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  1. 87bccaf Roll chromium_revision 79a14840b3..cef126f01d (1435904:1436082) by chromium-webrtc-autoroll · 6 months ago
  2. b906bbb Manual roll libvpx to 027bbee30a0103b99d86327b48d29567fed11688 by Ilya Nikolaevskiy · 6 months ago
  3. d84ab1d IWYU modules/congestion_controller by Philipp Hancke · 6 months ago
  4. 652c771 Move time_utils.h and crypto_random.h to webrtc namespace by Evan Shrubsole · 6 months ago
  5. 2fb82f7 Manually update android build tools to 36 by Christoffer Dewerin · 6 months ago
  6. 61090f1 Fix more shadow issues by Philipp Hancke · 6 months ago
  7. 080cdac Move forward includes in ice_transport_interface to webrtc namespace by Evan Shrubsole · 6 months ago
  8. 4979786 IWYU: fix missing abseil nullability dependency by Philipp Hancke · 6 months ago
  9. be1759b Roll chromium_revision a1723b7af1..60851a9f6e (1428676:1435379) by Jeremy Leconte · 6 months ago
  10. 7af24a9 Manual roll libaom to 616ba8c822fdefb7ede666962a98c1f914ddf615 by Ilya Nikolaevskiy · 6 months ago
  11. 699fe6f sdp munging: detect renomination being added to ice-options by Philipp Hancke · 6 months ago
  12. 08757d4 IWYU media/ by Philipp Hancke · 6 months ago
  13. 39c1c7e Move event.h to webrtc namespace by Evan Shrubsole · 7 months ago
  14. 4d67183 Remove forward declarations in environment.h by Danil Chapovalov · 7 months ago
  15. b0547a9 dtls-in-stun: Defer writable until Ice has become writable by Jonas Oreland · 7 months ago
  16. 2018357 Migrate event_tracer.h to webrtc namespace by Evan Shrubsole · 7 months ago
  17. d889e20 Migrate logging.h to webrtc namespace by Evan Shrubsole · 7 months ago
  18. e3fa737 Update WebRTC code version (2025-03-20T04:07:44). by webrtc-version-updater · 7 months ago
  19. bb7cd9e Add unittest for ScreenCapturerSck by Andreas Pehrson · 7 months ago
  20. 75abbe9 Revert "Remove cricket::MediaType as a separate enum definition" by Fredrik Solenberg · 7 months ago
  21. 05eb774 Fix shadowing issues in some unit tests by Philipp Hancke · 7 months ago
  22. 886c99a Remove cricket::MediaType as a separate enum definition by Harald Alvestrand · 7 months ago
  23. a337cab Revert "Checkout base/ only if fuzzing is enabled" by Jeremy Leconte · 7 months ago
  24. 5371588 PipeWire camera: drop redundant <pipewire/core.h> include by Jan Grulich · 7 months ago
  25. f8db664 Move codec mappings from RtpTransceiver to CodecLookupHelper by Harald Alvestrand · 7 months ago
  26. b745acf Revert "Work around an issue with clang-include-cleaner." by Jeremy Leconte · 7 months ago
  27. 03b6880 Move socket related files to webrtc namespace by Evan Shrubsole · 7 months ago
  28. dfe1d42 Deleting some parts of BUILD.gn that might not be needed by Harald Alvestrand · 7 months ago
  29. 839b657 Wayland screencast: drop dependency on libepoxy by Jan Grulich · 7 months ago
  30. 0ac0ba0 fec: Fast insertion of received fec/media packets by Andrei Volykhin · 7 months ago
  31. 5688fb8 Update WebRTC code version (2025-03-19T04:08:03). by webrtc-version-updater · 7 months ago
  32. c8e422d Fix more shadowing issues by Philipp Hancke · 7 months ago
  33. eefd2ab sdp munging: detect RRTR by Philipp Hancke · 7 months ago
