| /external/webrtc/webrtc/p2p/base/ |
| stunport.h | 21 namespace rtc { namespace
31 static UDPPort* Create(rtc::Thread* thread,
32 rtc::PacketSocketFactory* factory,
33 rtc::Network* network,
34 rtc::AsyncPacketSocket* socket,
48 static UDPPort* Create(rtc::Thread* thread,
49 rtc::PacketSocketFactory* factory,
50 rtc::Network* network,
51 const rtc::IPAddress& ip,
70 rtc::SocketAddress GetLocalAddress() const
[all...] |
| teststunserver.h | 23 static TestStunServer* Create(rtc::Thread* thread,
24 const rtc::SocketAddress& addr) {
25 rtc::AsyncSocket* socket =
27 rtc::AsyncUDPSocket* udp_socket =
28 rtc::AsyncUDPSocket::Create(socket, addr);
34 void set_fake_stun_addr(const rtc::SocketAddress& addr) {
39 explicit TestStunServer(rtc::AsyncUDPSocket* socket) : StunServer(socket) {}
42 const rtc::SocketAddress& remote_addr) override {
53 rtc::SocketAddress fake_stun_addr_;
|
| turnport.h | 24 namespace rtc { namespace
43 static TurnPort* Create(rtc::Thread* thread,
44 rtc::PacketSocketFactory* factory,
45 rtc::Network* network,
46 rtc::AsyncPacketSocket* socket,
57 static TurnPort* Create(rtc::Thread* thread,
58 rtc::PacketSocketFactory* factory,
59 rtc::Network* network,
60 const rtc::IPAddress& ip,
78 rtc::SocketAddress GetLocalAddress() const
[all...] |
| turnserver.h | 25 namespace rtc { namespace
44 TurnServerConnection(const rtc::SocketAddress& src,
46 rtc::AsyncPacketSocket* socket);
47 const rtc::SocketAddress& src() const { return src_; }
48 rtc::AsyncPacketSocket* socket() { return socket_; }
54 rtc::SocketAddress src_;
55 rtc::SocketAddress dst_;
57 rtc::AsyncPacketSocket* socket_;
65 class TurnServerAllocation : public rtc::MessageHandler,
69 rtc::Thread* thread
[all...] |
| /external/webrtc/webrtc/base/ |
| proxy_unittest.cc | 21 using rtc::Socket;
22 using rtc::Thread;
23 using rtc::SocketAddress;
33 class AutoDetectProxyRunner : public rtc::AutoDetectProxy {
46 ProxyTest() : ss_(new rtc::VirtualSocketServer(NULL)) {
48 socks_.reset(new rtc::SocksProxyServer(
50 https_.reset(new rtc::HttpListenServer());
57 rtc::SocketServer* ss() { return ss_.get(); }
59 rtc::ProxyType DetectProxyType(const SocketAddress& address) {
60 rtc::ProxyType type
[all...] |
| ssladapter_unittest.cc | 24 static rtc::AsyncSocket* CreateSocket(const rtc::SSLMode& ssl_mode) {
25 rtc::SocketAddress address(rtc::IPAddress(INADDR_ANY), 0);
27 rtc::AsyncSocket* socket = rtc::Thread::Current()->
29 address.family(), (ssl_mode == rtc::SSL_MODE_DTLS) ?
