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chromium / chromium / src / fb33cbfb3b32a97bc0a152e57b4baedd98f5a07f / content / renderer / p2p
  1. 5206d4e6 media: Simplify MediaPermissionDispatcher. by xhwang · 10 years ago
  2. 07945f63 Convert Pass()→std::move() in //content/renderer by dcheng · 10 years ago
  3. 1023d01 Switch to standard integer types in content/renderer/. by avi · 10 years ago
  4. c563e69 Reland of Removing references to webrtc::PortAllocatorFactoryInterface. by deadbeef · 10 years ago
  5. ed698a7 Revert of Removing references to webrtc::PortAllocatorFactoryInterface. (patchset #6 id:100001 of https://codereview.chromium.org/1500663003/ ) by hans · 10 years ago
  6. ad868b5e Removing references to webrtc::PortAllocatorFactoryInterface. by deadbeef · 10 years ago
  7. 9ce7c97 enable test case for ipc_network_manager. by guoweis · 10 years ago
  8. 72d02d7 Filter out default address if it's not on the enumerated list. by guoweis · 10 years ago
  9. 4ee4859 Reenable test cases for IP policy handling. by guoweis · 10 years ago
  10. 6c1939b Wire up the flag which disallows default local candidate. by guoweis · 10 years ago
  11. a8c3d89 Add support for default local address in IpcNetworkManager by guoweis · 10 years ago
  12. 1f2ab3f2 IWYU for net_util.h included in header files. by tfarina · 10 years ago
  13. 0fbb02dc Wire up transport sequence number and send time. by holmer · 10 years ago
  14. 5ffbdf62 Add retail logging for WebRTC IP permission debugability. by guoweis · 10 years ago
  15. a1ccdf21 Checking permission before granting local IP breaks extensions or apps which use WebRTC and have already access to such information through chrome app api. The enforcement of not allowing UDP is also redundant since manifest has its own security model. by guoweis · 10 years ago
  16. 8c1b8568 Check media permissions (mic/camera) before exposing local addresses to WebRTC. by guoweis · 10 years ago
  17. 287b625 Add three missing includes. by grunell · 10 years ago
  18. 5b380551 Simple variable renaming: remove transport. by guoweis · 10 years ago
  19. 80a44be Stop including network_interfaces.h from net_util.h by olli.raula · 10 years ago
  20. a84ace48 Create a new preference which disables UDP protocol for WebRTC. The preference will be used in an extension. The extension API CL will follow. by guoweis · 10 years ago
  21. c289dd4 Implement WebRTC Stun ping trial in Chrome renderer process. by guoweis · 10 years ago
  22. c9a6b72 Stop including ip_address_number.h from net_util.h by eroman · 10 years ago
  23. c66f8919 IpcPacketSocket returns incorrect value when not opened by guoweis · 10 years ago
  24. 236d317 Replace more ObserverList with base::ObserverList. by brettw · 10 years ago
  25. 2d3b5bd content/renderer: Remove use of MessageLoopProxy and deprecated MessageLoop APIs by skyostil · 10 years ago
  26. 12262cf content/child: Remove use of MessageLoopProxy and deprecated MessageLoop APIs by skyostil · 10 years ago
  27. b0201f5 Remove the same function definition and reuse the same one from WebRTC. by guoweis · 10 years ago
  28. b3b993a Enables passing Origin value with WebRTC MediaConstraints (flag controlled). by pthatcher · 10 years ago
  29. 5f066054 Only allow non-deprecated IPv6 addresses which don't contain MAC to be used in WebRTC. by guoweis · 10 years ago
  30. 30816d9 Only allow temporary IPv6 address. by guoweis · 10 years ago
  31. 7c98bab0 Add a Preference to allow WebRTC only bind to "any address" (all 0s). This way, no local IP or private ISP's public IP leaked when VPN is the default route. by guoweis · 11 years ago
  32. 2b11677 Use network counting from webrtc for UMA. by guoweis · 11 years ago
