Google Git
Sign in
webrtc / src / webrtc / ab1c7907dc77972086ceab9a9dbf2d0621b11a80 / pc
  1. 37342b9 Report timing frames info in GetStats. by ilnik · 8 years ago
  2. bc32410 Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  3. 60154fd Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  4. 39f7f7a Enable the injection of an APM into a peerconnection by peah · 8 years ago
  5. 56f7815 Enable -Wunused-function warning everywhere. by Henrik Kjellander · 8 years ago
  6. 150ba79 Fix -Wcomment warning in webrtcsdp.cc by kjellander · 8 years ago
  7. 0d58090 Support encrypted RTP extensions (RFC 6904) by jbauch · 8 years ago
  8. 588f761 Allow an external audio processing module to be used in WebRTC by peah · 8 years ago
  9. 0214bf4 Fixing RTCIceCandidatePairStats.nominated for ICE controlling agent. by deadbeef · 8 years ago
  10. 6efbd3c Support getting external HMAC auth context with libsrtp 2.1.0. by jbauch · 8 years ago
  11. 51a0b43 Remove unused "crypto_options_" field. by jbauch · 8 years ago
  12. db65e09 Fix the binary size regression on Chromium Windows. by zhihuang · 8 years ago
  13. 0f3c15e Reland of Try to fix the binary size increase issue on Chromium. (patchset #1 id:1 of https://codereview.webrtc.org/2949953003/ ) by zhihuang · 8 years ago
  14. 2d5cadc Revert of Try to fix the binary size increase issue on Chromium. (patchset #1 id:1 of https://codereview.webrtc.org/2945233002/ ) by zhihuang · 8 years ago
  15. d901125 Try to fix the binary size increase issue on Chromium. by zhihuang · 8 years ago
  16. 3943e59 Fix uploading of available send bitrate statistics. by Alex Narest · 8 years ago
  17. 10a13e1 Fixing incorrect use of erase/remove idiom. by deadbeef · 8 years ago
  18. 8c18d91 Enable SNI in ssl adapter. by Emad Omara · 8 years ago
  19. 5e343ec Delete SignalSrtpError. by nisse · 8 years ago
  20. f91805c Support building WebRTC without audio and video. by zhihuang · 8 years ago
  21. f94a820 Allow WebRtcMediaEngine to be created from any thread. by deadbeef · 8 years ago
  22. 5aaa670 Fix Chromium style checker warnings for MockAudioDecoder by kwiberg · 8 years ago
  23. 8f7d284 Remove DCHECK from PeerConnectionFactory::worker_thread. by zstein · 8 years ago
  24. 3d545a2 s/WebRtcVideoChannel2/WebRtcVideoChannel and s/WebRtcVideoEngine2/WebRtcVideoEngine by eladalon · 8 years ago
  25. 33944ff Relanding: Adds PeerConnectionInterface::UpdateCallBitrate to give clients more control of the bandwidth estimator. PeerConnection implements this method by passing a BitrateConfigMask to its associated Call, which is combined with the existing BitrateConfig and passed on to the SendSideCongestionController as necessary. The existing BitrateConfig generally comes from the x-google-{min,start,max}-bitrate params in the SDP. by zstein · 8 years ago
  26. fb8fb2d Revert of Add PeerConnectionInterface::UpdateCallBitrate. (patchset #7 id:120001 of https://codereview.webrtc.org/2888303005/ ) by charujain · 8 years ago
  27. 9f8b6f3 Adds PeerConnectionInterface::UpdateCallBitrate to give clients more control of the bandwidth estimator. PeerConnection implements this method by passing a BitrateConfigMask to its associated Call, which is combined with the existing BitrateConfig and passed on to the SendSideCongestionController as necessary. The existing BitrateConfig generally comes from the x-google-{min,start,max}-bitrate params in the SDP. by zstein · 8 years ago
  28. 4e13fcb Revert "Revert of Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. (patchset #8 id:140001 of https://codereview.webrtc.org/2863123002/ )" by stefan · 8 years ago
