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webrtc / src / webrtc / c1a5e1b9535b8c499df575c8927b0c4437f4451d / pc
  1. 13d9326 RTC_[D]CHECK_op: Remove superfluous casts by kwiberg · 8 years ago
  2. 9ed6bfe Don't cache video codec list in VideoEngine2. by brandtr · 8 years ago
  3. a1562c3 Let MediaSession generate a FlexFEC SSRC when FlexFEC is active. by brandtr · 8 years ago
  4. abef9e9 Remove all references to GYP by Henrik Kjellander · 8 years ago
  5. 155a5b4 Negotiate H264 profiles in SDP by magjed · 8 years ago
  6. 8261e17 Optimize FindCodecById and ReferencedCodecsMatch by magjed · 8 years ago
  7. e843185 Reland of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2471783002/ ) by nisse · 8 years ago
  8. a349842 Revert of Stop caching supported codecs in WebRtcVideoEngine2 (patchset #1 id:1 of https://codereview.webrtc.org/2492473002/ ) by magjed · 8 years ago
  9. 605bf2c Stop caching supported codecs in WebRtcVideoEngine2 by magjed · 8 years ago
  10. 2454551 Set actual transport overhead in rtp_rtcp by michaelt · 8 years ago
  11. a5dffff Revert of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #7 id:120001 of https://codereview.webrtc.org/2383093002/ ) by nisse · 8 years ago
  12. 6d63773 Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. by nisse · 8 years ago
  13. c6f0d15 Introduce rtc::PacketTransportInterface and let cricket::TransportChannel inherit. by johan · 8 years ago
  14. 3f983f0 Remove remnants of libsrtp1 by mattdr · 8 years ago
  15. 39c334a Delete unused file screencastid.h. by nisse · 8 years ago
  16. a5e117a Remove cricket::VideoCodec with, height and framerate properties by perkj · 8 years ago
  17. a6c6623 GN: Exclude suppressions of Chromium Clang warnings for Chromium builds. by kjellander · 8 years ago
  18. 58922f7 Remove useless debugging code by mattdr · 8 years ago
  19. d4026d5 Fix externalhmac.h/.cc to compile with libsrtp 1 and 2 by mattdr · 8 years ago
  20. 043abc4 - Filter data channel codecs based on codec name instead of payload type, which may have been remapped. by solenberg · 8 years ago
  21. 14ccce8 Refactoring: move ownership of RtcEventLog from Call to PeerConnection by skvlad · 8 years ago
  22. 0d86b80 Delete unused class cricket::MediaSinkInterface, and mediasink.h. by nisse · 8 years ago
  23. d402b24 Update WebRTC to build against libsrtp 2.0 by mattdr · 8 years ago
  24. 2cf29b5 GN: Change rtc_source_set targets --> rtc_static_library by kjellander · 8 years ago
  25. 435c7b3 GYP: Remove targets inside include_tests==1 that are converted to GN. by kjellander · 8 years ago
  26. 4c5e7eb GN: Change group deps to public_deps. by kjellander · 8 years ago
  27. 9eb213a OWNERS: Make everyone able to change *.gn,*.gni files. by Henrik Kjellander · 8 years ago
  28. 80e6692 GN Templates: Move common_inherited_config to the template. by ehmaldonado · 9 years ago
  29. 1d49219 GN Templates: Move common_config to the template. by ehmaldonado · 9 years ago
  30. f0532f3 GN: Introduce templates. by ehmaldonado · 9 years ago
  31. 774bc41 Add signaling to support ICE renomination. by Honghai Zhang · 9 years ago
  32. cf82062 Move webrtc/audio_*.h to webrtc/api/call by kjellander · 9 years ago
  33. ff96df5 Combining "SetTransportChannel" and "SetRtcpTransportChannel". by deadbeef · 9 years ago
  34. a9d5eb4 Renaming BaseChannel methods and adding comments for added clarity. by Taylor Brandstetter · 9 years ago
  35. c66c78a Fixing off-by-one error with max SCTP id. by Taylor Brandstetter · 9 years ago
  36. 70e7234 Some cleanup in BaseChannel RTCP mux code. by deadbeef · 9 years ago
  37. 89694d9 Adding deadbeef and honghaiz as owners of webrtc/pc. by deadbeef · 9 years ago
  38. 7a3fc83 Add kjellander@webrtc.org to more BUILD.gn OWNERS files. by kjellander · 9 years ago
  39. 4443dc4 Add support for GCM cipher suites from RFC 7714. by jbauch · 9 years ago
  40. 38dba54 Un-flaking TestSrtpError by using a fake clock. by Taylor Brandstetter · 9 years ago
  41. 96053e5 Don't stop sending media on EWOULDBLOCK by skvlad · 9 years ago
  42. 23ea12e Reland of "Move RtcEventLog object from inside VoiceEngine to Call.", "Fix to make the start/stop functions for the Rtc Eventlog non-virtual." and "Fix for RtcEventLog ObjC interface" by ivoc · 9 years ago
  43. 822f09e Revert of Move RtcEventLog object from inside VoiceEngine to Call. (patchset #16 id:420001 of https://codereview.webrtc.org/1748403002/ ) by ivoc · 9 years ago
  44. 1e2f1e5 Move RtcEventLog object from inside VoiceEngine to Call. by Ivo Creusen · 9 years ago
  45. 7790e22 Reland: Remove global list of SRTP sessions. by Joachim Bauch · 9 years ago
  46. d6e7474 Fixing bug where Connection drops packets when presumed writable. by Taylor Brandstetter · 9 years ago
