sam Zackrisson · 6 years ago
  45. 80bf775 Roll chromium_revision 2499289737..f34485ffde (598606:598711) by chromium-webrtc-autoroll · 6 years ago
  46. f7fcaf0 Use zero octets for rtp packet padding by Danil Chapovalov · 6 years ago
  47. 3d25530 Reland "Export symbols needed by the Chromium component build (part 1)." by Mirko Bonadei · 6 years ago
  48. 3e335d1 Add ability to specify if rate controller of video encoder is trusted. by Erik Språng · 6 years ago
  49. 88be972 Delete post_encode_callback by Niels Möller · 6 years ago
  50. 74f6c7e AEC3: Cleanup test code for platforms with clock-drift by Per Åhgren · 6 years ago
  51. d6b0796 AEC3: Ensure that the usage of stationary signal properties is not unset by Per Åhgren · 6 years ago
  52. 23b2a25 Remove unlimited retransmission for screenshare experiment code by Ilya Nikolaevskiy · 6 years ago
  53. cc21e61 Use unique_ptr and ArrayView in SSLFingerprint by Steve Anton · 6 years ago
  54. e8d2b1b Roll chromium_revision 8afdf16764..2499289737 (598496:598606) by chromium-webrtc-autoroll · 6 years ago
  55. f7dd9df Change TurnPort::Create to return a unique_ptr by Steve Anton · 6 years ago
  56. 9cfce17 Roll chromium_revision 0d09089dd5..8afdf16764 (598349:598496) by chromium-webrtc-autoroll · 6 years ago
  57. 0854eb6 Respond to SDP request extmap-allow-mixed. by Johannes Kron · 6 years ago
  58. a8f1e56 Change Port::Create methods to return a unique_ptr by Steve Anton · 6 years ago
  59. 7940da0 Integration of media_transport in JSepTransportController by Anton Sukhanov · 6 years ago
  60. 6cc9cca Don't reset streams for the FrameEncryptor / FrameDecryptor unless they changed. by Benjamin Wright · 6 years ago
  61. da67c16 Roll chromium_revision 8a25f94ac2..0d09089dd5 (598237:598349) by chromium-webrtc-autoroll · 6 years ago
  62. ca27091 Remove rtc_base:rtc_base_approved_generic. by Mirko Bonadei · 6 years ago
  63. ede8796 Print per-frame VMAF score instead of average. by Paulina Hensman · 6 years ago
  64. b3b0179 Fix backwards logic in rtc::Buffer::OnMovedFrom() by Karl Wiberg · 6 years ago
  65. 0213786 Add certificate gen/set functionality to bring Android closer to JS API by Michael Iedema · 6 years ago
  66. dcc0238 Don't increment timestamp on drop/reencode in LibvpxVp8Encoder. by Erik Språng · 6 years ago
  67. 5526e45 vp9: change x-google-profile-id to profile-id by Philipp Hancke · 6 years ago
  68. 028248c Add `rtc_enable_symbol_export` to incrementally create a WebRTC component. by Mirko Bonadei · 6 years ago
  69. b686396 Makes AudioSendStream signal that it's part of allocation. by Sebastian Jansson · 6 years ago
  70. 99a70a2 Remove rtc_base_approved_objc and introduce rtc_base:logging_mac. by Mirko Bonadei · 6 years ago
  71. edc49c1 [Cleanup] Remove unused swap function. by Yves Gerey · 6 years ago
  72. a4c8514 Add JSON parsing and corresponding ToString to EchoCanceller3Config by Sam Zackrisson · 6 years ago
  73. 2558c4e Remove ortc folder. by Mirko Bonadei · 6 years ago
  74. 88b68ac Create field trial for setting a minimum value for Opus encoder packet loss rate by Jakob Ivarsson · 6 years ago
  75. f08dd9d Disable flaky tests on mac perf bot by Ilya Nikolaevskiy · 6 years ago
  76. 1bca65b Makes RtpSender indicate allocation and feedback status on packets. by Sebastian Jansson · 6 years ago
  77. 81125f0 Implement (mostly) standards-compliant RTCIceTransportState. by Jonas Olsson · 6 years ago
  78. 5f35e96 Roll chromium_revision 476ae6d661..8a25f94ac2 (598136:598237) by chromium-webrtc-autoroll · 6 years ago
  79. c87b8c1 Moves GoogCC factory to API. by Sebastian Jansson · 6 years ago
  80. 0d8c100 AEC3: Decrease the suppression during the echo-only case by Per Åhgren · 6 years ago
  81. 463c764 Roll chromium_revision cfe6e706d0..476ae6d661 (598018:598136) by chromium-webrtc-autoroll · 6 years ago
  82. aabf204 Remove container typedefs from RelayServer by Steve Anton · 6 years ago
  83. 11358fe Use unique_ptr in port_unittest by Steve Anton · 6 years ago
  84. 13d392d AEC3: Utilize dominant nearend functionality to increase transparency by Per Åhgren · 6 years ago
  85. 3a3f027 Roll chromium_revision 0cf8926390..cfe6e706d0 (597915:598018) by chromium-webrtc-autoroll · 6 years ago
  86. 0378997 Adds flags indicating presence in allocation and feedback per packet. by Sebastian Jansson · 6 years ago
  87. 30e2d6e Moves locking outside function in RtpSender. by Sebastian Jansson · 6 years ago
  88. 789f459 Adds fields for unacknowledged data to transport feedback. by Sebastian Jansson · 6 years ago
  89. 20a49f3 Don't try to use CN if voice codec isn't mono by Karl Wiberg · 6 years ago
  90. 5fcc4de Roll chromium_revision f362b3e857..0cf8926390 (597811:597915) by chromium-webrtc-autoroll · 6 years ago
  91. 759f959 Refactor tests with ConfigurableFrameSizeEncoder by Niels Möller · 6 years ago
  92. 040f87f AEC3: Allow a more stable filter during double-talk by Gustaf Ullberg · 6 years ago
  93. 7730193 Remove SetExecutablePath, simplify ResourcePath by Patrik Höglund · 6 years ago
  94. 7004571 AEC3: Decrease the modelling of the reverb by Per Åhgren · 6 years ago
  95. d76a0fc Throttle the RTP decryption error messages in the SrtpSession and SrtpTransport by erikvarga@webrtc.org · 6 years ago
  96. b674cd1 Enable multithreading in libvpx VP9 decoder. by Sergey Silkin · 6 years ago
  97. d0bc462 Check if __IPHONE_OS_VERSION_MAX_ALLOWED is defined before reference by Joel Sutherland · 6 years ago
  98. 0414040 Fix race condition for SupportsFlexfecWithMultithreadedH264/0 test. by Yves Gerey · 6 years ago
  99. bf47198 Roll chromium_revision ba2e073e2c..f362b3e857 (597606:597811) by chromium-webrtc-autoroll · 6 years ago
  100. 4ff7214 Using TaskQueue for congestion controller by default. by Sebastian Jansson · 6 years ago