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webrtc / src / / 70483a59aacc2196d6685ee7100c817d42cea7e6 / pc / remote_audio_source.h
  1. c0d44d9 Split audio and video channels into Send and Receive APIs. by Harald Alvestrand · 2 years, 4 months ago
  2. c6c346d Remove usage of rtc::MessageHandler in pc/remote_audio_source by Danil Chapovalov · 2 years, 7 months ago
  3. 65685a6 Move pc/channel.h to only be used in .cc files by Harald Alvestrand · 3 years ago
  4. c335b0e [Unified Plan] Don't end audio tracks when SSRC changes. by Henrik Boström · 4 years ago
  5. 5761e7b Running apply-iwyu on ~all files in pc/ by Harald Alvestrand · 4 years, 2 months ago
  6. 6fcd0f8 Migrate pc/ to webrtc::Mutex. by Markus Handell · 4 years, 9 months ago
  7. 749f660 Enable SSRC 0 in MediaChannel methods by Saurav Das · 5 years ago
  8. 428dcb2 Remove SetLatency/GetLatency from MediaSourceInterface API level by Ruslan Burakov · 6 years ago
  9. 493a650 Propagate base minimum delay from video jitter buffer to webrtc/api. by Ruslan Burakov · 6 years ago
  10. 7ea4605 Add latency to remote source api. by Ruslan Burakov · 6 years ago
  11. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  12. 1c05765 (3) Rename files to snake_case: move the files by Steve Anton · 6 years ago[Renamed from pc/remoteaudiosource.h]
  13. d367921 Configure media flow correctly with Unified Plan by Steve Anton · 7 years ago
  14. 6077675 Change RtpReceivers to interact with the media channel directly by Steve Anton · 7 years ago
  15. 3b80aac Fix flaky memory leak in RemoteAudioSource by Steve Anton · 7 years ago
  16. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 8 years ago
  17. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 8 years ago[Renamed from webrtc/pc/remoteaudiosource.h]
  18. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  19. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  20. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  21. 7bb87ee Create //webrtc/api:libjingle_peerconnection_api + refactorings. by ossu · 8 years ago[Renamed (94%) from webrtc/api/remoteaudiosource.h]
  22. a69d973 Move webrtc/audio_*.h to webrtc/api/call by kjellander · 9 years ago
  23. ba29c6a Reland 2 of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 9 years ago
  24. 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
  25. 2d54917 Reland of: Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 9 years ago
  26. 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
  27. bc58319 Use VoiceChannel/VideoChannel directly from RtpSender/RtpReceiver. by Taylor Brandstetter · 9 years ago
  28. 1a018dc Prevent a voice channel from sending data before a source is set. by Taylor Brandstetter · 9 years ago
  29. 7ffeab5 Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies." by kjellander@webrtc.org · 9 years ago
  30. 7324eb9 Revert of Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. (patchset #2 id:40001 of https://codereview.webrtc.org/1737593002/ ) by kjellander · 9 years ago
  31. 99b345c Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. by kjellander@webrtc.org · 9 years ago
  32. b24317b Fix license headers in webrtc/api. by kjellander · 9 years ago
  33. 15583c1 Move talk/app/webrtc to webrtc/api by Henrik Kjellander · 9 years ago[Renamed (93%) from talk/app/webrtc/remoteaudiosource.h]
  34. a96e2d7 Move talk/media to webrtc/media by kjellander · 9 years ago
  35. 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
  36. f888bb5 Support for unmixed remote audio into tracks. by Tommi · 9 years ago
  37. 14665ff Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro by kjellander@webrtc.org · 10 years ago
  38. 5f93d0a Update libjingle license statements at top of talk files for consistency by jlmiller@webrtc.org · 10 years ago
  39. d4e598d (Auto)update libjingle 72097588-> 72159069 by buildbot@webrtc.org · 11 years ago
  40. b9a088b Update talk to 61538839. by wu@webrtc.org · 11 years ago
  41. 0de2950 Revert 5545 "Update libjingle to 61514460" by wu@webrtc.org · 11 years ago
  42. e749c9e Update libjingle to 61514460 by xians@webrtc.org · 11 years ago