  47. 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
  48. 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
  49. 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
  50. 6061fcc Delete GetExecutablePath and related unused code. by Niels Möller · 9 years ago
  51. a2636be Forward the SignalFirstPacketReceived to RtpReceiver. by zhihuang · 9 years ago
  52. e4800a7 Removed unused GetOutputVolume() and SetOutputVolume() from MediaEngineInterface. by solenberg · 9 years ago
  53. c0c552c Support for two audio codec lists down into WebRtcVoiceEngine. by ossu · 9 years ago
  54. d8878f5 Initial asymmetric codec support in MediaSessionDescription by ossu · 9 years ago
  55. b7f0831 Removing obsolete method from channel.h. by deadbeef · 9 years ago
  56. a996c6a GN: Add rtc_pc_unittests by kjellander · 9 years ago
  57. c0bec8f Adding more detail to MessageQueue::Dispatch logging. by Taylor Brandstetter · 9 years ago
  58. 3944b5a Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} by kjellander · 9 years ago
  59. 4e20ddd Combining SetVideoSend and SetSource into one method. by deadbeef · 9 years ago
  60. 6bd8e1a Remove metrics_default from rtc_media dependencies. by kjellander · 9 years ago
  61. 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
  62. d5e69cf Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} by kjellander · 9 years ago
  63. 549b014 Add RtpHeaderExtension to avoid client breakage by isheriff · 9 years ago
  64. c4921f4 Remove use of RtpHeaderExtension and clean up by isheriff · 9 years ago
  65. 9e1f65d Do not create a temporary transport channel when using max-bundle by skvlad · 9 years ago
  66. 80b957b Adding getParameters/setParameters APIs to RtpReceiver. by Taylor Brandstetter · 9 years ago
  67. bcf3191 Remove webrtc/base/scoped_ptr.h by kwiberg · 9 years ago
  68. 03917c4 Fix BaseChannel destructor when network thread differ from worker thread by Danil Chapovalov · 9 years ago
  69. 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
  70. 030734f Reland of GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} by kjellander · 9 years ago
  71. 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
  72. 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
  73. a0362d2 GN: Add BUILD.gn files for webrtc/{api,media,libjingle,p2p,pc} by kjellander · 9 years ago
  74. 1ebe930 Refactoring some tests in peerconnectioninterface_unittest.cc. by Taylor Brandstetter · 9 years ago
  75. 1ebd87f Renames TransportController worker_thread to network_thread. by Danil Chapovalov · 9 years ago
  76. 782a3f7 Fix all -Wnon-virtual-dtor warnings. by Henrik Kjellander · 9 years ago
  77. 5e67499 Adds network thread to rtc::BaseChannel by Danil Chapovalov · 9 years ago
  78. 97aa5c2 Revert of Remove webrtc/base/scoped_ptr.h (patchset #3 id:100001 of https://codereview.webrtc.org/1942823002/ ) by kwiberg · 9 years ago
  79. 8b348aa Remove webrtc/base/scoped_ptr.h by kwiberg · 9 years ago
  80. 8b8f8ff Only generate one CNAME per PeerConnection. by zhihuang · 9 years ago
  81. 3f20ddb Change default timestamp to 64 bits in all webrtc directories. by Honghai Zhang · 9 years ago
  82. 930147a Accept all the media profiles required by JSEP. by zhihuang · 9 years ago
  83. b670f85 Replace scoped_ptr with unique_ptr everywhere by kwiberg · 9 years ago
  84. 5fb5bd2 #include "webrtc/base/constructormagic.h" where appropriate by kwiberg · 9 years ago
  85. bfc1d55 Simple lint fixes by terelius · 9 years ago
  86. 1286d0e Replace scoped_ptr with unique_ptr in webrtc/base/ by jbauch · 9 years ago
  87. 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
  88. 5a9038f Remove the rtc_relative_path GYP variable and similar defines by kjellander · 9 years ago
  89. 512897c Update the call when the network route changes by Honghai Zhang · 9 years ago
  90. 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
  91. 397934d Removing `preference` field from `cricket::Codec`. by deadbeef · 9 years ago
  92. d2a52a7 Accept all the media profiles required by JSEP. by zhihuang · 9 years ago
  93. cc4f458 Replace SetCapturer and SetCaptureDevice by SetSource. Drop return value. by nisse · 9 years ago
  94. 46c4295 Allow applications to control audio send bitrate through RtpParameters. by skvlad · 9 years ago
  95. 56d9dfc Add the last_sent_packet_id to the candidate pair change signal by Honghai Zhang · 9 years ago
  96. 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
  97. d25cdbb - Add temporary VoEBase::audio_device_module() method. by solenberg · 9 years ago
  98. 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
  99. 463cc09 Remove {media,p2p,pc,xmllite,xmpp}_tests.gypi files. by kjellander · 9 years ago
  100. 82688e9 Reset the BWE when the network changes. by Honghai Zhang · 9 years ago