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webrtc / src / 4522317ed73ca77f43cfab3f3c2022b94c2bc890
  1. 4522317 Make PeerConnection::StartSctpTransport return an RTCError by Philipp Hancke · 7 months ago
  2. a903885 Fix some shadow issues by Philipp Hancke · 7 months ago
  3. 723e219 Deliver render resolution by Sergey Silkin · 7 months ago
  4. 7420f62 Remove StunProber by Harald Alvestrand · 7 months ago
  5. ae6dc96 Update WebRTC code version (2025-03-18T04:03:58). by webrtc-version-updater · 7 months ago
  6. 07960c0 Fix minor formatting issue in webrtc::TestClient by Joachim Reiersen · 7 months ago
  7. d68dd92 Retain video stats when calling RecreateWebRtcStream. by Jeremy Leconte · 7 months ago
  8. 60a2021 Don't shadow DcSctpTransport options from argument by Philipp Hancke · 7 months ago
  9. e4b09d7 Move tests and fakes to webrtc namespace by Evan Shrubsole · 7 months ago
  10. 340d7e1 Enable Rust in GN args on debug/release bots. by Mirko Bonadei · 7 months ago
  11. 7729906 Write frame level stats in all codec tests by Sergey Silkin · 7 months ago
  12. 076d914 Document WebRTC Committers expiration date in case of no activity. by Mirko Bonadei · 7 months ago
  13. 98d4b59 dtls-in-stun: Move period dtls sender into DtlsTransport by Jonas Oreland · 7 months ago
  14. 2f3b8b0 Update Rust toolchains. by Mirko Bonadei · 7 months ago
  15. 746698b Update WebRTC code version (2025-03-17T04:06:37). by webrtc-version-updater · 7 months ago
  16. 0e231c1 Update WebRTC code version (2025-03-16T04:06:29). by webrtc-version-updater · 7 months ago
  17. 9634696 Update WebRTC code version (2025-03-15T04:02:31). by webrtc-version-updater · 7 months ago
  18. 99d1807 Move CorruptionDetectionFilterSettings into webrtc namespace. by Joachim Reiersen · 7 months ago
  19. b401d3d Add a callback to use StreamStats::rtp_stats as the source of truth. by Jeremy Leconte · 7 months ago
  20. c839686 Introduce SctpOptions struct as argument to SctpTransport::Start by Philipp Hancke · 7 months ago
  21. a22e244 Enable test that there are no duplicates in a CodecList by Harald Alvestrand · 7 months ago
  22. c2f1bc7 Split EmulatedNetworkInterface into two by Danil Chapovalov · 7 months ago
  23. c3cb9b7 Add .rustfmt.toml file by Mirko Bonadei · 7 months ago
  24. 64e4714 Add field trial that change g2g metric to use abs. capture time. by Olov Brändström · 7 months ago
  25. b922597 dtls-in-stun: Fix late SDP answer by Jonas Oreland · 7 months ago
  26. 024ceea Run DataChannelIntegrationTests w/o media enginge (for most tests) by Jonas Oreland · 7 months ago
  27. d5d9526 Update freshness for the h-cc-pairs style document by Danil Chapovalov · 7 months ago
  28. 42a1cde Update WebRTC code version (2025-03-14T04:09:46). by webrtc-version-updater · 7 months ago
  29. 41bc570 Enable 0Hz capture for test::FrameGeneratorCapturer. by Jeremy Leconte · 7 months ago
  30. c2c052c Refresh abseil-in-webrtc rules and documentation by Danil Chapovalov · 7 months ago
  31. c13956e Move all files in p2p/test to webrtc namespace by Evan Shrubsole · 7 months ago
  32. 0d3b384 Avoids log spam in AudioDeviceGeneric by henrika · 7 months ago
  33. b6909ee dtls-in-stun: Remove dtls restart in dtls-in-stun handling by Jonas Oreland · 7 months ago
