/external/chromium_org/third_party/webrtc/base/ |
H A D | proxy_unittest.cc | 22 using rtc::Socket; 23 using rtc::Thread; 24 using rtc::SocketAddress; 34 class AutoDetectProxyRunner : public rtc::AutoDetectProxy { 47 ProxyTest() : ss_(new rtc::VirtualSocketServer(NULL)) { 49 socks_.reset(new rtc::SocksProxyServer( 51 https_.reset(new rtc::HttpListenServer()); 58 rtc::SocketServer* ss() { return ss_.get(); } 60 rtc::ProxyType DetectProxyType(const SocketAddress& address) { 61 rtc [all...] |
H A D | ssladapter_unittest.cc | 23 static rtc::AsyncSocket* CreateSocket(const rtc::SSLMode& ssl_mode) { 24 rtc::SocketAddress address(rtc::IPAddress(INADDR_ANY), 0); 26 rtc::AsyncSocket* socket = rtc::Thread::Current()-> 28 address.family(), (ssl_mode == rtc::SSL_MODE_DTLS) ? 35 static std::string GetSSLProtocolName(const rtc::SSLMode& ssl_mode) { 36 return (ssl_mode == rtc::SSL_MODE_DTLS) ? "DTLS" : "TLS"; 41 explicit SSLAdapterTestDummyClient(const rtc [all...] |
H A D | fakecpumonitor.h | 16 namespace rtc { namespace 18 class FakeCpuMonitor : public rtc::CpuMonitor { 26 virtual void OnMessage(rtc::Message* msg) { 30 } // namespace rtc
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H A D | windowpicker_unittest.cc | 24 if (!rtc::WindowPickerFactory::IsSupported()) { 28 rtc::scoped_ptr<rtc::WindowPicker> picker( 29 rtc::WindowPickerFactory::CreateWindowPicker()); 31 rtc::WindowDescriptionList descriptions; 39 if (!rtc::WindowPickerFactory::IsSupported()) { 43 rtc::scoped_ptr<rtc::WindowPicker> picker( 44 rtc::WindowPickerFactory::CreateWindowPicker()); 46 rtc [all...] |
H A D | macutils_unittest.cc | 15 rtc::MacOSVersionName ver = rtc::GetOSVersionName(); 17 EXPECT_NE(rtc::kMacOSUnknown, ver); 22 EXPECT_TRUE(rtc::GetQuickTimeVersion(&version)); 28 EXPECT_FALSE(rtc::RunAppleScript(script)); 33 EXPECT_FALSE(rtc::RunAppleScript(script)); 42 EXPECT_TRUE(rtc::RunAppleScript(script));
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/external/chromium_org/third_party/libjingle/source/talk/p2p/base/ |
H A D | stunport.h | 38 namespace rtc { namespace 48 static UDPPort* Create(rtc::Thread* thread, 49 rtc::PacketSocketFactory* factory, 50 rtc::Network* network, 51 rtc::AsyncPacketSocket* socket, 63 static UDPPort* Create(rtc::Thread* thread, 64 rtc::PacketSocketFactory* factory, 65 rtc::Network* network, 66 const rtc::IPAddress& ip, 81 rtc [all...] |
H A D | tcpport.h | 48 static TCPPort* Create(rtc::Thread* thread, 49 rtc::PacketSocketFactory* factory, 50 rtc::Network* network, 51 const rtc::IPAddress& ip, 72 virtual int GetOption(rtc::Socket::Option opt, int* value); 73 virtual int SetOption(rtc::Socket::Option opt, int value); 77 TCPPort(rtc::Thread* thread, rtc::PacketSocketFactory* factory, 78 rtc::Network* network, const rtc [all...] |
H A D | turnport.h | 40 namespace rtc { namespace 53 static TurnPort* Create(rtc::Thread* thread, 54 rtc::PacketSocketFactory* factory, 55 rtc::Network* network, 56 rtc::AsyncPacketSocket* socket, 67 static TurnPort* Create(rtc::Thread* thread, 68 rtc::PacketSocketFactory* factory, 69 rtc::Network* network, 70 const rtc::IPAddress& ip, 93 const rtc [all...] |
/external/chromium_org/third_party/libjingle/source/talk/app/webrtc/ |
