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webrtc / src / webrtc / 70977183e7dfea78739eeca94dc74b290f85c5da / pc
  1. 1e2f1e5 Move RtcEventLog object from inside VoiceEngine to Call. by Ivo Creusen · 9 years ago
  2. 7790e22 Reland: Remove global list of SRTP sessions. by Joachim Bauch · 9 years ago
  3. d6e7474 Fixing bug where Connection drops packets when presumed writable. by Taylor Brandstetter · 9 years ago
  4. 0e15799 Reland of "Revert of Update the BWE when the network route changes. (patchset #5 id:180001 of https://… (patchset #1 id:1 of https://codereview.webrtc.org/2098703004/ ) by honghaiz · 9 years ago
  5. 632995d Revert of Revert "Revert of Update the BWE when the network route changes. (patchset #5 id:180001 of https://… (patchset #5 id:120001 of https://codereview.webrtc.org/2041593002/ ) by honghaiz · 9 years ago
  6. cb624fd Revert "Revert of Update the BWE when the network route changes. (patchset #5 id:180001 of https://codereview.webrtc.org/2000063003/ )" by Honghai Zhang · 9 years ago
  7. 6061fcc Delete GetExecutablePath and related unused code. by Niels Möller · 9 years ago
  8. a2636be Forward the SignalFirstPacketReceived to RtpReceiver. by zhihuang · 9 years ago
  9. e4800a7 Removed unused GetOutputVolume() and SetOutputVolume() from MediaEngineInterface. by solenberg · 9 years ago
  10. c0c552c Support for two audio codec lists down into WebRtcVoiceEngine. by ossu · 9 years ago
  11. d8878f5 Initial asymmetric codec support in MediaSessionDescription by ossu · 9 years ago
  12. b7f0831 Removing obsolete method from channel.h. by deadbeef · 9 years ago
  13. a996c6a GN: Add rtc_pc_unittests by kjellander · 9 years ago
  14. c0bec8f Adding more detail to MessageQueue::Dispatch logging. by Taylor Brandstetter · 9 years ago
  15. 3944b5a Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} by kjellander · 9 years ago
  16. 4e20ddd Combining SetVideoSend and SetSource into one method. by deadbeef · 9 years ago
  17. 6bd8e1a Remove metrics_default from rtc_media dependencies. by kjellander · 9 years ago
  18. 951103e Revert of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} (patchset #5 id:80001 of https://codereview.webrtc.org/1979933002/ ) by kjellander · 9 years ago
  19. d5e69cf Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} by kjellander · 9 years ago
  20. 549b014 Add RtpHeaderExtension to avoid client breakage by isheriff · 9 years ago
  21. c4921f4 Remove use of RtpHeaderExtension and clean up by isheriff · 9 years ago
  22. 9e1f65d Do not create a temporary transport channel when using max-bundle by skvlad · 9 years ago
  23. 80b957b Adding getParameters/setParameters APIs to RtpReceiver. by Taylor Brandstetter · 9 years ago
  24. bcf3191 Remove webrtc/base/scoped_ptr.h by kwiberg · 9 years ago
  25. 03917c4 Fix BaseChannel destructor when network thread differ from worker thread by Danil Chapovalov · 9 years ago
  26. e139be4 Revert of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} (patchset #2 id:20001 of https://codereview.webrtc.org/1973313002/ ) by kjellander · 9 years ago
  27. 030734f Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} by kjellander · 9 years ago
  28. 18d8284 Add a parameter to set a maximum file size when starting an RTC event log on the PeerConnectionFactory API. by ivoc · 9 years ago
  29. 6fd71cc Revert of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} (patchset #2 id:140001 of https://codereview.webrtc.org/1929633002/ ) by kjellander · 9 years ago
  30. a0362d2 GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} by kjellander · 9 years ago
  31. 1ebe930 Refactoring some tests in peerconnectioninterface_unittest.cc. by Taylor Brandstetter · 9 years ago
  32. 1ebd87f Renames TransportController worker_thread to network_thread. by Danil Chapovalov · 9 years ago
  33. 782a3f7 Fix all -Wnon-virtual-dtor warnings. by Henrik Kjellander · 9 years ago
  34. 5e67499 Adds network thread to rtc::BaseChannel by Danil Chapovalov · 9 years ago
  35. 97aa5c2 Revert of Remove webrtc/base/scoped_ptr.h (patchset #3 id:100001 of https://codereview.webrtc.org/1942823002/ ) by kwiberg · 9 years ago
  36. 8b348aa Remove webrtc/base/scoped_ptr.h by kwiberg · 9 years ago
  37. 8b8f8ff Only generate one CNAME per PeerConnection. by zhihuang · 9 years ago
  38. 3f20ddb Change default timestamp to 64 bits in all webrtc directories. by Honghai Zhang · 9 years ago
  39. 930147a Accept all the media profiles required by JSEP. by zhihuang · 9 years ago
  40. b670f85 Replace scoped_ptr with unique_ptr everywhere by kwiberg · 9 years ago
  41. 5fb5bd2 #include "webrtc/base/constructormagic.h" where appropriate by kwiberg · 9 years ago
  42. bfc1d55 Simple lint fixes by terelius · 9 years ago
  43. 1286d0e Replace scoped_ptr with unique_ptr in webrtc/base/ by jbauch · 9 years ago
  44. 41ab517 Revert of Remove the rtc_relative_path GYP variable and similar defines (patchset #1 id:1 of https://codereview.webrtc.org/1903553003/ ) by kjellander · 9 years ago