36 static std::string GetSSLProtocolName(const rtc::SSLMode& ssl_mode) {
37 return (ssl_mode == rtc::SSL_MODE_DTLS) ? "DTLS" : "TLS";
42 explicit SSLAdapterTestDummyClient(const rtc::SSLMode& ssl_mode
[all...] |
| windowpicker_unittest.cc | 24 if (!rtc::WindowPickerFactory::IsSupported()) {
28 rtc::scoped_ptr<rtc::WindowPicker> picker(
29 rtc::WindowPickerFactory::CreateWindowPicker());
31 rtc::WindowDescriptionList descriptions;
39 if (!rtc::WindowPickerFactory::IsSupported()) {
43 rtc::scoped_ptr<rtc::WindowPicker> picker(
44 rtc::WindowPickerFactory::CreateWindowPicker());
46 rtc::DesktopDescriptionList descriptions
[all...] |
| /external/webrtc/talk/app/webrtc/test/ |
| fakedtlsidentitystore.h | 101 public rtc::MessageHandler {
103 typedef rtc::TypedMessageData<rtc::scoped_refptr<
116 rtc::KeyType key_type,
117 const rtc::scoped_refptr<webrtc::DtlsIdentityRequestObserver>&
120 RTC_DCHECK(key_type == rtc::KT_RSA || should_fail_);
122 rtc::scoped_refptr<webrtc::DtlsIdentityRequestObserver>(observer));
123 rtc::Thread::Current()->Post(
127 static rtc::scoped_refptr<rtc::RTCCertificate> GenerateCertificate()
[all...] |
| /external/webrtc/webrtc/common_audio/resampler/include/ |
| push_resampler.h | 39 rtc::scoped_ptr<PushSincResampler> sinc_resampler_;
40 rtc::scoped_ptr<PushSincResampler> sinc_resampler_right_;
44 rtc::scoped_ptr<T[]> src_left_;
45 rtc::scoped_ptr<T[]> src_right_;
46 rtc::scoped_ptr<T[]> dst_left_;
47 rtc::scoped_ptr<T[]> dst_right_;
|
| /external/webrtc/webrtc/p2p/stunprober/ |
| stunprober_unittest.cc | 31 const rtc::SocketAddress kLocalAddr("192.168.0.1", 0);
32 const rtc::SocketAddress kStunAddr1("1.1.1.1", 3478);
33 const rtc::SocketAddress kStunAddr2("1.1.1.2", 3478);
34 const rtc::SocketAddress kFailedStunAddr("1.1.1.3", 3478);
35 const rtc::SocketAddress kStunMappedAddr("77.77.77.77", 0);
42 : main_(rtc::Thread::Current()),
43 pss_(new rtc::PhysicalSocketServer),
44 ss_(new rtc::VirtualSocketServer(pss_.get())),
47 stun_server_1_(cricket::TestStunServer::Create(rtc::Thread::Current(),
49 stun_server_2_(cricket::TestStunServer::Create(rtc::Thread::Current()
[all...] |
| /external/webrtc/talk/app/webrtc/java/jni/ |
| androidnetworkmonitor_jni.h | 38 class AndroidNetworkMonitor : public rtc::NetworkMonitorBase {
52 rtc::ThreadChecker thread_checker_;
56 class AndroidNetworkMonitorFactory : public rtc::NetworkMonitorFactory {
60 rtc::NetworkMonitorInterface* CreateNetworkMonitor() override {
|
| /external/webrtc/webrtc/sound/ |
| platformsoundsystem.cc | 20 namespace rtc { namespace
31 } // namespace rtc
|
| /external/webrtc/webrtc/system_wrappers/source/ |
| cpu_info.cc | 18 return static_cast<uint32_t>(rtc::SystemInfo::GetMaxCpus());
|
| /external/webrtc/webrtc/test/ |
| test_main.cc | 27 if (rtc::LogMessage::GetLogToDebug() > rtc::LS_INFO)
28 rtc::LogMessage::LogToDebug(rtc::LS_INFO);
|
| /external/webrtc/talk/app/webrtc/objc/ |
| RTCLogging.mm | 32 rtc::LoggingSeverity RTCGetNativeLoggingSeverity(RTCLoggingSeverity severity) {
35 return rtc::LS_VERBOSE;
37 return rtc::LS_INFO;
39 return rtc::LS_WARNING;
41 return rtc::LS_ERROR;
53 rtc::LogMessage::LogToDebug(RTCGetNativeLoggingSeverity(severity));
|
| /external/webrtc/talk/app/webrtc/ |
| rtpreceiverinterface.h | 43 class RtpReceiverInterface : public rtc::RefCountInterface {
45 virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0;
59 PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
|
| dtlsidentitystore_unittest.cc | 54 void OnSuccess(rtc::scoped_ptr<rtc::SSLIdentity> identity) override {
