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gerrit-public.fairphone.software / platform / external / webrtc / 3b56ee73a389a71176ec0d3016e9d3f1ca9a8b0a
  1. 3b56ee7 Export symbols needed by the Chromium component build (part 2). by Mirko Bonadei · 6 years ago
  2. d4d5f8a Formatting and style guide improvements for opensslstreamadapter.cc by Benjamin Wright · 6 years ago
  3. f714ee1 Revert "Deprecates legacy transport feedback adapter." by Mirko Bonadei · 6 years ago
  4. a5778e0 Deprecates legacy transport feedback adapter. by Sebastian Jansson · 6 years ago
  5. 5c94f55 Removes analyzer dependency on legacy congestion controller. by Sebastian Jansson · 6 years ago
  6. 82c71af Revert "Modernize rtc::SSLCertificate" by Niklas Enbom · 6 years ago
  7. 1e3ed16 Fix force_fieldtrials documentation in video_loopback by Elad Alon · 6 years ago
  8. 0391446 Removing forward declarations in paced_sender.h. by Sebastian Jansson · 6 years ago
  9. cd0ca2d Adds unit test for RTT based backoff. by Sebastian Jansson · 6 years ago
  10. 74c066c Merges ControlHandler and PacerController. by Sebastian Jansson · 6 years ago
  11. 7341ab6 Moves functionality to TransportFeedbackAdapter. by Sebastian Jansson · 6 years ago
  12. ed04912 Stop simulations when a LOG_END event is reached. by Ivo Creusen · 6 years ago
  13. 961dbea NetEq fuzzer: Restrict fuzzer input to 90000 bytes by Henrik Lundin · 6 years ago
  14. d8a52b3 Make ivoc owner of audio_coding. by Ivo Creusen · 6 years ago
  15. 6932fb2 Revert "Reland: Use unique_ptr and ArrayView in SSLFingerprint" by Mirko Bonadei · 6 years ago
  16. 40a7a35 Extract functionality of test_main into separate library. by Artem Titov · 6 years ago
  17. d2d2ecb Add command-line flag for setting the max number of packets in the buffer. by Ivo Creusen · 6 years ago
  18. c84cd95 Move MockVideoDecoder to api/test. by Erik Språng · 6 years ago
  19. 11539f0 AEC3: Simplify render buffering by Gustaf Ullberg · 6 years ago
  20. e07864e Moves rtc::SentPacket to separate target. by Sebastian Jansson · 6 years ago
  21. 76ad154 New method for precise packet reception time measurement. by Christoffer Rodbro · 6 years ago
  22. 2c7149b Add field trial to disable unsignalled video. by Åsa Persson · 6 years ago
  23. 6003e7a Fix FakeEncoder to produce correct bitrate for several temporal layers by Ilya Nikolaevskiy · 6 years ago
  24. a85995a Set frame duration per spatial layer. by Sergey Silkin · 6 years ago
  25. 9ac3c91 Refactor of extmap-allow-mixed in SessionDescription by Johannes Kron · 6 years ago
  26. cae8802 Delete force_mic_volume_max. by Patrik Höglund · 6 years ago
  27. 83bd37c Add field trials for configuring Opus encoder packet loss rate. by Jakob Ivarsson · 6 years ago
  28. fcebe0e in RtpPacketizers separate case 'frame fits into single packet'. by Danil Chapovalov · 6 years ago
  29. 1a35fbd Add field trial for normalized simulcast size. by Åsa Persson · 6 years ago
  30. 09256c1 Remove ios32_sim_ios9_dbg from CQ. by Mirko Bonadei · 6 years ago
  31. 147038c cq: explicitly mark presubmit tryjob as not re-usable in CQ. by Oleh Prypin · 6 years ago
  32. 9c18d21 Remove rtc_base/Dummy.java. by Mirko Bonadei · 6 years ago
  33. 28887a5 Roll chromium_revision 03013c95df..0df2607f98 (599460:599562) by chromium-webrtc-autoroll · 6 years ago
