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gerrit-public.fairphone.software / platform / external / webrtc / 988cc0870bb49c855b4547f929462eeb38186d12
  1. 988cc08 [Cleanup] Add missing #include. Remove useless ones. by Yves Gerey · 6 years ago
  2. 8ee06a7 Increase coverage of AEC3 JSON config unit tests, fix bugs by Sam Zackrisson · 6 years ago
  3. f7a7c8a Stop adding RTT delay if there was not packet loss for enough time by Gustavo Garcia · 7 years ago
  4. 0627e21 Removes unused DeliverPacket from CallClient. by Sebastian Jansson · 6 years ago
  5. 9581bc4 Rename too long variable name to extmap_allow_mixed by Johannes Kron · 6 years ago
  6. 2edab4c Delete use of STR_CASE_CMP, replaced with absl::EqualsIgnoreCase. by Niels Möller · 6 years ago
  7. 01cf44d AEC3: Adding missing elements to the json parser by Per Åhgren · 6 years ago
  8. 3583693 3 TLs: add full stack test for short pattern + base heavy alloc. by Rasmus Brandt · 6 years ago
  9. 6ed37ba AEC3: Enable fuzzer testing of old render buffering code. by Gustaf Ullberg · 6 years ago
  10. d34597c Update test::CreateVideoStreams to use num_temporal_layers. by Åsa Persson · 6 years ago
  11. 98f5f6c In RtcpTransceiver functions with callback avoid relying on PostTaskAndReply by Danil Chapovalov · 6 years ago
  12. b0ab2ce Reland "Remove the HighPassFilter interface" by Sam Zackrisson · 6 years ago
  13. c9f9b87 AEC3: Improve dominant nearend detection by Gustaf Ullberg · 6 years ago
  14. ac19414 Export symbols needed by the Chromium component build (part 6). by Mirko Bonadei · 6 years ago
  15. 9a5da49 Split out a separate target for SimulcastEncoderAdapter by Jonathan Yu · 6 years ago
  16. 165148d Reland "Remove deprecated barcode scanning functionality" by Magnus Jedvert · 6 years ago
  17. 39feabe Enables FrameDecryptor to do an initial key request on frame decryption. by Benjamin Wright · 6 years ago
  18. 201596f Make packet max buffer size configurable via field trial flag by Johannes Kron · 6 years ago
  19. 68b2df7 Make protection_overhead_rate configurable through field trial. by Ying Wang · 6 years ago
  20. 6d21650 Don't decode frames with an older timestamp than the last decoded timestamp. by philipel · 6 years ago
  21. 38a3419 Revert "Remove deprecated barcode scanning functionality" by Alessio Bazzica · 6 years ago
  22. 67b011d Use BitrateAllocatorInterface in AudioSendStream and VideoSendStream by Niels Möller · 6 years ago
  23. ff292f3 Remove deprecated barcode scanning functionality by Magnus Jedvert · 6 years ago
  24. 635474e Compute RTCConnectionState and RTCIceConnectionState. by Jonas Olsson · 6 years ago
  25. 800e121 Adds support to change transport routes in Scenario tests. by Sebastian Jansson · 6 years ago
  26. 8d33c0c Adds field trial to do safer reset on route change. by Sebastian Jansson · 6 years ago
  27. c98849c AEC3: changes the signal used for deciding when to update the erle so the reverb render signal is now used by Jesús de Vicente Peña · 6 years ago
  28. ecdd432 Routing unacknowledged data in TransportFeedbackAdapter. by Sebastian Jansson · 6 years ago
  29. e482ff8 Audio codecs API: Remove some weasel words in the docs by Karl Wiberg · 6 years ago
  30. 57dd881 Delete dead code in webrtc_libyuv.cc by Niels Möller · 6 years ago
  31. 9acf1c1 Reland "Make sure Chromium will pick the correct field_trial/metric impl." by Mirko Bonadei · 6 years ago
  32. 648d28a Media engine and channel support for per-channel dscp values, specified by RtpParameter by Tim Haloun · 6 years ago
