| /external/chromium_org/third_party/webrtc/video/ |
| H A D | transport_adapter.h | 33 newapi::Transport *transport_; member in class:webrtc::internal::TransportAdapter
|
| /external/chromium_org/third_party/webrtc/video_engine/ |
| H A D | vie_sender.h | 51 Transport* transport_; member in class:webrtc::ViESender
|
| /external/chromium_org/third_party/webrtc/voice_engine/test/auto_test/fixtures/ |
| H A D | before_streaming_fixture.h | 47 LoopBackTransport* transport_; member in class:BeforeStreamingFixture
|
| /external/chromium_org/media/cast/net/rtp/ |
| H A D | rtp_sender.h | 76 PacedSender* const transport_; member in class:media::cast::RtpSender
|
| H A D | rtp_packetizer.h | 60 PacedSender* const transport_; // Not owned by this class. member in class:media::cast::RtpPacketizer
|
| H A D | rtp_packetizer_unittest.cc | 110 transport_.reset(new TestRtpPacketTransport(config_));
115 transport_.get(),
141 scoped_ptr<TestRtpPacketTransport> transport_; member in class:media::cast::RtpPacketizerTest
151 transport_->set_expected_number_of_packets(expected_num_of_packets);
152 transport_->set_rtp_timestamp(video_frame_.rtp_timestamp);
158 EXPECT_EQ(expected_num_of_packets, transport_->number_of_packets_received());
166 transport_->set_expected_number_of_packets(expected_num_of_packets);
167 transport_->set_rtp_timestamp(video_frame_.rtp_timestamp);
175 EXPECT_EQ(expected_num_of_packets, transport_->number_of_packets_received());
|
| /external/chromium_org/jingle/glue/ |
| H A D | proxy_resolving_client_socket.h | 89 scoped_ptr<net::ClientSocketHandle> transport_; member in class:jingle_glue::ProxyResolvingClientSocket
|
| /external/chromium_org/media/cast/test/utility/ |
| H A D | in_process_receiver.h | 75 // Helper method that creates |transport_| and |cast_receiver_|, starts
76 // |transport_| receiving, and requests the first audio/video frame.
80 // Helper method that destroys |transport_| and |cast_receiver_|.
109 scoped_ptr<UdpTransport> transport_; member in class:media::cast::InProcessReceiver
|
| /external/chromium_org/media/cast/net/ |
| H A D | cast_transport_sender_impl.h | 140 scoped_ptr<UdpTransport> transport_; member in class:media::cast::CastTransportSenderImpl
|
| H A D | cast_transport_sender_impl_unittest.cc | 90 &transport_));
111 &transport_));
126 &transport_));
161 FakePacketSender transport_; member in class:media::cast::CastTransportSenderImplTest
197 EXPECT_EQ(4, transport_.packets_sent());
205 transport_.SetPaused(true);
220 transport_.SetPaused(false);
225 EXPECT_EQ(6, transport_.packets_sent());
242 EXPECT_EQ(4, transport_.packets_sent());
248 transport_
[all...] |
| /external/chromium_org/media/cast/sender/ |
| H A D | audio_sender_unittest.cc | 89 &transport_));
102 TestPacketSender transport_; member in class:media::cast::AudioSenderTest
120 EXPECT_LE(1, transport_.number_of_rtp_packets());
121 EXPECT_LE(1, transport_.number_of_rtcp_packets());
140 EXPECT_LE(1, transport_.number_of_rtp_packets());
141 EXPECT_LE(1, transport_.number_of_rtcp_packets());
|
| H A D | video_sender_unittest.cc | 158 &transport_));
246 TestPacketSender transport_; member in class:media::cast::VideoSenderTest
265 EXPECT_LE(1, transport_.number_of_rtp_packets());
266 EXPECT_LE(1, transport_.number_of_rtcp_packets());
309 EXPECT_LE(1, transport_.number_of_rtp_packets());
310 EXPECT_LE(1, transport_.number_of_rtcp_packets());
317 EXPECT_LE(1, transport_.number_of_rtcp_packets());
345 transport_.number_of_rtp_packets() + transport_.number_of_rtcp_packets());