H A D | peerconnectionfactory.h | 41 public rtc::MessageHandler { 47 virtual rtc::scoped_refptr<PeerConnectionInterface> 57 virtual rtc::scoped_refptr<MediaStreamInterface> 60 virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource( 63 virtual rtc::scoped_refptr<VideoSourceInterface> CreateVideoSource( 67 virtual rtc::scoped_refptr<VideoTrackInterface> 71 virtual rtc::scoped_refptr<AudioTrackInterface> 75 virtual bool StartAecDump(rtc::PlatformFile file); 78 virtual rtc::Thread* signaling_thread(); 79 virtual rtc [all...] |
/external/chromium_org/third_party/libjingle/source/talk/examples/call/ |
H A D | console.h | 39 class Console : public rtc::MessageHandler { 41 Console(rtc::Thread *thread, CallClient *client); 49 virtual void OnMessage(rtc::Message *msg); 65 rtc::Thread *client_thread_; 66 rtc::scoped_ptr<rtc::Thread> console_thread_;
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/external/chromium_org/third_party/libjingle/source/talk/session/media/ |
H A D | soundclip.h | 34 namespace rtc { namespace 44 class Soundclip : private rtc::MessageHandler { 46 Soundclip(rtc::Thread* thread, SoundclipMedia* soundclip_media); 62 virtual void OnMessage(rtc::Message* message); 64 rtc::Thread* worker_thread_; 65 rtc::scoped_ptr<SoundclipMedia> soundclip_media_;
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H A D | mediarecorder.h | 40 namespace rtc { namespace 57 explicit RtpDumpSink(rtc::StreamInterface* stream); 72 rtc::scoped_ptr<rtc::StreamInterface> stream_; 73 rtc::scoped_ptr<RtpDumpWriter> writer_; 74 rtc::CriticalSection critical_section_; 85 rtc::StreamInterface* send_stream, 86 rtc::StreamInterface* recv_stream, 89 rtc::StreamInterface* send_stream, 90 rtc [all...] |
/external/chromium_org/jingle/glue/ |
H A D | logging_unittest.cc | 28 static const char* AsString(rtc::LoggingSeverity severity) { 30 case rtc::LS_ERROR: 32 case rtc::LS_WARNING: 34 case rtc::LS_INFO: 36 case rtc::LS_VERBOSE: 38 case rtc::LS_SENSITIVE: 78 LOG_V(rtc::LS_ERROR) << AsString(rtc::LS_ERROR); 79 LOG_V(rtc::LS_WARNING) << AsString(rtc [all...] |
H A D | mock_task.cc | 9 MockTask::MockTask(TaskParent* parent) : rtc::Task(parent) {}
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/external/chromium_org/third_party/libjingle/source/talk/p2p/client/ |
H A D | socketmonitor.h | 40 class SocketMonitor : public rtc::MessageHandler, 44 rtc::Thread* worker_thread, 45 rtc::Thread* monitor_thread); 51 rtc::Thread* monitor_thread() { return monitoring_thread_; } 57 void OnMessage(rtc::Message* message); 62 rtc::Thread* channel_thread_; 63 rtc::Thread* monitoring_thread_; 64 rtc::CriticalSection crit_;
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H A D | connectivitychecker.h | 20 namespace rtc { namespace 63 NicId(const rtc::IPAddress& ip, 64 const rtc::SocketAddress& proxy_address) 68 rtc::IPAddress ip; 69 rtc::SocketAddress proxy_address; 96 rtc::IPAddress ip; 97 rtc::ProxyInfo proxy_info; 98 rtc::SocketAddress external_address; 100 rtc::SocketAddress media_server_address; 122 const std::vector<rtc [all...] |
/external/chromium_org/content/browser/renderer_host/p2p/ |
H A D | socket_host_throttler.h | 11 namespace rtc { namespace 27 void SetTiming(scoped_ptr<rtc::Timing> timing); 32 scoped_ptr<rtc::Timing> timing_; 33 scoped_ptr<rtc::RateLimiter> rate_limiter_;
