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webrtc / src / / e3e030e54e96d7934dc70428b9fc7cf412c4c748 / media / base / media_constants.cc
  1. 44943c8 Add H265 codec name and profile/tier/level utils. by Qiu Jianlin · 1 year, 7 months ago
  2. 031ebc4 Increase RTP send buffer size from 64kb to 256kb. by Erik Språng · 2 years ago
  3. db1fae4 Reland "Remove ISAC media constant and payload type mapping" by Alessio Bazzica · 2 years, 1 month ago
  4. b79b74e Revert "Remove ISAC media constant and payload type mapping" by Björn Terelius · 2 years, 1 month ago
  5. 4c7271a Remove ISAC media constant and payload type mapping by Philipp Hancke · 2 years, 1 month ago
  6. 179f40e add 422 8 and 10 bit decoding support by Sergio Garcia Murillo · 2 years, 9 months ago
  7. 7194d83 Make AV1X constants private by Emil Lundmark · 3 years, 5 months ago
  8. 7145a14 red: fix fmtp payload type collision handling by Philipp Hancke · 3 years, 6 months ago
  9. 6b19d82 Replace AV1X with AV1 by Sergey Silkin · 3 years, 7 months ago
  10. 48171ec Remove more mentions of RTP datachannels by Harald Alvestrand · 4 years ago
  11. 006206d rtx-time implementation by Philipp Hancke · 4 years ago
  12. e71b55f build: merge media_constants and engine_constants by Philipp Hancke · 4 years, 2 months ago
  13. 1e98f95 sdp: remove some unused x-google attributes by Philipp Hancke · 4 years, 3 months ago
  14. afee708 do not set rtp datachannel b=AS for SCTP by Philipp Hancke · 4 years, 5 months ago
  15. 1b06876 Delete kHEVCCodecName by Emil Lundmark · 4 years, 7 months ago
  16. 2127aaa Add new fmtp parameter for H.264 by Eldar Rello · 4 years, 8 months ago
  17. ee8c246 Reland "sdp: parse and serialize b=TIAS" by Taylor Brandstetter · 4 years, 8 months ago
  18. e43648a Add constrained high profile level for h264 codec to media_constants by Andrey Logvin · 4 years, 9 months ago
  19. b59f337 Remove leftover SCTP "codec name" constants by Harald Alvestrand · 4 years, 10 months ago
  20. f026592 Add HEVC codec name. by Andrey Logvin · 4 years, 10 months ago
  21. c46385c Add Av1 Decoder wrapper behind a build flag by Danil Chapovalov · 5 years ago
  22. 479a3c0 Add support for enabling and negotiating raw RTP packetization. by Mirta Dvornicic · 6 years ago
  23. fadb181 Negotiate use of RTCP loss notification feedback (LNTF) by Elad Alon · 6 years ago
  24. 48cce4d Parse "max-message-size" parameter from SCTP SDP description by Harald Alvestrand · 6 years ago
  25. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  26. 1c05765 (3) Rename files to snake_case: move the files by Steve Anton · 6 years ago[Renamed from media/base/mediaconstants.cc]
  27. 3e70781 [Cleanup] Add missing #include. Remove useless ones. IWYU part 2. by Yves Gerey · 6 years ago
  28. 665174f Reformat the WebRTC code base by Yves Gerey · 7 years ago
  29. 634a777 Add RRTR parameter to media engine and pass it to video receive stream by Ilya Nikolaevskiy · 7 years ago
  30. f18072e Enable SVC based on number of SSRCs. by Sergey Silkin · 7 years ago
  31. d7ae3c3 Reland "Rename stereo video codec to multiplex" by Emircan Uysaler · 7 years ago
  32. 1204448 Revert "Reland "Rename stereo video codec to multiplex"" by Taylor Brandstetter · 7 years ago
  33. 4954a77 Reland "Rename stereo video codec to multiplex" by Emircan Uysaler · 7 years ago
  34. 6bc7bb6 Revert "Rename stereo video codec to multiplex" by Ivo Creusen · 7 years ago
  35. bbdabe5 Rename stereo video codec to multiplex by Emircan Uysaler · 7 years ago
  36. 0a37547 Add optional stereo codec to SDP negotiation by Emircan Uysaler · 7 years ago
  37. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 8 years ago
  38. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 8 years ago[Renamed from webrtc/media/base/mediaconstants.cc]
  39. 66753c3 Normalize codec names to those used by AcmCodecDatabase. by ossu · 8 years ago
  40. 5dfac56 Keep all codec parameters in VideoReceiveStream::Decoder by magjed · 8 years ago
  41. 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
