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webrtc / src / 769ef2030985a62c4f447c9c05c60f1de1add85b
  1. 769ef20 [rct_tools/video_encoder] Support yuv input and PSNR/Bitrate validation. by Zhaoliang Ma · 9 months ago
  2. e6ad337 PipeWire capture: hide cursor when it goes off screen or is not visible by Jan Grulich · 9 months ago
  3. 2d6c397 Update WebRTC code version (2024-06-27T04:06:16). by webrtc-version-updater · 8 months ago
  4. c2c6442 Add RTC_EXPORT to TimeUTCMicros (2nd try) by Olov Brändström · 9 months ago
  5. 20b8e33 Add AudioEncoderOpus constructors that use field trials from Environment by Danil Chapovalov · 9 months ago
  6. 81a3d95 Add checks and explicit buffer inititalization for FrameCombiner by Tommi · 9 months ago
  7. 0bbc8ce Enable flexible mode by default by Sergey Silkin · 9 months ago
  8. 1030eaa Provide Environment to create an audio encoder in tests by Danil Chapovalov · 9 months ago
  9. eb3da2b Extract video writing into separate target by Artem Titov · 9 months ago
  10. 0592d2b Revert "Roll chromium_revision 536609c347..ba1ae79f58 (1316213:1319128)" by Björn Terelius · 9 months ago
  11. a6c34d1 Introduce an empty target for video_frame_writer by Artem Titov · 9 months ago
  12. f86c247 Remove api/wrapping_async_dns_resolver.cc. by Mirko Bonadei · 9 months ago
  13. accef6a Allow for reordering around IRAPs. by philipel · 9 months ago
  14. 889402e Update WebRTC code version (2024-06-26T04:04:55). by webrtc-version-updater · 9 months ago
  15. 3be745a Roll chromium_revision 536609c347..ba1ae79f58 (1316213:1319128) by Björn Terelius · 9 months ago
  16. 2086ff5 Mac SCK capturer: Set per-frame capture_time_ms and DPI values. by Lambros Lambrou · 9 months ago
  17. 3fede87 Remove rtc_base/helpers.h and crypto_random include in port allocator by Philipp Hancke · 9 months ago
  18. 27abd69 Add RTC_EXPORT to TimeUTCMicros by Olov Brändström · 9 months ago
  19. eaea3e2 Extend AudioEncoderFactoryTemplate to pass Environment to AudioEncoder factory traits by Danil Chapovalov · 9 months ago
  20. d03ce76 Add support for pred_weight_table by Sergio Garcia Murillo · 9 months ago
  21. e71fa4e Revert "Clean up SRTP helper functions" by Björn Terelius · 9 months ago
  22. 46b43e0 Update support for missing HIGH profiles and 1080p by Sergio Garcia Murillo · 9 months ago
  23. 12b861e Delete FieldTrialsView parameter for AudioEncoderFactoryTemplate as unused by Danil Chapovalov · 9 months ago
  24. 0adf973 Update WebRTC code version (2024-06-25T04:06:01). by webrtc-version-updater · 9 months ago
  25. c47f649 Clean up SRTP helper functions by Philipp Hancke · 9 months ago
  26. 26d3e56 Add AV1 screencast perf test by Sergey Silkin · 9 months ago
  27. c8b857f Always use SEA in video quality tests by Sergey Silkin · 9 months ago
  28. 9603aa1 Update WebRTC code version (2024-06-23T04:02:54). by webrtc-version-updater · 9 months ago
  29. af1f3a8 Update WebRTC code version (2024-06-22T04:02:50). by webrtc-version-updater · 9 months ago
  30. 3a45801 Make Unit types factories from float numbers be constexpr by Danil Chapovalov · 9 months ago
  31. e226676 Update WebRTC code version (2024-06-21T04:02:45). by webrtc-version-updater · 9 months ago
  32. 3069c60 Add desktop-capture option for ScreenCaptureKit on macOS. by Lambros Lambrou · 9 months ago
  33. dedb03e Fix RTCMTLNSVideoView undefined symbol error by Anton Barkov · 1 year ago
  34. 0fd6731 Reset the speech encoder when creating a comfort noise encoder. by Jakob Ivarsson · 9 months ago
