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gerrit-public.fairphone.software / platform / external / webrtc / 5ae3a028c862bd97a7b52c3d798f5f06a4f65214 / pc / remoteaudiosource.cc
  1. 918f50c Use absl::make_unique and absl::WrapUnique directly by Karl Wiberg · 6 years ago
  2. d367921 Configure media flow correctly with Unified Plan by Steve Anton · 7 years ago
  3. 6077675 Change RtpReceivers to interact with the media channel directly by Steve Anton · 7 years ago
  4. 675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 7 years ago
  5. 3b80aac Fix flaky memory leak in RemoteAudioSource by Steve Anton · 7 years ago
  6. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  7. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/pc/remoteaudiosource.cc]
  8. ee89e78 Replace CHECK(x && y) with two separate CHECK() calls by kwiberg · 7 years ago
  9. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  10. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  11. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  12. 7bb87ee Create //webrtc/api:libjingle_peerconnection_api + refactorings. by ossu · 8 years ago[Renamed (98%) from webrtc/api/remoteaudiosource.cc]
  13. ba29c6a Reland 2 of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 9 years ago
  14. 3784b4a Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (patchset #3 id:40001 of https://codereview.webrtc.org/2046173002/ ) by tkchin · 9 years ago
  15. 2d54917 Reland of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 9 years ago
  16. 1a7162d Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (patchset #3 id:40001 of https://codereview.webrtc.org/2046173002/ ) by deadbeef · 9 years ago
  17. bc58319 Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 9 years ago
  18. 5d97a9a Adding more detail to MessageQueue::Dispatch logging. by Taylor Brandstetter · 9 years ago
  19. d1fe281 Replace scoped_ptr with unique_ptr in webrtc/api/ by kwiberg · 9 years ago
  20. 4485ffb #include "webrtc/base/constructormagic.h" where appropriate by kwiberg · 9 years ago
  21. b24317b Fix license headers in webrtc/api. by kjellander · 9 years ago
  22. 15583c1 Move talk/app/webrtc to webrtc/api by Henrik Kjellander · 9 years ago[Renamed (97%) from talk/app/webrtc/remoteaudiosource.cc]
  23. 2d110be Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ ) by deadbeef · 9 years ago
  24. e591f93 Storing raw audio sink for default audio track. by deadbeef · 9 years ago
  25. 6eca7e3 Add a 'remote' property to MediaSourceInterface. Also adding an implementation to the relevant sources we have (audio/video) and an extra check where we're casting a source into a local audio source :( by tommi · 9 years ago
  26. cb95f54e Remove pointless move() to fix build on clang/win. by Tommi · 9 years ago
  27. f888bb5 Support for unmixed remote audio into tracks. by Tommi · 9 years ago
  28. 5f93d0a Update libjingle license statements at top of talk files for consistency by jlmiller@webrtc.org · 10 years ago
  29. d4e598d (Auto)update libjingle 72097588-> 72159069 by buildbot@webrtc.org · 10 years ago
  30. b9a088b Update talk to 61538839. by wu@webrtc.org · 11 years ago
  31. 0de2950 Revert 5545 "Update libjingle to 61514460" by wu@webrtc.org · 11 years ago
  32. e749c9e Update libjingle to 61514460 by xians@webrtc.org · 11 years ago