Gitiles
Code Review Sign In
gerrit-public.fairphone.software / platform / external / webrtc / 96acb43b2affdd80eef8d0e63cafd9fdef8ff086 / pc
  1. 990d6b8 Revert "Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API" by Mirko Bonadei · 7 years ago
  2. 90bace0 Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API by henrika · 7 years ago
  3. 36b29d1 Enable cpplint in pc/ by Steve Anton · 7 years ago
  4. b2d355e Reland: Reject the description with fewer m= sections. by Zhi Huang · 7 years ago
  5. 074dece Fix flaky DataChannel integration test by Steve Anton · 7 years ago
  6. d5585ca Move almost all references from WebRtcSession to PeerConnection by Steve Anton · 7 years ago
  7. c4faa9c Remove QUIC transport/data channel by Steve Anton · 7 years ago
  8. ef48df9 Fix the issues in SrtpTransport. by Zhi Huang · 7 years ago
  9. 8a63f78 Rewrite the remaining few WebRtcSession tests. by Steve Anton · 7 years ago
  10. da6c095 Rewrite WebRtcSession data channel tests as PeerConnection tests by Steve Anton · 7 years ago
  11. 6f25b09 Reland "Rewrite WebRtcSession BUNDLE tests as PeerConnection tests" by Steve Anton · 7 years ago
  12. 8d3444d Reland "Rewrite WebRtcSession media tests as PeerConnection tests" by Steve Anton · 7 years ago
  13. f2662f0 Revert "Rewrite WebRtcSession media tests as PeerConnection tests" by Olga Sharonova · 7 years ago
  14. b49b661 Revert "Rewrite WebRtcSession BUNDLE tests as PeerConnection tests" by Olga Sharonova · 7 years ago
  15. 78609d5 Reland of BWE allocation strategy by Alex Narest · 7 years ago
  16. 6f72f56 Change return types of refcount methods. by Niels Möller · 7 years ago
  17. 096e367 Rewrite WebRtcSession BUNDLE tests as PeerConnection tests by Steve Anton · 7 years ago
  18. 3df5dca Rewrite WebRtcSession media tests as PeerConnection tests by Steve Anton · 7 years ago
  19. 3b80aac Fix flaky memory leak in RemoteAudioSource by Steve Anton · 7 years ago
  20. dc9ca93 Revert "BWE allocation strategy" by Alex Narest · 7 years ago
  21. a5fbc23 BWE allocation strategy by Alex Narest · 7 years ago
  22. 39260c4 Revert "BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic." by Lu Liu · 7 years ago
  23. 54d1da1 BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic. by Alex Narest · 7 years ago
  24. 1b0eae3 Don't call deprecated CreatePeerConnectionFactory() overloads by Karl Wiberg · 7 years ago
  25. 6592f2c Removes more unused ADM APIs: by henrika · 7 years ago
  26. 8b35df7 Try re-enabling VoiceChannel::TestInit. by Kári Tristan Helgason · 7 years ago
  27. ede9ca5 Rewrite WebRtcSession ICE integration tests as PeerConnection tests by Steve Anton · 7 years ago
  28. d6b4819 PeerConnection::StartRtcEventLog: Improve callback memory safety by Karl Wiberg · 7 years ago
  29. 919dc2e Removes fallback from Linux PulseAudio to ALSA. by henrika · 7 years ago
  30. 589ae45 Revert "Reject the subsequent offer with fewer m= sections." by Tommi · 7 years ago
  31. a8264db Reject the subsequent offer with fewer m= sections. by Zhi Huang · 7 years ago
  32. f1c6db1 Rewrite WebRtcSession ICE tests as PeerConnection tests by Steve Anton · 7 years ago
  33. 99c3fe5 Add PeerConnection::StartRtcEventLog version that takes RtcEventLogOutput as parameter by Elad Alon · 7 years ago
  34. 80cfb52 RTC_CHECK'ing content type before static_casting descriptions. by Taylor Brandstetter · 7 years ago
  35. b140b9f Keep count of libsrtp clients, and only deinitialize when it goes to 0. by Taylor Brandstetter · 7 years ago
  36. 9e6565b Fix PeerConnectionInterfaceTest_StartAndStopLoggingAfterPeerConnectionClosed by Elad Alon · 7 years ago
  37. c5bb00b PeerConnection end-to-end test with a non-builtin codec by Karl Wiberg · 7 years ago
  38. bdcee28 TurnCustomizer - an interface for modifying stun messages sent by TurnPort by Jonas Oreland · 7 years ago
  39. 933d8b0 Reland "Added PeerConnectionObserver::OnRemoveTrack." by Henrik Boström · 7 years ago
  40. 6c0c55c Revert "Added PeerConnectionObserver::OnRemoveTrack." by Alex Loiko · 7 years ago
  41. ba97ba7 Added PeerConnectionObserver::OnRemoveTrack. by Henrik Boström · 7 years ago
  42. 604427b Revert "TurnCustomizer - an interface for modifying stun messages sent by TurnPort" by Guido Urdaneta · 7 years ago
  43. b23ed7f TurnCustomizer - an interface for modifying stun messages sent by TurnPort by Jonas Oreland · 7 years ago
  44. 6b63cd5 Rewrite WebRtcSession DTLS/SDES crypto tests as PeerConnection tests by Steve Anton · 7 years ago
  45. 97a9f76 Add sdputils.h with useful functions for working with session descriptions by Steve Anton · 7 years ago