  35. 85c1db0 Update WebRTC code version (2024-06-20T04:02:53). by webrtc-version-updater · 9 months ago
  36. defafcb Pass random seed to SchedulableNetworkBehavior. by Jeremy Leconte · 9 months ago
  37. d5238b0 Support running gn_check_autofix.py on a local build dir (e.g. out/Default) by Björn Terelius · 9 months ago
  38. aefed55 [iwyu][1\n] Applying to api/[a-s]* by Dor Hen · 9 months ago
  39. e7a305d Update WebRTC code version (2024-06-19T04:02:52). by webrtc-version-updater · 9 months ago
  40. d4a6c3f New macOS screen-capturer which uses ScreenCaptureKit. by Lambros Lambrou · 9 months ago
  41. 0f86252 Video encoding: allow to use system OpenH264 by Jan Grulich · 10 months ago
  42. 578905e Provide Environment to create audio encoders in both prod code paths by Danil Chapovalov · 9 months ago
  43. fc6df05 Computing and propagating the audio stats totalprocessingdelay. by Jesús de Vicente Peña · 9 months ago
  44. 418bcf2 Expose a PeerConnection's NetworkControllerInterface instance by Tony Herre · 9 months ago
  45. 799c8e6 Update WebRTC code version (2024-06-18T04:02:44). by webrtc-version-updater · 9 months ago
  46. a93d5a0 Roll chromium_revision 5a273f36b5..536609c347 (1316042:1316213) by chromium-webrtc-autoroll · 9 months ago
  47. eed9422 Reset VTCompressionSession when underlying CVPixelBufferPoolRef isn't valid by Brian Clymer · 9 months ago
  48. 7115de6 Roll chromium_revision c72aa689a7..5a273f36b5 (1315734:1316042) by chromium-webrtc-autoroll · 9 months ago
  49. feea82f Fix issue with SchedulableNetworkBehavior::UpdateConfigAndReschedule returning negative delay by Per K · 9 months ago
  50. 6948d84 Change AudioEncoderFactory api to provide Environment to construct AudioEncoders by Danil Chapovalov · 9 months ago
  51. da4d496 IWYU api/audio_codecs (not subdirectories) by Harald Alvestrand · 9 months ago
  52. 6056976 Updates to AudioFrameView and VectorFloatFrame by Tommi · 9 months ago
  53. e19ce9b Fix is_first_packet_in_frame when receiving multiple slices per H264 frame by Sergio Garcia Murillo · 9 months ago
  54. a0b22af Revert "Temporary add 'RTPVideoHeaderH264::nalus_length'." by Jeremy Leconte · 9 months ago
  55. c24b2d5 Roll chromium_revision e80ae6ea68..c72aa689a7 (1315265:1315734) by chromium-webrtc-autoroll · 9 months ago
  56. 05c6e74 Better capture the goal of TurnPortTest.TestChannelBindGetErrorResponse by Mirko Bonadei · 9 months ago
  57. 04dd95f Temporary add 'RTPVideoHeaderH264::nalus_length'. by Jeremy Leconte · 9 months ago
  58. 72302cc Include-what-you-use rtc_base/numerics/ by Björn Terelius · 9 months ago
  59. 08b649b Include-what-you-use api/rtc_event_log_output* by Björn Terelius · 9 months ago
  60. 77ffbd3 Include-what-you-use api/rtc_event_log/ by Björn Terelius · 9 months ago
  61. 504f323 Update WebRTC code version (2024-06-15T04:02:13). by webrtc-version-updater · 9 months ago
  62. 537543b Roll chromium_revision 837c81d9f7..e80ae6ea68 (1315145:1315265) by chromium-webrtc-autoroll · 9 months ago
  63. 469e698 Remove kMaxNalusPerPacket hard limit for H264 frames by Sergio Garcia Murillo · 9 months ago
  64. 94fa6bf Roll chromium_revision 5aae97f666..837c81d9f7 (1315026:1315145) by chromium-webrtc-autoroll · 9 months ago
  65. da9ef00 Use iOS 17.5.1 for perf by Christoffer Dewerin · 9 months ago
  66. 025d69b PipeWire video capture: mmap() PipeWire buffers with MAP_SHARED by Jan Grulich · 9 months ago
