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gerrit-public.fairphone.software / platform / external / webrtc / 73674f80644c0e4bbf8436be8f3b8dedf959d695 / talk / session / media
  1. d66b44d Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87. by ivoc · 9 years ago
  2. 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
  3. e591f93 Storing raw audio sink for default audio track. by deadbeef · 9 years ago
  4. f475d36 Properly handle different transports having different SSL roles. by Taylor Brandstetter · 9 years ago
  5. e6bf587 Deleted VideoCapturer::screencast_max_pixels, together with by nisse · 9 years ago
  6. a4df27b Revert of Reland "Added option to specify a maximum file size when recording an AEC dump." (patchset #2 id:20001 of https://codereview.webrtc.org/1541633002/ ) by ivoc · 9 years ago
  7. f4f5cb0 Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87. by ivoc · 9 years ago
  8. 36d4c54 Revert of Added option to specify a maximum file size when recording an AEC dump. (patchset #5 id:120001 of https://codereview.webrtc.org/1413483003/ ) by ivoc · 9 years ago
  9. ae2c5ad Added option to specify a maximum file size when recording an AEC dump. by ivoc · 9 years ago
  10. 77fa59d Fix build break in google3 import caused by https://codereview.webrtc.org/1532543003 by guoweis · 9 years ago
  11. 4638331 DTLS-SRTP set up is bypassed when the channel has been writable. by guoweis · 9 years ago
  12. 0eb15ed Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector by kwiberg · 9 years ago
  13. 44f0819 Fixing bug where "mid" wasn't preserved across re-offers. by deadbeef · 9 years ago
  14. f888bb5 Support for unmixed remote audio into tracks. by Tommi · 9 years ago
  15. 822bdf9 Remove cricket::VideoEncoderConfig. by Peter Boström · 9 years ago
  16. 1387149 Adding reduced size RTCP configuration down to the video stream level. by deadbeef · 9 years ago
  17. 1a9d615 Add tracing to public PeerConnection methods. by Peter Boström · 9 years ago
  18. 246b817 Refactor handling of AudioOptions. by solenberg · 9 years ago
  19. 9f45a45 Add tracing to upper-level WebRTC calls. by Peter Boström · 9 years ago
  20. 46ad542 Revert of "Create rtc::AtomicInt POD struct." (patchset #3 id:40001 of https://codereview.webrtc.org/1498953002/ ) by pbos · 9 years ago
  21. 6f28cf0 Implement standalone event tracing in AppRTCDemo. by Peter Boström · 9 years ago
  22. 84f0970 Reland of "Create rtc::AtomicInt POD struct." by Peter Boström · 9 years ago
  23. 9d69c3f Return a copy of the supported RTP header extensions instead of a reference. by Stefan Holmer · 9 years ago
  24. 1218d7a Allow remote fingerprint update during a call by Guo-wei Shieh · 9 years ago
  25. 86aaa4b Revert "Allow remote fingerprint update during a call" by Guo-wei Shieh · 9 years ago
  26. 9c38c2d Allow remote fingerprint update during a call by Guo-wei Shieh · 9 years ago
  27. 1d63dd0 - Remove cricket::VoiceChannel::PressDtmf(); AFAICT unused. by solenberg · 9 years ago
  28. b5cb19b Fixing direction attribute in answer for non-RTP protocols. by deadbeef · 9 years ago
  29. bd13838 Remove SetVideoLogging/SetAudioLogging from ChannelManager and down the stack. by solenberg · 9 years ago
  30. 521ed7b Reland Convert internal representation of Srtp cryptos from string to int by Guo-wei Shieh · 9 years ago
  31. 318166b Revert of Convert internal representation of Srtp cryptos from string to int. (patchset #10 id:180001 of https://codereview.webrtc.org/1416673006/ ) by guoweis · 9 years ago
  32. 2764e10 Convert internal representation of Srtp cryptos from string to int. by guoweis · 9 years ago
  33. 482b12e Remove BundleFilter filtering of RTCP. by pbos · 9 years ago