  34. 37cf245 Revert "Propagate media transport to media channel." by Oleh Prypin · 6 years ago
  35. f409246 Roll chromium_revision 3b54b6aa8b..03013c95df (599343:599460) by chromium-webrtc-autoroll · 6 years ago
  36. 8c16f74 Propagate media transport to media channel. by Anton Sukhanov · 6 years ago
  37. dbc2ea7 Roll chromium_revision c12ec9eedc..3b54b6aa8b (599188:599343) by chromium-webrtc-autoroll · 6 years ago
  38. 55cd3ac Modernize rtc::SSLCertificate by Steve Anton · 6 years ago
  39. 47f3240 Reland: Use unique_ptr and ArrayView in SSLFingerprint by Steve Anton · 6 years ago
  40. 5e23a41 Removes backwards compatability CryptoOptions support. by Benjamin Wright · 6 years ago
  41. 23e48fb Move expectations from eventlog unittests to helper functions. by Bjorn Terelius · 6 years ago
  42. f7fee39 Remove rtc_base:rtc_base_generic. by Mirko Bonadei · 6 years ago
  43. b354f74 Roll chromium_revision d47784f23e..c12ec9eedc (599082:599188) by chromium-webrtc-autoroll · 6 years ago
  44. 6af1c92 Add mock_video_encoder.h to api/test by Erik Språng · 6 years ago
  45. 3b4b4f5 Mitigate miscalculation of rtp packet size by Danil Chapovalov · 6 years ago
  46. 781b2bd Restore "device type" for iOS internal.client.webrtc by Artem Titarenko · 6 years ago
  47. 62b1345 Get rid of thread_darwin file. by Kári Tristan Helgason · 6 years ago
  48. c34cf71 Revert "Remove old video_bitrate_allocator.h" by Oleh Prypin · 6 years ago
  49. 93428bf Move SdpType from/to string definition close to declaration. by Mirko Bonadei · 6 years ago
  50. 55d1af1 Remove support for microsecond resolution in RtcEventLogs. by Bjorn Terelius · 6 years ago
  51. 4529fbc Move TemporalLayers to api/video_codecs. by Erik Språng · 6 years ago
  52. 28d200c Roll chromium_revision 37b6d53f02..d47784f23e (598967:599082) by chromium-webrtc-autoroll · 6 years ago
  53. a54daf1 Reland "Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h" by Benjamin Wright · 6 years ago
  54. edd204e Roll chromium_revision 9d052f4b6f..37b6d53f02 (598839:598967) by chromium-webrtc-autoroll · 6 years ago
  55. 8f4bc41 Revert "Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h" by Oleh Prypin · 6 years ago
  56. 1cd39fa make CreateOffer/CreateAnswer use ice credentials of pooled sessions. by Jonas Oreland · 6 years ago
  57. df1bf00 Headers shouldn't include themselves. by Yves Gerey · 6 years ago
  58. ac2f3d1 Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h by Benjamin Wright · 6 years ago
  59. 8285841 Adds handling of untracked data to congestion controller. by Sebastian Jansson · 6 years ago
  60. ca51189 Roll chromium_revision f34485ffde..9d052f4b6f (598711:598839) by chromium-webrtc-autoroll · 6 years ago
  61. 0d399a8 Removes socket addresses from PacketInfo struct. by Sebastian Jansson · 6 years ago
  62. 20ad254 Adds tracking of allocated but unacknowledged bitrate. by Sebastian Jansson · 6 years ago
  63. 26968ba Delete unused utf8 conversion utilities by Niels Möller · 6 years ago
  64. e8038e9 Adds IP overhead info to PacketInfo. by Sebastian Jansson · 6 years ago
  65. 74cd1ef AEC3: Enabling by default the use of the stationarity properties at render at init by Jesús de Vicente Peña · 6 years ago
  66. 5350d1c RtcEventLogSource no longer uses deprecated parsing functions. by Bjorn Terelius · 6 years ago
  67. 499bc6c Fix race conditions for ReofferDoesNotCallOnTrack test. by Yves Gerey · 6 years ago
