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gerrit-public.fairphone.software / platform / external / webrtc / 3d255309e9775c3d5064a6e4ba55f8373ac01ff9 / media / base / rtputils.h
  1. 3d25530 Reland "Export symbols needed by the Chromium component build (part 1)." by Mirko Bonadei · 6 years ago
  2. 16fe3f2 Revert "Export symbols needed by the Chromium component build (part 1)." by Mirko Bonadei · 6 years ago
  3. 99eea42 Reland "Reland "Export symbols needed by the Chromium component build (part 1)."" by Mirko Bonadei · 6 years ago
  4. b49520b Revert "Reland "Export symbols needed by the Chromium component build (part 1)."" by Mirko Bonadei · 6 years ago
  5. 588f464 Reland "Export symbols needed by the Chromium component build (part 1)." by Mirko Bonadei · 6 years ago
  6. 2ea9af2 Revert "Export symbols needed by the Chromium component build (part 1)." by Mirko Bonadei · 6 years ago
  7. 9e24dcf Export symbols needed by the Chromium component build (part 1). by Mirko Bonadei · 6 years ago
  8. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  9. 365381f Replace BundleFilter with RtpDemuxer in RtpTransport. by Zhi Huang · 6 years ago
  10. 95e7dbb Revert "Reland "Replace BundleFilter with RtpDemuxer in RtpTransport."" by Zhi Huang · 6 years ago
  11. 27f3bf5 Reland "Replace BundleFilter with RtpDemuxer in RtpTransport." by Zhi Huang · 6 years ago
  12. 97d5e5b Revert "Replace BundleFilter with RtpDemuxer in RtpTransport." by Zhi Huang · 6 years ago
  13. ea8b62a Replace BundleFilter with RtpDemuxer in RtpTransport. by Zhi Huang · 6 years ago
  14. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  15. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/media/base/rtputils.h]
  16. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  17. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  18. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  19. 3dcf0e9 Move RTP/RTCP demuxing logic from BaseChannel to RtpTransport. by zstein · 7 years ago
  20. 1afca73 Change to WebRTC license in webrtc/media by kjellander · 9 years ago
  21. a96e2d7 Move talk/media to webrtc/media by kjellander · 9 years ago[Renamed (96%) from talk/media/base/rtputils.h]
  22. dc305db Add ApplyPacketOptions() by Sergey Ulanov · 9 years ago
  23. 0c4e06b Use suffixed {uint,int}{8,16,32,64}_t types. by Peter Boström · 9 years ago
  24. e9facf8 Add range checks in a variety of places where the values will subsequently be by pkasting@chromium.org · 10 years ago
  25. 0e81fdf Avoid implicit type truncations by inserting explicit casts or modifying prototypes to avoid needless up- and then down-casting. by pkasting@chromium.org · 10 years ago
  26. d4e598d (Auto)update libjingle 72097588-> 72159069 by buildbot@webrtc.org · 10 years ago
  27. 1ef789d (Auto)update libjingle 69568113-> 69587333 by buildbot@webrtc.org · 10 years ago
  28. e3cdd99 Revert "Fix the "Failed unprotect audio RTP packet" error when SCTP is bundled with audio." by jiayl@webrtc.org · 10 years ago
  29. 745a39c Fix the "Failed unprotect audio RTP packet" error when SCTP is bundled with audio. by jiayl@webrtc.org · 10 years ago
  30. 28e2075 Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk by henrike@webrtc.org · 11 years ago