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gerrit-public.fairphone.software / platform / external / webrtc / c92781737ccb1c9d205d872b77688e3bbf732b8e / samples / js
  1. 83ffb0d Added functionality in apprtc demo to close the capture device on hangup. by vikasmarwaha@webrtc.org · 11 years ago
  2. 5a27e49 This CL will add support of passing all turn urls returned by the CEOD to PeerConnection object. by mallinath@webrtc.org · 11 years ago
  3. 6e7c203 Modified apprtc demo code to detect browser by checking user_agent in apprtc.py. Now we will use Mozilla stun server if FF is detected as the browser. The CL is an improvement to r4388. by vikasmarwaha@webrtc.org · 11 years ago
  4. 10bbfef Apprtc: add 'event' parameter to onkeydown event handler. by braveyao@webrtc.org · 11 years ago
  5. b63c29f Minor bug fix in r4388, had to change pc_config variable to pcConfig for apprtc demo. by vikasmarwaha@webrtc.org · 11 years ago
  6. 59fb7a6 Use Mozilla STUN server in apprtc demo for FF. Currently FF cannot work with Google STUN server as it expects XOR-MAPPED address while Google STUN server provides MAPPED address. by vikasmarwaha@webrtc.org · 11 years ago
  7. d4d9480 Added gum4.html, a multiple camera opening demo, each opening with a different resolution and/or frame rate. by mcasas@webrtc.org · 11 years ago
  8. bb25256 Added changes in apprtc demo to ignore turn address through query string for FF. Also made sure that the iceServers array doesnot include transport parameter in turn url for FF. Finally removed turn: from the turn_url when creating iceservers for pre-M28 chrome. by vikasmarwaha@webrtc.org · 11 years ago
  9. a193339 Apprtc CSS: Add flip to local view of FireFox and remove warning of Canary by braveyao@webrtc.org · 11 years ago
  10. fe6b571 AppRTCDemo: delete hosted android_channel.html now that it's no longer necessary. by fischman@webrtc.org · 11 years ago
  11. 5ed7051 Apprtc: not to start the call until we get Turn response. by braveyao@webrtc.org · 11 years ago
  12. fddf6be Updated apprtc to use new TURN format for chrome versions M28 & above. by vikasmarwaha@webrtc.org · 11 years ago
  13. 5f8f112 Not to request to TURN server for local tests. Follow-up work to issue1197. by braveyao@webrtc.org · 11 years ago
  14. 5e2a1bb AppRTC: make requestTurn() failure non-fatal to call establishment. by fischman@webrtc.org · 11 years ago
  15. 59a0667 Updated apprtc demo to interop with firefox. by vikasmarwaha@webrtc.org · 11 years ago
  16. 40298d4 Added webaudio-and-webtrc.html to the demos index.html. by vikasmarwaha@webrtc.org · 11 years ago
  17. 1993a55 Added Stereo url paramter to apprtc demo. by vikasmarwaha@webrtc.org · 11 years ago
  18. 7a5615b New WebAudio-WebRTC demo. by henrika@webrtc.org · 11 years ago
  19. 77ac848 Added new demo states.html & updated existing demos to work on firefox. by vikasmarwaha@webrtc.org · 11 years ago
  20. a39a8fe Add owner to Apprtc by braveyao@webrtc.org · 11 years ago
  21. ceaedc0 Remove executable bit from dc1.html. by andrew@webrtc.org · 11 years ago
  22. f1bf3a0 A device switcher code example, with fake. by hta@webrtc.org · 11 years ago
  23. 4c44fe0 Updated pranswer, dtmf demos & deleted pc1-deprecated.html. by vikasmarwaha@webrtc.org · 11 years ago
  24. b4a0623 Fix of lint script errors in apprtc.py by pbos@webrtc.org · 11 years ago
  25. 37bf584 Show stats from both sides by hta@webrtc.org · 11 years ago
  26. 222e994 Migrating Apprtc to use new TURN service which supports time-limited TURN credentials. by vikasmarwaha@webrtc.org · 11 years ago
  27. 3ed599a Bandwidth stats display in constraints-and-stats. by hta@webrtc.org · 12 years ago
  28. f354e1f Add audio/video only option in apprtc by braveyao@webrtc.org · 12 years ago
  29. ebf49da Url option to change the resolution. by vikasmarwaha@webrtc.org · 12 years ago
  30. ecfd328 Changed stats reporting to not use local/remote by hta@webrtc.org · 12 years ago
  31. eddc5a6 Updated local-audio-rendering.html to remove unmute. by vikasmarwaha@webrtc.org · 12 years ago
  32. da0f708 Update demos to have local audio control muted by default. by vikasmarwaha@webrtc.org · 12 years ago
  33. a33037e Added an android_channel.html reflector page to allow Android apps to use a by fischman@webrtc.org · 12 years ago
  34. 3137a21 Dtmf twinkle-twinkle. by wu@webrtc.org · 12 years ago
  35. 5d371393 Fixed a ton of Python lint errors, enabled python lint checking. by phoglund@webrtc.org · 12 years ago
  36. 488d4c9 Submit symlink in apprtc from Linux since it fails from Win by braveyao@webrtc.org · 12 years ago
  37. 07db4a6 Add symlink of adapter.js from apprtc to base by braveyao@webrtc.org · 12 years ago
  38. db3f427 Using adapter.js and getRemoteStreams by hta@webrtc.org · 12 years ago
  39. a856db2 Moved trace function to adapter.js and removed from pc1 & multiple.html. by vikasmarwaha@webrtc.org · 12 years ago
  40. 7881b57 Updated path of adapter.js for dtmf & pc1-audio demos. by vikasmarwaha@webrtc.org · 12 years ago
  41. 99f1346 Typo in index.html and updated svn propset for dtmf & pc1-audio demos. by vikasmarwaha@webrtc.org · 12 years ago
  42. 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
  43. 98fce15 Adding webrtc-sample demos under trunk/samples. by vikasmarwaha@webrtc.org · 12 years ago