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gerrit-public.fairphone.software / platform / external / webrtc / d895f42bfb24738333e39c174f078d544d68dccf
  1. d895f42 Revert "Remove the HighPassFilter interface" by Niklas Enbom · 6 years ago
  2. c1bfe1a Avoids creating empty call_order file when no call order data is written by Per Åhgren · 6 years ago
  3. 6026f05 Calculate max payload size for an rtp packet to fit full video frame by Danil Chapovalov · 6 years ago
  4. f5e767d Don't send max allocation probe unless allocation changed. by Sebastian Jansson · 6 years ago
  5. a1c9312 Update proto for new event log format. by Bjorn Terelius · 6 years ago
  6. aba0633 Delete wrappers for snprintf and vsnprintf by Niels Möller · 6 years ago
  7. 3100fc1 Use color aligning in video quality analysis tool by Magnus Jedvert · 6 years ago
  8. 3e7b7b1 AEC3: Changes to initial behavior and handling of saturated echo by Per Åhgren · 6 years ago
  9. 276827c Export symbols needed by the Chromium component build (part 3). by Mirko Bonadei · 6 years ago
  10. 0753675 Using more specific dependencies in rtc_base. by Sebastian Jansson · 6 years ago
  11. 6c78ff4 Always verify packet wasn't resend recently before resending it. by Danil Chapovalov · 6 years ago
  12. 2d0c687 Remove |hw_encoder| and |hw_decoder| from VideoCodecTestFixture::Config. by Rasmus Brandt · 6 years ago
  13. f907c49 Delete unused code in rtc_base/stringutils.* by Niels Möller · 6 years ago
  14. 84d2827 Add generate_ios_coverage_command.py script by Artem Titarenko · 6 years ago
  15. 1298541 Removing unnecessary dependencies on socket.h. by Sebastian Jansson · 6 years ago
  16. 03d2801 Roll chromium_revision 0cecb6ce10..f8cad916e6 (599821:599923) by chromium-webrtc-autoroll · 6 years ago
  17. be65d48 Remove AECM comfort noise setting from API by Sam Zackrisson · 6 years ago
  18. e2405c1 Remove the HighPassFilter interface by Sam Zackrisson · 6 years ago
  19. d419db9 Adding support for logging severity LS_NONE. by Peter Hanspers · 6 years ago
  20. 2e47f7c Implement test class LoopbackMediaTransport by Niels Möller · 6 years ago
  21. f06bacc Add test that verifies that VideoEncoderConfig max_framerate is reported to source. by Åsa Persson · 6 years ago
  22. 2560e2e Removes Clock instance from RoundRobinPacketQueue. by Sebastian Jansson · 6 years ago
  23. 1927dfa Add tool for aligning color space of video files by Magnus Jedvert · 6 years ago
  24. f0e926f Add missing #include and deps to absl/memory by tzik · 6 years ago
  25. 1b26a0a Roll chromium_revision 0e821c2fa2..0cecb6ce10 (599702:599821) by chromium-webrtc-autoroll · 6 years ago
  26. a39a007 Reland "Deprecates legacy transport feedback adapter." by Sebastian Jansson · 6 years ago
  27. acaed83 Roll chromium_revision 0df2607f98..0e821c2fa2 (599562:599702) by chromium-webrtc-autoroll · 6 years ago
  28. c9e6b96 Add necessary frameworks to sdk objc audio targets. by Jiawei Ou · 6 years ago
  29. 3b56ee7 Export symbols needed by the Chromium component build (part 2). by Mirko Bonadei · 6 years ago
  30. d4d5f8a Formatting and style guide improvements for opensslstreamadapter.cc by Benjamin Wright · 6 years ago
  31. f714ee1 Revert "Deprecates legacy transport feedback adapter." by Mirko Bonadei · 6 years ago
  32. a5778e0 Deprecates legacy transport feedback adapter. by Sebastian Jansson · 6 years ago
