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gerrit-public.fairphone.software / platform / external / webrtc / fae640003cf028477819309777ecd29a74c25eff / pc / srtp_filter.cc
  1. a4d8737 Format almost everything. by Jonas Olsson · 5 years ago
  2. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  3. 1c05765 (3) Rename files to snake_case: move the files by Steve Anton · 6 years ago[Renamed from pc/srtpfilter.cc]
  4. 3e70781 [Cleanup] Add missing #include. Remove useless ones. IWYU part 2. by Yves Gerey · 6 years ago
  5. 26968ba Delete unused utf8 conversion utilities by Niels Möller · 6 years ago
  6. a76af0c Move base64.h to the proper location. by Artem Titov · 6 years ago
  7. 66cadcc Replace rtc::Optional with absl::optional in pc by Danil Chapovalov · 6 years ago
  8. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  9. 5b32f23 Securely clear memory containing key information / passwords before freeing. by Joachim Bauch · 6 years ago
  10. e818b6e Create the JsepTransportController and JsepTransport2. by Zhi Huang · 6 years ago
  11. 45cc890 Assorted logging pedantry by Jonas Olsson · 6 years ago
  12. 36f8f3e Optional: Use nullopt and implicit construction in /pc by Oskar Sundbom · 7 years ago
  13. 675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 7 years ago
  14. b140b9f Keep count of libsrtp clients, and only deinitialize when it goes to 0. by Taylor Brandstetter · 7 years ago
  15. cf990f5 Reland: Completed the functionalities of SrtpTransport. by Zhi Huang · 7 years ago
  16. eb23e17 Revert of Completed the functionalities of SrtpTransport. (patchset 7 id:320001 of https://codereview.webrtc.org/2997983002/ ) by zhihuang · 7 years ago
  17. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  18. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/pc/srtpfilter.cc]
  19. e683c68 Completed the functionalities of SrtpTransport. by zhihuang · 7 years ago
  20. 4dde3df Move SrtpSession and tests to their own files. by zstein · 7 years ago
  21. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  22. a80c16a Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)" by Henrik Kjellander · 7 years ago
  23. c3771cc Update includes for webrtc/{base => rtc_base} rename (2/3) by kjellander · 7 years ago
  24. 5869f50 Support encrypted RTP extensions (RFC 6904) by jbauch · 7 years ago
  25. 03fa534 Support getting external HMAC auth context with libsrtp 2.1.0. by jbauch · 7 years ago
  26. af99b6d Delete SignalSrtpError. by nisse · 7 years ago
  27. eaa9c1d Remove HAVE_SRTP define and unmaintained code. by jbauch · 7 years ago
  28. dfcab72 Reland: Improve testing of SRTP external auth code paths. by jbauch · 7 years ago
  29. d81f121 Revert of Improve testing of SRTP external auth code paths. (patchset #2 id:20001 of https://codereview.webrtc.org/2722423003/ ) by jbauch · 7 years ago
  30. ac170d5 Improve testing of SRTP external auth code paths. by jbauch · 7 years ago
  31. d48f488 Support GCM ciphers even if ENABLE_EXTERNAL_AUTH is defined. by jbauch · 7 years ago
  32. 7d25426 Delete unneeded includes of base/common.h. by nisse · 8 years ago
  33. 79e0588 Set actual transport overhead in rtp_rtcp by michaelt · 8 years ago
  34. 0d8ade5 Remove remnants of libsrtp1 by mattdr · 8 years ago
  35. 8ff52cc Remove useless debugging code by mattdr · 8 years ago
  36. 51f2919 Update WebRTC to build against libsrtp 2.0 by mattdr · 8 years ago
  37. cb56065 Add support for GCM cipher suites from RFC 7714. by jbauch · 8 years ago
  38. 37bb54e Reland: Remove global list of SRTP sessions. by Joachim Bauch · 8 years ago
  39. 82d7862 Change default timestamp to 64 bits in all webrtc directories. by Honghai Zhang · 8 years ago
