Gitiles
Code Review Sign In
gerrit-public.fairphone.software / platform / external / webrtc / 22f9925b3eea255ea61edae6bd364249db2fc574 / media / base / media_engine.cc
  1. 22f9925 webrtc: Remove semicolons. by Nico Weber · 6 years ago
  2. 2297d33 Rejected simulcast layers will no longer appear in GetParameters(). by Amit Hilbuch · 6 years ago
  3. c1a0bcb Implement the encoding RtpParameter scaleResolutionDownBy by Florent Castelli · 6 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/mediaengine.cc]
  6. e1301a8 Revert "Implement read-only codecPayloadType in RtpParameters" by Henrik Grunell · 6 years ago
  7. 806e06d Implement read-only codecPayloadType in RtpParameters by Florent Castelli · 6 years ago
  8. 3e70781 [Cleanup] Add missing #include. Remove useless ones. IWYU part 2. by Yves Gerey · 6 years ago
  9. 6eb8a16 Exposing audio and video engines directly. by Sebastian Jansson · 6 years ago
  10. fa0aa39 Removes templating from CompositeMediaEngine. by Sebastian Jansson · 6 years ago
  11. 23eba22 Add support for RtpEncodingParameters num_temporal_layers. by Åsa Persson · 6 years ago
  12. 892acf0 Add support for send_encodings parameters in addTransceiver by Florent Castelli · 6 years ago
  13. dacec71 Add Rtcp parameters for PeerConnection senders by Florent Castelli · 6 years ago
  14. 11b34f4 Remove chromium clang style errors affecting sdk/android/media_jni by Paulina Hensman · 6 years ago
  15. 3ca452b Create an RtpEncodingParameters struct for each simulcast stream by Zach Stein · 7 years ago
  16. 8d0f1db Removing cricket::MediaEngineFactory. by Mirko Bonadei · 7 years ago
  17. aba85d1 Resolve circular dependency in rtc_media_base. by Patrik Höglund · 7 years ago
  18. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  19. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/media/base/mediaengine.cc]
  20. b252856 Remove all uses of the HAVE_CONFIG_H define. by Henrik Kjellander · 8 years ago
  21. dc1c62c Enable setting the maximum bitrate limit in RtpSender. by skvlad · 8 years ago
  22. 5ad1297 Rename webrtc/media/webrtc -> webrtc/media/engine by kjellander@webrtc.org · 9 years ago
  23. 1afca73 Change to WebRTC license in webrtc/media by kjellander · 9 years ago
  24. a96e2d7 Move talk/media to webrtc/media by kjellander · 9 years ago[Renamed (95%) from talk/media/base/mediaengine.cc]
  25. 81ea54e Remove WebRtcVideoEngine. by Peter Boström · 9 years ago
  26. 5f93d0a Update libjingle license statements at top of talk files for consistency by jlmiller@webrtc.org · 10 years ago
  27. 0481f15 (Auto)update libjingle 73399579-> 73626167 by henrike@webrtc.org · 10 years ago
  28. bb2d658 (Auto)update libjingle 69617317-> 69623266 by buildbot@webrtc.org · 10 years ago
  29. 58e7c86 (Auto)update libjingle 69588980-> 69589535 by buildbot@webrtc.org · 10 years ago
  30. 40b3b68 Update libjingle 62364298->62472237 by henrike@webrtc.org · 11 years ago
  31. 1bbfb57 Rollback of r5629 "(Auto)update libjingle 62364298-> 62368661". by henrike@webrtc.org · 11 years ago
  32. 31413dc (Auto)update libjingle 62364298-> 62368661 by henrike@webrtc.org · 11 years ago
  33. 28e2075 Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk by henrike@webrtc.org · 11 years ago