  34. cbe9f51 Revert of Remove global list of SRTP sessions. (patchset #4 id:60001 of https://codereview.webrtc.org/1416093010/ ) by phoglund · 9 years ago
  35. 5237aaf Convert usage of ARRAY_SIZE to arraysize. by tfarina · 9 years ago
  36. 9cafd97 Remove global list of SRTP sessions. by jbauch · 9 years ago
  37. be57983 Rename Maybe to Optional by Karl Wiberg · 9 years ago
  38. 102c6a6 Replace rtc::cricket::Settable with rtc::Maybe by kwiberg · 9 years ago
  39. ec9d187 Added override keyword to overridden methods to stop compiler warnings. by rlester · 9 years ago
  40. ff134eb talk: Use NDEBUG macro. by tfarina · 9 years ago
  41. c80741f Fixing some issues with the direction attribute of m-lines in offers. by deadbeef · 9 years ago
  42. 797ef12 Added StopAecDump function to PeerConnectionFactory. by ivoc · 9 years ago
  43. 112a3d8 Added functions on libjingle API to start and stop the recording of an RtcEventLog. by ivoc · 9 years ago
  44. c1aeaf0 Wire up packet_id / send time callbacks to webrtc via libjingle. by stefan · 9 years ago
  45. d59daf8 Merging BaseSession code into WebRtcSession. by deadbeef · 9 years ago
  46. 1ac5614 Remove default receive channel from WVoE; baby step 3. by solenberg · 9 years ago
  47. d4cec0d Remove MediaChannel::SetRemoteRenderer(). by solenberg · 9 years ago
  48. 4bac9c5 Change SetOutputScaling to set a single level, not left/right levels. by solenberg · 9 years ago
  49. 0c4e06b Use suffixed {uint,int}{8,16,32,64}_t types. by Peter Boström · 9 years ago
  50. 5629a1d Fix flaky test TestSrtpError, introduced in https://codereview.webrtc.org/1362913004. by solenberg · 9 years ago
  51. 5b14b42 Remove unused SignalMediaError and infrastructure. by solenberg · 9 years ago
  52. dfc8f4f Change 'mute' parameter of MediaChannel::SetAudioSend()/SetVideoSend() to 'enable'. by solenberg · 9 years ago
  53. 456696a Reland Change WebRTC SslCipher to be exposed as number only by Guo-wei Shieh · 9 years ago
  54. 27dc29b Revert of Change WebRTC SslCipher to be exposed as number only. (patchset #20 id:750001 of https://codereview.webrtc.org/1337673002/ ) by guoweis · 9 years ago
  55. 4fe3c9a Change WebRTC SslCipher to be exposed as number only. by guoweis · 9 years ago
  56. 34fbfff Remove VideoMediaChannel::SetRender(). by Peter Boström · 9 years ago
  57. 4a3ccad Remove SetAudioDelayOffset() and friends. by solenberg · 9 years ago
  58. 61e933e Remove ChannelManager::GetCapabilities() by solenberg · 9 years ago
  59. facbbec Remove use of DeviceManager from ChannelManager. by solenberg · 9 years ago
  60. cbecd35 Reland of TransportController refactoring. (patchset #1 id:1 of https://codereview.webrtc.org/1358413003/ ) by deadbeef · 9 years ago
  61. 7d17336 Remove the [Un]RegisterVoiceProcessor() API. by Fredrik Solenberg · 9 years ago
  62. a81a42f Revert of TransportController refactoring. (patchset #6 id:100001 of https://codereview.webrtc.org/1350523003/ ) by torbjorng · 9 years ago
  63. 47ee2f3 TransportController refactoring. by deadbeef · 9 years ago
  64. c1a1b35 Remove the SetLocalMonitor() API. by solenberg · 9 years ago
  65. 22011c1 Remove Channel::SetRingbackTone() and Channel::PlayRingbackTone(), and the code beneath it (within libjingle). by solenberg · 9 years ago
  66. 8902433 Revert "TransportController refactoring." by Guo-wei Shieh · 9 years ago
  67. 9af63f4 TransportController refactoring. by deadbeef · 9 years ago
  68. 7cbd188 Remove GICE (again). by Peter Thatcher · 9 years ago
  69. b071a19 Full use of NnChannel::SetSendParameters and NnChannel::SetRecvParameters. by Fredrik Solenberg · 9 years ago
  70. 91d6ede Add RTC_ prefix to (D)CHECKs and related macros. by henrikg · 9 years ago
