/external/webrtc/pc/ |
D | rtc_stats_traversal_unittest.cc | 28 transport_ = new RTCTransportStats("transport", 0); in RTCStatsTraversalTest() 33 initial_report_->AddStats(std::unique_ptr<const RTCStats>(transport_)); in RTCStatsTraversalTest() 73 RTCTransportStats* transport_; member in webrtc::RTCStatsTraversalTest 94 TakeReferencedStats({transport_}); in TEST_F() 95 EXPECT_VISITED(transport_); in TEST_F() 102 transport_->rtcp_transport_stats_id = "transport"; in TEST_F() 103 TakeReferencedStats({transport_}); in TEST_F() 104 EXPECT_VISITED(transport_); in TEST_F() 111 transport_->rtcp_transport_stats_id = "bogus-reference"; in TEST_F() 112 TakeReferencedStats({transport_}); in TEST_F() [all …]
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D | sctp_transport_unittest.cc | 109 SctpTransport* transport() { return transport_.get(); } in transport() 115 transport_ = new rtc::RefCountedObject<SctpTransport>( in CreateTransport() 125 transport_->SetDtlsTransport(dtls_transport_); in AddDtlsTransport() 138 return static_cast<FakeCricketSctpTransport*>(transport_->internal()); in CricketSctpTransport() 141 rtc::scoped_refptr<SctpTransport> transport_; member in webrtc::SctpTransportTest
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D | dtls_transport_unittest.cc | 55 DtlsTransport* transport() { return transport_.get(); } in transport() 65 transport_ = in CreateTransport() 70 auto fake_dtls1 = static_cast<FakeDtlsTransport*>(transport_->internal()); in CompleteDtlsHandshake() 82 rtc::scoped_refptr<DtlsTransport> transport_; member in webrtc::DtlsTransportTest
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D | rtp_transport_unittest.cc | 38 transport_ = transport; in SignalObserver() 63 if (packet_transport == transport_->rtp_packet_transport()) { in OnSentPacket() 66 ASSERT_EQ(transport_->rtcp_packet_transport(), packet_transport); in OnSentPacket() 78 RtpTransport* transport_ = nullptr; member in webrtc::SignalObserver
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/external/libbrillo/brillo/http/ |
D | http_request_test.cc | 45 transport_ = std::make_shared<MockTransport>(); in SetUp() 46 connection_ = std::make_shared<MockConnection>(transport_); in SetUp() 59 testing::Mock::VerifyAndClearExpectations(transport_.get()); in TearDown() 60 transport_.reset(); in TearDown() 64 std::shared_ptr<MockTransport> transport_; member in brillo::http::HttpRequestTest 69 Request request{"http://www.foo.bar", request_type::kPost, transport_}; in TEST_F() 77 Request request2{"http://www.foo.bar/baz", request_type::kGet, transport_}; in TEST_F() 83 Request request{"http://www.foo.bar", request_type::kPost, transport_}; in TEST_F() 89 Request request{"http://www.foo.bar", request_type::kPost, transport_}; in TEST_F() 95 Request request{"http://www.foo.bar", request_type::kPost, transport_}; in TEST_F() [all …]
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D | http_transport_curl_test.cc | 34 transport_ = std::make_shared<Transport>(curl_api_); in SetUp() 50 transport_.reset(); in TearDown() 56 std::shared_ptr<Transport> transport_; member in brillo::http::curl::HttpCurlTransportTest 72 auto connection = transport_->CreateConnection("http://foo.bar/get", in TEST_F() 120 auto connection = transport_->CreateConnection( in TEST_F() 134 auto connection = transport_->CreateConnection( in TEST_F() 150 auto connection = transport_->CreateConnection( in TEST_F() 170 auto connection = transport_->CreateConnection( in TEST_F() 188 auto connection = transport_->CreateConnection( in TEST_F() 201 transport_ = std::make_shared<Transport>(curl_api_); in SetUp() [all …]