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/external/chromium_org/third_party/webrtc/sound/ |
H A D | platformsoundsystem.cc | 20 namespace rtc { namespace 31 } // namespace rtc
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/external/chromium_org/remoting/protocol/ |
H A D | chromium_socket_factory_unittest.cc | 24 rtc::SocketAddress("127.0.0.1", 0), 0, 0)); 26 EXPECT_EQ(socket_->GetState(), rtc::AsyncPacketSocket::STATE_BOUND); 31 void OnPacket(rtc::AsyncPacketSocket* socket, 33 const rtc::SocketAddress& address, 34 const rtc::PacketTime& packet_time) { 41 void VerifyCanSendAndReceive(rtc::AsyncPacketSocket* sender) { 47 rtc::PacketOptions options; 63 scoped_ptr<rtc::PacketSocketFactory> socket_factory_; 64 scoped_ptr<rtc::AsyncPacketSocket> socket_; 67 rtc [all...] |
/external/chromium_org/third_party/libjingle/source/talk/media/base/ |
H A D | filemediaengine.cc | 62 rtc::FileStream* input_file_stream = NULL; 63 rtc::FileStream* output_file_stream = NULL; 68 input_file_stream = rtc::Filesystem::OpenFile( 69 rtc::Pathname(voice_input_filename_), "rb"); 77 output_file_stream = rtc::Filesystem::OpenFile( 78 rtc::Pathname(voice_output_filename_), "wb"); 92 rtc::FileStream* input_file_stream = NULL; 93 rtc::FileStream* output_file_stream = NULL; 99 input_file_stream = rtc::Filesystem::OpenFile( 100 rtc [all...] |
/external/chromium_org/content/renderer/p2p/ |
H A D | ipc_network_manager.cc | 18 rtc::AdapterType ConvertConnectionTypeToAdapterType( 22 return rtc::ADAPTER_TYPE_UNKNOWN; 24 return rtc::ADAPTER_TYPE_ETHERNET; 26 return rtc::ADAPTER_TYPE_WIFI; 30 return rtc::ADAPTER_TYPE_CELLULAR; 32 return rtc::ADAPTER_TYPE_UNKNOWN; 74 // rtc::Network uses these prefix_length to compare network 76 std::vector<rtc::Network*> networks; 84 address = rtc::NetworkToHost32(address); 85 rtc [all...] |
/external/chromium_org/remoting/test/ |
H A D | fake_socket_factory.h | 21 class FakePacketSocketFactory : public rtc::PacketSocketFactory, 52 // rtc::PacketSocketFactory interface. 53 virtual rtc::AsyncPacketSocket* CreateUdpSocket( 54 const rtc::SocketAddress& local_address, 56 virtual rtc::AsyncPacketSocket* CreateServerTcpSocket( 57 const rtc::SocketAddress& local_address, 60 virtual rtc::AsyncPacketSocket* CreateClientTcpSocket( 61 const rtc::SocketAddress& local_address, 62 const rtc::SocketAddress& remote_address, 63 const rtc [all...] |
/external/chromium_org/third_party/libjingle/source/talk/examples/stunserver/ |
H A D | stunserver_main.cc | 45 rtc::SocketAddress server_addr; 51 rtc::Thread *pthMain = rtc::Thread::Current(); 53 rtc::AsyncUDPSocket* server_socket = 54 rtc::AsyncUDPSocket::Create(pthMain->socketserver(), server_addr);
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/external/chromium_org/third_party/libjingle/source/talk/session/tunnel/ |
H A D | securetunnelsessionclient.h | 69 void SetIdentity(rtc::SSLIdentity* identity); 80 rtc::SSLIdentity& GetIdentity() const; 99 Session* session, rtc::Thread* stream_thread, 107 rtc::scoped_ptr<rtc::SSLIdentity> identity_; 126 rtc::Thread* stream_thread, 132 virtual rtc::StreamInterface* GetStream(); 141 rtc::StreamInterface* MakeSecureStream( 142 rtc::StreamInterface* stream); 158 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...] |