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gerrit-public.fairphone.software / platform / external / webrtc / 10542f21c8e4e2d60b136fab45338f2b1e132dde / pc / media_session.cc
  1. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  2. 1c05765 (3) Rename files to snake_case: move the files by Steve Anton · 6 years ago[Renamed from pc/mediasession.cc]
  3. 2bed397 Support changing the tagged BUNDLE media section section by Steve Anton · 6 years ago
  4. c63ddb2 Negotiating Simulcast in the initial offer/answer - Part1. by Amit Hilbuch · 6 years ago
  5. 77938e6 Simulcast work to enable RID mux. by Amit Hilbuch · 6 years ago
  6. 31d8b52 Delete unneeded includes of rtc_base/stringutils.h. by Niels Möller · 6 years ago
  7. 06817cd [Unified Plan] Support legacy endpoints that do not use a=mid by Steve Anton · 6 years ago
  8. 6fe1fba Convert MediaSessionFactory to return unique_ptrs by Steve Anton · 6 years ago
  9. 1a9d3c3 Convert TransportDescriptionFactory to return unique_ptrs by Steve Anton · 6 years ago
  10. 8f66ddb Move is_unified_plan flag to a member variable by Steve Anton · 6 years ago
  11. 5c72e71 [Unified Plan] Fix issues with recycling m= sections by Steve Anton · 6 years ago
  12. 89f874e Add offer_extmap_allow_mixed to RTCConfiguration by Johannes Kron · 6 years ago
  13. 039743e Reland "Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase" by Niels Möller · 6 years ago
  14. 6e8e299 Revert "Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase" by Oleh Prypin · 6 years ago
  15. 80cd25b Delete CodecNamesEq, replaced with absl::EqualsIgnoreCase by Niels Möller · 6 years ago
  16. 9581bc4 Rename too long variable name to extmap_allow_mixed by Johannes Kron · 6 years ago
  17. 2edab4c Delete use of STR_CASE_CMP, replaced with absl::EqualsIgnoreCase. by Niels Möller · 6 years ago
  18. 9ac3c91 Refactor of extmap-allow-mixed in SessionDescription by Johannes Kron · 6 years ago
  19. a54daf1 Reland "Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h" by Benjamin Wright · 6 years ago
  20. 8f4bc41 Revert "Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h" by Oleh Prypin · 6 years ago
  21. 1cd39fa make CreateOffer/CreateAnswer use ice credentials of pooled sessions. by Jonas Oreland · 6 years ago
  22. ac2f3d1 Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h by Benjamin Wright · 6 years ago
  23. 0854eb6 Respond to SDP request extmap-allow-mixed. by Johannes Kron · 6 years ago
  24. 07ba2b9 Parse two-byte header extensions. by Johannes Kron · 6 years ago
  25. 3a66edf Use C++11 for range loop in pc/mediasession.cc (and test) by Steve Anton · 6 years ago
  26. a76af0c Move base64.h to the proper location. by Artem Titov · 6 years ago
  27. 66cadcc Replace rtc::Optional with absl::optional in pc by Danil Chapovalov · 6 years ago
  28. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  29. 2199580 Remove deprecated std::bind2nd and std::ptr_fun by tzik · 7 years ago
  30. 0ab5651 Fix handling of empty BUNDLE groups. by Taylor Brandstetter · 7 years ago
  31. 1b8773d Negotiate the MID header extension for Unified Plan by Steve Anton · 7 years ago
  32. 5b4f075 Reland "Reland "Adds support for multiple or no media stream ids."" by Seth Hampson · 7 years ago
  33. 191bf5c Revert "Reland "Adds support for multiple or no media stream ids."" by Tomas Gunnarsson · 7 years ago
  34. f351c34 Reland "Adds support for multiple or no media stream ids." by Seth Hampson · 7 years ago
  35. bc609ea Revert "Adds support for multiple or no media stream ids." by Emircan Uysaler · 7 years ago
  36. 1550292 Adds support for multiple or no media stream ids. by Seth Hampson · 7 years ago
  37. 5e55fe8 Adding flag to enable/disable use of SRTP_AES128_CM_SHA1_32 crypto suite. by Taylor Brandstetter · 7 years ago
  38. 845e878 Name change from stream label to stream id for spec compliance. by Seth Hampson · 7 years ago
  39. 5a26a3a Remove public sync_label from StreamParams by Steve Anton · 7 years ago
  40. 45cc890 Assorted logging pedantry by Jonas Olsson · 7 years ago
  41. 8e545ee Revert "Use SRTP_AES128_CM_SHA1_80 by default instead of SRTP_AES128_CM_SHA1_32." by Tommi · 7 years ago
  42. 6780c51 Use SRTP_AES128_CM_SHA1_80 by default instead of SRTP_AES128_CM_SHA1_32. by Joachim Bauch · 7 years ago
  43. e831b8c Add MSID signaling compatibility for Unified Plan endpoints by Steve Anton · 7 years ago
  44. 3518e7b Add the rejected TransportInfo when creating an answer. by Zhi Huang · 7 years ago
  45. ad7bffc Parameterize PeerConnection media tests for Unified Plan by Steve Anton · 7 years ago
  46. fa2260d Add support for data channels with Unified Plan by Steve Anton · 7 years ago
  47. dcc3c02 Add support for JSEP offer/answer with transceivers by Steve Anton · 7 years ago
  48. b1c1de1 Use the SDP ContentInfo helpers to avoid downcasting by Steve Anton · 7 years ago