  33. 9a77a72 Pass FROM_HERE to ObserverListThreadSafe::Notify to improve profile. by reillyg · 11 years ago
  34. 9e38d55 Mechanical rename of tracing includes for /content [2/3] by primiano · 11 years ago
  35. e29b688 replace COMPILE_ASSERT with static_assert in content/ by mostynb · 11 years ago
  36. 467bab46 Added a CHECK assert which will cause crash when missing send completion signal is detected. by guoweis · 11 years ago
  37. feb79eb Change the UMA histogram bucket to help track large video encoded output. The previous change shouldn't use the same UMA as it mixed old and new buckets. by guoweis · 11 years ago
  38. 83883c8 Make callers of CommandLine use it via the base:: namespace. by avi · 11 years ago
  39. 24c1e12 Change the UMA histogram bucket to help track large video encoded output by guoweis · 11 years ago
  40. c5f656f Hide the proxy socket address from the clients of ProxyResolvingClientSocket. by jiayl · 11 years ago
  41. 070ff085 Make the ipc_socket_factory logging show up in release build. by jiayl · 11 years ago
  42. 4c60e323 Rename network_prefix to prefix_length in NetworkInterface structure. by guoweis · 11 years ago
  43. f0ba8deca [content/renderer] Convert VLOGs to DVLOGs by anujk.sharma · 11 years ago
  44. 47041bb1e Introduce a new experiment to control the application send socket buffer size for UDP case only. by guoweis · 11 years ago
  45. 7aba582a IpcPacketSocket should SignalConnect AFTER updating the remote address. by jiayl · 11 years ago
  46. f9567f85 Update Chromium for webrtc r7656. by pkasting · 11 years ago
  47. f8c2060 Add juberti@ to OWNERS files for p2p socket by guoweis · 11 years ago
  48. ecb39f13 Roll WebRTC 7546:7549. by hellner · 11 years ago
  49. 6ca1a64 Revert "Roll WebRTC 7546:7549." by Mike Wittman · 11 years ago
  50. 4e40bbe Roll WebRTC 7546:7549. by hellner · 11 years ago
  51. 0811418 Increase WebRTC socket send buffer to 256k. by guoweis · 11 years ago
  52. fd669ae Implement UMA and internal data structure for tracking EWOULDBLOCK. by guoweis · 11 years ago
  53. 6d18e40 Standardize usage of virtual/override/final in content/renderer/ by dcheng · 11 years ago
  54. ee0b42a Replace FINAL and OVERRIDE with their C++11 counterparts in content/renderer by mohan.reddy · 11 years ago
  55. 6118ccc Remove support for legacy relay servers from P2PPortAllocator. by sergeyu · 11 years ago
  56. 0079fba Use of base::StringPairs appropriately in content by mohan.reddy · 11 years ago
  57. 45d667c Interface change for net_util.h for WebRTC IPv6 support. by guoweis · 11 years ago
  58. bb4f3cc IpcNetworkManager in renderer/p2p didn't specify the right network prefix, by guoweis · 11 years ago
  59. 1f028b7 Fix the hostname used to setup a TURN/TSL connection. by [email protected] · 11 years ago
  60. 1ee82e2e Support multiple STUN servers for PeerConnection. by [email protected] · 11 years ago
  61. e758d4c Update webrtc&libjingle 6774:6825. by [email protected] · 11 years ago
  62. d08baed Revert "Update webrtc&libjingle 6774:6799." by [email protected] · 11 years ago
  63. bf45dff Update webrtc&libjingle 6774:6799. by [email protected] · 11 years ago
  64. f3784e3 Add IPv4/IPv6 counters to UMA even when count is 0. by [email protected] · 11 years ago
  65. 2fc94c74 Provide "hostname:port" to the P2P Browser process sockets via P2PHostMsg_CreateSocket. by [email protected] · 11 years ago
  66. 5a04ee80 Handle unresolved remote hostname for TCP sockets. by [email protected] · 11 years ago
  67. c52b1b2 Add WebRtc logging in IpcPacketSocket when the socket is blocked or unblocked. by [email protected] · 11 years ago
  68. d1549b8 Decouple IPC::MessageFilter from IPC::Channel by [email protected] · 11 years ago
