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webrtc / src.git / d48a2b14e7545d0a0778df753e062075c044e2a1
  1. d48a2b1 Prepare to avoid hops to worker for network events. by Tomas Gunnarsson · 4 years, 2 months ago
  2. 16ab60c Use CallbackList in DtlsHandshakeError in dtls_transport. by Lahiru Ginnaliya Gamathige · 4 years, 2 months ago
  3. c8421c4 Replace rtc::ThreadChecker with webrtc::SequenceChecker by Artem Titov · 4 years, 2 months ago
  4. 7358b40 Remove usage of AsyncInvoker in test class FakeNetworkSocket by Danil Chapovalov · 4 years, 2 months ago
  5. eee0e9e Remove passing rtp packet metadata through webrtc as array of bytes by Danil Chapovalov · 4 years, 2 months ago
  6. a1ca64c Roll chromium_revision 24c8af75a9..e753f3f38e (849426:849529) by chromium-webrtc-autoroll · 4 years, 2 months ago
  7. 3f41294 Update WebRTC code version (2021-02-02T04:02:53). by webrtc-version-updater · 4 years, 2 months ago
  8. 8463502 Roll chromium_revision 23e1598eba..24c8af75a9 (849278:849426) by chromium-webrtc-autoroll · 4 years, 2 months ago
  9. 2ab9b28 Get rid of unnecessary network thread Invoke in BaseChannel. by Taylor Brandstetter · 4 years, 2 months ago
  10. d1d2dc7 Roll chromium_revision 6aa02eda73..23e1598eba (849160:849278) by chromium-webrtc-autoroll · 4 years, 2 months ago
  11. 0f009dc Roll chromium_revision 402f104a74..6aa02eda73 (849004:849160) by chromium-webrtc-autoroll · 4 years, 2 months ago
  12. e7ded68 Fix integer overflow. by Jakob Ivarsson · 4 years, 2 months ago
  13. 78f87ab Delete use of RecursiveCriticalSection in JsepTransport by Niels Möller · 4 years, 2 months ago
  14. 51e5c4b Fix data race for config_ in AudioSendStream by Artem Titov · 4 years, 2 months ago
  15. e7c79fd Remove from chromium build targets that are not compatible with it. by Andrey Logvin · 4 years, 2 months ago
  16. d6604df Revert "Enable Video-QualityScaling experiment by default" by Ilya Nikolaevskiy · 4 years, 2 months ago
  17. 3faea70 allow empty scalability mode in AV1 encoder by Sergio Garcia Murillo · 4 years, 2 months ago
  18. c91c423 LibvpxVp9Encoder: add option to configure resolution_bitrate_limits. by Åsa Persson · 4 years, 2 months ago
  19. 989e6e7 Switch WebRTC's MB to RBE-CAS. by Mirko Bonadei · 4 years, 2 months ago
  20. b853d72 Update Apple device list by Dave Cowart · 4 years, 2 months ago
  21. 1f1e190 Roll chromium_revision 878a605f67..402f104a74 (848896:849004) by chromium-webrtc-autoroll · 4 years, 2 months ago
  22. 22b5efa Update WebRTC code version (2021-02-01T04:03:13). by webrtc-version-updater · 4 years, 2 months ago
  23. 41bfcf4 Inject network thread to Call. by Tomas Gunnarsson · 4 years, 2 months ago
  24. cedc3c7 Update WebRTC code version (2021-01-31T04:03:17). by webrtc-version-updater · 4 years, 2 months ago
  25. 5ac4212 Roll chromium_revision 2e446035f5..878a605f67 (848796:848896) by chromium-webrtc-autoroll · 4 years, 2 months ago
  26. 692f565 Update WebRTC code version (2021-01-30T04:04:03). by webrtc-version-updater · 4 years, 2 months ago
  27. b28d6ca Roll chromium_revision a635fd2809..2e446035f5 (848661:848796) by chromium-webrtc-autoroll · 4 years, 2 months ago
  28. d0acbd8 Revert "Do all BaseChannel operations within a single Thread::Invoke." by Taylor Brandstetter · 4 years, 2 months ago
  29. 271adff Roll chromium_revision 3353629fad..a635fd2809 (848531:848661) by chromium-webrtc-autoroll · 4 years, 2 months ago
  30. f9a6148 Roll chromium_revision bbd3f0121d..3353629fad (848401:848531) by chromium-webrtc-autoroll · 4 years, 2 months ago
  31. 7864600 Add absl_deps field for rtc_test and rtc_executable by Andrey Logvin · 4 years, 2 months ago
  32. b79acd8 Format webrtc/modules/audio_processing/transient/BUILD.gn file by Andrey Logvin · 4 years, 2 months ago
