/external/chromium_org/third_party/webrtc/video/ |
transport_adapter.h | 33 newapi::Transport *transport_; member in class:webrtc::internal::TransportAdapter
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/external/chromium_org/third_party/webrtc/video_engine/ |
vie_sender.h | 51 Transport* transport_; member in class:webrtc::ViESender
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/external/chromium_org/third_party/webrtc/voice_engine/test/auto_test/fixtures/ |
before_streaming_fixture.h | 47 LoopBackTransport* transport_; member in class:BeforeStreamingFixture
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/external/chromium_org/media/cast/net/rtp/ |
rtp_sender.h | 76 PacedSender* const transport_; member in class:media::cast::RtpSender
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rtp_packetizer.h | 60 PacedSender* const transport_; // Not owned by this class. member in class:media::cast::RtpPacketizer
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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());
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/external/chromium_org/jingle/glue/ |
proxy_resolving_client_socket.h | 89 scoped_ptr<net::ClientSocketHandle> transport_; member in class:jingle_glue::ProxyResolvingClientSocket
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/external/chromium_org/media/cast/test/utility/ |
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
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/external/chromium_org/media/cast/net/ |
cast_transport_sender_impl.h | 140 scoped_ptr<UdpTransport> transport_; member in class:media::cast::CastTransportSenderImpl
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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_.SetPaused(true) [all...] |
/external/chromium_org/media/cast/sender/ |
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());
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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_.number_of_rtp_packets() [all...] |
/external/chromium_org/net/socket/ |
socks_client_socket.h | 100 scoped_ptr<ClientSocketHandle> transport_; member in class:net::SOCKSClientSocket
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socks5_client_socket.h | 121 scoped_ptr<ClientSocketHandle> transport_; member in class:net::SOCKS5ClientSocket
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ssl_client_socket_nss.h | 166 scoped_ptr<ClientSocketHandle> transport_; member in class:net::SSLClientSocketNSS
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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::__anon14160::SSLClientSocketOpenSSLClientAuthTest
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/external/chromium_org/extensions/browser/api/cast_channel/ |
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_->StartReadLoop() [all...] |
/external/chromium_org/net/http/ |
http_proxy_client_socket.h | 147 scoped_ptr<ClientSocketHandle> transport_; member in class:net::HttpProxyClientSocket
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/external/chromium_org/third_party/libjingle/source/talk/p2p/base/ |
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_->CreateChannel(1) != NULL) [all...] |
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_.
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
rtp_sender.h | 349 Transport *transport_; member in class:webrtc::RTPSender
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/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/testAPI/ |
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
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/external/chromium_org/third_party/webrtc/voice_engine/test/auto_test/standard/ |
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
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rtp_rtcp_test.cc | 84 transport_ = new LoopBackTransport(voe_network_); 86 *transport_)); 100 delete transport_; 104 LoopBackTransport* transport_; member in class:RtpRtcpTest
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/external/chromium_org/media/cast/net/pacing/ |
paced_sender.h | 168 PacketSender* transport_; // Not owned by this class. member in class:media::cast::PacedSender
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