| /external/chromium_org/third_party/webrtc/base/ |
| socketfactory.h | 17 namespace rtc { namespace
36 } // namespace rtc
|
| thread_checker.h | 33 namespace rtc { namespace
89 } // namespace rtc
|
| thread_checker_impl.h | 18 namespace rtc { namespace
49 } // namespace rtc
|
| timing.h | 18 namespace rtc { namespace
58 } // namespace rtc
|
| sharedexclusivelock.cc | 13 namespace rtc { namespace
22 shared_count_is_zero_.Wait(rtc::kForever);
44 } // namespace rtc
|
| 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); \
|
| 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
|
| 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
|
| /external/chromium_org/third_party/webrtc/sound/ |
| platformsoundsystemfactory.cc | 16 namespace rtc { namespace
40 } // namespace rtc
|
| soundsysteminterface.cc | 15 namespace rtc { namespace
29 } // namespace rtc
|
| /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)) {
|
| 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> >
|
| /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",
|
| /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",
|
| /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),
|
| /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 {
|
| /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);
|