  45. 5a9038f Remove the rtc_relative_path GYP variable and similar defines by kjellander · 9 years ago
  46. 512897c Update the call when the network route changes by Honghai Zhang · 9 years ago
  47. cc35cae Revert of Accept all the media profiles required by JSEP. (patchset #5 id:80001 of https://codereview.webrtc.org/1880913002/ ) by zhihuang · 9 years ago
  48. 397934d Removing `preference` field from `cricket::Codec`. by deadbeef · 9 years ago
  49. d2a52a7 Accept all the media profiles required by JSEP. by zhihuang · 9 years ago
  50. cc4f458 Replace SetCapturer and SetCaptureDevice by SetSource. Drop return value. by nisse · 9 years ago
  51. 46c4295 Allow applications to control audio send bitrate through RtpParameters. by skvlad · 9 years ago
  52. 56d9dfc Add the last_sent_packet_id to the candidate pair change signal by Honghai Zhang · 9 years ago
  53. e1e16f0 Reland of move {media,p2p,pc,xmllite,xmpp}_tests.gypi files. (patchset #1 id:1 of https://codereview.webrtc.org/1846693002/ ) by kjellander · 9 years ago
  54. d25cdbb - Add temporary VoEBase::audio_device_module() method. by solenberg · 9 years ago
  55. 1f0f771 Revert of Remove {media,p2p,pc,xmllite,xmpp}_tests.gypi files. (patchset #1 id:1 of https://codereview.webrtc.org/1839763004/ ) by kjellander · 9 years ago
  56. 463cc09 Remove {media,p2p,pc,xmllite,xmpp}_tests.gypi files. by kjellander · 9 years ago
  57. 82688e9 Reset the BWE when the network changes. by Honghai Zhang · 9 years ago
  58. 12cfa58 Remove all uses of the HAVE_CONFIG_H define. by Henrik Kjellander · 9 years ago
  59. bb6decf Reland Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. by jbauch · 9 years ago
  60. 6a4e627 Revert of Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. (patchset #4 id:60001 of https://codereview.webrtc.org/1785713005/ ) by kjellander · 9 years ago
  61. 0912ecc Use CopyOnWriteBuffer instead of Buffer to avoid unnecessary copies. by jbauch · 9 years ago
  62. b1cf6ad Enabling rtcp-rsize negotiation and fixing some issues with it. by Taylor Brandstetter · 9 years ago
  63. 0b54e5a Reland "Add check_deps rules in DEPS files." by kjellander@webrtc.org · 9 years ago
  64. e7e56c0 Fix for intermittent tsan2 errors from SendRtpToRtpOnThread and SendSrtpToSrtpOnThread. by ossu · 9 years ago
  65. 4ac195e Enable setting the maximum bitrate limit in RtpSender. by skvlad · 9 years ago
  66. 10d5f8c Drop VideoOptions from VideoSendParameters. by nisse · 9 years ago
  67. de82d23 Revert of Add check_deps rules in DEPS files. (patchset #2 id:60001 of https://codereview.webrtc.org/1796413002/ ) by kjellander · 9 years ago
  68. 181e867 Add check_deps rules in DEPS files. by kjellander@webrtc.org · 9 years ago
  69. 0128c5a Replace scoped_ptr with unique_ptr in webrtc/pc/ by kwiberg · 9 years ago
  70. c20887e Add tracing to interesting media-related methods. by Peter Boström · 9 years ago
  71. e38b09a Prevent a voice channel from sending data before a source is set. by Taylor Brandstetter · 9 years ago
  72. 78f3581 Remove CaptureManager and related calls in ChannelManager. by perkj · 9 years ago
  73. e4e4306 Fixing some issues with payload type mappings. by Taylor Brandstetter · 9 years ago
  74. 5ec6244 Rename constants files in webrtc/{media,p2p} by kjellander · 9 years ago
  75. 4ff9968 Reland Remove unused cricket::VideoCapturer methods. Originally reviewed and landed as patchset #2 id:30001 of https://codereview.webrtc.org/1733673002/) by Per · 9 years ago
  76. caa8176 Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies." by kjellander@webrtc.org · 9 years ago
  77. 306b1ad Replace scoped_ptr with unique_ptr in webrtc/media/ by kwiberg · 9 years ago
  78. 5559034 Revert of Removed unused cricket::VideoCapturer methods (patchset #2 id:30001 of https://codereview.webrtc.org/1733673002/ ) by perkj · 9 years ago
  79. 2d78c68 Removed unused cricket::VideoCapturer methods: by perkj · 9 years ago
  80. 52cf08c 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
  81. 0bb951e Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies. by kjellander@webrtc.org · 9 years ago
  82. 516d536 Track pending ICE restarts independently for different media sections. by deadbeef · 9 years ago
  83. 3417189 Remove conference_mode flag from AudioOptions and VideoOptions. by nisse · 9 years ago
  84. 2412399 Rename libjingle_p2p_unittest -> rtc_pc_unittests by kjellander@webrtc.org · 9 years ago
  85. 5ec5665 Add OWNERS file in webrtc/pc by kjellander@webrtc.org · 9 years ago
  86. 886513ef This reland https://codereview.webrtc.org/1655793003/ with the change that cricket::VideoCapturer::SignalVideoFrame is added back and used for frame forwarding. It is used in Chrome remoting. by Per · 9 years ago
  87. e1a2ad1 Introduce struct MediaConfig, with construction-time settings. by nisse · 9 years ago
  88. 96ce682 Fix license headers in webrtc/pc by kjellander · 9 years ago
  89. 4056bb6 Move talk/session/media -> webrtc/pc by kjellander@webrtc.org · 9 years ago