/external/chromium_org/third_party/webrtc/base/ |
H A D | ssladapter.cc | 35 namespace rtc { namespace 97 } // namespace rtc
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H A D | sslstreamadapter.cc | 34 namespace rtc { namespace 77 } // namespace rtc
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H A D | unixfilesystem.h | 18 namespace rtc { namespace 124 } // namespace rtc
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H A D | win32regkey.h | 29 namespace rtc { namespace 335 } // namespace rtc
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H A D | asyncresolverinterface.h | 17 namespace rtc { namespace 45 } // namespace rtc
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H A D | asyncsocket.cc | 13 namespace rtc { namespace 44 } // namespace rtc
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H A D | asynctcpsocket_unittest.cc | 19 namespace rtc { namespace 26 : pss_(new rtc::PhysicalSocketServer), 27 vss_(new rtc::VirtualSocketServer(pss_.get())), 35 void OnReadyToSend(rtc::AsyncPacketSocket* socket) { 53 } // namespace rtc
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H A D | asyncudpsocket_unittest.cc | 19 namespace rtc { namespace 26 : pss_(new rtc::PhysicalSocketServer), 27 vss_(new rtc::VirtualSocketServer(pss_.get())), 35 void OnReadyToSend(rtc::AsyncPacketSocket* socket) { 53 } // namespace rtc
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H A D | atomicops_unittest.cc | 14 namespace rtc { namespace 30 rtc::FixedSizeLockFreeQueue<int> queue; 39 rtc::FixedSizeLockFreeQueue<int> queue(5); 47 rtc::FixedSizeLockFreeQueue<int> queue(2); 68 rtc::FixedSizeLockFreeQueue<int> queue(2);
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H A D | bandwidthsmoother_unittest.cc | 16 namespace rtc { namespace 116 } // namespace rtc
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H A D | crc32.cc | 15 namespace rtc { namespace 51 } // namespace rtc
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H A D | event.h | 25 namespace rtc { namespace 49 } // namespace rtc
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/external/chromium_org/third_party/webrtc/sound/ |
H A D | alsasoundsystem.h | 18 namespace rtc { namespace 101 } // namespace rtc
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H A D | alsasymboltable.h | 18 namespace rtc { namespace 47 } // namespace rtc
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H A D | nullsoundsystemfactory.cc | 15 namespace rtc { namespace 32 } // namespace rtc
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H A D | platformsoundsystemfactory.cc | 16 namespace rtc { namespace 40 } // namespace rtc
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H A D | pulseaudiosoundsystem.h | 20 namespace rtc { namespace 173 } // namespace rtc
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H A D | pulseaudiosymboltable.h | 23 namespace rtc { namespace 85 } // namespace rtc
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H A D | soundinputstreaminterface.h | 17 namespace rtc { namespace 21 // for rtc::Worker. 66 } // namespace rtc
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H A D | soundoutputstreaminterface.h | 17 namespace rtc { namespace 21 // DisableBufferMonitoring() are the same as for rtc::Worker. 70 } // namespace rtc
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/external/chromium_org/remoting/signaling/ |
H A D | jingle_info_request.h | 21 namespace rtc { namespace 23 } // namespace rtc 42 const std::vector<rtc::SocketAddress>&
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/external/chromium_org/third_party/libjingle/source/talk/media/base/ |
H A D | executablehelpers.h | 40 namespace rtc { namespace 52 return rtc::Pathname(); 56 return rtc::Pathname(); 61 rtc::Pathname path(dir_tmp); 63 rtc::Pathname path(exe_path_buffer); 71 return rtc::Pathname(); 81 return rtc::Pathname(); 86 return rtc::Pathname(); 90 rtc::Pathname path(exe_path_buffer); 92 rtc [all...] |
/external/chromium_org/third_party/libjingle/source/talk/p2p/base/ |
H A D | packetsocketfactory.h | 33 namespace rtc { namespace 67 } // namespace rtc
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H A D | portproxy.h | 34 namespace rtc { namespace 49 virtual rtc::Network* Network() const; 68 const rtc::SocketAddress& remote_addr); 71 const rtc::SocketAddress& addr, 72 const rtc::PacketOptions& options, 74 virtual int SetOption(rtc::Socket::Option opt, int value); 75 virtual int GetOption(rtc::Socket::Option opt, int* value); 81 const rtc::SocketAddress& addr); 83 StunMessage* request, const rtc::SocketAddress& addr, 91 const rtc [all...] |
/external/chromium_org/third_party/libjingle/source/talk/xmpp/ |
H A D | xmppsocket.h | 41 namespace rtc { namespace 45 extern rtc::AsyncSocket* cricket_socket_; 58 virtual bool Connect(const rtc::SocketAddress& addr); 69 void OnReadEvent(rtc::AsyncSocket * socket); 70 void OnWriteEvent(rtc::AsyncSocket * socket); 71 void OnConnectEvent(rtc::AsyncSocket * socket); 72 void OnCloseEvent(rtc::AsyncSocket * socket, int error); 74 void OnEvent(rtc::StreamInterface* stream, int events, int err); 77 rtc::AsyncSocket * cricket_socket_; 79 rtc [all...] |