Gitiles
Code Review Sign In
gerrit-public.fairphone.software / platform / external / webrtc / 2db1778d38064855f79ff76420de211e4c283540 / api / rtpparameters.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/rtpparameters.h]
  3. 0a2ed5f Removing webrtc/config.h by Mirko Bonadei · 7 years ago
  4. 84f6a3f Move optional.h to webrtc/api/ by kwiberg · 7 years ago
  5. 1acbd68 Move RtpExtension to api/ directory and config.h/.cc to call/. by Stefan Holmer · 7 years ago
  6. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  7. d3501ad Create the SrtpTransportInterface. by zhihuang · 7 years ago
  8. e814a0d Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc. by deadbeef · 7 years ago
  9. e702b30 Adding C++ versions of currently spec'd "RtpParameters" structs. by deadbeef · 7 years ago
  10. 1fd9595 Pass VideoDecoderParams to VideoDecoderFactory and add SSRC to RtpEncodingParameters. by sakal · 8 years ago
  11. db0cd9e Adding getParameters/setParameters APIs to RtpReceiver. by Taylor Brandstetter · 8 years ago
  12. 0cd086b Adding codecs to the RtpParameters returned by an RtpSender. by Taylor Brandstetter · 8 years ago
  13. dbe2b87 Adding support for RTCRtpEncodingParameters.active flag. by deadbeef · 8 years ago
  14. dc1c62c Enable setting the maximum bitrate limit in RtpSender. by skvlad · 8 years ago