  68. 53e2211 AEC3: Kill kill-switches by Gustaf Ullberg · 6 years ago
  69. 8b3cc49 Adds default values for feedback/allocation indicators. by Sebastian Jansson · 6 years ago
  70. fb226af Remove some old logging in goog_cc for congestion window. by Ying Wang · 6 years ago
  71. a1d9ca4 Revert "Add ability to specify if rate controller of video encoder is trusted." by Oleh Prypin · 6 years ago
  72. cdc959f Compute video freeze metrics on rendered frames instead of on decoded by Ilya Nikolaevskiy · 6 years ago
  73. 3bdbc84 Moves pushback controller to GoogCC by Sebastian Jansson · 6 years ago
  74. f81170b Add error logs to RtpPacketHistory::GetBestFittingPacket when no packet is found. by Per Kjellander · 6 years ago
  75. ade98c9 Adds srte to WATCHLISTS. by Sebastian Jansson · 6 years ago
  76. 2b15626 Revert "Use unique_ptr and ArrayView in SSLFingerprint" by Henrik Grunell · 6 years ago
  77. 703259c Don't CHECK when parsing AEC3 parameters from json by Sam Zackrisson · 6 years ago
  78. 80bf775 Roll chromium_revision 2499289737..f34485ffde (598606:598711) by chromium-webrtc-autoroll · 6 years ago
  79. f7fcaf0 Use zero octets for rtp packet padding by Danil Chapovalov · 6 years ago
  80. 3d25530 Reland "Export symbols needed by the Chromium component build (part 1)." by Mirko Bonadei · 6 years ago
  81. 3e335d1 Add ability to specify if rate controller of video encoder is trusted. by Erik Språng · 6 years ago
  82. 88be972 Delete post_encode_callback by Niels Möller · 6 years ago
  83. 74f6c7e AEC3: Cleanup test code for platforms with clock-drift by Per Åhgren · 6 years ago
  84. d6b0796 AEC3: Ensure that the usage of stationary signal properties is not unset by Per Åhgren · 6 years ago
  85. 23b2a25 Remove unlimited retransmission for screenshare experiment code by Ilya Nikolaevskiy · 6 years ago
  86. cc21e61 Use unique_ptr and ArrayView in SSLFingerprint by Steve Anton · 6 years ago
  87. e8d2b1b Roll chromium_revision 8afdf16764..2499289737 (598496:598606) by chromium-webrtc-autoroll · 6 years ago
  88. f7dd9df Change TurnPort::Create to return a unique_ptr by Steve Anton · 6 years ago
  89. 9cfce17 Roll chromium_revision 0d09089dd5..8afdf16764 (598349:598496) by chromium-webrtc-autoroll · 6 years ago
  90. 0854eb6 Respond to SDP request extmap-allow-mixed. by Johannes Kron · 6 years ago
  91. a8f1e56 Change Port::Create methods to return a unique_ptr by Steve Anton · 6 years ago
  92. 7940da0 Integration of media_transport in JSepTransportController by Anton Sukhanov · 6 years ago
  93. 6cc9cca Don't reset streams for the FrameEncryptor / FrameDecryptor unless they changed. by Benjamin Wright · 6 years ago
  94. da67c16 Roll chromium_revision 8a25f94ac2..0d09089dd5 (598237:598349) by chromium-webrtc-autoroll · 6 years ago
  95. ca27091 Remove rtc_base:rtc_base_approved_generic. by Mirko Bonadei · 6 years ago
  96. ede8796 Print per-frame VMAF score instead of average. by Paulina Hensman · 6 years ago
  97. b3b0179 Fix backwards logic in rtc::Buffer::OnMovedFrom() by Karl Wiberg · 6 years ago
  98. 0213786 Add certificate gen/set functionality to bring Android closer to JS API by Michael Iedema · 6 years ago
  99. dcc0238 Don't increment timestamp on drop/reencode in LibvpxVp8Encoder. by Erik Språng · 6 years ago
  100. 5526e45 vp9: change x-google-profile-id to profile-id by Philipp Hancke · 6 years ago