  33. 51cc30c Fix a null reference bug in NetworkMonitorAutoDetect.getNetworkState. by Qingsi Wang · 6 years ago
  34. cb21ffe Add blob-encoding support for RTC event logs by Elad Alon · 6 years ago
  35. 2dfa998 Reland "Prefix flag macros with WEBRTC_." by Mirko Bonadei · 6 years ago
  36. c538fc7 Revert "Prefix flag macros with WEBRTC_." by Mirko Bonadei · 6 years ago
  37. 1cb20de Revert "Make sure Chromium will pick the correct field_trial/metric impl." by Niklas Enbom · 6 years ago
  38. 3c7d599 Replace _stricmp with absl::EqualsIgnoreCase by Niels Möller · 6 years ago
  39. 53347b7 Mute failed tests when no sanitizer defects. by Yves Gerey · 6 years ago
  40. 2baa3c4 Roll chromium_revision c66210d3ab..b1cb85713b (601019:601125) by chromium-webrtc-autoroll · 6 years ago
  41. 0d26c99 Set renderThreadHandler to null on uncaught exception in EglRenderer. by Sami Kalliomäki · 6 years ago
  42. 5ccdc13 Prefix flag macros with WEBRTC_. by Mirko Bonadei · 6 years ago
  43. 8dc280d Make sure Chromium will pick the correct field_trial/metric impl. by Mirko Bonadei · 6 years ago
  44. 1ddc5b6 Export symbols needed by the Chromium component build (part 5). by Mirko Bonadei · 6 years ago
  45. cf58bf7 Move the SocketStream class to test target by Niels Möller · 6 years ago
  46. bc6a06c Adding missing #include on absl/memory/memory.h. by Mirko Bonadei · 6 years ago
  47. 82d4329 Delete unused test class StreamSource by Niels Möller · 6 years ago
  48. 2461c31 Roll chromium_revision 343f58e4df..c66210d3ab (600903:601019) by chromium-webrtc-autoroll · 6 years ago
  49. 97fc11f Fix the 'SetConfiguration(RTCConfiguration::use_media_transport)' setting. by Piotr (Peter) Slatala · 6 years ago
  50. 28c437c Roll chromium_revision 834490b775..343f58e4df (600802:600903) by chromium-webrtc-autoroll · 6 years ago
  51. aad5d36 Roll chromium_revision fc405b495a..834490b775 (600654:600802) by chromium-webrtc-autoroll · 6 years ago
  52. cb06cac Moves fake media engine implementation to cc file. by Sebastian Jansson · 6 years ago
  53. 7dc9774 Delete unused code from media/base/testutils.{cc,h} by Niels Möller · 6 years ago
  54. 192eeec Enable End-to-End Encrypted Video Frames. by Benjamin Wright · 6 years ago
  55. 6714bf9 Fix up OpenSSL/BoringSSL forward declarations. by David Benjamin · 6 years ago
  56. 50b1e6b Add fixed-size delta-encoding/decoding code for WebRTC event logs by Elad Alon · 6 years ago
  57. 608298b Move RtcEventLog::CreateNull implementation near declaration. by Danil Chapovalov · 6 years ago
  58. 78416b6 Adds time to initial config in analyzer code. by Sebastian Jansson · 6 years ago
  59. f203d73 Correctly slice MediaBitrateRecieved on content type in ReceiveStatisticsProxy by Ilya Nikolaevskiy · 6 years ago
  60. d28efe5 Adds field trial to AudioPriorityBitrateAllocationStrategy. by Sebastian Jansson · 6 years ago
  61. 65faede AEC3: Introduce partial adaptive filter resets at echo path changes by Per Åhgren · 6 years ago
  62. 1ffee36 AEC3: Remove ERLE uncertainty code that has no effect by Per Åhgren · 6 years ago
  63. 4b7a412 Relieve perkj@ of some OWNER duties by Karl Wiberg · 6 years ago
  64. 6347bda Remove expat from generate_licenses.py. by Mirko Bonadei · 6 years ago
  65. d0be002 Add missing #include to absl/memory/memory.h by tzik · 6 years ago
  66. d65d179 Export symbols needed by the Chromium component build (part 4). by Mirko Bonadei · 6 years ago