  33. 5c94f55 Removes analyzer dependency on legacy congestion controller. by Sebastian Jansson · 6 years ago
  34. 82c71af Revert "Modernize rtc::SSLCertificate" by Niklas Enbom · 6 years ago
  35. 1e3ed16 Fix force_fieldtrials documentation in video_loopback by Elad Alon · 6 years ago
  36. 0391446 Removing forward declarations in paced_sender.h. by Sebastian Jansson · 6 years ago
  37. cd0ca2d Adds unit test for RTT based backoff. by Sebastian Jansson · 6 years ago
  38. 74c066c Merges ControlHandler and PacerController. by Sebastian Jansson · 6 years ago
  39. 7341ab6 Moves functionality to TransportFeedbackAdapter. by Sebastian Jansson · 6 years ago
  40. ed04912 Stop simulations when a LOG_END event is reached. by Ivo Creusen · 6 years ago
  41. 961dbea NetEq fuzzer: Restrict fuzzer input to 90000 bytes by Henrik Lundin · 6 years ago
  42. d8a52b3 Make ivoc owner of audio_coding. by Ivo Creusen · 6 years ago
  43. 6932fb2 Revert "Reland: Use unique_ptr and ArrayView in SSLFingerprint" by Mirko Bonadei · 6 years ago
  44. 40a7a35 Extract functionality of test_main into separate library. by Artem Titov · 6 years ago
  45. d2d2ecb Add command-line flag for setting the max number of packets in the buffer. by Ivo Creusen · 6 years ago
  46. c84cd95 Move MockVideoDecoder to api/test. by Erik Språng · 6 years ago
  47. 11539f0 AEC3: Simplify render buffering by Gustaf Ullberg · 6 years ago
  48. e07864e Moves rtc::SentPacket to separate target. by Sebastian Jansson · 6 years ago
  49. 76ad154 New method for precise packet reception time measurement. by Christoffer Rodbro · 6 years ago
  50. 2c7149b Add field trial to disable unsignalled video. by Åsa Persson · 6 years ago
  51. 6003e7a Fix FakeEncoder to produce correct bitrate for several temporal layers by Ilya Nikolaevskiy · 6 years ago
  52. a85995a Set frame duration per spatial layer. by Sergey Silkin · 6 years ago
  53. 9ac3c91 Refactor of extmap-allow-mixed in SessionDescription by Johannes Kron · 6 years ago
  54. cae8802 Delete force_mic_volume_max. by Patrik Höglund · 6 years ago
  55. 83bd37c Add field trials for configuring Opus encoder packet loss rate. by Jakob Ivarsson · 6 years ago
  56. fcebe0e in RtpPacketizers separate case 'frame fits into single packet'. by Danil Chapovalov · 6 years ago
  57. 1a35fbd Add field trial for normalized simulcast size. by Åsa Persson · 6 years ago
  58. 09256c1 Remove ios32_sim_ios9_dbg from CQ. by Mirko Bonadei · 6 years ago
  59. 147038c cq: explicitly mark presubmit tryjob as not re-usable in CQ. by Oleh Prypin · 6 years ago
  60. 9c18d21 Remove rtc_base/Dummy.java. by Mirko Bonadei · 6 years ago
  61. 28887a5 Roll chromium_revision 03013c95df..0df2607f98 (599460:599562) by chromium-webrtc-autoroll · 6 years ago
  62. 37cf245 Revert "Propagate media transport to media channel." by Oleh Prypin · 6 years ago
  63. f409246 Roll chromium_revision 3b54b6aa8b..03013c95df (599343:599460) by chromium-webrtc-autoroll · 6 years ago
  64. 8c16f74 Propagate media transport to media channel. by Anton Sukhanov · 6 years ago
  65. dbc2ea7 Roll chromium_revision c12ec9eedc..3b54b6aa8b (599188:599343) by chromium-webrtc-autoroll · 6 years ago
  66. 55cd3ac Modernize rtc::SSLCertificate by Steve Anton · 6 years ago
  67. 47f3240 Reland: Use unique_ptr and ArrayView in SSLFingerprint by Steve Anton · 6 years ago