  67. f13a0e9 Update WebRTC code version (2024-06-14T04:04:42). by webrtc-version-updater · 9 months ago
  68. 6118951 Roll chromium_revision b86ab04138..5aae97f666 (1314628:1315026) by chromium-webrtc-autoroll · 9 months ago
  69. 3252f5d PipeWire capture: fix mmap arguments by Jan Grulich · 9 months ago
  70. 093824c Switch away from hz to samples per channel for FrameCombiner et al by Tommi · 9 months ago
  71. da485a1 Implement delayed start of Scheduled network configuration by Per K · 9 months ago
  72. 2da85bc Roll chromium_revision c4e011d5c7..b86ab04138 (1314523:1314628) by chromium-webrtc-autoroll · 9 months ago
  73. f9f631c Add terelius@ as owner of test/network by Per K · 9 months ago
  74. b244727 [Android] Add RtcError class and use it in RtpTransceiver.setCodecPreferences by Byoungchan Lee · 9 months ago
  75. 6724f1b Fix default link capacity in standalone loopback tests by Johannes Kron · 9 months ago
  76. c3aeffd PipeWire camera: add support for BGRA/RGBA formats by Jan Grulich · 9 months ago
  77. 94a6b92 Comment out device_status for ios internal perf for now and see if the tests run by Christoffer Dewerin · 9 months ago
  78. 8c0e628 Roll chromium_revision 05621b945d..c4e011d5c7 (1313445:1314523) by Christoffer Dewerin · 9 months ago
  79. ed18014 Remove more (D)TLS1.0 legacy code by Philipp Hancke · 9 months ago
  80. f79120a Update iOS perf dimensions to 16.7.5. by Christoffer Dewerin · 9 months ago
  81. b0a1d8b Support WebRTC-DataChannelMessageInterleaving by Victor Boivie · 9 months ago
  82. 7ee37cf Deprecate WebRTC-Audio-GainController2 fieldtrial by Hanna Silen · 9 months ago
  83. b2c4f54 Remove cores dimensions for perf bots by Christoffer Dewerin · 9 months ago
  84. 6e37ee3 Reuse QP limits from the main encoder config by Sergey Silkin · 9 months ago
  85. ff2bf4b Update FrameCombiner to use audio view methods for interleaved buffers by Tommi · 9 months ago
  86. 6dfb8c1 Update WebRTC code version (2024-06-12T04:05:13). by webrtc-version-updater · 9 months ago
  87. c2c5817 Roll chromium_revision f929cc54e6..05621b945d (1313332:1313445) by chromium-webrtc-autoroll · 9 months ago
  88. 633a41f PipeWire camera: check for node existence before adding it to the list by Jan Grulich · 9 months ago
  89. 3f91288 Roll chromium_revision afe6645537..f929cc54e6 (1313185:1313332) by chromium-webrtc-autoroll · 9 months ago
  90. 03ebfdf Create Environment for VoipCore by Danil Chapovalov · 9 months ago
  91. 6f3103f Add AGC2 input volume controller mode in audioproc_f by Hanna Silen · 9 months ago
  92. 546d15a Roll chromium_revision ab2dcf34af..afe6645537 (1304907:1313185) by chromium-webrtc-autoroll · 9 months ago
  93. 403220e add arm64 to perf dimensions to override x86 by Christoffer Dewerin · 9 months ago
  94. 41b934f Fix GoogCcNetworkController::OnNetworkStateEstimate behaviour by Per K · 9 months ago
  95. 63c3809 Use new M2 macmini and iPhone15 for iOS perf test by Christoffer Dewerin · 9 months ago
  96. c24ccd8 Remove instrumented_libraries_release=focal for msan. by Mirko Bonadei · 9 months ago
  97. 021160d Update WebRTC code version (2024-06-09T04:04:04). by webrtc-version-updater · 9 months ago
  98. 33e6e80 Actually skip AudioDecoderG722StereoTest.EncodeDecode on UBSan. by Mirko Bonadei · 9 months ago
  99. 21bfa5f Add RTC_EXPORT to API structs needed for RTCRtpTransport JS API by Tony Herre · 9 months ago
  100. 11e366d Skip tests failing with the new version of UBSan. by Mirko Bonadei · 9 months ago