| /external/chromium_org/third_party/libjingle/source/talk/p2p/base/ |
| rawtransportchannel.h | 40 namespace rtc { namespace
57 public rtc::MessageHandler {
62 rtc::Thread *worker_thread,
68 const rtc::PacketOptions& options, int flags);
69 virtual int SetOption(rtc::Socket::Option opt, int value);
94 void OnRemoteAddress(const rtc::SocketAddress& remote_address);
117 virtual bool GetSslRole(rtc::SSLRole* role) const {
121 virtual bool SetSslRole(rtc::SSLRole role) {
136 virtual bool GetLocalIdentity(rtc::SSLIdentity** identity) const {
140 virtual bool GetRemoteCertificate(rtc::SSLCertificate** cert) const
[all...] |
| portinterface.h | 36 namespace rtc { namespace
61 virtual rtc::Network* Network() const = 0;
84 const rtc::SocketAddress& remote_addr) = 0;
92 virtual int SetOption(rtc::Socket::Option opt, int value) = 0;
93 virtual int GetOption(rtc::Socket::Option opt, int* value) = 0;
101 const rtc::SocketAddress& addr,
102 const rtc::PacketOptions& options, bool payload) = 0;
107 sigslot::signal6<PortInterface*, const rtc::SocketAddress&,
115 const rtc::SocketAddress& addr) = 0;
117 StunMessage* request, const rtc::SocketAddress& addr
[all...] |
| stunserver.cc | 35 StunServer::StunServer(rtc::AsyncUDPSocket* socket) : socket_(socket) {
44 rtc::AsyncPacketSocket* socket, const char* buf, size_t size,
45 const rtc::SocketAddress& remote_addr,
46 const rtc::PacketTime& packet_time) {
48 rtc::ByteBuffer bbuf(buf, size);
69 StunMessage* msg, const rtc::SocketAddress& remote_addr) {
88 const StunMessage& msg, const rtc::SocketAddress& addr,
103 const StunMessage& msg, const rtc::SocketAddress& addr) {
104 rtc::ByteBuffer buf;
106 rtc::PacketOptions options
[all...] |
| tcpport.cc | 36 TCPPort::TCPPort(rtc::Thread* thread,
37 rtc::PacketSocketFactory* factory,
38 rtc::Network* network, const rtc::IPAddress& ip,
56 rtc::SocketAddress(ip(), 0), min_port(), max_port(),
110 if (rtc::AsyncPacketSocket* socket =
128 if (socket_->GetState() == rtc::AsyncPacketSocket::STATE_BOUND ||
129 socket_->GetState() == rtc::AsyncPacketSocket::STATE_CLOSED)
131 rtc::SocketAddress(),
138 AddAddress(rtc::SocketAddress(ip(), 0)
[all...] |
| basicpacketsocketfactory.cc | 42 namespace rtc { namespace
66 rtc::AsyncSocket* socket =
78 return new rtc::AsyncUDPSocket(socket);
90 rtc::AsyncSocket* socket =
107 socket = new rtc::AsyncSSLSocket(socket);
112 socket->SetOption(rtc::Socket::OPT_NODELAY, 1);
117 return new rtc::AsyncTCPSocket(socket, true);
124 rtc::AsyncSocket* socket =
138 if (proxy_info.type == rtc::PROXY_SOCKS5) {
139 socket = new rtc::AsyncSocksProxySocket
[all...] |
| transportchannelproxy.cc | 47 worker_thread_ = rtc::Thread::Current();
58 ASSERT(rtc::Thread::Current() == worker_thread_);
104 const rtc::PacketOptions& options,
106 ASSERT(rtc::Thread::Current() == worker_thread_);
114 int TransportChannelProxy::SetOption(rtc::Socket::Option opt, int value) {
115 ASSERT(rtc::Thread::Current() == worker_thread_);
124 ASSERT(rtc::Thread::Current() == worker_thread_);
132 ASSERT(rtc::Thread::Current() == worker_thread_);
140 ASSERT(rtc::Thread::Current() == worker_thread_);
147 bool TransportChannelProxy::GetSslRole(rtc::SSLRole* role) const
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/media/base/ |
| rtpdump.cc | 56 void RtpDumpFileHeader::WriteToByteBuffer(rtc::ByteBuffer* buf) {
114 rtc::StreamResult RtpDumpReader::ReadPacket(RtpDumpPacket* packet) {
115 if (!packet) return rtc::SR_ERROR;
117 rtc::StreamResult res = rtc::SR_SUCCESS;
121 if (res != rtc::SR_SUCCESS) {
130 if (res != rtc::SR_SUCCESS) {
133 rtc::ByteBuffer buf(header, sizeof(header));
153 if (res == rtc::SR_SUCCESS &&
156 rtc::SetBE32(&packet->data[kRtpSsrcOffset], ssrc_override_)
[all...] |
| rtpdataengine.h | 54 void SetTiming(rtc::Timing* timing) {
60 rtc::scoped_ptr<rtc::Timing> timing_;
