/external/chromium_org/third_party/webrtc/base/ |
socketfactory.h | 17 namespace rtc { namespace 36 } // namespace rtc
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thread_checker.h | 33 namespace rtc { namespace 89 } // namespace rtc
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thread_checker_impl.h | 18 namespace rtc { namespace 49 } // namespace rtc
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timing.h | 18 namespace rtc { namespace 58 } // namespace rtc
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sharedexclusivelock.cc | 13 namespace rtc { namespace 22 shared_count_is_zero_.Wait(rtc::kForever); 44 } // namespace rtc
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gunit.h | 24 for (uint32 start = rtc::Time(); \ 25 !(ex) && rtc::Time() < start + timeout;) \ 26 rtc::Thread::Current()->ProcessMessages(1); 33 uint32 start = rtc::Time(); \ 35 while (!res && rtc::Time() < start + timeout) { \ 36 rtc::Thread::Current()->ProcessMessages(1); \
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autodetectproxy_unittest.cc | 17 namespace rtc { namespace 90 void OnWorkDone(rtc::SignalThread *thread) { 92 static_cast<rtc::AutoDetectProxy *>(thread); 104 TestCopesWithProxy(rtc::SocketAddress("localhost", 9999)); 108 TestCopesWithProxy(rtc::SocketAddress("invalid", 9999)); 112 TestCopesWithProxy(rtc::SocketAddress("::1", 9999)); 116 TestCopesWithProxy(rtc::SocketAddress("127.0.0.1", 9999)); 132 } // namespace rtc
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sslfingerprint.cc | 20 namespace rtc { namespace 23 const std::string& algorithm, const rtc::SSLIdentity* identity) { 32 const std::string& algorithm, const rtc::SSLCertificate* cert) { 46 if (algorithm.empty() || !rtc::IsFips180DigestAlgorithm(algorithm)) 53 char value[rtc::MessageDigest::kMaxSize]; 54 value_len = rtc::hex_decode_with_delimiter(value, sizeof(value), 82 rtc::hex_encode_with_delimiter( 96 } // namespace rtc
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/external/chromium_org/third_party/webrtc/sound/ |
platformsoundsystemfactory.cc | 16 namespace rtc { namespace 40 } // namespace rtc
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soundsysteminterface.cc | 15 namespace rtc { namespace 29 } // namespace rtc
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/external/chromium_org/third_party/libjingle/source/talk/p2p/base/ |
asyncstuntcpsocket.cc | 56 rtc::AsyncSocket* socket, 57 const rtc::SocketAddress& bind_address, 58 const rtc::SocketAddress& remote_address) { 64 rtc::AsyncSocket* socket, bool listen) 65 : rtc::AsyncTCPSocketBase(socket, listen, kBufSize) { 69 const rtc::PacketOptions& options) { 104 rtc::SocketAddress remote_addr(GetRemoteAddress()); 129 rtc::CreatePacketTime(0)); 139 rtc::AsyncSocket* socket) { 147 rtc::GetBE16(static_cast<const char*>(data) + kPacketLenOffset) [all...] |
turnserver.cc | 75 class TurnServer::Allocation : public rtc::MessageHandler, 79 rtc::Thread* thread, const Connection& conn, 80 rtc::AsyncPacketSocket* server_socket, 108 void OnExternalPacket(rtc::AsyncPacketSocket* socket, 110 const rtc::SocketAddress& addr, 111 const rtc::PacketTime& packet_time); 114 bool HasPermission(const rtc::IPAddress& addr); 115 void AddPermission(const rtc::IPAddress& addr); 116 Permission* FindPermission(const rtc::IPAddress& addr) const; 118 Channel* FindChannel(const rtc::SocketAddress& addr) const [all...] |
