1. f88487c Explicitly forbid dynamic_cast by Harald Alvestrand · 2 years, 5 months ago
  2. 666c333 Stop revalidating STUN packets with the wrong password by Harald Alvestrand · 2 years, 5 months ago
  3. 0e3cd63 stats: add missing ice candidate stats by Philipp Hancke · 2 years, 6 months ago
  4. 11174e7 Clarify how to reference WebRTC bugs in TODOs by Emil Lundmark · 2 years, 9 months ago
  5. 4b97928 Refactor RepeatingTaskHandle to use absl::AnyInvocable by Danil Chapovalov · 2 years, 9 months ago
  6. c4a3589 Add documentation for field trials by Emil Lundmark · 2 years, 9 months ago
  7. c1be89f Define Factory, Builder and Manager by Harald Alvestrand · 2 years, 9 months ago
  8. 4c29ca6 Remove mention of MSVC support from docs by Byoungchan Lee · 2 years, 11 months ago
  9. 832657f Recommend using [[deprecated]] over ABSL_DEPRECATED by Harald Alvestrand · 2 years, 11 months ago
  10. e610610 Fix fd leak in ifaddrs_android.cc by Danil Chapovalov · 3 years, 1 month ago
  11. 23bb9d7 Allow designated initializers in WebRTC by Danil Chapovalov · 3 years, 1 month ago
  12. 8ad0be0 Update link to chromium modern c++ page by Danil Chapovalov · 3 years, 2 months ago
  13. 80569ea Refresh WebRTC style guide by Danil Chapovalov · 3 years, 2 months ago
  14. 28cd164 Update style guide about C++17 by Mirko Bonadei · 3 years, 3 months ago
  15. 31b03e9 Add static AsString functions for PeerConnectionInterface enums by Harald Alvestrand · 3 years, 5 months ago
  16. 6ab8fc1 Add section about getting try job access by Artem Titov · 3 years, 5 months ago
  17. 9cd326c Update missed link to chromium C++11 styleguide by Joe Mason · 3 years, 6 months ago
  18. 4893dbe Update link to Chromium Modern C++ style guide by Joe Mason · 3 years, 6 months ago
  19. 09fb787 Use absl instead of self-made function for low-level bit counting by Danil Chapovalov · 3 years, 7 months ago
  20. f80ea7a Add a link to template to create a ticket to become WebRTC committer. by Artem Titov · 3 years, 8 months ago
  21. b0ed120 Update links to point at main branch by Tony Herre · 3 years, 8 months ago
  22. 41e98ba Add a documentation on how to become WebRTC committer by Artem Titov · 3 years, 8 months ago
  23. 0d2dc1f Reference "main" branches instead of "master" branches. by Fanny Linderborg · 3 years, 8 months ago
  24. 06b8f7e Move supported_platforms.md under g3doc/. by Mirko Bonadei · 3 years, 9 months ago
  25. 0fe60bd Add RecursiveCriticalSection to the don't-use list of primitives by Harald Alvestrand · 3 years, 9 months ago
  26. d25af8ce doc: document rtp payload type mapping behaviour by Philipp Hancke · 3 years, 10 months ago
  27. ab229b0 Add documentation for RTC event log by Björn Terelius · 3 years, 10 months ago
  28. 5cb983b Add basic synchronization function info to g3doc by Harald Alvestrand · 3 years, 10 months ago
  29. fec79b7 add srtp docs by Philipp Hancke · 3 years, 10 months ago
  30. 46f5c11 Move style guide and abseil-in-webrtc into g3doc subfolder by Danil Chapovalov · 3 years, 10 months ago
  31. ada0012 add dtls docs by Philipp Hancke · 3 years, 11 months ago
  32. e316b86 Write some doc about PeerConnection by Harald Alvestrand · 3 years, 11 months ago
  33. 39e2385 Add conceptual documentation for Audio - Mixer by Alessio Bazzica · 4 years ago
  34. f981cb3 Add video/g3doc/stats.md to the doc site menu by Artem Titov · 4 years ago
  35. b291da8 Add conceptual docs for modules/video_coding by Rasmus Brandt · 4 years ago
  36. 3ab7a55 Reformat pacer doc and add it into sitemap by Artem Titov · 4 years ago
  37. 5744b7f Fix formatting in sitemap.md by Artem Titov · 4 years ago
  38. 08d30a2 Add documentation for video/adaptation by Evan Shrubsole · 4 years ago
  39. dea5721 Adding g3doc for AudioProcessingModule (APM) by Per Åhgren · 4 years ago
  40. 8181b4f Add conceptual documentation for NetEq. by Jakob Ivarsson · 4 years ago
  41. 0498519 Add g3doc for audio coding module. by Minyue Li · 4 years ago
  42. 77d73a6 Document SctpTransport by Harald Alvestrand · 4 years ago
  43. 09c7f1e Add architecture section about PeerConnection test framework by Artem Titov · 4 years ago
  44. 3db3a06 Adding g3doc for AudioDeviceModule (ADM) - part of the AudioEngine by henrika · 4 years ago
  45. a168bb9 Add index.md documentation page for PC level test framework by Artem Titov · 4 years ago
  46. 0aa1a19 Add module overview of ICE by Jonas Oreland · 4 years ago
  47. 9de39f6 Add titovartem@webrtc.org as owner for /g3doc by Artem Titov · 4 years ago
  48. 4af6f2b Move threading documentation for API into g3doc structure by Harald Alvestrand · 4 years ago
  49. 967d4cd Improve webrtc documentation infra. Preview at: by Xavier Décoret · 4 years ago
  50. 8647340 Introduce WebRTC documentation structure and how-to by Artem Titov · 4 years ago
  51. 6512af0 Add root doc site definition for WebRTC documentation by Artem Titov · 4 years, 1 month ago