  69. c047ef0d Remove --enable-webrtc-tcp-server-socket by [email protected] · 11 years ago
  70. 81140c8c Revert 272379 "Revert 272371 "Add UMA Counts for number of IPv4 ..." by [email protected] · 11 years ago
  71. f916f7c Revert 272371 "Add UMA Counts for number of IPv4 and IPv6 interf..." by [email protected] · 11 years ago
  72. c01a446 Add UMA Counts for number of IPv4 and IPv6 interfaces detected for PeerConnection. by [email protected] · 11 years ago
  73. 887dc5d8 Fix more cases of unreachable code on Windows, mostly added recently. by [email protected] · 11 years ago
  74. 24b2481b Send SignalClose to the clients of P2PSocket, if TCP socket is failed to by [email protected] · 11 years ago
  75. e4916df Wiring network interface from Chromium net to libjingle Network. by [email protected] · 11 years ago
  76. 7412204 Move IPC::MessageFilter and router to a separate file by [email protected] · 11 years ago
  77. 0ca7e8a Remove net_log.h from net_util, as it's no longer needed by net_log.h. by [email protected] · 11 years ago
  78. cd03e8a Remove DCHECK for delegate object on IPC thread, as it must be accessed by [email protected] · 11 years ago
  79. 4938295c Replace SetIP with SetReolveddIP after address is resolved. by [email protected] · 11 years ago
  80. fc239be Push remote hostname to P2P socket host. by [email protected] · 11 years ago
  81. 779f58f Store hostname before address is resolved. by [email protected] · 11 years ago
  82. 8af69c6c Move TrimWhitespace to the base namespace. by [email protected] · 11 years ago
  83. 79a21df6 Add ipv4 version of loopback for peer connection when --allow-loopback-in-peer-connection is set. by [email protected] · 11 years ago
  84. f5f1707 Relanding PacketOptions enabling CL https://codereview.chromium.org/177603002 by [email protected] · 11 years ago
  85. aa207ab Add a command line flag to include loopback interface in network list. by [email protected] · 11 years ago
  86. c4261102 Moving sigslot out of content::P2PAsyncAddressResolver as it's a by [email protected] · 11 years ago
  87. 4b0416e Remove p2p socket API from content/public by [email protected] · 11 years ago
  88. 4f1ac21 Revert 252408 "Adding talk_base::PacketOptions to P2P IPC Send m..." by [email protected] · 12 years ago
  89. 905b49e Adding talk_base::PacketOptions to P2P IPC Send message. by [email protected] · 12 years ago
  90. e6b239a Roll webrtc to r5549. Major change in this revision is addition of PacketOptions structure to every packet sent over network in AsyncPacketSocket::Send methods. by [email protected] · 12 years ago
  91. a49576b1 Update webrtc/libjingle 5523:5548. by [email protected] · 12 years ago
  92. c8c56758 Add transfer size paramater to didFinishLoading [2/3] by [email protected] · 12 years ago
  93. af1e730 Implement SetOption and GetOption methods defined in talk_base::AsyncPacketSocket. by [email protected] · 12 years ago
  94. 8469c8b Revert 246612 "Implement SetOption and GetOption methods defined..." by [email protected] · 12 years ago
  95. 37c7eab7 Implement SetOption and GetOption methods defined in talk_base::AsyncPacketSocket. by [email protected] · 12 years ago
  96. ea8b7b6 Wire talk_base::AsyncResolverInterface in chrome. by [email protected] · 12 years ago
  97. 863369f7 Update webrtc/libjingle 5268:5301. by [email protected] · 12 years ago
  98. 91dce6f7 Pass packet received timestamp in microseconds from by [email protected] · 12 years ago
  99. 9da582b This CL is in preparation for making a public way for the media/cast library to access the p2p sockets. by [email protected] · 12 years ago
  100. 106e9838 Roll webrtc 5104:5122 by [email protected] · 12 years ago