/external/chromium_org/third_party/libjingle/source/talk/app/webrtc/ |
portallocatorfactory.cc | 38 using rtc::scoped_ptr; 40 rtc::scoped_refptr<PortAllocatorFactoryInterface> 42 rtc::Thread* worker_thread) { 43 rtc::RefCountedObject<PortAllocatorFactory>* allocator = 44 new rtc::RefCountedObject<PortAllocatorFactory>(worker_thread); 48 PortAllocatorFactory::PortAllocatorFactory(rtc::Thread* worker_thread) 49 : network_manager_(new rtc::BasicNetworkManager()), 50 socket_factory_(new rtc::BasicPacketSocketFactory(worker_thread)) {
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streamcollection.h | 41 static rtc::scoped_refptr<StreamCollection> Create() { 42 rtc::RefCountedObject<StreamCollection>* implementation = 43 new rtc::RefCountedObject<StreamCollection>(); 47 static rtc::scoped_refptr<StreamCollection> Create( 49 rtc::RefCountedObject<StreamCollection>* implementation = 50 new rtc::RefCountedObject<StreamCollection>(streams); 118 typedef std::vector<rtc::scoped_refptr<MediaStreamInterface> >
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/external/chromium_org/third_party/libjingle/source/talk/app/webrtc/test/ |
fakedtlsidentityservice.h | 68 public rtc::MessageHandler { 76 rtc::scoped_refptr<DTLSIdentityRequestObserver> observer; 78 typedef rtc::TypedMessageData<Request> MessageData; 92 rtc::Thread::Current()->Post(this, MSG_FAILURE, msg); 94 rtc::Thread::Current()->Post(this, MSG_SUCCESS, msg); 105 // rtc::MessageHandler implementation. 106 void OnMessage(rtc::Message* msg) { 128 rtc::SSLIdentity::PemToDer("CERTIFICATE", kCERT_PEM, der_cert); 129 rtc::SSLIdentity::PemToDer("RSA PRIVATE KEY",
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/external/chromium_org/third_party/libjingle/source/talk/media/devices/ |
libudevsymboltable.cc | 44 bool IsWrongLibUDevAbiVersion(rtc::DllHandle libudev_0) { 45 rtc::DllHandle libudev_1 = dlopen("libudev.so.1",
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/external/chromium_org/jingle/notifier/communicator/ |
connection_settings_unittest.cc | 47 rtc::SocketAddress("supports_ssltcp.com", 100), 52 rtc::SocketAddress("supports_ssltcp.com", 443), 57 rtc::SocketAddress("does_not_support_ssltcp.com", 200), 77 rtc::SocketAddress("supports_ssltcp.com", 443), 82 rtc::SocketAddress("supports_ssltcp.com", 100), 87 rtc::SocketAddress("does_not_support_ssltcp.com", 200),
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/external/chromium_org/third_party/libjingle/source/talk/app/webrtc/objc/ |
RTCMediaSource.mm | 37 rtc::scoped_refptr<webrtc::MediaSourceInterface> _mediaSource; 49 (rtc::scoped_refptr<webrtc::MediaSourceInterface>)mediaSource { 61 - (rtc::scoped_refptr<webrtc::MediaSourceInterface>)mediaSource {
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/external/chromium_org/remoting/client/plugin/ |
pepper_packet_socket_factory.cc | 86 class UdpPacketSocket : public rtc::AsyncPacketSocket { 93 bool Init(const rtc::SocketAddress& local_address, 97 // rtc::AsyncPacketSocket interface. 98 virtual rtc::SocketAddress GetLocalAddress() const OVERRIDE; 99 virtual rtc::SocketAddress GetRemoteAddress() const OVERRIDE; 101 const rtc::PacketOptions& options) OVERRIDE; 104 const rtc::SocketAddress& address, 105 const rtc::PacketOptions& options) OVERRIDE; 108 virtual int GetOption(rtc::Socket::Option opt, int* value) OVERRIDE; 109 virtual int SetOption(rtc::Socket::Option opt, int value) OVERRIDE [all...] |