  34. 2cf4492 Log VideoEncoderConfig::simulcast_layers[] by Sergey Silkin · 7 months ago
  35. 8990f2a Move rtc_certificate and rtc_certificate_generator to webrtc namespace by Evan Shrubsole · 7 months ago
  36. c713071 dtls_transport: Extend logging a tiny bit by Jonas Oreland · 7 months ago
  37. dd99114 Add RtcEventProcessor TieBreaker for LoggedRtcpPacketSenderReport. by Rasmus Brandt · 7 months ago
  38. d200661 Move sanitizer.h to webrtc namespace by Evan Shrubsole · 7 months ago
  39. 64b076f4 Move socket_address.h to webrtc namespace by Evan Shrubsole · 7 months ago
  40. e1912c5 Fix issue with protobuf that is blocking perf tests. by Mirko Bonadei · 7 months ago
  41. 63f9cc0 Apply include-cleaner to call/ by Danil Chapovalov · 7 months ago
  42. 1bd545a Revert "fix: h26x packet buffer video artifacts" by Philip Eliasson · 7 months ago
  43. e0cbfe4 Update WebRTC code version (2025-03-13T04:04:53). by webrtc-version-updater · 7 months ago
  44. 01f2ec6 Prevent VideoQualityMetricsReporter from crashing because of negative DataRate. by Jeremy Leconte · 7 months ago
  45. 7e54bb9 Remove unused constant by Rasmus Brandt · 7 months ago
  46. 762d77e Revert the deletion of WebRTC-VideoH26xPacketBuffer flag. by Henrik Boström · 7 months ago
  47. b2e88f9 Improve test outcomes for WebRTC-PayloadTypesInTransport by Harald Alvestrand · 7 months ago
  48. cc5db7a Remove deprecated non-Optional datachannel parameters by Harald Alvestrand · 7 months ago
  49. 31d20a6 Move moving_max_counter.h to webrtc namespace by Evan Shrubsole · 7 months ago
  50. 9278968 Delete deprecated functions in test EmulatedNetworkManagerInterface by Danil Chapovalov · 7 months ago
  51. 9ca15b6 Increase precision of high pass filter by Gustaf Ullberg · 7 months ago
  52. eb835d0 Move ssl_stream_adapter.h to webrtc namespace by Evan Shrubsole · 7 months ago
  53. 97feff2 Move function_view.h to webrtc namespace by Evan Shrubsole · 7 months ago
  54. b1d704b Move port_interface.h to webrtc namespace by Evan Shrubsole · 7 months ago
  55. 9f114ece Delete sigslot_tester by Evan Shrubsole · 7 months ago
  56. c9f82e6 Update WebRTC code version (2025-03-12T04:09:25). by webrtc-version-updater · 7 months ago
  57. 851bcba dtls-in-stun: Redo handling of when ice complete before dtls by Jonas Oreland · 7 months ago
  58. 6202bf1 Move ip_address.h to webrtc namespace by Evan Shrubsole · 7 months ago
  59. a0a83aa Allow test::EmulatedEndpoint to rebind to a different network by Danil Chapovalov · 7 months ago
  60. 0a76875 Skip assembling frames if the stream has been decoded past that point. by Philip Eliasson · 7 months ago
  61. a3866aa Refactoring: Look up codec vendor via RtpTransceiver by Harald Alvestrand · 7 months ago
  62. 8a9e08b Add QP threshold experiment for H265. by Henrik Boström · 7 months ago
  63. e1dc6d5 Deprecate sigslot_tester by Evan Shrubsole · 7 months ago
  64. b000e8e Move test_client.h to webrtc namespace by Evan Shrubsole · 7 months ago
  65. f13201d Temporarily disable two failing PCFullStackTests that hit DCHECKs. by Henrik Boström · 7 months ago
  66. 576ec5a PipeWire capture: Clean latest_available_frame on capture stop by Jan Grulich · 7 months ago