89 explicit RtpDataMediaChannel(rtc::Timing* timing);
95 void set_timing(rtc::Timing* timing) {
119 virtual void OnPacketReceived(rtc::Buffer* packet,
120 const rtc::PacketTime& packet_time);
121 virtual void OnRtcpReceived(rtc::Buffer* packet,
122 const rtc::PacketTime& packet_time) {}
126 const rtc::Buffer& payload
[all...] |
| rtpdump_unittest.cc | 43 rtc::ByteBuffer rtp_buf;
64 rtc::ByteBuffer rtcp_buf;
78 rtc::MemoryStream stream;
80 rtc::scoped_ptr<RtpDumpReader> reader;
87 EXPECT_EQ(rtc::SR_SUCCESS, reader->ReadPacket(&packet));
92 EXPECT_EQ(rtc::SR_SUCCESS,
96 EXPECT_EQ(rtc::SR_SUCCESS, reader->ReadPacket(&packet));
101 EXPECT_EQ(rtc::SR_SUCCESS,
105 EXPECT_EQ(rtc::SR_ERROR, reader->ReadPacket(&packet));
110 EXPECT_EQ(rtc::SR_SUCCESS
[all...] |
| filemediaengine_unittest.cc | 52 FileNetworkInterface(rtc::StreamInterface* output, MediaChannel* ch)
61 virtual bool SendPacket(rtc::Buffer* packet,
62 rtc::DiffServCodePoint dscp) {
66 media_channel_->OnPacketReceived(packet, rtc::PacketTime());
69 rtc::SR_SUCCESS != dump_writer_->WriteRtpPacket(
78 virtual bool SendRtcp(rtc::Buffer* packet,
79 rtc::DiffServCodePoint dscp) { return false; }
81 rtc::Socket::Option opt, int option) {
84 virtual void SetDefaultDSCPCode(rtc::DiffServCodePoint dscp) {}
90 rtc::scoped_ptr<RtpDumpWriter> dump_writer_
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/session/tunnel/ |
| pseudotcpchannel.h | 37 namespace rtc { namespace
67 public rtc::MessageHandler,
71 PseudoTcpChannel(rtc::Thread* stream_thread,
77 rtc::StreamInterface* GetStream();
96 rtc::StreamState GetState() const;
97 rtc::StreamResult Read(void* buffer, size_t buffer_len,
99 rtc::StreamResult Write(const void* data, size_t data_len,
104 void OnMessage(rtc::Message* pmsg);
114 const rtc::PacketTime& packet_time, int flags);
126 rtc::Thread* signal_thread_, * worker_thread_, * stream_thread_
[all...] |
| tunnelsessionclient_unittest.cc | 48 public rtc::MessageHandler,
52 : local_pa_(rtc::Thread::Current(), NULL),
53 remote_pa_(rtc::Thread::Current(), NULL),
54 local_sm_(&local_pa_, rtc::Thread::Current()),
55 remote_sm_(&remote_pa_, rtc::Thread::Current()),
107 rtc::Thread::Current()->Post(this, MSG_LSIGNAL,
108 rtc::WrapMessageData(*stanza));
110 rtc::Thread::Current()->Post(this, MSG_RSIGNAL,
111 rtc::WrapMessageData(*stanza));
117 virtual void OnMessage(rtc::Message* message)
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/app/webrtc/ |
| peerconnectionfactory.cc | 47 using rtc::scoped_refptr;
51 typedef rtc::TypedMessageData<bool> InitMessageData;
53 struct CreatePeerConnectionParams : public rtc::MessageData {
74 struct CreateAudioSourceParams : public rtc::MessageData {
83 struct CreateVideoSourceParams : public rtc::MessageData {
94 struct StartAecDumpParams : public rtc::MessageData {
95 explicit StartAecDumpParams(rtc::PlatformFile aec_dump_file)
98 rtc::PlatformFile aec_dump_file;
115 rtc::scoped_refptr<PeerConnectionFactoryInterface>
117 rtc::scoped_refptr<PeerConnectionFactory> pc_factory
[all...] |
| dtmfsender.h | 44 namespace rtc { namespace
73 public rtc::MessageHandler {
75 static rtc::scoped_refptr<DtmfSender> Create(
77 rtc::Thread* signaling_thread,
93 rtc::Thread* signaling_thread,
101 virtual void OnMessage(rtc::Message* msg);
110 rtc::scoped_refptr<AudioTrackInterface> track_;
112 rtc::Thread* signaling_thread_;
|
| /external/chromium_org/jingle/glue/ |
| thread_wrapper.cc | 16 PendingSend(const rtc::Message& message_value)
24 rtc::Message message;
40 DCHECK_EQ(rtc::Thread::Current(), current());
50 : rtc::Thread(new rtc::NullSocketServer()),
57 DCHECK(!rtc::Thread::Current());
59 rtc::MessageQueueManager::Add(this);
64 Clear(NULL, rtc::MQID_ANY, NULL);
68 DCHECK_EQ(rtc::Thread::Current(), current());
71 rtc::ThreadManager::Instance()->SetCurrentThread(NULL)
[all...] |
| /external/chromium_org/jingle/notifier/base/ |