  34. ddd9c60 dtls-in-stun: Enable more tests by Jonas Oreland · 7 months ago
  35. de8429e Migrate absl variant to std in WebRTC by Danil Chapovalov · 7 months ago
  36. 970b9bc Enable adaptive convergence monitor for VP8 screenshare by Johannes Kron · 7 months ago
  37. 4522317 Make PeerConnection::StartSctpTransport return an RTCError by Philipp Hancke · 7 months ago
  38. a903885 Fix some shadow issues by Philipp Hancke · 7 months ago
  39. 723e219 Deliver render resolution by Sergey Silkin · 7 months ago
  40. 7420f62 Remove StunProber by Harald Alvestrand · 7 months ago
  41. ae6dc96 Update WebRTC code version (2025-03-18T04:03:58). by webrtc-version-updater · 7 months ago
  42. 07960c0 Fix minor formatting issue in webrtc::TestClient by Joachim Reiersen · 7 months ago
  43. d68dd92 Retain video stats when calling RecreateWebRtcStream. by Jeremy Leconte · 7 months ago
  44. 60a2021 Don't shadow DcSctpTransport options from argument by Philipp Hancke · 7 months ago
  45. e4b09d7 Move tests and fakes to webrtc namespace by Evan Shrubsole · 7 months ago
  46. 340d7e1 Enable Rust in GN args on debug/release bots. by Mirko Bonadei · 7 months ago
  47. 7729906 Write frame level stats in all codec tests by Sergey Silkin · 7 months ago
  48. 076d914 Document WebRTC Committers expiration date in case of no activity. by Mirko Bonadei · 7 months ago
  49. 98d4b59 dtls-in-stun: Move period dtls sender into DtlsTransport by Jonas Oreland · 7 months ago
  50. 2f3b8b0 Update Rust toolchains. by Mirko Bonadei · 7 months ago
  51. 746698b Update WebRTC code version (2025-03-17T04:06:37). by webrtc-version-updater · 7 months ago
  52. 0e231c1 Update WebRTC code version (2025-03-16T04:06:29). by webrtc-version-updater · 7 months ago
  53. 9634696 Update WebRTC code version (2025-03-15T04:02:31). by webrtc-version-updater · 7 months ago
  54. 99d1807 Move CorruptionDetectionFilterSettings into webrtc namespace. by Joachim Reiersen · 7 months ago
  55. b401d3d Add a callback to use StreamStats::rtp_stats as the source of truth. by Jeremy Leconte · 7 months ago
  56. c839686 Introduce SctpOptions struct as argument to SctpTransport::Start by Philipp Hancke · 7 months ago
  57. a22e244 Enable test that there are no duplicates in a CodecList by Harald Alvestrand · 7 months ago
  58. c2f1bc7 Split EmulatedNetworkInterface into two by Danil Chapovalov · 7 months ago
  59. c3cb9b7 Add .rustfmt.toml file by Mirko Bonadei · 7 months ago
  60. 64e4714 Add field trial that change g2g metric to use abs. capture time. by Olov Brändström · 7 months ago
  61. b922597 dtls-in-stun: Fix late SDP answer by Jonas Oreland · 7 months ago
  62. 024ceea Run DataChannelIntegrationTests w/o media enginge (for most tests) by Jonas Oreland · 7 months ago
  63. d5d9526 Update freshness for the h-cc-pairs style document by Danil Chapovalov · 7 months ago
  64. 42a1cde Update WebRTC code version (2025-03-14T04:09:46). by webrtc-version-updater · 7 months ago
  65. 41bc570 Enable 0Hz capture for test::FrameGeneratorCapturer. by Jeremy Leconte · 7 months ago
  66. c2c052c Refresh abseil-in-webrtc rules and documentation by Danil Chapovalov · 7 months ago