  29. df24c99 Move RTP/RTCP demuxing logic from BaseChannel to RtpTransport. by zstein · 8 years ago
  30. 64ccd10 enabling `gn check` on the whole WebRTC repo by mbonadei · 8 years ago
  31. b6a2c8f Revert of Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. (patchset #8 id:140001 of https://codereview.webrtc.org/2863123002/ ) by charujain · 8 years ago
  32. ddb82e2 Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. by Stefan Holmer · 8 years ago
  33. f82a638 Reland of Removes usage of native base::android::GetApplicationContext() by sakal · 8 years ago
  34. f78d11d Support "UDP/DTLS/SCTP" and "TCP/DTLS/SCTP" profile strings. by deadbeef · 8 years ago
  35. 6511376 Revert of https://codereview.webrtc.org/2889183002/ by lliuu · 8 years ago
  36. 4873e3b Reland of moves usage of native base::android::GetApplicationContext() (patchset #1 id:1 of https://codereview.webrtc.org/2894593002/ ) by sakal · 8 years ago
  37. 9f58c6f Add media related stats (audio level etc.) to unsignaled streams. by zhihuang · 8 years ago
  38. cfc8721 Revert of Removes usage of native base::android::GetApplicationContext() (patchset #6 id:120001 of https://codereview.webrtc.org/2888093004/ ) by sakal · 8 years ago
  39. 98eef6c Removes usage of native base::android::GetApplicationContext() by sakal · 8 years ago
  40. 8357b2c Remove temporary include of builtin_audio_encoder_factory.h. by ossu · 8 years ago
  41. 92b90e4 Remove VirtualSocketServer's dependency on PhysicalSocketServer. by deadbeef · 8 years ago
  42. 61de4a2 Get tests working on systems that only support IPv6. by deadbeef · 8 years ago
  43. 0e31205 Initialize PeerConnection members in declaration order and destroy them in reverse order. by terelius · 8 years ago
  44. 59a36ed Delete method MessageQueue::set_socketserver by nisse · 8 years ago
  45. e6bd325 Update comments for removal of MediaController. by nisse · 8 years ago
  46. 886a9f8 Unflaking PeerConnectionIntegrationTest.DtmfSenderObserver. by deadbeef · 8 years ago
  47. 7f952f3 Fix webrtcsdp_unittest. by ehmaldonado · 8 years ago
  48. c0ff88b Delete MediaController class, move Call ownership to PeerConnection. by nisse · 8 years ago
  49. f26202b Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. by ossu · 8 years ago
  50. 45c4cdc Run some peer connection end-to-end tests with an empty audio decoder factory by kwiberg · 8 years ago
  51. 48cf9c8 Fix comment about remote restart being requested in createOffer by philipp.hancke · 8 years ago
  52. 4a5e818 Enabling 'gn check' on webrtc/video. by mbonadei · 8 years ago
  53. b5a7da7 Move ready to send logic from BaseChannel to RtpTransport. by zstein · 8 years ago
  54. ad4ba26 Negotiate the same SRTP crypto suites for every DTLS association formed. by deadbeef · 8 years ago
  55. cee9a93 Add "ice-option:trickle" to generated offers/answers. by deadbeef · 8 years ago
  56. 73c0f2a Enabling 'gn check' on webrtc/sdk by mbonadei · 8 years ago
  57. ef476cc Test CreatePeerConnectionFactory() with a forwarding mock AudioDecoderFactory by kwiberg · 8 years ago
  58. 17d1c10 Adding integration test for unsignaled inbound RTP stream stats. by deadbeef · 8 years ago
  59. ff493f2 Fix SDP stream ID mismatch issue when a track's stream changes. by deadbeef · 8 years ago
  60. acebebe Make RtpTransport actually implement RtpTransportInterface by zstein · 8 years ago
  61. f4509f8 Removing unnecessary parameters from initializeAndroidGlobals. by deadbeef · 8 years ago
  62. 69579d7 Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ ) by hbos · 8 years ago
  63. fee642e Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ ) by olka · 8 years ago