394 const int number_of_packets_sent = transport_
[all...] |
| /external/chromium_org/net/socket/ |
| H A D | socks_client_socket.h | 100 scoped_ptr<ClientSocketHandle> transport_; member in class:net::SOCKSClientSocket
|
| H A D | socks5_client_socket.h | 121 scoped_ptr<ClientSocketHandle> transport_; member in class:net::SOCKS5ClientSocket
|
| H A D | ssl_client_socket_nss.h | 166 scoped_ptr<ClientSocketHandle> transport_; member in class:net::SSLClientSocketNSS
|
| H A D | ssl_client_socket_openssl_unittest.cc | 129 transport_.reset(new TCPClientSocket(
132 transport_->Connect(callback_.callback()));
161 sock_ = CreateSSLClientSocket(transport_.Pass(),
192 scoped_ptr<StreamSocket> transport_; member in class:net::__anon9535::SSLClientSocketOpenSSLClientAuthTest
|
| /external/chromium_org/extensions/browser/api/cast_channel/ |
| H A D | cast_transport_unittest.cc | 181 transport_.reset(new CastTransport(&mock_socket_, &delegate_, logger_));
189 scoped_ptr<CastTransport> transport_; member in class:extensions::core_api::cast_channel::CastTransportTest
208 transport_->SendMessage(
236 transport_->SendMessage(
255 transport_->SendMessage(
273 transport_->SendMessage(
297 transport_->SendMessage(
310 transport_->SendMessage(
346 transport_->StartReadLoop();
391 transport_
[all...] |
| /external/chromium_org/net/http/ |
| H A D | http_proxy_client_socket.h | 147 scoped_ptr<ClientSocketHandle> transport_; member in class:net::HttpProxyClientSocket
|
| /external/chromium_org/third_party/libjingle/source/talk/p2p/base/ |
| H A D | transport_unittest.cc | 63 transport_(new FakeTransport(
69 transport_->SignalConnecting.connect(this, &TransportTest::OnConnecting);
70 transport_->SignalCompleted.connect(this, &TransportTest::OnCompleted);
71 transport_->SignalFailed.connect(this, &TransportTest::OnFailed);
74 transport_->DestroyAllChannels();
82 transport_->CreateChannel(component));
85 transport_->DestroyChannel(1);
101 rtc::scoped_ptr<FakeTransport> transport_; member in class:TransportTest
139 transport_->ConnectChannels();
147 EXPECT_TRUE(transport_
[all...] |
| H A D | dtlstransportchannel.h | 194 return transport_;
247 Transport* transport_; // The transport_ that created us. member in class:cricket::DtlsTransportChannelWrapper
249 TransportChannelImpl* channel_; // Underlying channel, owned by transport_.
|
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| H A D | rtp_sender.h | 349 Transport *transport_; member in class:webrtc::RTPSender
|
| /external/chromium_org/ui/gfx/ |
| H A D | native_widget_types.h | 231 GLSurfaceHandle(PluginWindowHandle handle_, SurfaceType transport_) argument
233 transport_type(transport_),
|
| /external/chromium_org/media/cast/net/pacing/ |
| H A D | paced_sender.h | 168 PacketSender* transport_; // Not owned by this class. member in class:media::cast::PacedSender
|
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/testAPI/ |
| H A D | test_api_video.cc | 40 transport_ = new LoopBackTransport();
47 configuration.outgoing_transport = transport_;
59 transport_->SetSendModule(video_module_, &rtp_payload_registry_,
123 delete transport_;
132 LoopBackTransport* transport_; member in class:webrtc::RtpRtcpVideoTest
|
| /external/chromium_org/third_party/webrtc/voice_engine/test/auto_test/standard/ |
| H A D | mixing_test.cc | 38 transport_ = new LoopBackTransport(voe_network_);
41 delete transport_;
184 EXPECT_EQ(0, voe_network_->RegisterExternalTransport(stream, *transport_));
216 LoopBackTransport* transport_; member in class:webrtc::MixingTest
|