  42. 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
  43. 42043b9 Stop using hardcoded payload types for video codecs by Magnus Jedvert · 8 years ago
  44. 725e484 Use different RTX payload types for different H264 profiles by magjed · 8 years ago
  45. 87d7d77 Add new codec for FlexFEC. by brandtr · 8 years ago
  46. 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
  47. 2675274 Remove cricket::VideoCodec with, height and framerate properties by perkj · 8 years ago
  48. 9fa4975 - Filter data channel codecs based on codec name instead of payload type, which may have been remapped. by solenberg · 8 years ago
  49. 68979ab H264 codec: Check profile-level-id when matching by magjed · 8 years ago
  50. 6f8d686 Remove use of RtpHeaderExtension and clean up by isheriff · 9 years ago
  51. b031a2e Allow WebRTC to offer receiving capability for 120ms Opus packets. by minyuel · 9 years ago
  52. a6b9944 Generate FMTP parameters for the H.264 codec. by hta · 9 years ago
  53. f475277 Rename constants files in webrtc/{media,p2p} by kjellander · 9 years ago[Renamed (98%) from webrtc/media/base/constants.cc]
  54. 5711c8d Change transport sequence number extension strings to specify what revision is implemented. by Stefan Holmer · 9 years ago
  55. 1e01660 Add support for rtx with h264. by stefan · 9 years ago
  56. 1afca73 Change to WebRTC license in webrtc/media by kjellander · 9 years ago
  57. a96e2d7 Move talk/media to webrtc/media by kjellander · 9 years ago[Renamed (98%) from talk/media/base/constants.cc]
  58. 1088001 Support multiple rtx codecs. by Stefan Holmer · 9 years ago
  59. b163c3f Delete unused members from VideoOptions by nisse · 9 years ago
  60. 43edf0f Require negotiation to send transport cc feedback over RTCP. by stefan · 9 years ago
  61. c1aeaf0 Wire up packet_id / send time callbacks to webrtc via libjingle. by stefan · 9 years ago
  62. 71f6f44 iOS HW H264 support. by Zeke Chin · 10 years ago
  63. e62202f Support handling multiple RTX but only generate SDP with RTX associated with VP8. by Shao Changbin · 10 years ago
  64. 7100dcd Adding "usedtx" as Opus codec parameter. by Minyue Li · 10 years ago
  65. 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 · 10 years ago
  66. fdd1057 Add CVO support to Vie layer. by guoweis@webrtc.org · 10 years ago
  67. d324546 Misc. cleanup split out of https://webrtc-codereview.appspot.com/37699004/ : by pkasting@chromium.org · 10 years ago
  68. 5d639b3 (Auto)update libjingle 75141932-> 75179475 by buildbot@webrtc.org · 11 years ago
  69. b5a22b1 Revert r6110 and r6109. by pbos@webrtc.org · 11 years ago
  70. 17911dc (Auto)update libjingle 66798415-> 66813165 by buildbot@webrtc.org · 11 years ago
  71. d266a20 Initial wiring of new webrtc API in libjingle. by pbos@webrtc.org · 11 years ago
  72. ed97bb0 (Auto)update libjingle 66340694-> 66388864 by buildbot@webrtc.org · 11 years ago
  73. 79047f9 (Auto)update libjingle 62691533-> 62713454 by henrike@webrtc.org · 11 years ago
  74. 704bf9e (Auto)update libjingle 62063505-> 62278774 by henrike@webrtc.org · 11 years ago
  75. d43aa9d Update libjingle 61901702->61966318 by henrike@webrtc.org · 11 years ago
  76. aebb1ad pRevert 5371 "Revert 5367 "Update talk to 59410372."" by henrika@webrtc.org · 11 years ago
  77. 44461fa Revert 5367 "Update talk to 59410372." by henrika@webrtc.org · 11 years ago
  78. 0f3356e Update talk to 59410372. by mallinath@webrtc.org · 11 years ago
  79. 32f485b Fix constants.[h|cc] to avoid static initializer in webrtcvideoengine.cc. by sergeyu@chromium.org · 11 years ago
  80. 57a5f64 revert r5230 by sergeyu@chromium.org · 11 years ago
  81. a1b21cd Fix constants.[h|cc] to avoid static initializer in webrtcvideoengine.cc. by sergeyu@chromium.org · 11 years ago
  82. 7818752 Update libjingle to 53856368. by wu@webrtc.org · 11 years ago
  83. 1e09a71 Update talk folder to revision=49952949 by henrike@webrtc.org · 12 years ago
  84. 28e2075 Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk by henrike@webrtc.org · 12 years ago