/external/chromium_org/third_party/webrtc/base/ |
H A D | asynctcpsocket_unittest.cc | 19 namespace rtc { namespace 26 : pss_(new rtc::PhysicalSocketServer), 27 vss_(new rtc::VirtualSocketServer(pss_.get())), 35 void OnReadyToSend(rtc::AsyncPacketSocket* socket) { 53 } // namespace rtc
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H A D | asyncudpsocket_unittest.cc | 19 namespace rtc { namespace 26 : pss_(new rtc::PhysicalSocketServer), 27 vss_(new rtc::VirtualSocketServer(pss_.get())), 35 void OnReadyToSend(rtc::AsyncPacketSocket* socket) { 53 } // namespace rtc
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H A D | systeminfo_unittest.cc | 17 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 [all...] |
H A D | scoped_autorelease_pool.mm | 15 namespace rtc { 25 } // namespace rtc
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/external/chromium_org/third_party/webrtc/sound/ |
H A D | linuxsoundsystem.h | 16 namespace rtc { namespace 39 } // namespace rtc
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H A D | nullsoundsystemfactory.cc | 15 namespace rtc { namespace 32 } // namespace rtc
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/external/chromium_org/third_party/libjingle/source/talk/session/media/ |
H A D | mediamonitor.h | 42 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)
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H A D | typingmonitor.h | 34 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_;
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/external/chromium_org/third_party/libjingle/source/talk/p2p/client/ |
H A D | connectivitychecker_unittest.cc | 43 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 [all...] |
/external/chromium_org/third_party/libjingle/source/talk/p2p/base/ |
H A D | dtlstransport.h | 34 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 [all...] |
H A D | 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 [all...] |
H A D | 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 [all...] |
H A D | portproxy.h | 34 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 [all...] |
/external/chromium_org/third_party/libjingle/source/talk/examples/peerconnection/client/ |
H A D | peer_connection_client.h | 57 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 [all...] |
/external/chromium_org/third_party/libjingle/source/talk/media/base/ |
H A D | 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 [all...] |
H A D | 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 [all...] |
/external/chromium_org/third_party/libjingle/source/talk/app/webrtc/ |
H A D | 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 [all...] |
/external/chromium_org/jingle/glue/ |
H A D | 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 [all...] |
/external/chromium_org/third_party/libjingle/source/talk/app/webrtc/objc/ |
H A D | RTCMediaSource+Internal.h | 35 rtc::scoped_refptr<webrtc::MediaSourceInterface> mediaSource; 38 (rtc::scoped_refptr<webrtc::MediaSourceInterface>)mediaSource;
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H A D | RTCMediaStream+Internal.h | 35 rtc::scoped_refptr<webrtc::MediaStreamInterface> mediaStream; 38 (rtc::scoped_refptr<webrtc::MediaStreamInterface>)mediaStream;
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H A D | RTCMediaStreamTrack+Internal.h | 35 rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> mediaTrack; 38 (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)mediaTrack;
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/external/chromium_org/remoting/client/plugin/ |
H A D | pepper_address_resolver.h | 14 // 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_;
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/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/ |
H A D | audio_checksum.h | 45 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];
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/external/chromium_org/remoting/test/ |
H A D | 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 [all...] |
/external/chromium_org/third_party/libjingle/source/talk/examples/turnserver/ |
H A D | 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 [all...] |