1. 0abc754 Change RtpSender to have multiple stream_ids by Steve Anton · 7 years ago
  2. ea15d68 Replace CHECK(x && y) with two separate CHECK() calls by kwiberg · 7 years ago
  3. 76de83e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  4. bc32410 Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  5. 60154fd Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  6. 633cb75 Protect APM in webkit builds. by agouaillard · 8 years ago
  7. d4996cd Move DTMF sender to RtpSender (as opposed to WebRtcSession). by deadbeef · 8 years ago
  8. 25e4ac7 Create //webrtc/api:libjingle_peerconnection_api + refactorings. by ossu · 8 years ago[Renamed (98%) from api/rtpsender.cc]
  9. a875ff8 Replace ASSERT by RTC_DCHECK in all non-test code. by nisse · 8 years ago
  10. 1709986 Replace RTC_DCHECK(false) with RTC_NOTREACHED(). by nisse · 8 years ago
  11. 37d7b3f Add support for content hints to VideoTrack. by pbos · 8 years ago
  12. bc1f6bd Reland 2 of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 9 years ago
  13. 00dbf61 Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (patchset #3 id:40001 of https://codereview.webrtc.org/2046173002/ ) by tkchin · 9 years ago
  14. 23492b9 Reland of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 9 years ago
  15. a7ba978 Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (patchset #3 id:40001 of https://codereview.webrtc.org/2046173002/ ) by deadbeef · 9 years ago
  16. f29b3f1 Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 9 years ago
  17. 0fec214 Separating internal and external methods of RtpSender/RtpReceiver. by deadbeef · 9 years ago
  18. 4e20ddd Combining SetVideoSend and SetSource into one method. by deadbeef · 9 years ago
  19. 80b957b Adding getParameters/setParameters APIs to RtpReceiver. by Taylor Brandstetter · 9 years ago
  20. f6eaeac Fixing a segfault that can occur when changing the track of an RtpSender. by deadbeef · 9 years ago
  21. e38f5d6 Add missing tracing to RtpSender objects. by Peter Boström · 9 years ago
  22. cc4f458 Replace SetCapturer and SetCaptureDevice by SetSource. Drop return value. by nisse · 9 years ago
  23. 3c4cb59 Change VideoSourceInterface::needs_denoising() to return rtc::Optional<bool> by Per · 9 years ago
  24. 5f44bf3 Reland of Delete empty API files and cleaned up includes. (patchset #1 id:1 of https://codereview.webrtc.org/1813083002/ ) by perkj · 9 years ago
  25. 8396577 Revert of Delete empty API files and cleaned up includes. (patchset #2 id:20001 of https://codereview.webrtc.org/1809053002/ ) by deadbeef · 9 years ago
  26. 2350721 Delete empty API files and cleaned up includes. by perkj · 9 years ago
  27. 4ac195e Enable setting the maximum bitrate limit in RtpSender. by skvlad · 9 years ago
  28. f4524c4 Add implementation of VideoTrackSource and make VideoCapturerTrackSource inherit from it. by perkj · 9 years ago
  29. e38b09a Prevent a voice channel from sending data before a source is set. by Taylor Brandstetter · 9 years ago
  30. 2a4ff98 Renamed VideoSourceInterface to VideoTrackSourceInterface. by perkj · 9 years ago
  31. 95ed4e6 Fix license headers in webrtc/api. by kjellander · 9 years ago
  32. bd0ae45 Move talk/app/webrtc to webrtc/api by Henrik Kjellander · 9 years ago