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webrtc / src / webrtc / 03e0f87478e99d4afbc416b2df8915c06a41dce1 / pc
  1. 3f3e7f8 In PC integration tests, create tracks/streams with random IDs. by deadbeef · 8 years ago
  2. 0abc754 Change RtpSender to have multiple stream_ids by Steve Anton · 8 years ago
  3. bb3bbd7 Add a PeerConnection integration test for adding an audio track mid-call by deadbeef · 8 years ago
  4. efa3277 Thread-checkers for PeerConnectionFactory::worker_thread_ by eladalon · 8 years ago
  5. 6ae8262 Add reporting of googContentType via GetStats on send side by ilnik · 8 years ago
  6. 05f793e Reland of Make RtcEventLogImpl to use a TaskQueue instead of a helper-thread ( https://codereview.webrtc.org/3007473002/ ) by eladalon · 8 years ago
  7. 5b18967 Move optional.h to webrtc/api/ by kwiberg · 8 years ago
  8. 17ced3f Revert of Make RtcEventLogImpl to use a TaskQueue instead of a helper-thread (patchset #4 id:200001 of https://codereview.webrtc.org/3005153002/ ) by eladalon · 8 years ago
  9. 5fbad34 Reland of Make RtcEventLogImpl to use a TaskQueue instead of a helper-thread (patchset #1 id:1 of https://codereview.webrtc.org/3010143002/ ) by eladalon · 8 years ago
  10. 753e08c Avoid construction of unused RtcEventLogNullImpl object by eladalon · 8 years ago
  11. 1e08e04 Reland of Trace the stats report as JSON instead of each stat separately. (patchset #1 id:1 of https://codereview.webrtc.org/3001683002/ ) by ehmaldonado · 8 years ago
  12. 1f11d1a Implement googContentType GetStats metric reported on receive side. by ilnik · 8 years ago
  13. 23aa43d Move array_view.h to webrtc/api/ by kwiberg · 8 years ago
  14. 36189cd Move RtpExtension to api/ directory and config.h/.cc to call/. by Stefan Holmer · 8 years ago
  15. 01599fe Completed the functionalities of SrtpTransport. by zhihuang · 8 years ago
  16. d25b84c Removes unused APIs from the ADM (part II). by henrika · 8 years ago
  17. beb8e45 Change WebRtcSession to have a vector of channels by Steve Anton · 8 years ago
  18. 710df7d Change ChannelManager to use unique_ptr by Steve Anton · 8 years ago
  19. 32612eb Fix RTCP transport not destroyed when channel creation fails by Steve Anton · 8 years ago
  20. 00ed864 Recently we moved webrtc/base to webrtc/rtc_base, so these by mbonadei · 8 years ago
  21. 7a02da1 Fix the Chromium crash when creating an answer without a remote description. by zhihuang · 8 years ago
  22. adaf22f Support a user-provided string for the TLS ALPN extension. by Diogo Real · 8 years ago
  23. 320b4cd Removes unused WaveOut APIs from ADM. by henrika · 8 years ago
  24. 94c183c Revert of Add logging of host lookups made by TurnPort to the RtcEventLog. (patchset #11 id:200001 of https://codereview.webrtc.org/2996933003/ ) by maxmorin · 8 years ago
  25. 4282b1f Add logging host lookups made by TurnPort to the RtcEventLog. by jonaso · 8 years ago
  26. 443f9a9 Wiring discard rate of audio FEC/RED packets up to StatsReport. by minyue-webrtc · 8 years ago
  27. 94ac82f Add stats totalSamplesReceived and concealedSamples by Steve Anton · 8 years ago
  28. 775b7c5 Fix places that trigger no-unused-lambda-capture - change to using static-constexpr. by eladalon · 8 years ago
  29. 37f8e0d Delete unneeded include of videocapturer.h by Niels Möller · 8 years ago
  30. 1329144 Use fake audio device in peerconnectioninterface_unittest.cc. by deadbeef · 8 years ago
  31. 6a5ac8f Report max interframe delay over window insdead of interframe delay sum by ilnik · 8 years ago
  32. 7d8db23 Fix places that trigger no-unused-lambda-capture by eladalon · 8 years ago
  33. 4e4b0a2 Relanding: Adding support for Unified Plan offer/answer negotiation to the mediasession layer. by zhihuang · 8 years ago
  34. b38cd1f Revert of Adding support for Unified Plan offer/answer negotiation. (patchset #9 id:500001 of https://codereview.webrtc.org/2991693002/ ) by olka · 8 years ago
  35. c65a896 Revert of Trace the stats report as JSON instead of each stat separately. (patchset #3 id:100001 of https://codereview.webrtc.org/2986453002/ ) by mbonadei · 8 years ago
  36. 0947a65 Adding support for Unified Plan offer/answer negotiation to the mediasession layer. by zhihuang · 8 years ago
  37. 273a2ea Trace the stats report as JSON instead of each stat separately. by ehmaldonado · 8 years ago
  38. ea15d68 Replace CHECK(x && y) with two separate CHECK() calls by kwiberg · 8 years ago
  39. 6ab33cf Wire up RTP keep-alive in ortc api. by sprang · 8 years ago
  40. 7db8343 Ignore "b=AS:-1" instead of treating as a hard error. by deadbeef · 8 years ago
  41. 58b7d4e Reject negative values for "b=AS". by deadbeef · 8 years ago
  42. cf8bdaf Reland of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt by perkj · 8 years ago
  43. 3b5bf18 Make RTCStatsReport::ToString() return JSON-parseable string. by ehmaldonado · 8 years ago
  44. ac6511e Remove libsrtp 2.0.0 compatibility code. by jbauch · 8 years ago
  45. d29f8b0 Add "max_ipv6_networks" field to RTCConfiguration. by deadbeef · 8 years ago
  46. fdad635 When a track is added/removed directly to MediaStream notify observer->OnRenegotionNeeded by korniltsev.anatoly · 8 years ago
  47. 17741a4 Introduce RtpTransportInternal and SrtpTransport. by zstein · 8 years ago
  48. 2bc62f3 Remove remains of webrtc/base by ehmaldonado · 8 years ago
  49. f0633bf Enable tracing on rtcstats_integrationtest.cc by ehmaldonado · 8 years ago
  50. ed15115 Increase the size of the buffer for type.name.id. by ehmaldonado · 8 years ago
  51. ae14476 Trace stats in RTCStatsCollector. by ehmaldonado · 8 years ago
  52. 85d0c05 Reinstate "Add additional check when setting RTCConfiguration" by Steve Anton · 8 years ago
  53. fbd3f7b Reinstate "API for periodically regathering ICE candidates" by Steve Anton · 8 years ago
  54. a0d6f39 Reland of Make the default ctor of rtc::Thread, protected by tommi · 8 years ago
  55. 90809b6 SignalPacketReceived should pass packet as a pointer instead of a non-const reference. by zstein · 8 years ago
  56. d1701d0 Adding stats that can be used to compute output audio levels as described here https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy. by zstein · 8 years ago
  57. beaccf9 Revert "API for periodically regathering ICE candidates" by Magnus Jedvert · 8 years ago
  58. 3c277e7 Revert "Add additional check when setting RTCConfiguration" by Magnus Jedvert · 8 years ago
  59. 03ba3a6 Add additional check when setting RTCConfiguration by Steve Anton · 8 years ago
  60. 40bb491 Revert of Make the default ctor of rtc::Thread, protected (patchset #3 id:40001 of https://codereview.webrtc.org/2981623002/ ) by charujain · 8 years ago
  61. 0924ae9 Make the default ctor of rtc::Thread, protected. by tommi · 8 years ago
  62. 4685ae4 RTCStatsCollector: Get track IDs from senders/receivers instead of streams. by hbos · 8 years ago
  63. faf0fff Make BaseChannel::rtp_transport_ a unique_ptr. by zstein · 8 years ago
  64. 13e59ec API for periodically regathering ICE candidates by Steve Anton · 8 years ago
  65. 59d5575 Use relative paths in GN files. by jianjun.zhu · 8 years ago
  66. 61b128a Don't call CreateDtlsTransport_n from non-network thread in WebRtcSession by deadbeef · 8 years ago
  67. 4f870fc Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ ) by ehmaldonado · 8 years ago
  68. c6c814d Remove remains of webrtc/base by ehmaldonado · 8 years ago
  69. 7b2b061 Revert of Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on wt (patchset #2 id:20001 of https://codereview.webrtc.org/2964863002/ ) by mbonadei · 8 years ago
  70. 90ae41e Change VideoTrack implementation to invoke VideoTrackSourceInterface::AddOrUpdateSink on the worker thread. by perkj · 8 years ago
  71. eb0ab39 Move SrtpSession and tests to their own files. by zstein · 8 years ago
  72. 4b941f0 Report interframe delay sum in old GetStats by ilnik · 8 years ago
  73. 76de83e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  74. 37342b9 Report timing frames info in GetStats. by ilnik · 8 years ago
  75. bc32410 Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  76. 60154fd Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  77. 39f7f7a Enable the injection of an APM into a peerconnection by peah · 8 years ago
  78. 56f7815 Enable -Wunused-function warning everywhere. by Henrik Kjellander · 8 years ago
  79. 150ba79 Fix -Wcomment warning in webrtcsdp.cc by kjellander · 8 years ago
  80. 0d58090 Support encrypted RTP extensions (RFC 6904) by jbauch · 8 years ago
  81. 588f761 Allow an external audio processing module to be used in WebRTC by peah · 8 years ago
  82. 0214bf4 Fixing RTCIceCandidatePairStats.nominated for ICE controlling agent. by deadbeef · 8 years ago
  83. 6efbd3c Support getting external HMAC auth context with libsrtp 2.1.0. by jbauch · 8 years ago
  84. 51a0b43 Remove unused "crypto_options_" field. by jbauch · 8 years ago
  85. db65e09 Fix the binary size regression on Chromium Windows. by zhihuang · 8 years ago
  86. 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
  87. 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
  88. d901125 Try to fix the binary size increase issue on Chromium. by zhihuang · 8 years ago
  89. 3943e59 Fix uploading of available send bitrate statistics. by Alex Narest · 8 years ago
  90. 10a13e1 Fixing incorrect use of erase/remove idiom. by deadbeef · 8 years ago
  91. 8c18d91 Enable SNI in ssl adapter. by Emad Omara · 8 years ago
  92. 5e343ec Delete SignalSrtpError. by nisse · 8 years ago
  93. f91805c Support building WebRTC without audio and video. by zhihuang · 8 years ago
  94. f94a820 Allow WebRtcMediaEngine to be created from any thread. by deadbeef · 8 years ago
  95. 5aaa670 Fix Chromium style checker warnings for MockAudioDecoder by kwiberg · 8 years ago
  96. 8f7d284 Remove DCHECK from PeerConnectionFactory::worker_thread. by zstein · 8 years ago
  97. 3d545a2 s/WebRtcVideoChannel2/WebRtcVideoChannel and s/WebRtcVideoEngine2/WebRtcVideoEngine by eladalon · 8 years ago
  98. 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
  99. fb8fb2d Revert of Add PeerConnectionInterface::UpdateCallBitrate. (patchset #7 id:120001 of https://codereview.webrtc.org/2888303005/ ) by charujain · 8 years ago
  100. 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