  33. ee8c275 Make DVQA CPU usage tests more stable by Andrey Logvin · 4 years, 2 months ago
  34. 5761e7b Running apply-iwyu on ~all files in pc/ by Harald Alvestrand · 4 years, 2 months ago
  35. 5e227ab Move under enable_google_benchmarks targets that rely on the benchmarks by Andrey Logvin · 4 years, 2 months ago
  36. 133c052 Make the config_ member of JsepTransportController const by Harald Alvestrand · 4 years, 2 months ago
  37. 9673ca4 Add field trial for bitrate limit interpolation for simulcast resolutions <180p. by Rasmus Brandt · 4 years, 2 months ago
  38. c5bdac6 Fix call_tests target dependencies by Andrey Logvin · 4 years, 2 months ago
  39. cd467b5 sdp: check that sctp is on a application content type by Philipp Hancke · 4 years, 2 months ago
  40. 1a29a5d Delete rtc::Bind by Niels Möller · 4 years, 2 months ago
  41. 4ea26e5 Update WebRTC code version (2021-01-29T04:02:57). by webrtc-version-updater · 4 years, 2 months ago
  42. 5e32fb8 Roll chromium_revision 0584f34f9c..bbd3f0121d (848277:848401) by chromium-webrtc-autoroll · 4 years, 2 months ago
  43. 9e9bf75 Add comment about setting transport_name field for RemoveIceCandidates. by Taylor Brandstetter · 4 years, 2 months ago
  44. 8bd0f97 Address CL comments from 200161. by Austin Orion · 4 years, 2 months ago
  45. 79f6452 Roll chromium_revision 3042ccda4e..0584f34f9c (848168:848277) by chromium-webrtc-autoroll · 4 years, 2 months ago
  46. 76bbc98 Adding MockVoipEngine for downstream project's tests by Tim Na · 4 years, 2 months ago
  47. 4f5322c Roll chromium_revision ba13ceb157..3042ccda4e (848044:848168) by chromium-webrtc-autoroll · 4 years, 2 months ago
  48. 066b5b6 Enable Video-QualityScaling experiment by default by Ilya Nikolaevskiy · 4 years, 2 months ago
  49. 54b925c add metrics for bundle usage by Philipp Hancke · 4 years, 2 months ago
  50. 1aa1d64 Ensure VideoLayersAllocation.frame_rate_fps can not overflow by Per Kjellander · 4 years, 2 months ago
  51. 37dfddd Avoid treating VP8 key frame in simulcast as delta frame by Danil Chapovalov · 4 years, 2 months ago
  52. 075fd4b Roll chromium_revision 84c1288826..ba13ceb157 (847739:848044) by chromium-webrtc-autoroll · 4 years, 2 months ago
  53. 3b68aa3 Move some RTC_LOG to RTC_DLOG. by Mirko Bonadei · 4 years, 2 months ago
  54. 70f9e24 Remove DtlsHandShakeError and replace it with a Function Pointer. by Lahiru Ginnaliya Gamathige · 4 years, 2 months ago
  55. b70c953 sdp: cross-check media type and protocol earlier by Philipp Hancke · 4 years, 2 months ago
  56. 2aad812 Refactor and implement WgcCapturerWin, a source agnostic capturer. by Austin Orion · 4 years, 2 months ago
  57. ca5d4a4 Roll chromium_revision 61b8ff5c89..84c1288826 (847529:847739) by chromium-webrtc-autoroll · 4 years, 2 months ago
  58. d19d0cf Reland: Add ability to load CreateDirect3DDeviceFromDXGIDevice from d3d11.dll by Austin Orion · 4 years, 2 months ago
  59. 49dbad0 Fixing audio timestamp stall during inactivation (under a kill switch) by Minyue Li · 4 years, 2 months ago
  60. 14b0e73 Roll chromium_revision e3ed290da5..61b8ff5c89 (846763:847529) by Artem Titov · 4 years, 2 months ago
  61. 934b5e2 Update WebRTC code version (2021-01-27T04:02:39). by webrtc-version-updater · 4 years, 2 months ago
  62. fae4fb1 video_replay: add support for IVF file output by Philipp Hancke · 4 years, 2 months ago
  63. 1d77c3e Fix roll chromium_revision 18311e2720..e3ed290da5 (844473:846763) by Mirko Bonadei · 4 years, 2 months ago
  64. 103876f av1: turn off a few tools that are not used for rtc by Jerome Jiang · 4 years, 2 months ago
  65. cc8a1f8 Add API to get current time mode from NetworkEmulationManager by Artem Titov · 4 years, 2 months ago
  66. 08f4690 Protect DefaultVideoQualityAnalyzer::peers_ with lock by Artem Titov · 4 years, 2 months ago
