1. b62f41a Rename ChannelGroup to CongestionController and move to webrtc/call/. by mflodman · 9 years ago
  2. bf9f73c Split webrtc/video into webrtc/{audio,call,video}. by Peter Boström · 10 years ago
  3. 1f9001e Make sure rtp_rtcp module doesn't directly reference anything in the pacer module, and remove build dependencies on it. by sprang · 10 years ago
  4. bd3fce5 Includes webrtc/build/protoc.gypi instead of build/protoc.gypi by Bjorn Terelius · 10 years ago
  5. 8c456f5 Revert "Renamed the ACMDump to RtcEventLog and moved it to webrtc/video, since it is not specific to the audio coding module. Updated .gyp and .gn files accordingly." by Bjorn Terelius · 10 years ago
  6. 44be068 Renamed the ACMDump to RtcEventLog and moved it to webrtc/video, since it is not specific to the audio coding module. Updated .gyp and .gn files accordingly. by Bjorn Terelius · 10 years ago
  7. d5d98ba Move frame input (ViECapturer) to webrtc/video/. by Peter Boström · 10 years ago
  8. 5910eeb Merge video_engine_core into webrtc target. by Peter Boström · 10 years ago
  9. 3a03799 PRESUBMIT: Improve PyLint check and add GN format check. by Henrik Kjellander · 10 years ago
  10. 6859ada Add HW fallback option to software encoding. by Peter Boström · 10 years ago
  11. 5691fcf Add HW fallback option to software decoding. by Peter Boström · 10 years ago
  12. ad86786 Add AudioReceiveStream to Call API. by Fredrik Solenberg · 10 years ago
  13. c4e2cd0 Fix style violations in common_types.h and config.h by kwiberg@webrtc.org · 10 years ago
  14. 296871b GN: Add common configs to all targets. by kjellander@webrtc.org · 10 years ago
  15. f8698ce GN: Fix webrtc/video/BUILD.gn for Chromium build. by kjellander@webrtc.org · 11 years ago
  16. 0de7d38 GN: Implement video_engine, video_capture and video_render. by kjellander@webrtc.org · 11 years ago
  17. 3610f63 GN: Add BUILD.gn files + kjellander to OWNERS by kjellander@webrtc.org · 11 years ago