  71. 3c089d7 Add RTC_ prefix to contructormagic macros. by henrikg · 9 years ago
  72. 709ed67 Move instantiation of webrtc::Call into a MediaController class so that it can be used for both audio and video media channels. by Fredrik Solenberg · 9 years ago
  73. d12140a Revert change which removes GICE. by guoweis · 9 years ago
  74. fab882b Remove obsolete typingmonitor.cc/.h files. by solenberg · 9 years ago
  75. 1dd98f3 - Rename VoiceChannel::MuteStream() -> SetAudioSend() (incl. media channel) by solenberg · 9 years ago
  76. 66f4339 Remove [Voice|Video]MediaChannel::GetOptions(). by solenberg · 9 years ago
  77. 8006f07 Remove unused TypingMonitor class. by solenberg · 9 years ago
  78. e9ad18b Remove obsolete soundclip.cc/.h files. by solenberg · 9 years ago
  79. 3a14bf3 Replacing SSLIdentity* with scoped_refptr<RTCCertificate> in the cricket::TransportDescriptionFactory layers. by Henrik Boström · 9 years ago
  80. d828198 Replaces SSLIdentity* with scoped_refptr<RTCCertificate> in the cricket::Transport layer. by Henrik Boström · 9 years ago
  81. 2159b89 Reland "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots. by Peter Thatcher · 9 years ago
  82. 5bdafd4 Revert "Reland "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots."" by minyuel · 9 years ago
  83. c232096 Remove cricket::VideoProcessor and AddVideoProcessor() functionality by Magnus Jedvert · 9 years ago
  84. bfab5cb Fix some minor errors with the voice engine caused by the refactor CL https://codereview.webrtc.org/1229283003/. by Peter Thatcher · 9 years ago
  85. a5b273a Fixing problems with RTP extension ID conflict resolution by deadbeef · 9 years ago
  86. 081f34b Reland "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots." by Peter Thatcher · 9 years ago
  87. dbe5bd9 Delete unused function SetSessionError. by Nico Weber · 9 years ago
  88. b6d4ec4 Support generation of EC keys using P256 curve and support ECDSA certs. by Torbjorn Granlund · 9 years ago
  89. fa30180 Revert "Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever)." becauese remoting code is using dead constants and breaks the FYI bots. by pthatcher · 9 years ago
  90. 3449faa Remove GICE (gone forever!) and PORTALLOCATOR_ENABLE_SHARED_UFRAG (enabled forever). by Peter Thatcher · 9 years ago
  91. c2ee2c8 Refactor the relationship between BaseChannel and MediaChannel so that we send over all the parameters in one method call rather then having them broken up into multiple method calls. This should allow future refactorings of the WebRtcVideoEngine2 to not recreate configurations so many times, and have more simple code as well. by Peter Thatcher · 9 years ago
  92. 0c02264 Get rid of media_engine_ from BaseChannel; only VoiceChannel needs it. by Fredrik Solenberg · 9 years ago
  93. a9b4c32 Nuke buffered latency mode. It's not actually working, and it's not used. It's just dead code complexity. by Peter Thatcher · 9 years ago
  94. 083b73f Use std::string references instead of copying contents. by jbauch · 9 years ago
  95. f393829 Use "UDP/TLS/RTP/SAVPF" profile in offer when DTLS-SRTP is used. by deadbeef · 9 years ago
  96. a6d2444 Remove BaseSession::SignalNewDescription. It was only used by GTP and now just clutters the code. by Peter Thatcher · 10 years ago
  97. 3b1e647 Remove media sinks from Channel. by pbos · 10 years ago
  98. c28a896 VoE: Initialize WebRtcVoiceMediaChannel with AudioOptions during creation by Jelena Marusic · 10 years ago
  99. e70028e Protect access to shared list of SRTP sessions. by Joachim Bauch · 10 years ago
  100. fec2c6d Prevent potential double-free if srtp_create fails. by Joachim Bauch · 10 years ago