/external/chromium_org/content/renderer/p2p/ |
ipc_socket_factory.h | 17 // IpcPacketSocketFactory implements rtc::PacketSocketFactory 20 // rtc::Thread) and also has associated base::MessageLoop. Each 23 class IpcPacketSocketFactory : public rtc::PacketSocketFactory { 29 virtual rtc::AsyncPacketSocket* CreateUdpSocket( 30 const rtc::SocketAddress& local_address, 32 virtual rtc::AsyncPacketSocket* CreateServerTcpSocket( 33 const rtc::SocketAddress& local_address, 37 virtual rtc::AsyncPacketSocket* CreateClientTcpSocket( 38 const rtc::SocketAddress& local_address, 39 const rtc::SocketAddress& remote_address [all...] |
/external/chromium_org/remoting/client/plugin/ |
pepper_packet_socket_factory.h | 14 class PepperPacketSocketFactory : public rtc::PacketSocketFactory { 19 virtual rtc::AsyncPacketSocket* CreateUdpSocket( 20 const rtc::SocketAddress& local_address, 22 virtual rtc::AsyncPacketSocket* CreateServerTcpSocket( 23 const rtc::SocketAddress& local_address, 27 virtual rtc::AsyncPacketSocket* CreateClientTcpSocket( 28 const rtc::SocketAddress& local_address, 29 const rtc::SocketAddress& remote_address, 30 const rtc::ProxyInfo& proxy_info, 33 virtual rtc::AsyncResolverInterface* CreateAsyncResolver() OVERRIDE [all...] |
pepper_util.h | 17 namespace rtc { namespace 26 const rtc::SocketAddress& address, 30 const rtc::SocketAddress& address, 33 rtc::SocketAddress* address);
|
/external/chromium_org/jingle/glue/ |
jingle_glue_mock_objects.h | 13 class MockStream : public rtc::StreamInterface { 18 MOCK_CONST_METHOD0(GetState, rtc::StreamState()); 20 MOCK_METHOD4(Read, rtc::StreamResult(void*, size_t, size_t*, int*)); 21 MOCK_METHOD4(Write, rtc::StreamResult(const void*, size_t, 26 MOCK_METHOD3(PostEvent, void(rtc::Thread*, int, int));
|
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::LS_WARNING) [all...] |
thread_wrapper.h | 19 // JingleThreadWrapper implements rtc::Thread interface on top of 31 public rtc::Thread { 57 // rtc::MessageQueue overrides. 58 virtual void Post(rtc::MessageHandler *phandler, 60 rtc::MessageData *pdata, 63 rtc::MessageHandler* handler, 65 rtc::MessageData* data) OVERRIDE; 66 virtual void Clear(rtc::MessageHandler* handler, 68 rtc::MessageList* removed) OVERRIDE; 69 virtual void Send(rtc::MessageHandler *handler [all...] |
mock_task.cc | 9 MockTask::MockTask(TaskParent* parent) : rtc::Task(parent) {}
|
/external/chromium_org/third_party/libjingle/source/talk/app/webrtc/ |
sctputils.h | 35 namespace rtc { namespace 37 } // namespace rtc 42 bool ParseDataChannelOpenMessage(const rtc::Buffer& payload, 46 bool ParseDataChannelOpenAckMessage(const rtc::Buffer& payload); 50 rtc::Buffer* payload); 52 void WriteDataChannelOpenAckMessage(rtc::Buffer* payload);
|
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::Thread* worker_thread() [all...] |
/external/chromium_org/third_party/libjingle/source/talk/p2p/base/ |
teststunserver.h | 40 TestStunServer(rtc::Thread* thread, 41 const rtc::SocketAddress& addr) 44 udp_socket_(rtc::AsyncUDPSocket::Create(socket_, addr)), 48 rtc::AsyncSocket* socket_; 49 rtc::AsyncUDPSocket* udp_socket_;
|
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::SocketAddress GetLocalAddress() const [all...] |
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::SocketAddress& addr [all...] |
turnserver.h | 42 namespace rtc { namespace 69 virtual bool ShouldRedirect(const rtc::SocketAddress& address, 70 rtc::SocketAddress* out) = 0; 80 explicit TurnServer(rtc::Thread* thread); 101 void AddInternalSocket(rtc::AsyncPacketSocket* socket, 106 void AddInternalServerSocket(rtc::AsyncSocket* socket, 109 void SetExternalSocketFactory(rtc::PacketSocketFactory* factory, 110 const rtc::SocketAddress& address); 117 Connection(const rtc::SocketAddress& src, 119 rtc::AsyncPacketSocket* socket) [all...] |
relayport.h | 52 typedef std::pair<rtc::Socket::Option, int> OptionValue; 56 rtc::Thread* thread, rtc::PacketSocketFactory* factory, 57 rtc::Network* network, const rtc::IPAddress& ip, 74 virtual int SetOption(rtc::Socket::Option opt, int value); 75 virtual int GetOption(rtc::Socket::Option opt, int* value); 86 RelayPort(rtc::Thread* thread, rtc::PacketSocketFactory* factory, 87 rtc::Network*, const rtc::IPAddress& ip [all...] |
/external/chromium_org/third_party/libjingle/source/talk/session/media/ |
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::StreamInterface* recv_stream [all...] |
/external/chromium_org/third_party/webrtc/base/ |
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::ProxyType type [all...] |
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::SSLMode& ssl_mode [all...] |
/external/chromium_org/third_party/libjingle/source/talk/examples/call/ |
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_;
|
/external/chromium_org/third_party/libjingle/source/talk/p2p/client/ |
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_;
|
/external/chromium_org/third_party/webrtc/sound/ |
nullsoundsystemfactory.h | 16 namespace rtc { namespace 31 } // namespace rtc
|
platformsoundsystem.cc | 20 namespace rtc { namespace 31 } // namespace rtc
|
platformsoundsystemfactory.h | 16 namespace rtc { namespace 31 } // namespace rtc
|
/external/chromium_org/remoting/test/ |
fake_network_manager.h | 16 class FakeNetworkManager : public rtc::NetworkManager { 18 FakeNetworkManager(const rtc::IPAddress& address); 21 // rtc::NetworkManager interface. 30 scoped_ptr<rtc::Network> network_;
|
/external/chromium_org/remoting/protocol/ |
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::SocketAddress last_address_ [all...] |
/external/chromium_org/third_party/libjingle/source/talk/media/base/ |
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::Pathname(video_input_filename_), "rb") [all...] |