/external/chromium_org/third_party/libjingle/source/talk/media/base/ |
rtpdump.h | 60 void WriteToByteBuffer(rtc::ByteBuffer* buf); 107 explicit RtpDumpReader(rtc::StreamInterface* stream) 118 virtual rtc::StreamResult ReadPacket(RtpDumpPacket* packet); 121 rtc::StreamResult ReadFileHeader(); 130 rtc::StreamInterface* stream_; 146 explicit RtpDumpLoopReader(rtc::StreamInterface* stream); 147 virtual rtc::StreamResult ReadPacket(RtpDumpPacket* packet); 189 explicit RtpDumpWriter(rtc::StreamInterface* stream); 195 rtc::StreamResult WriteRtpPacket(const void* data, size_t data_len) { 198 rtc::StreamResult WriteRtcpPacket(const void* data, size_t data_len) [all...] |
videocapturer.h | 129 public rtc::MessageHandler { 136 explicit VideoCapturer(rtc::Thread* thread); 277 screencast_max_pixels_ = rtc::_max(0, p); 313 void OnMessage(rtc::Message* message); 365 const rtc::RollingAccumulator<T>& data, 368 rtc::Thread* thread_; 371 rtc::scoped_ptr<VideoFrameFactory> frame_factory_; 372 rtc::scoped_ptr<VideoFormat> capture_format_; 374 rtc::scoped_ptr<VideoFormat> max_format_; 390 rtc::Timing frame_length_time_reporter_ [all...] |
/external/chromium_org/third_party/libjingle/source/talk/p2p/client/ |
httpportallocator.h | 39 namespace rtc { namespace 54 HttpPortAllocatorBase(rtc::NetworkManager* network_manager, 56 HttpPortAllocatorBase(rtc::NetworkManager* network_manager, 57 rtc::PacketSocketFactory* socket_factory, 69 void SetStunHosts(const std::vector<rtc::SocketAddress>& hosts) { 81 const std::vector<rtc::SocketAddress>& stun_hosts() const { 98 std::vector<rtc::SocketAddress> stun_hosts_; 114 const std::vector<rtc::SocketAddress>& stun_hosts, 144 std::vector<rtc::SocketAddress> stun_hosts_; 152 HttpPortAllocator(rtc::NetworkManager* network_manager [all...] |
httpportallocator.cc | 98 rtc::NetworkManager* network_manager, 99 rtc::PacketSocketFactory* socket_factory, 104 rtc::SocketAddress("stun.l.google.com", 19302)); 108 rtc::NetworkManager* network_manager, 113 rtc::SocketAddress("stun.l.google.com", 19302)); 127 const std::vector<rtc::SocketAddress>& stun_hosts, 146 for (std::vector<rtc::SocketAddress>::iterator it = stun_hosts_.begin(); 183 SendSessionRequest(host, rtc::HTTP_SECURE_PORT); 191 url = url + "?username=" + rtc::s_url_encode(username()) + 192 "&password=" + rtc::s_url_encode(password()) [all...] |
/external/chromium_org/third_party/libjingle/source/talk/session/media/ |
mediarecorder_unittest.cc | 42 rtc::StreamInterface* Open(const std::string& path) { 43 return rtc::Filesystem::OpenFile( 44 rtc::Pathname(path), "wb"); 53 EXPECT_TRUE(rtc::Filesystem::GetTemporaryFolder(path_, true, NULL)); 65 EXPECT_TRUE(rtc::Filesystem::DeleteFile(path_)); 70 rtc::ByteBuffer buf; 75 rtc::StreamResult ReadPacket(RtpDumpPacket* packet) { 78 stream_.reset(rtc::Filesystem::OpenFile(path_, "rb")); 84 rtc::Pathname path_; 85 rtc::scoped_ptr<RtpDumpSink> sink_ [all...] |
typingmonitor.cc | 37 rtc::Thread* worker_thread, 55 rtc::MessageList messages; 78 muted_at_ = rtc::Time(); 92 rtc::MessageList removed; 105 void TypingMonitor::OnMessage(rtc::Message* msg) { 111 << "ms ago, unmuting after " << rtc::TimeSince(muted_at_) 119 rtc::Thread::Current()->PostDelayed(expiry_time, this, 0);
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