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webrtc / src / e86af2c75fe6ac8bfc0b1ee31b1bdf10da14fe52 / p2p / base / relay_server.cc
  1. ae226f6 Use Abseil container algorithms in p2p/ by Steve Anton · 6 years ago
  2. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  3. 1c05765 (3) Rename files to snake_case: move the files by Steve Anton · 6 years ago[Renamed from p2p/base/relayserver.cc]
  4. 4415315 Move ServerSocket code to separate files and into test target by Niels Möller · 6 years ago
  5. e693381 Delete struct rtc::PacketTime. by Niels Möller · 6 years ago
  6. aabf204 Remove container typedefs from RelayServer by Steve Anton · 6 years ago
  7. 665174f Reformat the WebRTC code base by Yves Gerey · 7 years ago
  8. a40f824 Explicitly specify is_clang=false for Win MSVC bots by Oleh Prypin · 7 years ago
  9. 6c38cc7 Fix cpplint errors in p2p/ by Steve Anton · 7 years ago
  10. 675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 7 years ago
  11. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  12. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/p2p/base/relayserver.cc]
  13. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  14. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  15. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  16. f42cc9d Add MakeUnique from chromium and change StunMessage::AddAttribute to take a unique_ptr. by zstein · 8 years ago
  17. ede5da4 Replace ASSERT by RTC_DCHECK in all non-test code. by nisse · 8 years ago
  18. c80e741 Replace ASSERT(false) by RTC_NOTREACHED(). by nisse · 8 years ago
  19. 5d97a9a Adding more detail to MessageQueue::Dispatch logging. by Taylor Brandstetter · 9 years ago
  20. 1bffc1d Rename rtc::Time64 --> rtc::TimeMillis. by nisse · 9 years ago
  21. f1f8720 Split ByteBuffer into writer/reader objects. by jbauch · 9 years ago
  22. 34b11eb Using 64-bit timestamp to replace the 32-bit one in webrtc/p2p. by honghaiz · 9 years ago
  23. 0c4e06b Use suffixed {uint,int}{8,16,32,64}_t types. by Peter Boström · 9 years ago
  24. ff689be Use std::min and std::max instead of self-defined functions such as rtc::_min/_max. by andresp@webrtc.org · 10 years ago
  25. 269fb4b move xmpp and p2p to webrtc by henrike@webrtc.org · 10 years ago
  26. 28100cb Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p." by henrike@webrtc.org · 10 years ago
  27. d1ba6d9 Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p. by henrike@webrtc.org · 10 years ago