| /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...] |