  64. 04834ca Fix compilation issues of std::unique_ptr by steweg · 8 years ago
  65. b5eeeee Added the GetSources() to the RtpReceiverInterface and implemented by zhihuang · 8 years ago
  66. 138c150 Injectable audio encoders: BuiltinAudioEncoderFactory by ossu · 8 years ago
  67. 9b1cf9e Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ ) by ilnik · 8 years ago
  68. 91673d1 Add a minimal RtpTransport class for use by BaseChannel. by zstein · 8 years ago
  69. 6dcff13 Fixing some case-sensitive codec name comparisons. by deadbeef · 8 years ago
  70. bbca310 Revert of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #8 id:140001 of https://codereview.webrtc.org/2780943003/ ) by guidou · 8 years ago
  71. 71dd2e5 Move video_encoder.h and video_decoder.h to /api and create GN targets for them by ilnik · 8 years ago
  72. cd05a9e Rewrite PeerConnection integration tests using better testing practices. by deadbeef · 8 years ago
  73. 31279bc Change minimum DTMF event duration to be 40 milliseconds by dminor · 8 years ago
  74. cd95bda Pass ownership of candidate to PeerConnection::OnIceCandidate by jbauch · 8 years ago
  75. 88b0869 Fix the fuzz test. by zhihuang · 8 years ago
  76. d0da0c0 Parse the connection data in SDP (c= line). by zhihuang · 8 years ago
  77. d33e561 RTCStatsCollector: Remove closed channels from opened set. by hbos · 8 years ago
  78. 6c5c54e Making candidate pool size behave as decided in JSEP. by deadbeef · 8 years ago
  79. 236b5f7 Delete deprecated PeerConnection methods, and corresponding using declarations. by nisse · 8 years ago
  80. b7b48ae Add the option to disable IPv6 ICE candidates on WiFi. by zhihuang · 8 years ago
  81. d9a29f6 Rename PeerConnection::OnIceConnectionChange to OnIceConnectionStateChange by zstein · 8 years ago
  82. 97befe8 Remove HAVE_SRTP define and unmaintained code. by jbauch · 8 years ago
  83. 2a4557d Reland: Improve testing of SRTP external auth code paths. by jbauch · 8 years ago
  84. c716659 Revert of Improve testing of SRTP external auth code paths. (patchset #2 id:20001 of https://codereview.webrtc.org/2722423003/ ) by jbauch · 8 years ago
  85. 4e9904b Improve testing of SRTP external auth code paths. by jbauch · 8 years ago
  86. 125c0f3 Support GCM ciphers even if ENABLE_EXTERNAL_AUTH is defined. by jbauch · 8 years ago
  87. 27e573e Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect. by hbos · 8 years ago
  88. 8ab9f62 Test field trial group with startswith rather than equals. by sprang · 8 years ago
  89. 95aa50b Rename RTCCodecStats.codec -> mimeType, parameters -> sdpFmtpLine. by hbos · 8 years ago
  90. c9b58ba Make hbos and hta rtcstats* OWNERS of webrtc/pc, not webrtc/api. by hbos · 8 years ago
  91. fca0fe0 RTCIceCandidatePairStats.[total/current]RoundTripTime collected. by hbos · 8 years ago
  92. 3528419 Fix TSAN race in webrtc::voe::Channel. by hbos · 8 years ago
  93. cf2b8c1 RTCIceCandidatePairStats.nominated collected. by hbos · 8 years ago
  94. f17d926 Disable flaky tests on iOS by kjellander · 8 years ago
  95. cd91bae Roll chromium_revision 33a7a547b9..0e44c5e141 (452838:453130) by kjellander · 8 years ago
  96. ad5d1e6 Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc. by deadbeef · 8 years ago
  97. 60350d4 Fix occasional race in VideoCapturerTrackSource seen by memcheck bot. by deadbeef · 8 years ago
  98. 47f0111 Rewrite rtc::Bind using variadic templates. by deadbeef · 8 years ago
  99. 1f3c4f8 Disable RTCStatsIntegrationTest on TSAN bots. by philipel · 8 years ago
  100. e2628ce Accept SDP with TRANSPORT attributes missing from bundled m= sections. by deadbeef · 8 years ago