  68. 5e23a41 Removes backwards compatability CryptoOptions support. by Benjamin Wright · 6 years ago
  69. 23e48fb Move expectations from eventlog unittests to helper functions. by Bjorn Terelius · 6 years ago
  70. f7fee39 Remove rtc_base:rtc_base_generic. by Mirko Bonadei · 6 years ago
  71. b354f74 Roll chromium_revision d47784f23e..c12ec9eedc (599082:599188) by chromium-webrtc-autoroll · 6 years ago
  72. 6af1c92 Add mock_video_encoder.h to api/test by Erik Språng · 6 years ago
  73. 3b4b4f5 Mitigate miscalculation of rtp packet size by Danil Chapovalov · 6 years ago
  74. 781b2bd Restore "device type" for iOS internal.client.webrtc by Artem Titarenko · 6 years ago
  75. 62b1345 Get rid of thread_darwin file. by Kári Tristan Helgason · 6 years ago
  76. c34cf71 Revert "Remove old video_bitrate_allocator.h" by Oleh Prypin · 6 years ago
  77. 93428bf Move SdpType from/to string definition close to declaration. by Mirko Bonadei · 6 years ago
  78. 55d1af1 Remove support for microsecond resolution in RtcEventLogs. by Bjorn Terelius · 6 years ago
  79. 4529fbc Move TemporalLayers to api/video_codecs. by Erik Språng · 6 years ago
  80. 28d200c Roll chromium_revision 37b6d53f02..d47784f23e (598967:599082) by chromium-webrtc-autoroll · 6 years ago
  81. a54daf1 Reland "Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h" by Benjamin Wright · 6 years ago
  82. edd204e Roll chromium_revision 9d052f4b6f..37b6d53f02 (598839:598967) by chromium-webrtc-autoroll · 6 years ago
  83. 8f4bc41 Revert "Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h" by Oleh Prypin · 6 years ago
  84. 1cd39fa make CreateOffer/CreateAnswer use ice credentials of pooled sessions. by Jonas Oreland · 6 years ago
  85. df1bf00 Headers shouldn't include themselves. by Yves Gerey · 6 years ago
  86. ac2f3d1 Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h by Benjamin Wright · 6 years ago
  87. 8285841 Adds handling of untracked data to congestion controller. by Sebastian Jansson · 6 years ago
  88. ca51189 Roll chromium_revision f34485ffde..9d052f4b6f (598711:598839) by chromium-webrtc-autoroll · 6 years ago
  89. 0d399a8 Removes socket addresses from PacketInfo struct. by Sebastian Jansson · 6 years ago
  90. 20ad254 Adds tracking of allocated but unacknowledged bitrate. by Sebastian Jansson · 6 years ago
  91. 26968ba Delete unused utf8 conversion utilities by Niels Möller · 6 years ago
  92. e8038e9 Adds IP overhead info to PacketInfo. by Sebastian Jansson · 6 years ago
  93. 74cd1ef AEC3: Enabling by default the use of the stationarity properties at render at init by Jesús de Vicente Peña · 6 years ago
  94. 5350d1c RtcEventLogSource no longer uses deprecated parsing functions. by Bjorn Terelius · 6 years ago
  95. 499bc6c Fix race conditions for ReofferDoesNotCallOnTrack test. by Yves Gerey · 6 years ago
  96. 53e2211 AEC3: Kill kill-switches by Gustaf Ullberg · 6 years ago
  97. 8b3cc49 Adds default values for feedback/allocation indicators. by Sebastian Jansson · 6 years ago
  98. fb226af Remove some old logging in goog_cc for congestion window. by Ying Wang · 6 years ago
  99. a1d9ca4 Revert "Add ability to specify if rate controller of video encoder is trusted." by Oleh Prypin · 6 years ago
  100. cdc959f Compute video freeze metrics on rendered frames instead of on decoded by Ilya Nikolaevskiy · 6 years ago