| weak_xmpp_client.h | 18 namespace rtc { namespace
25 // from rtc::Task, whose destructor *is* marked virtual, so we
29 explicit WeakXmppClient(rtc::TaskParent* parent);
|
| /external/chromium_org/third_party/libjingle/source/talk/app/webrtc/objc/ |
| RTCMediaSource+Internal.h | 35 rtc::scoped_refptr<webrtc::MediaSourceInterface> mediaSource;
38 (rtc::scoped_refptr<webrtc::MediaSourceInterface>)mediaSource;
|
| RTCMediaStream+Internal.h | 35 rtc::scoped_refptr<webrtc::MediaStreamInterface> mediaStream;
38 (rtc::scoped_refptr<webrtc::MediaStreamInterface>)mediaStream;
|
| RTCMediaStreamTrack+Internal.h | 35 rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> mediaTrack;
38 (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)mediaTrack;
|
| /external/chromium_org/third_party/webrtc/base/ |
| scoped_autorelease_pool.mm | 15 namespace rtc {
25 } // namespace rtc
|
| /external/chromium_org/remoting/test/ |
| fake_socket_factory.cc | 37 class FakeUdpSocket : public rtc::AsyncPacketSocket {
41 const rtc::SocketAddress& local_address);
44 void ReceivePacket(const rtc::SocketAddress& from,
45 const rtc::SocketAddress& to,
49 // rtc::AsyncPacketSocket interface.
50 virtual rtc::SocketAddress GetLocalAddress() const OVERRIDE;
51 virtual rtc::SocketAddress GetRemoteAddress() const OVERRIDE;
53 const rtc::PacketOptions& options) OVERRIDE;
55 const rtc::SocketAddress& address,
56 const rtc::PacketOptions& options) OVERRIDE
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/examples/turnserver/ |
| turnserver_main.cc | 52 size_t len = rtc::hex_decode(buf, sizeof(buf), hex);
58 rtc::OptionsFile file_;
68 rtc::SocketAddress int_addr;
74 rtc::IPAddress ext_addr;
80 rtc::Thread* main = rtc::Thread::Current();
81 rtc::AsyncUDPSocket* int_socket =
82 rtc::AsyncUDPSocket::Create(main->socketserver(), int_addr);
95 server.SetExternalSocketFactory(new rtc::BasicPacketSocketFactory(),
96 rtc::SocketAddress(ext_addr, 0))
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/session/media/ |
| audiomonitor.cc | 40 rtc::Thread *monitor_thread) {
62 void AudioMonitor::OnMessage(rtc::Message *message) {
63 rtc::CritScope cs(&crit_);
67 assert(rtc::Thread::Current() == voice_channel_->worker_thread());
75 assert(rtc::Thread::Current() == voice_channel_->worker_thread());
83 assert(rtc::Thread::Current() == voice_channel_->worker_thread());
89 assert(rtc::Thread::Current() == monitoring_thread_);
100 rtc::CritScope cs(&crit_);
101 assert(rtc::Thread::Current() == voice_channel_->worker_thread());
117 rtc::Thread *AudioMonitor::monitor_thread()
[all...] |
| /external/chromium_org/remoting/protocol/ |
| chromium_socket_factory.cc | 33 class UdpPacketSocket : public rtc::AsyncPacketSocket {
38 bool Init(const rtc::SocketAddress& local_address,
41 // rtc::AsyncPacketSocket interface.
42 virtual rtc::SocketAddress GetLocalAddress() const OVERRIDE;
43 virtual rtc::SocketAddress GetRemoteAddress() const OVERRIDE;
45 const rtc::PacketOptions& options) OVERRIDE;
47 const rtc::SocketAddress& address,
48 const rtc::PacketOptions& options) OVERRIDE;
51 virtual int GetOption(rtc::Socket::Option option, int* value) OVERRIDE;
52 virtual int SetOption(rtc::Socket::Option option, int value) OVERRIDE
[all...] |
| /external/chromium_org/third_party/libjingle/source/talk/media/sctp/ |
| sctpdataengine.h | 110 public rtc::MessageHandler {
135 explicit SctpDataMediaChannel(rtc::Thread* thread);
152 virtual void OnMessage(rtc::Message* msg);
157 const rtc::Buffer& payload,
160 virtual void OnPacketReceived(rtc::Buffer* packet,
161 const rtc::PacketTime& packet_time);
164 rtc::Thread* worker_thread() const { return worker_thread_; }
183 virtual void OnRtcpReceived(rtc::Buffer* packet,
184 const rtc::PacketTime& packet_time) {}
216 void OnPacketFromSctpToNetwork(rtc::Buffer* buffer)
[all...] |