  46. 82eb3c4 Remove dead version of StartRtcEventLog by Elad Alon · 7 years ago
  47. acb2417 Fix apparent copy/paste error in comment (PeerConnection) by Elad Alon · 7 years ago
  48. 84255bb Add explicit includes of refcountedobject.h where it is used. by Niels Möller · 7 years ago
  49. fb26f85 Revert "Reland "Make rtc_base/refcount.h self contained, not including refcountedobject.h."" by Niels Moller · 7 years ago
  50. bf6937a Reland "Make rtc_base/refcount.h self contained, not including refcountedobject.h." by Niels Möller · 7 years ago
  51. e2d6a06 Reland "Clean up libjingle API dependencies." by Patrik Höglund · 7 years ago
  52. 1af3d82 Revert "Reland "Clean up libjingle API dependencies."" by Henrik Kjellander · 7 years ago
  53. 9185aca Reland "Clean up libjingle API dependencies." by Patrik Höglund · 7 years ago
  54. 04eaa15 Change the flag when RtpTransport objects send packet. by Zhi Huang · 7 years ago
  55. a32dd01 Reland "Remove AudioDeviceObserver and make ADM not inherit from the Module interface." by Fredrik Solenberg · 7 years ago
  56. 83ccca1 Create and use RtcEventLogOutput for output by Elad Alon · 7 years ago
  57. 98ea2da Removing logging in unit test that was committed accidentally. by Taylor Brandstetter · 7 years ago
  58. 1c34974 Fixing invalid calls to FindMatchingCodec. by Taylor Brandstetter · 7 years ago
  59. 8c0f7a7 Add GetRemoteAudioSSLCertificate() to PeerConnection by Steve Anton · 7 years ago
  60. 4a87e1c Remove encoding code from RtcEventLogImpl and use RtcEventLogEncoder instead by Elad Alon · 7 years ago
  61. d25fa78 Revert "Make rtc_base/refcount.h self contained, not including refcountedobject.h." by Niels Moller · 7 years ago
  62. b7239a9 Make rtc_base/refcount.h self contained, not including refcountedobject.h. by Niels Möller · 7 years ago
  63. 978b876 Move clients of WebRtcSession to use PeerConnection by Steve Anton · 7 years ago
  64. d4404c2 Revert "Remove AudioDeviceObserver and make ADM not inherit from the Module interface." by Fredrik Solenberg · 7 years ago
  65. 34cdd2d Remove AudioDeviceObserver and make ADM not inherit from the Module interface. by Fredrik Solenberg · 7 years ago
  66. b0a0207 Added RTCMediaStreamTrackStats.jitterBufferDelay for audio by Gustaf Ullberg · 7 years ago
  67. 581df61 Revert "Reland "Clean up libjingle API dependencies."" by Patrik Höglund · 7 years ago
  68. 5117b04 Reland "Clean up libjingle API dependencies." by Patrik Höglund · 7 years ago
  69. b526158 Move the TransportController from p2p/base to pc/. by Zhi Huang · 7 years ago
  70. d8970db Delete unneeded includes of fileutils.h by Niels Möller · 7 years ago
  71. 7bcfc3b Revert "Clean up libjingle API dependencies." by Patrik Höglund · 7 years ago
  72. bf66794 Revert "Move clients of WebRtcSession to use PeerConnection" by Alex Loiko · 7 years ago
  73. 57fb315 Clean up libjingle API dependencies. by Patrik Höglund · 7 years ago
  74. 3dc4d4a Move clients of WebRtcSession to use PeerConnection by Steve Anton · 7 years ago
  75. 94286cb Add base fixture and PeerConnection wrapper for unit tests by Steve Anton · 7 years ago
  76. 02e7a19 Remove unnecessary video factory references in PeerConnectionFactory by Magnus Jedvert · 7 years ago
  77. cf990f5 Reland: Completed the functionalities of SrtpTransport. by Zhi Huang · 7 years ago
  78. 835cc0c Remove unnecessary audio references in PeerConnectionFactory by Magnus Jedvert · 7 years ago
  79. 4e2deab Only return stats for the most recent unsignaled audio stream. by deadbeef · 7 years ago
  80. b19012e Remove the support of fallback from DTLS to SDES. by zhihuang · 7 years ago
  81. eb23e17 Revert of Completed the functionalities of SrtpTransport. (patchset 7 id:320001 of https://codereview.webrtc.org/2997983002/ ) by zhihuang · 7 years ago
  82. 1d4db39 Revert of If SRTP sessions exist, don't create new ones when applying answer. (patchset #1 id:1 of https://codereview.webrtc.org/3019443002/ ) by henrika · 7 years ago
  83. 9a2e906 Added RTCMediaStreamTrackStats.concealmentEvents by Gustaf Ullberg · 7 years ago
  84. d45aea8 Serialize "a=x-google-flag:conference". by deadbeef · 7 years ago
  85. 5ada7ac If SRTP sessions exist, don't create new ones when applying answer. by deadbeef · 7 years ago
  86. 58b0316 Expose new video codec factories in the PeerConnectionFactory API by Magnus Jedvert · 7 years ago
  87. 7120742 Adding NOLINT for typedefs.h and common_types.h by Mirko Bonadei · 7 years ago
  88. 563934e Clean up dependencies of peerconnection_unittest. by Patrik Höglund · 7 years ago
  89. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  90. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago