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gerrit-public.fairphone.software / platform / external / webrtc / fae640003cf028477819309777ecd29a74c25eff / pc / remote_audio_source.cc
  1. 749f660 Enable SSRC 0 in MediaChannel methods by Saurav Das · 4 years, 8 months ago
  2. 317a1f0 Use std::make_unique instead of absl::make_unique. by Mirko Bonadei · 4 years, 10 months ago
  3. a4d8737 Format almost everything. by Jonas Olsson · 5 years ago
  4. 428dcb2 Remove SetLatency/GetLatency from MediaSourceInterface API level by Ruslan Burakov · 5 years ago
  5. 493a650 Propagate base minimum delay from video jitter buffer to webrtc/api. by Ruslan Burakov · 5 years ago
  6. 7ea4605 Add latency to remote source api. by Ruslan Burakov · 5 years ago
  7. 3d02384 Fix inverted DCHECK conditional by Steve Anton · 5 years ago
  8. 64b626b Use Abseil container algorithms in pc/ by Steve Anton · 5 years ago
  9. d970807 Remove rtc_base/scoped_ref_ptr.h. by Mirko Bonadei · 6 years ago
  10. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  11. 1c05765 (3) Rename files to snake_case: move the files by Steve Anton · 6 years ago[Renamed from pc/remoteaudiosource.cc]
  12. 3e70781 [Cleanup] Add missing #include. Remove useless ones. IWYU part 2. by Yves Gerey · 6 years ago
  13. 918f50c Use absl::make_unique and absl::WrapUnique directly by Karl Wiberg · 6 years ago
  14. d367921 Configure media flow correctly with Unified Plan by Steve Anton · 7 years ago
  15. 6077675 Change RtpReceivers to interact with the media channel directly by Steve Anton · 7 years ago
  16. 675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 7 years ago
  17. 3b80aac Fix flaky memory leak in RemoteAudioSource by Steve Anton · 7 years ago
  18. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  19. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/pc/remoteaudiosource.cc]
  20. ee89e78 Replace CHECK(x && y) with two separate CHECK() calls by kwiberg · 7 years ago
  21. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  22. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  23. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  24. 7bb87ee Create //webrtc/api:libjingle_peerconnection_api + refactorings. by ossu · 8 years ago[Renamed (98%) from webrtc/api/remoteaudiosource.cc]
  25. ba29c6a Reland 2 of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 8 years ago
  26. 3784b4a Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (patchset #3 id:40001 of https://codereview.webrtc.org/2046173002/ ) by tkchin · 8 years ago
  27. 2d54917 Reland of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 8 years ago
  28. 1a7162d Revert of Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. (patchset #3 id:40001 of https://codereview.webrtc.org/2046173002/ ) by deadbeef · 8 years ago
  29. bc58319 Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 8 years ago
  30. 5d97a9a Adding more detail to MessageQueue::Dispatch logging. by Taylor Brandstetter · 8 years ago
  31. d1fe281 Replace scoped_ptr with unique_ptr in webrtc/api/ by kwiberg · 8 years ago
  32. 4485ffb #include "webrtc/base/constructormagic.h" where appropriate by kwiberg · 8 years ago
  33. b24317b Fix license headers in webrtc/api. by kjellander · 8 years ago
  34. 15583c1 Move talk/app/webrtc to webrtc/api by Henrik Kjellander · 8 years ago[Renamed (97%) from talk/app/webrtc/remoteaudiosource.cc]
  35. 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
  36. e591f93 Storing raw audio sink for default audio track. by deadbeef · 9 years ago
  37. 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
  38. cb95f54e Remove pointless move() to fix build on clang/win. by Tommi · 9 years ago
  39. f888bb5 Support for unmixed remote audio into tracks. by Tommi · 9 years ago
  40. 5f93d0a Update libjingle license statements at top of talk files for consistency by jlmiller@webrtc.org · 10 years ago
  41. d4e598d (Auto)update libjingle 72097588-> 72159069 by buildbot@webrtc.org · 10 years ago
  42. b9a088b Update talk to 61538839. by wu@webrtc.org · 10 years ago
  43. 0de2950 Revert 5545 "Update libjingle to 61514460" by wu@webrtc.org · 10 years ago
  44. e749c9e Update libjingle to 61514460 by xians@webrtc.org · 10 years ago