1. f93eda1 Move some video codec constants to separate file. by Erik Språng · 6 years ago
  2. 988cc08 [Cleanup] Add missing #include. Remove useless ones. by Yves Gerey · 6 years ago
  3. 276827c Export symbols needed by the Chromium component build (part 3). by Mirko Bonadei · 6 years ago
  4. 44b384d Delete support for VoIP metrics (RFC 3611 4.7) by Niels Möller · 6 years ago
  5. db12856 Cleanup modules_common_types by Danil Chapovalov · 6 years ago
  6. 01a8990 Remove deprecated type alias for RtpVideoCodecTypes. by Kári Tristan Helgason · 6 years ago
  7. 9a29c03 Fix random crashes - invariant broken in LinkedSet (LRU) implementation. by Yves Gerey · 6 years ago
  8. a12c42a Delete root header file typedef.h. by Niels Möller · 6 years ago
  9. 1a4746a Change RTPVideoTypeHeader to absl::variant and move RTPVideoHeader into its own h/cc file. by philipel · 6 years ago
  10. 5ab67a5 Add accessors to the types in the RTPVideoTypeHeader in RTPVideoHeader. by philipel · 6 years ago
  11. 011dc64 Remove RTPTypeHeader union and use RTPVideoHeader directly. by philipel · 6 years ago
  12. 8b23dba Add RTPVideoHeader const accessor. by philipel · 6 years ago
  13. 196100e Replace rtc::Optional with absl::optional by Danil Chapovalov · 6 years ago
  14. 7b55c73 Add RTPVideoHeader accessor. by philipel · 6 years ago
  15. 0a5fe77 Clean up in module_common_types.h by removing the unused struct RTPAudioHeader. by philipel · 6 years ago
  16. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  17. 8643b78 Moved NackModule and VCMPacket to their own targets by Ilya Nikolaevskiy · 6 years ago
  18. 520ca4e Delete enum RtpVideoCodecTypes, replaced with VideoCodecType. by Niels Möller · 6 years ago
  19. c6c4426 Moves network control interface to API. by Sebastian Jansson · 6 years ago
  20. ae8d8a1 Remove audio_frame.h from module_common_types.h by Fredrik Solenberg · 7 years ago
  21. 0dd1b0a Revert "Revert "Enables PeerConnectionFactory using external fec controller"" by Ying Wang · 7 years ago
  22. 0073301 Revert "Enables PeerConnectionFactory using external fec controller" by Taylor Brandstetter · 7 years ago
  23. 4f07bdb Enables PeerConnectionFactory using external fec controller by Ying Wang · 7 years ago
  24. d377f04 Move AudioFrame to its own header file and target in api/. by Niels Möller · 7 years ago
  25. 9bb8f05 Cleanup of unused RTP structs and packetizer for stereo codec by Emircan Uysaler · 7 years ago
  26. 3e11343 Fix circular dependencies in webrtc_common. by Patrik Höglund · 7 years ago
  27. 90612a6 Reland "Add stereo codec header and pass it through RTP" by Emircan Uysaler · 7 years ago
  28. deb8663 Revert "Add stereo codec header and pass it through RTP" by Philip Eliasson · 7 years ago
  29. 20f2133 Add stereo codec header and pass it through RTP by Emircan Uysaler · 7 years ago
  30. 248ccf8 Optional: Use nullopt and implicit construction in / by Oskar Sundbom · 7 years ago
  31. e40468b Move some numeric utility code from rtc_base/ to rtc_base/numerics/ by Karl Wiberg · 7 years ago
  32. feee08f Marked UnwrapWithoutUpdate function as const by Sebastian Jansson · 7 years ago
  33. 558cabf Refactor RtpToNtpEstimator and MovingMedianFilter by Ilya Nikolaevskiy · 7 years ago
  34. a32dd01 Reland "Remove AudioDeviceObserver and make ADM not inherit from the Module interface." by Fredrik Solenberg · 7 years ago
  35. d4404c2 Revert "Remove AudioDeviceObserver and make ADM not inherit from the Module interface." by Fredrik Solenberg · 7 years ago
  36. 34cdd2d Remove AudioDeviceObserver and make ADM not inherit from the Module interface. by Fredrik Solenberg · 7 years ago
  37. c7b4a45 Remove various IDs: by solenberg · 7 years ago
  38. e423a9de Revert of Remove various IDs (patchset #7 id:120001 of https://codereview.webrtc.org/3019543002/ ) by solenberg · 7 years ago
  39. 2d0f775 Remove various IDs: by solenberg · 7 years ago
  40. fb08994 Adding time profiling support to AudioFrame by henrika · 7 years ago
  41. 7120742 Adding NOLINT for typedefs.h and common_types.h by Mirko Bonadei · 7 years ago
  42. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  43. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago