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gerrit-public.fairphone.software / platform / external / webrtc / be2e5f78b3498858cd8694fa6d00a28ba8093f26 / pc / channel.cc
  1. be2e5f7 Make payload type demux conditional on media direction by Steve Anton · 5 years ago
  2. 6563934 Revert "Sanitize the codec list before sending it to the media engine" by Artem Titov · 5 years ago
  3. add7ef9 Sanitize the codec list before sending it to the media engine by Steve Anton · 5 years ago
  4. e1795f4 Adds remote estimate RTCP packet. by Sebastian Jansson · 5 years ago
  5. 479a3c0 Add support for enabling and negotiating raw RTP packetization. by Mirta Dvornicic · 6 years ago
  6. 3a1b927 Remove rtp_ and rtcp_packet_transport() from the RtpTransport interface. by Bjorn A Mellem · 6 years ago
  7. 4f08faa Introduce MediaTransportConfig by Anton Sukhanov · 6 years ago
  8. 5fc28b1 Reland "Reland "Version 2 "Refactoring DataContentDescription class""" by Harald Alvestrand · 6 years ago
  9. 46afbf9 Revert "Reland "Version 2 "Refactoring DataContentDescription class""" by Steve Anton · 6 years ago
  10. 37f2b43 Reland "Version 2 "Refactoring DataContentDescription class"" by Harald Alvestrand · 6 years ago
  11. 141c0ad Revert "Version 2 "Refactoring DataContentDescription class"" by Harald Alvestrand · 6 years ago
  12. 14b2758 Version 2 "Refactoring DataContentDescription class" by Harald Alvestrand · 6 years ago
  13. edd2054 Minor fixes and refactoring for RtpTransport until the Demux. by Amit Hilbuch · 6 years ago
  14. e7a5f7b Modifying MediaChannel to accept CopyOnWriteBuffer by value. by Amit Hilbuch · 6 years ago
  15. fe6e50f Allow more than one registered network change callback in MediaTransport by Niels Möller · 6 years ago
  16. eee110d Remove nogncheck from pc/. by Mirko Bonadei · 6 years ago
  17. 64b626b Use Abseil container algorithms in pc/ by Steve Anton · 6 years ago
  18. bcd39d4 Creating Simulcast offer and answer in Peer Connection. by Amit Hilbuch · 6 years ago
  19. 309aafe Add 'AudioPacket' notification to media transport interface. by Piotr (Peter) Slatala · 6 years ago
  20. c2c733e Remove unused methods from cricket::BaseChannel. by Mirko Bonadei · 6 years ago
  21. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  22. 29e13fd Delete rtc::PacketTime (was an alias for int64_t) by Niels Möller · 6 years ago
  23. 5f8b5fd Use for range loop in pc/channel.cc by Steve Anton · 6 years ago
  24. 179a392 Implement TargetBitrate, NetworkRoute and overhead features of media transport interface. by Piotr (Peter) Slatala · 6 years ago
  25. dd9390c Prevent channels being set on stopped transceiver. by Amit Hilbuch · 6 years ago
  26. e693381 Delete struct rtc::PacketTime. by Niels Möller · 6 years ago
  27. 15ca5a9 Add implicit conversion between rtc:PacketTime and int64_t. by Niels Möller · 6 years ago
  28. 9190b82 Propagate SDP negotiation of extmap-allow-mixed to RtpHeaderExtensionMap by Johannes Kron · 6 years ago
  29. 98a462c Reland "Reland "Propagate media transport to media channel."" by Anton Sukhanov · 6 years ago
  30. 9accc9f Revert "Reland "Propagate media transport to media channel."" by Oleh Prypin · 6 years ago
  31. da65ed2 Reland "Propagate media transport to media channel." by Anton Sukhanov · 6 years ago
  32. 37cf245 Revert "Propagate media transport to media channel." by Oleh Prypin · 6 years ago
  33. 8c16f74 Propagate media transport to media channel. by Anton Sukhanov · 6 years ago
  34. a54daf1 Reland "Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h" by Benjamin Wright · 6 years ago
  35. 8f4bc41 Revert "Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h" by Oleh Prypin · 6 years ago
  36. ac2f3d1 Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h by Benjamin Wright · 6 years ago
  37. 366a50c Remove simple stringstream usages. by Jonas Olsson · 6 years ago
  38. ee01a83 Remove MetricsObserverInterface. by Qingsi Wang · 6 years ago
  39. 918f50c Use absl::make_unique and absl::WrapUnique directly by Karl Wiberg · 6 years ago
  40. 66cadcc Replace rtc::Optional with absl::optional in pc by Danil Chapovalov · 7 years ago
  41. 665174f Reformat the WebRTC code base by Yves Gerey · 7 years ago
  42. 365381f Replace BundleFilter with RtpDemuxer in RtpTransport. by Zhi Huang · 7 years ago
  43. 5897a6e Adds support for signaling a=msid lines without a=ssrc lines. by Seth Hampson · 7 years ago
  44. 0ffe03d Add Deinit() to the destructors of Voice/Video/RtpDataChannel. by Zhi Huang · 7 years ago
  45. e830e68 Use new TransportController implementation in PeerConnection. by Zhi Huang · 7 years ago
  46. 95e7dbb Revert "Reland "Replace BundleFilter with RtpDemuxer in RtpTransport."" by Zhi Huang · 7 years ago
  47. 27f3bf5 Reland "Replace BundleFilter with RtpDemuxer in RtpTransport." by Zhi Huang · 7 years ago
  48. 97d5e5b Revert "Replace BundleFilter with RtpDemuxer in RtpTransport." by Zhi Huang · 7 years ago
  49. ea8b62a Replace BundleFilter with RtpDemuxer in RtpTransport. by Zhi Huang · 7 years ago
  50. bb50ce5 Wire up MID send value to the PeerConnection API by Steve Anton · 7 years ago
  51. db67ba1 Report SRTP error codes to UMA by Steve Anton · 7 years ago
  52. 0807d15 Remove more dead code from BaseChannel by Steve Anton · 7 years ago
  53. f120cba Delete AudioMonitor and related code. by Niels Möller · 7 years ago
  54. 42805f3 Revert "Remove nogncheck and add proper dependencies." by Patrik Höglund · 7 years ago
  55. 9b045fa Remove nogncheck and add proper dependencies. by Patrik Höglund · 7 years ago
  56. e2a9318 Delete ConnectionMonitor. by Niels Möller · 7 years ago
  57. 0228485 Delete MediaMonitor. by Niels Möller · 7 years ago
  58. 053c1f8 Delete unused signal VoiceChannel::SignalAudioMonitor. by Niels Möller · 7 years ago
  59. 47136dd Change RtpSenders to interact with the media channel directly by Steve Anton · 7 years ago
  60. aaaf1cf Revert "Remove nogncheck and add proper dependencies." by Patrik Höglund · 7 years ago
  61. eefd543 Remove nogncheck and add proper dependencies. by Patrik Höglund · 7 years ago
  62. 6077675 Change RtpReceivers to interact with the media channel directly by Steve Anton · 7 years ago
  63. dc8b5ab Remove dead code for media channel errors by Steve Anton · 7 years ago
  64. b1c1de1 Use the SDP ContentInfo helpers to avoid downcasting by Steve Anton · 7 years ago
  65. 5634427 Remove unused properties from MediaContentDescription by Steve Anton · 7 years ago
  66. 3828c06 Replace cricket::ContentAction with webrtc::SdpType by Steve Anton · 7 years ago
  67. 2dfc42d Prepare to make BaseChannel depend on RtpTransportInternal only. by Zhi Huang · 7 years ago
  68. d745578 Call SrtpTransport::EnableExternalAuth when enabling SDES. by Zhi Huang · 7 years ago
  69. 2a4d70c Make the DtlsSrtpTransport cache the RtpAbsSendTimeHeaderExtension. by Zhi Huang · 7 years ago
  70. cd3fc5d Use the DtlsSrtpTransport in BaseChannel. by Zhi Huang · 7 years ago
  71. 4e70a72 Replace MediaContentDirection with RtpTransceiverDirection by Steve Anton · 7 years ago
  72. 36f8f3e Optional: Use nullopt and implicit construction in /pc by Oskar Sundbom · 7 years ago
  73. 1d88d74 Remove the unused code. by Zhi Huang · 7 years ago
  74. 801b868 Remove the CA_UPDATE and related code. by Zhi Huang · 7 years ago
  75. 5f5918f Merge MediaChannel's OnTransportOverheadChanged and OnNetworkRouteChanged callbacks. by Zhi Huang · 7 years ago
  76. 942bc2e Reland: Replaced the SignalSelectedCandidatePairChanged with a new signal. by Zhi Huang · 7 years ago
  77. 8c316c1 Revert "Replaced the SignalSelectedCandidatePairChanged with a new signal." by Zhi Huang · 7 years ago
  78. 7167745 Replaced the SignalSelectedCandidatePairChanged with a new signal. by Zhi Huang · 7 years ago
  79. c99b6c7 Remove the SetEncryptedHeaderExtensionIds methods. by Zhi Huang · 7 years ago
  80. 675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 7 years ago
  81. 8699a32 Have BaseChannel take MediaChannel as unique_ptr by Steve Anton · 7 years ago
  82. 04eaa15 Change the flag when RtpTransport objects send packet. by Zhi Huang · 7 years ago
  83. cf990f5 Reland: Completed the functionalities of SrtpTransport. by Zhi Huang · 7 years ago
  84. eb23e17 Revert of Completed the functionalities of SrtpTransport. (patchset 7 id:320001 of https://codereview.webrtc.org/2997983002/ ) by zhihuang · 7 years ago
  85. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  86. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/pc/channel.cc]
  87. 18ee1d5 Move SDP m= line matching from BaseChannel to WebRtcSession by Steve Anton · 7 years ago
  88. e683c68 Completed the functionalities of SrtpTransport. by zhihuang · 7 years ago
  89. 398c3fd Introduce RtpTransportInternal and SrtpTransport. by zstein · 7 years ago
  90. 634977b SignalPacketReceived should pass packet as a pointer instead of a non-const reference. by zstein · 7 years ago
  91. e8ab543 Make BaseChannel::rtp_transport_ a unique_ptr. by zstein · 7 years ago
  92. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  93. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 8 years ago
  94. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 8 years ago
  95. 5869f50 Support encrypted RTP extensions (RFC 6904) by jbauch · 8 years ago
  96. 38ede13 Support building WebRTC without audio and video. by zhihuang · 8 years ago
  97. f79ade1 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
  98. 3dcf0e9 Move RTP/RTCP demuxing logic from BaseChannel to RtpTransport. by zstein · 8 years ago
  99. d72098a 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
  100. e80f4c9 Wire up BWE stats through WebrtcSession so that they are filled in both for audio and video calls. by Stefan Holmer · 8 years ago