  67. 9e1b0ec Move stream.h to webrtc namespace by Evan Shrubsole · 7 months ago
  68. 377b69a Move test_certificate_verifier.h to webrtc namespace by Evan Shrubsole · 7 months ago
  69. 6738e57 Move base64.h to webrtc namespace by Evan Shrubsole · 7 months ago
  70. f36d45f Disable 'rtc_unittests' on iOS simulator. by Jeremy Leconte · 7 months ago
  71. d9232b3 Refactor VideoQualityMetricsReporter with no impact. by Jeremy Leconte · 7 months ago
  72. 47ddc6e Move test_echo_server.h to webrtc namespace by Evan Shrubsole · 7 months ago
  73. 4887bc3 Move socket_address_pair.h to webrtc namespace by Evan Shrubsole · 7 months ago
  74. 0a97878 Fix build with Pipewire 1.4 by K900 · 7 months ago
  75. 4203617 dtls-in-stun: Add new testcase by Jonas Oreland · 7 months ago
  76. 58c273b Move bit_buffer.h to webrtc namespace by Evan Shrubsole · 7 months ago
  77. c3fba45 Update WebRTC code version (2025-03-10T04:05:53). by webrtc-version-updater · 7 months ago
  78. 49ba3e2 Move byte_order.h to webrtc namespace by Evan Shrubsole · 7 months ago
  79. 5015b4d Move crc32.h to webrtc namespace by Evan Shrubsole · 7 months ago
  80. ccf5114 Update WebRTC code version (2025-03-09T04:08:42). by webrtc-version-updater · 7 months ago
  81. cec4b95 Update WebRTC code version (2025-03-08T04:06:00). by webrtc-version-updater · 7 months ago
  82. ee5e7e5 sdp munging: detect audio nack and changes to opus FEC and DTX by Philipp Hancke · 7 months ago
  83. b101a7e doc: SRTP_AES128_CM_HMAC_SHA1_32 is disabled by default by Philipp Hancke · 7 months ago
  84. f45f111 Sequence checker shouldn't guard itself by Artem Titov · 7 months ago
  85. 4df984d dtls-in-stun: Add callback for disabling of piggybacking by Jonas Oreland · 7 months ago
  86. 1816152 stats: do not expose ufrag before the first setRemoteDescription by Philipp Hancke · 7 months ago
  87. 2167733 Format fuzzer documentation by Björn Terelius · 7 months ago
  88. 3af24f3 Ensure one can build audioproc_f with apm_debug_dump=true by Lionel Koenig Gélas · 7 months ago
  89. a5da882 Update fuzzer documentation by Björn Terelius · 7 months ago
  90. e733ebc dtls1.3 - patch 6 by Jonas Oreland · 7 months ago
  91. 9789a44 Catch also IllegalStateException when creating hw encoder by Liad Rubin · 7 months ago
  92. 4a3dffe Add SurfaceEglRenderer ctor with VideoFrameDrawer and propagate to super class by Liad Rubin · 7 months ago
  93. 924b73c Checkout base/ only if fuzzing is enabled by Byoungchan Lee · 7 months ago
  94. fb8032e Move non-PT-assigning codec collection out of VoiceMediaEngine by Harald Alvestrand · 7 months ago
  95. 5568c98 Update WebRTC code version (2025-03-06T04:08:40). by webrtc-version-updater · 7 months ago
  96. 34bfdcd Roll chromium_revision d4f6168ad3..a1723b7af1 (1428571:1428676) by chromium-webrtc-autoroll · 7 months ago
  97. 0e8160a Roll chromium_revision 0b0f98e256..d4f6168ad3 (1428408:1428571) by chromium-webrtc-autoroll · 7 months ago
  98. 31f317e Roll chromium_revision a82e0ea13a..0b0f98e256 (1428241:1428408) by chromium-webrtc-autoroll · 7 months ago
  99. f1a3142 IWYU pc/test/ by Philipp Hancke · 7 months ago
  100. 72a1d7d Deprecate the cricket::Codec access functions in MediaEngine by Harald Alvestrand · 7 months ago