  49. 5adfafd Make ContentInfo/ContentDescription slightly more ergonomic by Steve Anton · 7 years ago
  50. 5634427 Remove unused properties from MediaContentDescription by Steve Anton · 7 years ago
  51. 4e70a72 Replace MediaContentDirection with RtpTransceiverDirection by Steve Anton · 7 years ago
  52. 1d03a75 Remove cricket::RtpTransceiverDirection by Steve Anton · 7 years ago
  53. 6f36747 Use local codec parameters in the answer. by Zhi Huang · 7 years ago
  54. 7aee3d5 Fix ortc_api circular deps. by Patrik Höglund · 7 years ago
  55. 675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 7 years ago
  56. 36b29d1 Enable cpplint in pc/ by Steve Anton · 7 years ago
  57. 80cfb52 RTC_CHECK'ing content type before static_casting descriptions. by Taylor Brandstetter · 7 years ago
  58. 1c34974 Fixing invalid calls to FindMatchingCodec. by Taylor Brandstetter · 7 years ago
  59. 7120742 Adding NOLINT for typedefs.h and common_types.h by Mirko Bonadei · 7 years ago
  60. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  61. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/pc/mediasession.cc]
  62. 8ffb9c3 Change RtpSender to have multiple stream_ids by Steve Anton · 7 years ago
  63. 84f6a3f Move optional.h to webrtc/api/ by kwiberg · 7 years ago
  64. 1c378ed Relanding: Adding support for Unified Plan offer/answer negotiation to the mediasession layer. by zhihuang · 7 years ago
  65. 3c74766 Revert of Adding support for Unified Plan offer/answer negotiation. (patchset #9 id:500001 of https://codereview.webrtc.org/2991693002/ ) by olka · 7 years ago
  66. a77e6bb Adding support for Unified Plan offer/answer negotiation to the mediasession layer. by zhihuang · 7 years ago
  67. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  68. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  69. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  70. 5869f50 Support encrypted RTP extensions (RFC 6904) by jbauch · 7 years ago
  71. 38ede13 Support building WebRTC without audio and video. by zhihuang · 7 years ago
  72. 8b7e9ad Support "UDP/DTLS/SCTP" and "TCP/DTLS/SCTP" profile strings. by deadbeef · 7 years ago
  73. 7914b8c Negotiate the same SRTP crypto suites for every DTLS association formed. by deadbeef · 8 years ago
  74. 2f425aa Fix SDP stream ID mismatch issue when a track's stream changes. by deadbeef · 8 years ago
  75. eaa9c1d Remove HAVE_SRTP define and unmaintained code. by jbauch · 8 years ago
  76. e814a0d Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc. by deadbeef · 8 years ago
  77. b789253 Accept SDP with TRANSPORT attributes missing from bundled m= sections. by deadbeef · 8 years ago
  78. 21e4e0b Delete webrtc/base/common.h by nisse · 8 years ago
  79. 4b2e082 Use the same draft version in SDP data channel answers as used in the offer. by zstein · 8 years ago
  80. c16fa5e Replace all use of the VERIFY macro. by nisse · 8 years ago
  81. 7bb87ee Create //webrtc/api:libjingle_peerconnection_api + refactorings. by ossu · 8 years ago
  82. ede5da4 Replace ASSERT by RTC_DCHECK in all non-test code. by nisse · 8 years ago
  83. c80e741 Replace ASSERT(false) by RTC_NOTREACHED(). by nisse · 8 years ago
  84. 7af91dd Removing "crypto_required" from MediaContentDescription. by deadbeef · 8 years ago
  85. 352444f RTC_[D]CHECK_op: Remove superfluous casts by kwiberg · 8 years ago
  86. 03d5fb1 Let MediaSession generate a FlexFEC SSRC when FlexFEC is active. by brandtr · 8 years ago
  87. f823ede Negotiate H264 profiles in SDP by magjed · 8 years ago
  88. b05fa24 Optimize FindCodecById and ReferencedCodecsMatch by magjed · 8 years ago
  89. 3cf8ece Revert of Stop caching supported codecs in WebRtcVideoEngine2 (patchset #1 id:1 of https://codereview.webrtc.org/2492473002/ ) by magjed · 8 years ago
  90. 9f71ec5 Stop caching supported codecs in WebRtcVideoEngine2 by magjed · 8 years ago
  91. 9fa4975 - Filter data channel codecs based on codec name instead of payload type, which may have been remapped. by solenberg · 8 years ago
  92. 4cedf2b Add signaling to support ICE renomination. by Honghai Zhang · 8 years ago
  93. 1d7a637 Fixing off-by-one error with max SCTP id. by Taylor Brandstetter · 8 years ago
  94. cb56065 Add support for GCM cipher suites from RFC 7714. by jbauch · 8 years ago
  95. dedfd28 Support for two audio codec lists down into WebRtcVoiceEngine. by ossu · 8 years ago
  96. 075af92 Initial asymmetric codec support in MediaSessionDescription by ossu · 8 years ago
  97. 6f8d686 Remove use of RtpHeaderExtension and clean up by isheriff · 8 years ago
  98. dc4eb8c Refactoring some tests in peerconnectioninterface_unittest.cc. by Taylor Brandstetter · 8 years ago
  99. 8f65cdf Only generate one CNAME per PeerConnection. by zhihuang · 8 years ago
  100. cf5b37c Accept all the media profiles required by JSEP. by zhihuang · 9 years ago