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webrtc / src / 0b10f4605e25c1b4b1ae17d3e57c7753afd6c6f0 / pc / sctp_utils_unittest.cc
  1. fd5ae7f Pass datachannel priority in DC open messages by Harald Alvestrand · 4 years, 11 months ago
  2. 6a489f2 Fully qualify googletest symbols. by Mirko Bonadei · 6 years ago
  3. f3736ed Datachannel: Use absl::optional for maxRetransmits and maxRetransmitTime. by Harald Alvestrand · 6 years ago
  4. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  5. 1c05765 (3) Rename files to snake_case: move the files by Steve Anton · 6 years ago[Renamed from pc/sctputils_unittest.cc]
  6. 3e70781 [Cleanup] Add missing #include. Remove useless ones. IWYU part 2. by Yves Gerey · 6 years ago
  7. 665174f Reformat the WebRTC code base by Yves Gerey · 7 years ago
  8. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 8 years ago
  9. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 8 years ago[Renamed from webrtc/pc/sctputils_unittest.cc]
  10. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  11. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  12. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  13. 7bb87ee Create //webrtc/api:libjingle_peerconnection_api + refactorings. by ossu · 8 years ago[Renamed (99%) from webrtc/api/sctputils_unittest.cc]
  14. d4d2f60 Reinstated sctputils_unittest.cc by ossu · 8 years ago
  15. b24317b Fix license headers in webrtc/api. by kjellander · 9 years ago
  16. 15583c1 Move talk/app/webrtc to webrtc/api by Henrik Kjellander · 9 years ago[Renamed (98%) from talk/app/webrtc/sctputils_unittest.cc]
  17. ab9b2d1 Reland of Moving MediaStreamSignaling logic into PeerConnection. (patchset #1 id:1 of https://codereview.webrtc.org/1403633005/ ) by deadbeef · 9 years ago
  18. fc648b6 Revert of Moving MediaStreamSignaling logic into PeerConnection. (patchset #10 id:180001 of https://codereview.webrtc.org/1393563002/ ) by deadbeef · 9 years ago
  19. 97c3929 Moving MediaStreamSignaling logic into PeerConnection. by deadbeef · 9 years ago
  20. 0c4e06b Use suffixed {uint,int}{8,16,32,64}_t types. by Peter Boström · 9 years ago
  21. a09a999 (Auto)update libjingle 73222930-> 73226398 by buildbot@webrtc.org · 11 years ago
  22. d4e598d (Auto)update libjingle 72097588-> 72159069 by buildbot@webrtc.org · 11 years ago
  23. aebb1ad pRevert 5371 "Revert 5367 "Update talk to 59410372."" by henrika@webrtc.org · 11 years ago[Renamed (80%) from talk/media/sctp/sctputils_unittest.cc]
  24. 44461fa Revert 5367 "Update talk to 59410372." by henrika@webrtc.org · 11 years ago[Renamed (80%) from talk/app/webrtc/sctputils_unittest.cc]
  25. 0f3356e Update talk to 59410372. by mallinath@webrtc.org · 11 years ago[Renamed (80%) from talk/media/sctp/sctputils_unittest.cc]
  26. 97077a3 Update libjingle to 55618622. Update libyuv to r826. by wu@webrtc.org · 11 years ago
  27. 1d1ffc9 Update talk to 54898858. by wu@webrtc.org · 11 years ago