  40. 7bc7c06 Revert of Remove the rtc_relative_path GYP variable and similar defines (patchset #1 id:1 of https://codereview.webrtc.org/1903553003/ ) by kjellander · 8 years ago
  41. e19cf59 Remove the rtc_relative_path GYP variable and similar defines by kjellander · 8 years ago
  42. b252856 Remove all uses of the HAVE_CONFIG_H define. by Henrik Kjellander · 8 years ago
  43. 65c7f67 Fix license headers in webrtc/pc by kjellander · 8 years ago
  44. 9b8df25 Move talk/session/media -> webrtc/pc by kjellander@webrtc.org · 8 years ago[Renamed (99%) from talk/session/media/srtpfilter.cc]
  45. a96e2d7 Move talk/media to webrtc/media by kjellander · 8 years ago
  46. 77fa59d Fix build break in google3 import caused by https://codereview.webrtc.org/1532543003 by guoweis · 9 years ago
  47. 4638331 DTLS-SRTP set up is bypassed when the channel has been writable. by guoweis · 9 years ago
  48. 46ad542 Revert of "Create rtc::AtomicInt POD struct." (patchset #3 id:40001 of https://codereview.webrtc.org/1498953002/ ) by pbos · 9 years ago
  49. 84f0970 Reland of "Create rtc::AtomicInt POD struct." by Peter Boström · 9 years ago
  50. 521ed7b Reland Convert internal representation of Srtp cryptos from string to int by Guo-wei Shieh · 9 years ago
  51. 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
  52. 2764e10 Convert internal representation of Srtp cryptos from string to int. by guoweis · 9 years ago
  53. cbe9f51 Revert of Remove global list of SRTP sessions. (patchset #4 id:60001 of https://codereview.webrtc.org/1416093010/ ) by phoglund · 9 years ago
  54. 9cafd97 Remove global list of SRTP sessions. by jbauch · 9 years ago
  55. ff134eb talk: Use NDEBUG macro. by tfarina · 9 years ago
  56. 0c4e06b Use suffixed {uint,int}{8,16,32,64}_t types. by Peter Boström · 9 years ago
  57. 456696a Reland Change WebRTC SslCipher to be exposed as number only by Guo-wei Shieh · 9 years ago
  58. 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
  59. 4fe3c9a Change WebRTC SslCipher to be exposed as number only. by guoweis · 9 years ago
  60. e70028e Protect access to shared list of SRTP sessions. by Joachim Bauch · 9 years ago
  61. fec2c6d Prevent potential double-free if srtp_create fails. by Joachim Bauch · 9 years ago
  62. 469c2c0 Make Config::default_value leak instead of having an exit-time destructor. by Andrew MacDonald · 9 years ago
  63. 9478437 rtc::Buffer improvements by Karl Wiberg · 9 years ago
  64. 4b3c0d6 Use WebRTC API to convert byteorder in srtpfilter. by Jiayang Liu · 9 years ago
  65. a197a5e Update libsrtp includes in preparation of roll into Chromium. by jiayl@webrtc.org · 9 years ago
  66. a9cf079 Rename external_hmac_ctx_t to ExternalHmacContext. by pbos@webrtc.org · 10 years ago
  67. 2e7ee4b Fix the SrtpFilter crash caused by two local offers. by pthatcher@webrtc.org · 10 years ago
  68. a09a999 (Auto)update libjingle 73222930-> 73226398 by buildbot@webrtc.org · 10 years ago
  69. d4e598d (Auto)update libjingle 72097588-> 72159069 by buildbot@webrtc.org · 10 years ago
  70. 0537634 (Auto)update libjingle 62713454-> 62865357 by henrike@webrtc.org · 10 years ago
  71. 371243d Remove std:: prefixes from C functions in talk/. by pbos@webrtc.org · 10 years ago
  72. d43aa9d Update libjingle 61901702->61966318 by henrike@webrtc.org · 10 years ago
  73. a7b9818 Unrevert 5590 "description"(=(Auto)update libjingle 61834300->61901702). by henrike@webrtc.org · 10 years ago
  74. ef22151 Revert 5590 "description" by xians@webrtc.org · 10 years ago
  75. 2643805 description by henrike@webrtc.org · 10 years ago
  76. 9dba525 * Update libjingle to 50389769. by wu@webrtc.org · 11 years ago
  77. 28e2075 Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk by henrike@webrtc.org · 11 years ago