81 : worker_thread_(new rtc::Thread()),
82 store_(new DtlsIdentityStoreImpl(rtc::Thread::Current(),
85 new rtc::RefCountedObject<MockDtlsIdentityRequestObserver>()) {
91 rtc::InitializeSSL();
94 rtc::CleanupSSL();
97 rtc::scoped_ptr<rtc::Thread> worker_thread_;
98 rtc::scoped_ptr<DtlsIdentityStoreImpl> store_
[all...] |
| peerconnectionfactoryproxy.h | 44 rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
47 rtc::scoped_ptr<cricket::PortAllocator> a3,
48 rtc::scoped_ptr<DtlsIdentityStoreInterface> a4,
50 return owner_thread_->Invoke<rtc::scoped_refptr<PeerConnectionInterface>>(
51 rtc::Bind(&PeerConnectionFactoryProxy::CreatePeerConnection_ot, this,
54 PROXY_METHOD1(rtc::scoped_refptr<MediaStreamInterface>,
56 PROXY_METHOD1(rtc::scoped_refptr<AudioSourceInterface>,
58 PROXY_METHOD2(rtc::scoped_refptr<VideoSourceInterface>,
61 PROXY_METHOD2(rtc::scoped_refptr<VideoTrackInterface>,
63 PROXY_METHOD2(rtc::scoped_refptr<AudioTrackInterface>
[all...] |
| audiotrack.h | 47 const rtc::scoped_refptr<AudioSourceInterface>& source);
51 static rtc::scoped_refptr<AudioTrack> Create(
53 const rtc::scoped_refptr<AudioSourceInterface>& source);
69 const rtc::scoped_refptr<AudioSourceInterface> audio_source_;
70 rtc::ThreadChecker thread_checker_;
|
| /external/webrtc/webrtc/base/objc/ |
| RTCLogging.mm | 15 rtc::LoggingSeverity RTCGetNativeLoggingSeverity(RTCLoggingSeverity severity) {
18 return rtc::LS_VERBOSE;
20 return rtc::LS_INFO;
22 return rtc::LS_WARNING;
24 return rtc::LS_ERROR;
36 rtc::LogMessage::LogToDebug(RTCGetNativeLoggingSeverity(severity));
|
| /external/webrtc/webrtc/common_video/include/ |
| i420_buffer_pool.h | 31 rtc::scoped_refptr<VideoFrameBuffer> CreateBuffer(int width, int height);
37 rtc::ThreadChecker thread_checker_;
38 std::list<rtc::scoped_refptr<I420Buffer>> buffers_;
|
| /external/tlsdate/src/ |
| routeup.c | 40 routeup_setup (struct routeup *rtc)
46 rtc->netlinkfd = socket (AF_NETLINK, SOCK_RAW, NETLINK_ROUTE);
47 if (rtc->netlinkfd < 0)
52 if (bind (rtc->netlinkfd, (struct sockaddr *) &sa, sizeof (sa)) < 0)
55 close (rtc->netlinkfd);
58 if (fcntl (rtc->netlinkfd, F_SETFL, O_NONBLOCK) < 0)
61 close (rtc->netlinkfd);
73 routeup_process (struct routeup *rtc)
79 if ( (ret = read (rtc->netlinkfd, buf, sizeof (buf))) < 0)
99 while (read (rtc->netlinkfd, buf, sizeof (buf)) > 0
149 struct routeup rtc; local
[all...] |
| /external/webrtc/webrtc/examples/stunserver/ |
| stunserver_main.cc | 28 rtc::SocketAddress server_addr;
34 rtc::Thread *pthMain = rtc::Thread::Current();
36 rtc::AsyncUDPSocket* server_socket =
37 rtc::AsyncUDPSocket::Create(pthMain->socketserver(), server_addr);
|
| /external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
| audio_checksum.h | 44 checksum_.Finish(checksum_result_, rtc::Md5Digest::kSize);
46 return rtc::hex_encode(checksum_result_, rtc::Md5Digest::kSize);
50 rtc::Md5Digest checksum_;
51 char checksum_result_[rtc::Md5Digest::kSize];
|
| /external/webrtc/webrtc/libjingle/xmpp/ |
| xmppsocket.h | 24 namespace rtc { namespace
28 extern rtc::AsyncSocket* cricket_socket_;
41 virtual bool Connect(const rtc::SocketAddress& addr);
52 void OnReadEvent(rtc::AsyncSocket * socket);
53 void OnWriteEvent(rtc::AsyncSocket * socket);
54 void OnConnectEvent(rtc::AsyncSocket * socket);
55 void OnCloseEvent(rtc::AsyncSocket * socket, int error);
57 void OnEvent(rtc::StreamInterface* stream, int events, int err);
60 rtc::AsyncSocket * cricket_socket_;
62 rtc::StreamInterface *stream_
[all...] |
