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gerrit-public.fairphone.software / platform / external / webrtc / dfaea9dd98728b16e7d67aec49543123e150ac8f / api / rtp_sender_interface.h
  1. e1e789b Removing non-const RtpSenderInterface::GetParameters(). by Amit Hilbuch · 6 years ago
  2. 22f9925 webrtc: Remove semicolons. by Nico Weber · 6 years ago
  3. 2297d33 Rejected simulcast layers will no longer appear in GetParameters(). by Amit Hilbuch · 6 years ago
  4. d970807 Remove rtc_base/scoped_ref_ptr.h. by Mirko Bonadei · 6 years ago
  5. 4a7b3ac Add DTLSTransport info into sender/receiver state. by Harald Alvestrand · 6 years ago
  6. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  7. 1c05765 (3) Rename files to snake_case: move the files by Steve Anton · 6 years ago[Renamed from api/rtpsenderinterface.h]
  8. 2e00abc Reland "[cleanup] Remove useless includes." by Yves Gerey · 6 years ago
  9. 96a0f61 Revert "[cleanup] Remove useless includes." by Oleh Prypin · 6 years ago
  10. be8b534 [cleanup] Remove useless includes. by Yves Gerey · 6 years ago
  11. 892acf0 Add support for send_encodings parameters in addTransceiver by Florent Castelli · 6 years ago
  12. d81ac95 Injects FrameEncryptorInterface into RtpSender. by Benjamin Wright · 6 years ago
  13. 79eb4dd Enabling clang::find_bad_constructs for libjingle_peerconnection_api. by Mirko Bonadei · 6 years ago
  14. 0bc58cf Replace rtc::Optional with absl::optional in api by Danil Chapovalov · 7 years ago
  15. 5565981 Add functionality to set min/max bitrate per simulcast layer through RtpEncodingParameters. by Åsa Persson · 7 years ago
  16. 665174f Reformat the WebRTC code base by Yves Gerey · 7 years ago
  17. 4c6390a Remove deprecated RtpSenderInterface::GetParameters() const method by Florent Castelli · 7 years ago
  18. cebf50f Reland "Implement RtpParameters.transaction_id for PC RtpSenderInterface" by Florent Castelli · 7 years ago
  19. 909338b Revert "Implement RtpParameters.transaction_id for PC RtpSenderInterface" by Max Morin · 7 years ago
  20. 5faf36e Implement RtpParameters.transaction_id for PC RtpSenderInterface by Florent Castelli · 7 years ago
  21. 5b4f075 Reland "Reland "Adds support for multiple or no media stream ids."" by Seth Hampson · 7 years ago
  22. 191bf5c Revert "Reland "Adds support for multiple or no media stream ids."" by Tomas Gunnarsson · 7 years ago
  23. f351c34 Reland "Adds support for multiple or no media stream ids." by Seth Hampson · 7 years ago
  24. bc609ea Revert "Adds support for multiple or no media stream ids." by Emircan Uysaler · 7 years ago
  25. 1550292 Adds support for multiple or no media stream ids. by Seth Hampson · 7 years ago
  26. 57858b3 Reland "Update RTCStatsCollector to work with RtpTransceivers" by Steve Anton · 7 years ago
  27. ee2388f Revert "Update RTCStatsCollector to work with RtpTransceivers" by Guido Urdaneta · 7 years ago
  28. 56bae8d Update RTCStatsCollector to work with RtpTransceivers by Steve Anton · 7 years ago
  29. ba37b4b Change return type of RtpSenderInterface::SetParameters from bool to RTCError by Zach Stein · 7 years ago
  30. c72af93 Reland "Move stats ID generation from SSRC to local ID" by Harald Alvestrand · 7 years ago
  31. c0092c3 Revert "Move stats ID generation from SSRC to local ID" by Erik Språng · 7 years ago
  32. e357a4d Move stats ID generation from SSRC to local ID by Harald Alvestrand · 7 years ago
  33. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  34. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/api/rtpsenderinterface.h]
  35. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  36. b10f32f Adding more comments to every header file in api/ subdirectory. by deadbeef · 8 years ago
  37. 20cb0c1 Move DTMF sender to RtpSender (as opposed to WebRtcSession). by deadbeef · 8 years ago
  38. 7bb87ee Create //webrtc/api:libjingle_peerconnection_api + refactorings. by ossu · 8 years ago
  39. d99a200 Adding some features to proxy.h, and restructuring the macros. by deadbeef · 8 years ago
  40. a601f5c Separating internal and external methods of RtpSender/RtpReceiver. by deadbeef · 9 years ago
  41. 72c8d2b Rename BEGIN_PROXY_MAP --> BEGIN_SIGNALLING_PROXY_MAP. by nisse · 9 years ago
  42. dc1c62c Enable setting the maximum bitrate limit in RtpSender. by skvlad · 9 years ago
  43. 9b8df25 Move talk/session/media -> webrtc/pc by kjellander@webrtc.org · 9 years ago
  44. b24317b Fix license headers in webrtc/api. by kjellander · 9 years ago
  45. 15583c1 Move talk/app/webrtc to webrtc/api by Henrik Kjellander · 9 years ago[Renamed (93%) from talk/app/webrtc/rtpsenderinterface.h]
  46. a96e2d7 Move talk/media to webrtc/media by kjellander · 9 years ago
  47. fac0655 Reland of Adding the ability to create an RtpSender without a track. by deadbeef · 9 years ago
  48. 5def7b9 Revert of Adding the ability to create an RtpSender without a track. (patchset #3 id:300001 of https://codereview.webrtc.org/1413983004/ ) by deadbeef · 9 years ago
  49. 6834fa1 Reland of Adding the ability to create an RtpSender without a track. (patchset #1 id:1 of https://codereview.webrtc.org/1426443007/ ) by deadbeef · 9 years ago
  50. 8f46c63 Revert of Adding the ability to create an RtpSender without a track. (patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ ) by deadbeef · 9 years ago
  51. ac9d92c Adding the ability to create an RtpSender without a track. by deadbeef · 9 years ago
  52. 70ab1a1 Exposing RtpSenders and RtpReceivers from PeerConnection. by deadbeef · 9 years ago
  53. 6979b02 Adding stub files for RtpSender/RtpReceiver. by deadbeef · 9 years ago