Searched refs:rtc (Results 76 - 100 of 936) sorted by relevance

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/external/chromium_org/third_party/webrtc/base/
H A Dasynctcpsocket_unittest.cc19 namespace rtc { namespace
26 : pss_(new rtc::PhysicalSocketServer),
27 vss_(new rtc::VirtualSocketServer(pss_.get())),
35 void OnReadyToSend(rtc::AsyncPacketSocket* socket) {
53 } // namespace rtc
H A Dasyncudpsocket_unittest.cc19 namespace rtc { namespace
26 : pss_(new rtc::PhysicalSocketServer),
27 vss_(new rtc::VirtualSocketServer(pss_.get())),
35 void OnReadyToSend(rtc::AsyncPacketSocket* socket) {
53 } // namespace rtc
H A Dsysteminfo_unittest.cc17 rtc::SystemInfo info;
25 rtc::SystemInfo info;
27 EXPECT_TRUE(rtc::string_match(info.GetCpuVendor().c_str(),
29 rtc::string_match(info.GetCpuVendor().c_str(),
32 EXPECT_TRUE(rtc::string_match(info.GetCpuVendor().c_str(), "ARM"));
39 rtc::SystemInfo info;
41 rtc::SystemInfo::Architecture architecture = info.GetCpuArchitecture();
44 EXPECT_EQ(rtc::SystemInfo::SI_ARCH_X64, architecture);
47 EXPECT_EQ(rtc::SystemInfo::SI_ARCH_ARM, architecture);
49 EXPECT_EQ(rtc
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H A Dscoped_autorelease_pool.mm15 namespace rtc {
25 } // namespace rtc
/external/chromium_org/third_party/webrtc/sound/
H A Dlinuxsoundsystem.h16 namespace rtc { namespace
39 } // namespace rtc
H A Dnullsoundsystemfactory.cc15 namespace rtc { namespace
32 } // namespace rtc
/external/chromium_org/third_party/libjingle/source/talk/session/media/
H A Dmediamonitor.h42 class MediaMonitor : public rtc::MessageHandler,
45 MediaMonitor(rtc::Thread* worker_thread,
46 rtc::Thread* monitor_thread);
53 void OnMessage(rtc::Message *message);
58 rtc::CriticalSection crit_;
59 rtc::Thread* worker_thread_;
60 rtc::Thread* monitor_thread_;
69 MediaMonitorT(MC* media_channel, rtc::Thread* worker_thread,
70 rtc::Thread* monitor_thread)
H A Dtypingmonitor.h34 namespace rtc { namespace
60 : public rtc::MessageHandler, public sigslot::has_slots<> {
62 TypingMonitor(VoiceChannel* channel, rtc::Thread* worker_thread,
72 void OnMessage(rtc::Message* msg);
75 rtc::Thread* worker_thread_;
/external/chromium_org/third_party/libjingle/source/talk/p2p/client/
H A Dconnectivitychecker_unittest.cc43 static const rtc::SocketAddress kClientAddr1("11.11.11.11", 0);
44 static const rtc::SocketAddress kClientAddr2("22.22.22.22", 0);
45 static const rtc::SocketAddress kExternalAddr("33.33.33.33", 3333);
46 static const rtc::SocketAddress kStunAddr("44.44.44.44", 4444);
47 static const rtc::SocketAddress kRelayAddr("55.55.55.55", 5555);
48 static const rtc::SocketAddress kProxyAddr("66.66.66.66", 6666);
49 static const rtc::ProxyType kProxyType = rtc::PROXY_HTTPS;
69 FakeRelayPort(rtc::Thread* thread,
70 rtc
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/external/chromium_org/third_party/libjingle/source/talk/p2p/base/
H A Ddtlstransport.h34 namespace rtc { namespace
46 DtlsTransport(rtc::Thread* signaling_thread,
47 rtc::Thread* worker_thread,
50 rtc::SSLIdentity* identity)
53 secure_role_(rtc::SSL_CLIENT) {
59 virtual void SetIdentity_w(rtc::SSLIdentity* identity) {
62 virtual bool GetIdentity_w(rtc::SSLIdentity** identity) {
72 rtc::SSLFingerprint* local_fp =
78 rtc::scoped_ptr<rtc
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H A Drelayport.h52 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
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H A Dportinterface.h36 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
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H A Dportproxy.h34 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
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/external/chromium_org/third_party/libjingle/source/talk/examples/peerconnection/client/
H A Dpeer_connection_client.h57 public rtc::MessageHandler {
87 void OnMessage(rtc::Message* msg);
94 void OnConnect(rtc::AsyncSocket* socket);
95 void OnHangingGetConnect(rtc::AsyncSocket* socket);
106 bool ReadIntoBuffer(rtc::AsyncSocket* socket, std::string* data,
109 void OnRead(rtc::AsyncSocket* socket);
111 void OnHangingGetRead(rtc::AsyncSocket* socket);
122 void OnClose(rtc::AsyncSocket* socket, int err);
124 void OnResolveResult(rtc::AsyncResolverInterface* resolver);
127 rtc
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/external/chromium_org/third_party/libjingle/source/talk/media/base/
H A Drtpdump.cc56 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
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H A Drtpdataengine.h54 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
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/external/chromium_org/third_party/libjingle/source/talk/app/webrtc/
H A Dpeerconnectionfactory.cc47 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
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/external/chromium_org/jingle/glue/
H A Dthread_wrapper.cc16 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
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/external/chromium_org/third_party/libjingle/source/talk/app/webrtc/objc/
H A DRTCMediaSource+Internal.h35 rtc::scoped_refptr<webrtc::MediaSourceInterface> mediaSource;
38 (rtc::scoped_refptr<webrtc::MediaSourceInterface>)mediaSource;
H A DRTCMediaStream+Internal.h35 rtc::scoped_refptr<webrtc::MediaStreamInterface> mediaStream;
38 (rtc::scoped_refptr<webrtc::MediaStreamInterface>)mediaStream;
H A DRTCMediaStreamTrack+Internal.h35 rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> mediaTrack;
38 (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)mediaTrack;
/external/chromium_org/remoting/client/plugin/
H A Dpepper_address_resolver.h14 // rtc::AsyncResolverInterface implementation that uses Pepper to resolve
16 class PepperAddressResolver : public rtc::AsyncResolverInterface {
21 // rtc::AsyncResolverInterface.
22 virtual void Start(const rtc::SocketAddress& addr) OVERRIDE;
24 rtc::SocketAddress* addr) const OVERRIDE;
33 rtc::SocketAddress ipv4_address_;
34 rtc::SocketAddress ipv6_address_;
/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/
H A Daudio_checksum.h45 checksum_.Finish(checksum_result_, rtc::Md5Digest::kSize);
47 return rtc::hex_encode(checksum_result_, rtc::Md5Digest::kSize);
51 rtc::Md5Digest checksum_;
52 char checksum_result_[rtc::Md5Digest::kSize];
/external/chromium_org/remoting/test/
H A Dfake_socket_factory.cc37 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
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/external/chromium_org/third_party/libjingle/source/talk/examples/turnserver/
H A Dturnserver_main.cc52 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
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