  67. 9d24795 rtc::ZeroOnFreeBuffer: Don't forget to zero memory we free in operator= by Karl Wiberg · 6 years ago
  68. b5541a0 Fix: Argv may be corrupted after InitGoogleMock found any related flags by Artem Titov · 6 years ago
  69. 576a333 Roll chromium_revision c926d3bb2f..fc405b495a (600547:600654) by chromium-webrtc-autoroll · 6 years ago
  70. f05cae3 Roll chromium_revision 8bef2e268b..c926d3bb2f (600433:600547) by chromium-webrtc-autoroll · 6 years ago
  71. 7fa6ee6 Adds support for "-" to a=ssrc msid lines. by Seth Hampson · 6 years ago
  72. 98a462c Reland "Reland "Propagate media transport to media channel."" by Anton Sukhanov · 6 years ago
  73. 55fab32 Roll chromium_revision c5242283d9..8bef2e268b (600305:600433) by chromium-webrtc-autoroll · 6 years ago
  74. bfb444c Adds new CryptoOption crypto_options.frame.require_frame_encryption. by Benjamin Wright · 6 years ago
  75. d932fba Track padding and header size in log event. by Bjorn Terelius · 6 years ago
  76. b9972fa Adds AudioNetworkAdaptation support to Scenario tests. by Sebastian Jansson · 6 years ago
  77. 09beff2 Add UseMediaTransport RTCConfiguration support in Java class by Piotr (Peter) Slatala · 6 years ago
  78. 2bff543 Removes undefined declarations in channel.h. by Sebastian Jansson · 6 years ago
  79. 4f3ce27 rtc::Buffer: Handle move self-assignment by Karl Wiberg · 6 years ago
  80. d189252 Delete more rtc_base/stringutils.* by Niels Möller · 6 years ago
  81. fab9129 Get frame type, width and height from the generic descriptor. by philipel · 6 years ago
  82. 34d990f Adding NetEq buffer full metric to UMA. by Minyue Li · 6 years ago
  83. a240daa Change verification of stream configs in RTC event log unittest. by Bjorn Terelius · 6 years ago
  84. 5a464d3 Add resolution to generic frame descriptor extension by Danil Chapovalov · 6 years ago
  85. 4744e5b Reland "Remove old video_bitrate_allocator.h" by Rasmus Brandt · 6 years ago
  86. dbb47b8 Roll chromium_revision d06a979d44..c5242283d9 (600199:600305) by chromium-webrtc-autoroll · 6 years ago
  87. f25303e Reland: Modernize rtc::SSLCertificate by Steve Anton · 6 years ago
  88. 28b6d1d Roll chromium_revision 2419220cab..d06a979d44 (600044:600199) by chromium-webrtc-autoroll · 6 years ago
  89. 9accc9f Revert "Reland "Propagate media transport to media channel."" by Oleh Prypin · 6 years ago
  90. c76b8ff Roll chromium_revision f8cad916e6..2419220cab (599923:600044) by chromium-webrtc-autoroll · 6 years ago
  91. aa1e7c2 Allow 'use_media_transport' to be modified on PeerConnection before local/remote description are set. by Piotr (Peter) Slatala · 6 years ago
  92. da65ed2 Reland "Propagate media transport to media channel." by Anton Sukhanov · 6 years ago
  93. 4905edb Reland: Use unique_ptr and ArrayView in SSLFingerprint by Steve Anton · 6 years ago
  94. 243cabe Formatting openssladapter to be more consistent. by Benjamin Wright · 6 years ago
  95. 4e5074e Add MediaTransportInterface factory to the Jni bindings by Piotr (Peter) Slatala · 6 years ago
  96. 9b1d679 Remove 'iOS32 Sim Debug (iOS 9.0)' from client.webrtc. by Mirko Bonadei · 6 years ago
  97. d895f42 Revert "Remove the HighPassFilter interface" by Niklas Enbom · 6 years ago
  98. c1bfe1a Avoids creating empty call_order file when no call order data is written by Per Åhgren · 6 years ago
  99. 6026f05 Calculate max payload size for an rtp packet to fit full video frame by Danil Chapovalov · 6 years ago
  100. f5e767d Don't send max allocation probe unless allocation changed. by Sebastian Jansson · 6 years ago