  67. c13956e Move all files in p2p/test to webrtc namespace by Evan Shrubsole · 7 months ago
  68. 0d3b384 Avoids log spam in AudioDeviceGeneric by henrika · 7 months ago
  69. b6909ee dtls-in-stun: Remove dtls restart in dtls-in-stun handling by Jonas Oreland · 7 months ago
  70. 2cf4492 Log VideoEncoderConfig::simulcast_layers[] by Sergey Silkin · 7 months ago
  71. 8990f2a Move rtc_certificate and rtc_certificate_generator to webrtc namespace by Evan Shrubsole · 7 months ago
  72. c713071 dtls_transport: Extend logging a tiny bit by Jonas Oreland · 7 months ago
  73. dd99114 Add RtcEventProcessor TieBreaker for LoggedRtcpPacketSenderReport. by Rasmus Brandt · 7 months ago
  74. d200661 Move sanitizer.h to webrtc namespace by Evan Shrubsole · 7 months ago
  75. 64b076f4 Move socket_address.h to webrtc namespace by Evan Shrubsole · 7 months ago
  76. e1912c5 Fix issue with protobuf that is blocking perf tests. by Mirko Bonadei · 7 months ago
  77. 63f9cc0 Apply include-cleaner to call/ by Danil Chapovalov · 7 months ago
  78. 1bd545a Revert "fix: h26x packet buffer video artifacts" by Philip Eliasson · 7 months ago
  79. e0cbfe4 Update WebRTC code version (2025-03-13T04:04:53). by webrtc-version-updater · 7 months ago
  80. 01f2ec6 Prevent VideoQualityMetricsReporter from crashing because of negative DataRate. by Jeremy Leconte · 7 months ago
  81. 7e54bb9 Remove unused constant by Rasmus Brandt · 7 months ago
  82. 762d77e Revert the deletion of WebRTC-VideoH26xPacketBuffer flag. by Henrik Boström · 7 months ago
  83. b2e88f9 Improve test outcomes for WebRTC-PayloadTypesInTransport by Harald Alvestrand · 7 months ago
  84. cc5db7a Remove deprecated non-Optional datachannel parameters by Harald Alvestrand · 7 months ago
  85. 31d20a6 Move moving_max_counter.h to webrtc namespace by Evan Shrubsole · 7 months ago
  86. 9278968 Delete deprecated functions in test EmulatedNetworkManagerInterface by Danil Chapovalov · 7 months ago
  87. 9ca15b6 Increase precision of high pass filter by Gustaf Ullberg · 7 months ago
  88. eb835d0 Move ssl_stream_adapter.h to webrtc namespace by Evan Shrubsole · 7 months ago
  89. 97feff2 Move function_view.h to webrtc namespace by Evan Shrubsole · 7 months ago
  90. b1d704b Move port_interface.h to webrtc namespace by Evan Shrubsole · 7 months ago
  91. 9f114ece Delete sigslot_tester by Evan Shrubsole · 7 months ago
  92. c9f82e6 Update WebRTC code version (2025-03-12T04:09:25). by webrtc-version-updater · 7 months ago
  93. 851bcba dtls-in-stun: Redo handling of when ice complete before dtls by Jonas Oreland · 7 months ago
  94. 6202bf1 Move ip_address.h to webrtc namespace by Evan Shrubsole · 7 months ago
  95. a0a83aa Allow test::EmulatedEndpoint to rebind to a different network by Danil Chapovalov · 7 months ago
  96. 0a76875 Skip assembling frames if the stream has been decoded past that point. by Philip Eliasson · 7 months ago
  97. a3866aa Refactoring: Look up codec vendor via RtpTransceiver by Harald Alvestrand · 7 months ago
  98. 8a9e08b Add QP threshold experiment for H265. by Henrik Boström · 7 months ago
  99. e1dc6d5 Deprecate sigslot_tester by Evan Shrubsole · 7 months ago
  100. b000e8e Move test_client.h to webrtc namespace by Evan Shrubsole · 7 months ago