Gitiles
Code Review Sign In
gerrit-public.fairphone.software / platform / external / webrtc / 2db1778d38064855f79ff76420de211e4c283540 / api / rtpsenderinterface.h
  1. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  2. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/api/rtpsenderinterface.h]
  3. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  4. b10f32f Adding more comments to every header file in api/ subdirectory. by deadbeef · 7 years ago
  5. 20cb0c1 Move DTMF sender to RtpSender (as opposed to WebRtcSession). by deadbeef · 7 years ago
  6. 7bb87ee Create //webrtc/api:libjingle_peerconnection_api + refactorings. by ossu · 8 years ago
  7. d99a200 Adding some features to proxy.h, and restructuring the macros. by deadbeef · 8 years ago
  8. a601f5c Separating internal and external methods of RtpSender/RtpReceiver. by deadbeef · 8 years ago
  9. 72c8d2b Rename BEGIN_PROXY_MAP --> BEGIN_SIGNALLING_PROXY_MAP. by nisse · 8 years ago
  10. dc1c62c Enable setting the maximum bitrate limit in RtpSender. by skvlad · 8 years ago
  11. 9b8df25 Move talk/session/media -> webrtc/pc by kjellander@webrtc.org · 8 years ago
  12. b24317b Fix license headers in webrtc/api. by kjellander · 8 years ago
  13. 15583c1 Move talk/app/webrtc to webrtc/api by Henrik Kjellander · 8 years ago[Renamed (93%) from talk/app/webrtc/rtpsenderinterface.h]
  14. a96e2d7 Move talk/media to webrtc/media by kjellander · 8 years ago
  15. fac0655 Reland of Adding the ability to create an RtpSender without a track. by deadbeef · 9 years ago
  16. 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
  17. 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
  18. 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
  19. ac9d92c Adding the ability to create an RtpSender without a track. by deadbeef · 9 years ago
  20. 70ab1a1 Exposing RtpSenders and RtpReceivers from PeerConnection. by deadbeef · 9 years ago
  21. 6979b02 Adding stub files for RtpSender/RtpReceiver. by deadbeef · 9 years ago