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gerrit-public.fairphone.software / platform / external / webrtc / 3ee13e4ac219338c861a5d08212ffba88001cdb1 / samples / js
  1. 5ed7051 Apprtc: not to start the call until we get Turn response. by braveyao@webrtc.org · 11 years ago
  2. fddf6be Updated apprtc to use new TURN format for chrome versions M28 & above. by vikasmarwaha@webrtc.org · 11 years ago
  3. 5f8f112 Not to request to TURN server for local tests. Follow-up work to issue1197. by braveyao@webrtc.org · 11 years ago
  4. 5e2a1bb AppRTC: make requestTurn() failure non-fatal to call establishment. by fischman@webrtc.org · 11 years ago
  5. 59a0667 Updated apprtc demo to interop with firefox. by vikasmarwaha@webrtc.org · 11 years ago
  6. 40298d4 Added webaudio-and-webtrc.html to the demos index.html. by vikasmarwaha@webrtc.org · 11 years ago
  7. 1993a55 Added Stereo url paramter to apprtc demo. by vikasmarwaha@webrtc.org · 11 years ago
  8. 7a5615b New WebAudio-WebRTC demo. by henrika@webrtc.org · 11 years ago
  9. 77ac848 Added new demo states.html & updated existing demos to work on firefox. by vikasmarwaha@webrtc.org · 11 years ago
  10. a39a8fe Add owner to Apprtc by braveyao@webrtc.org · 12 years ago
  11. ceaedc0 Remove executable bit from dc1.html. by andrew@webrtc.org · 12 years ago
  12. f1bf3a0 A device switcher code example, with fake. by hta@webrtc.org · 12 years ago
  13. 4c44fe0 Updated pranswer, dtmf demos & deleted pc1-deprecated.html. by vikasmarwaha@webrtc.org · 12 years ago
  14. b4a0623 Fix of lint script errors in apprtc.py by pbos@webrtc.org · 12 years ago
  15. 37bf584 Show stats from both sides by hta@webrtc.org · 12 years ago
  16. 222e994 Migrating Apprtc to use new TURN service which supports time-limited TURN credentials. by vikasmarwaha@webrtc.org · 12 years ago
  17. 3ed599a Bandwidth stats display in constraints-and-stats. by hta@webrtc.org · 12 years ago
  18. f354e1f Add audio/video only option in apprtc by braveyao@webrtc.org · 12 years ago
  19. ebf49da Url option to change the resolution. by vikasmarwaha@webrtc.org · 12 years ago
  20. ecfd328 Changed stats reporting to not use local/remote by hta@webrtc.org · 12 years ago
  21. eddc5a6 Updated local-audio-rendering.html to remove unmute. by vikasmarwaha@webrtc.org · 12 years ago
  22. da0f708 Update demos to have local audio control muted by default. by vikasmarwaha@webrtc.org · 12 years ago
  23. a33037e Added an android_channel.html reflector page to allow Android apps to use a by fischman@webrtc.org · 12 years ago
  24. 3137a21 Dtmf twinkle-twinkle. by wu@webrtc.org · 12 years ago
  25. 5d371393 Fixed a ton of Python lint errors, enabled python lint checking. by phoglund@webrtc.org · 12 years ago
  26. 488d4c9 Submit symlink in apprtc from Linux since it fails from Win by braveyao@webrtc.org · 12 years ago
  27. 07db4a6 Add symlink of adapter.js from apprtc to base by braveyao@webrtc.org · 12 years ago
  28. db3f427 Using adapter.js and getRemoteStreams by hta@webrtc.org · 12 years ago
  29. a856db2 Moved trace function to adapter.js and removed from pc1 & multiple.html. by vikasmarwaha@webrtc.org · 12 years ago
  30. 7881b57 Updated path of adapter.js for dtmf & pc1-audio demos. by vikasmarwaha@webrtc.org · 12 years ago
  31. 99f1346 Typo in index.html and updated svn propset for dtmf & pc1-audio demos. by vikasmarwaha@webrtc.org · 12 years ago
  32. b203540 Redirect webrtc-demos.appspot.com to svn site and added dtmf & pc1-audio demos. Also updated index page to include information about new demos. by vikasmarwaha@webrtc.org · 12 years ago
  33. 98fce15 Adding webrtc-sample demos under trunk/samples. by vikasmarwaha@webrtc.org · 12 years ago