  67. c57089a Add new peer to injector when adding it to analyzer. Removed unused injector by Artem Titov · 4 years, 2 months ago
  68. 4f3a2eb Destroy previous offer instead of leaking it in PeerConnectionInterfaceTest.ExtmapAllowMixedIsConfigurable by Artem Titov · 4 years, 2 months ago
  69. 0a03ed8 Update WebRTC code version (2021-01-26T04:02:59). by webrtc-version-updater · 4 years, 2 months ago
  70. 5312a8f Add option to attach custom object to an rtp packet by Danil Chapovalov · 4 years, 2 months ago
  71. ded6636 Cleanup RtcpSender from legacy functionality by Danil Chapovalov · 4 years, 2 months ago
  72. 54fb32a IvfFileReader: Fix SpatialIndex values by Florent Castelli · 4 years, 2 months ago
  73. 437843f Update WebRTC code version (2021-01-25T04:04:10). by webrtc-version-updater · 4 years, 2 months ago
  74. b1b79f7 Update WebRTC code version (2021-01-24T04:03:52). by webrtc-version-updater · 4 years, 2 months ago
  75. 90776cb Enable RTC_NO_UNIQUE_ADDRESS on MSan builds. by Mirko Bonadei · 4 years, 2 months ago
  76. ef53a7f Reset IO thread checker when iOS audio unit stops by Brian Dai · 4 years, 2 months ago
  77. bf95da8 Update WebRTC code version (2021-01-23T04:04:31). by webrtc-version-updater · 4 years, 2 months ago
  78. 8c007ff Restrict usage of resolution bitrate limits to singlecast by Sergey Silkin · 4 years, 2 months ago
  79. 461b1d9 Restart CPU overuse detection when encoder settings has changed. by Jakob Ivarsson · 4 years, 2 months ago
  80. 2803a2d Make audio device mocks publicly visible by Steve Anton · 4 years, 2 months ago
  81. 8df643b Introduce FinalRefCountedObject template class by Danil Chapovalov · 4 years, 2 months ago
  82. cbacec5 Monitor the "concealed samples" stat for the audio during negotiation. by Harald Alvestrand · 4 years, 2 months ago
  83. 11215fe Require scalability mode to initialize av1 encoder. by Danil Chapovalov · 4 years, 2 months ago
  84. 2ed56fe Update WebRTC code version (2021-01-22T04:03:26). by webrtc-version-updater · 4 years, 2 months ago
  85. d2dd732 Introduce network emulated endpoint optional name for better logging by Artem Titov · 4 years, 2 months ago
  86. e4fd1ba Delete mutable rtc::CopyOnWriteBuffer::data by Danil Chapovalov · 4 years, 2 months ago
  87. 6031b74 Implement a Neon optimized function to find the argmax element in an array. by Ivo Creusen · 4 years, 2 months ago
  88. 03eed7c Fixes issue triggered by WebRTC-VP9-PerformanceFlags trial. by Erik Språng · 4 years, 2 months ago
  89. a7e34d3 Add resolution_bitrate_limits to EncoderInfo field trial. by Åsa Persson · 4 years, 2 months ago
  90. 026ad9a Update WebRTC code version (2021-01-21T04:03:14). by webrtc-version-updater · 4 years, 2 months ago
  91. 49b20f9 Fix race with SctpTransport destruction and usrsctp timer thread. by Taylor Brandstetter · 4 years, 2 months ago
  92. c2ae4c8 Allow separate dump sets for the data dumper in APM by Per Åhgren · 4 years, 2 months ago
  93. 0be1846 Fix enabling DependencyDescriptor for VP9 with spatial layers by Danil Chapovalov · 4 years, 2 months ago
  94. 1657baf Add `.cache` to .gitignore. by Rasmus Brandt · 4 years, 2 months ago
  95. 4f281f1 Cleanup FakeRtcEventLog from thread awareness by Danil Chapovalov · 4 years, 2 months ago
  96. 812c73c Another ilbc cross correlation fix by Ivo Creusen · 4 years, 2 months ago
  97. 5eb43b4 Prefix HAVE_SCTP macro with WEBRTC_. by Mirko Bonadei · 4 years, 2 months ago
  98. 6dcbcea Update WebRTC code version (2021-01-20T04:04:26). by webrtc-version-updater · 4 years, 2 months ago
  99. 33c0ab4 Call MediaChannel::OnPacketReceived on the network thread. by Tomas Gunnarsson · 4 years, 2 months ago
  100. 1cbf21e ChannelStatistics RTT test case around remote SSRC change. by Tim Na · 4 years, 2 months ago