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gerrit-public.fairphone.software / platform / external / webrtc / 065348503c2ea5b0a6cb4f78144b13ca74777365 / media / base / media_constants.cc
  1. 479a3c0 Add support for enabling and negotiating raw RTP packetization. by Mirta Dvornicic · 5 years ago
  2. fadb181 Negotiate use of RTCP loss notification feedback (LNTF) by Elad Alon · 5 years ago
  3. 48cce4d Parse "max-message-size" parameter from SCTP SDP description by Harald Alvestrand · 5 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 media/base/mediaconstants.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 · 6 years ago
  8. 634a777 Add RRTR parameter to media engine and pass it to video receive stream by Ilya Nikolaevskiy · 6 years ago
  9. f18072e Enable SVC based on number of SSRCs. by Sergey Silkin · 7 years ago
  10. d7ae3c3 Reland "Rename stereo video codec to multiplex" by Emircan Uysaler · 7 years ago
  11. 1204448 Revert "Reland "Rename stereo video codec to multiplex"" by Taylor Brandstetter · 7 years ago
  12. 4954a77 Reland "Rename stereo video codec to multiplex" by Emircan Uysaler · 7 years ago
  13. 6bc7bb6 Revert "Rename stereo video codec to multiplex" by Ivo Creusen · 7 years ago
  14. bbdabe5 Rename stereo video codec to multiplex by Emircan Uysaler · 7 years ago
  15. 0a37547 Add optional stereo codec to SDP negotiation by Emircan Uysaler · 7 years ago
  16. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  17. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/media/base/mediaconstants.cc]
  18. 66753c3 Normalize codec names to those used by AcmCodecDatabase. by ossu · 7 years ago
  19. 5dfac56 Keep all codec parameters in VideoReceiveStream::Decoder by magjed · 8 years ago
  20. 509e4fe Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ ) by magjed · 8 years ago
  21. eacbaea Revert of Stop using hardcoded payload types for video codecs (patchset #6 id:210001 of https://codereview.webrtc.org/2493133002/ ) by magjed · 8 years ago
  22. 42043b9 Stop using hardcoded payload types for video codecs by Magnus Jedvert · 8 years ago
  23. 725e484 Use different RTX payload types for different H264 profiles by magjed · 8 years ago
  24. 87d7d77 Add new codec for FlexFEC. by brandtr · 8 years ago
  25. 06c8e1e Revert of H264 codec: Check profile-level-id when matching (patchset #2 id:60001 of https://codereview.webrtc.org/2347863003/ ) by Magnus Jedvert · 8 years ago
  26. 2675274 Remove cricket::VideoCodec with, height and framerate properties by perkj · 8 years ago
  27. 9fa4975 - Filter data channel codecs based on codec name instead of payload type, which may have been remapped. by solenberg · 8 years ago
  28. 68979ab H264 codec: Check profile-level-id when matching by magjed · 8 years ago
  29. 6f8d686 Remove use of RtpHeaderExtension and clean up by isheriff · 8 years ago
  30. b031a2e Allow WebRTC to offer receiving capability for 120ms Opus packets. by minyuel · 8 years ago
  31. a6b9944 Generate FMTP parameters for the H.264 codec. by hta · 8 years ago
  32. f475277 Rename constants files in webrtc/{media,p2p} by kjellander · 9 years ago[Renamed (98%) from webrtc/media/base/constants.cc]
  33. 5711c8d Change transport sequence number extension strings to specify what revision is implemented. by Stefan Holmer · 9 years ago
  34. 1e01660 Add support for rtx with h264. by stefan · 9 years ago
  35. 1afca73 Change to WebRTC license in webrtc/media by kjellander · 9 years ago
  36. a96e2d7 Move talk/media to webrtc/media by kjellander · 9 years ago[Renamed (98%) from talk/media/base/constants.cc]
  37. 1088001 Support multiple rtx codecs. by Stefan Holmer · 9 years ago
  38. b163c3f Delete unused members from VideoOptions by nisse · 9 years ago
  39. 43edf0f Require negotiation to send transport cc feedback over RTCP. by stefan · 9 years ago
  40. c1aeaf0 Wire up packet_id / send time callbacks to webrtc via libjingle. by stefan · 9 years ago
  41. 71f6f44 iOS HW H264 support. by Zeke Chin · 9 years ago
  42. e62202f Support handling multiple RTX but only generate SDP with RTX associated with VP8. by Shao Changbin · 9 years ago
  43. 7100dcd Adding "usedtx" as Opus codec parameter. by Minyue Li · 9 years ago
  44. 5225dd8 If audio ptime is negotiated in SDP, then we would set the audio codec with negotiated packet size if it's allowed. If the negotiated packet size is not supported by the working codec, then we would use the next smallest size. by Brave Yao · 9 years ago
  45. fdd1057 Add CVO support to Vie layer. by guoweis@webrtc.org · 10 years ago
  46. d324546 Misc. cleanup split out of https://webrtc-codereview.appspot.com/37699004/ : by pkasting@chromium.org · 10 years ago
  47. 5d639b3 (Auto)update libjingle 75141932-> 75179475 by buildbot@webrtc.org · 10 years ago
  48. b5a22b1 Revert r6110 and r6109. by pbos@webrtc.org · 10 years ago
  49. 17911dc (Auto)update libjingle 66798415-> 66813165 by buildbot@webrtc.org · 10 years ago
  50. d266a20 Initial wiring of new webrtc API in libjingle. by pbos@webrtc.org · 10 years ago
  51. ed97bb0 (Auto)update libjingle 66340694-> 66388864 by buildbot@webrtc.org · 10 years ago
  52. 79047f9 (Auto)update libjingle 62691533-> 62713454 by henrike@webrtc.org · 11 years ago
  53. 704bf9e (Auto)update libjingle 62063505-> 62278774 by henrike@webrtc.org · 11 years ago
  54. d43aa9d Update libjingle 61901702->61966318 by henrike@webrtc.org · 11 years ago
  55. aebb1ad pRevert 5371 "Revert 5367 "Update talk to 59410372."" by henrika@webrtc.org · 11 years ago
  56. 44461fa Revert 5367 "Update talk to 59410372." by henrika@webrtc.org · 11 years ago
  57. 0f3356e Update talk to 59410372. by mallinath@webrtc.org · 11 years ago
  58. 32f485b Fix constants.[h|cc] to avoid static initializer in webrtcvideoengine.cc. by sergeyu@chromium.org · 11 years ago
  59. 57a5f64 revert r5230 by sergeyu@chromium.org · 11 years ago
  60. a1b21cd Fix constants.[h|cc] to avoid static initializer in webrtcvideoengine.cc. by sergeyu@chromium.org · 11 years ago
  61. 7818752 Update libjingle to 53856368. by wu@webrtc.org · 11 years ago
  62. 1e09a71 Update talk folder to revision=49952949 by henrike@webrtc.org · 11 years ago
  63. 28e2075 Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk by henrike@webrtc.org · 11 years ago