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D | http_request.cc | 110 : transport_(transport), request_url_(url), method_(method) { in Request() 112 if (!transport_) in Request() 113 transport_ = http::Transport::CreateDefault(); in Request() 121 DCHECK(transport_) << "Request already sent"; in AddRange() 130 DCHECK(transport_) << "Request already sent"; in AddRange() 140 transport_.reset(); // Indicate that the response has been received in GetResponseAndBlock() 148 transport_->RunCallbackAsync( in GetResponse() 155 transport_.reset(); // Indicate that the request has been dispatched. in GetResponse() 160 DCHECK(transport_) << "Request already sent"; in SetAccept() 169 DCHECK(transport_) << "Request already sent"; in SetContentType() [all …]
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D | http_connection.h | 47 : transport_(transport) {} in Connection() 99 std::shared_ptr<Transport> transport_;
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D | http_connection_curl_test.cc | 155 transport_ = std::make_shared<MockTransport>(); in SetUp() 159 handle_, request_type::kPost, curl_api_, transport_); in SetUp() 166 transport_.reset(); in TearDown() 172 std::shared_ptr<MockTransport> transport_; member in brillo::http::curl::HttpCurlConnectionTest 216 EXPECT_CALL(*transport_, StartAsyncTransfer(connection_.get(), _, _)) in TEST_F()
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D | http_connection_fake.cc | 45 fake::Transport* transport = static_cast<fake::Transport*>(transport_.get()); in FinishRequest() 71 transport_->RunCallbackAsync(FROM_HERE, in FinishRequestAsync()
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/external/webrtc/modules/rtp_rtcp/source/ |
D | rtp_sender_unittest.cc | 360 config.outgoing_transport = &transport_; in SetUpRtpSender() 390 LoopbackTransportTest transport_; member in webrtc::RtpSenderTest 609 config.outgoing_transport = &transport_; in TEST_P() 645 config.outgoing_transport = &transport_; in TEST_P() 674 const auto& packet = transport_.last_sent_packet(); in TEST_P() 678 EXPECT_EQ(transport_.last_options_.packet_id, transport_seq_no); in TEST_P() 679 EXPECT_TRUE(transport_.last_options_.included_in_allocation); in TEST_P() 685 config.outgoing_transport = &transport_; in TEST_P() 696 EXPECT_FALSE(transport_.last_options_.is_retransmit); in TEST_P() 706 EXPECT_TRUE(transport_.last_options_.included_in_feedback); in TEST_P() [all …]
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D | rtp_sender_audio_unittest.cc | 75 config.outgoing_transport = &transport_; in RtpSenderAudioTest() 84 LoopbackTransportTest transport_; member in webrtc::RtpSenderAudioTest 101 auto sent_payload = transport_.last_sent_packet().payload(); in TEST_F() 122 auto sent_payload = transport_.last_sent_packet().payload(); in TEST_F() 127 EXPECT_TRUE(transport_.last_sent_packet().GetExtension<AudioLevel>( in TEST_F() 145 EXPECT_FALSE(transport_.last_sent_packet() in TEST_F() 164 transport_.last_sent_packet() in TEST_F() 209 EXPECT_TRUE(transport_.last_sent_packet().Marker()); in TEST_F() 216 EXPECT_FALSE(transport_.last_sent_packet().Marker()); in TEST_F()
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D | rtp_sender_video_unittest.cc | 172 config.outgoing_transport = &transport_; in RtpSenderVideoTest() 193 LoopbackTransportTest transport_; member in webrtc::__anonc30e10920111::RtpSenderVideoTest 212 transport_.last_sent_packet().GetExtension<VideoOrientation>(&rotation)); in TEST_P() 237 EXPECT_TRUE(transport_.last_sent_packet().GetExtension<VideoTimingExtension>( in TEST_P() 264 transport_.last_sent_packet().GetExtension<VideoOrientation>(&rotation)); in TEST_P() 287 transport_.last_sent_packet().GetExtension<VideoOrientation>(&rotation)); in TEST_P() 520 ASSERT_EQ(transport_.packets_sent(), 1); in TEST_P() 522 ASSERT_TRUE(transport_.last_sent_packet() in TEST_P() 546 EXPECT_EQ(transport_.packets_sent(), 2); in TEST_P() 549 transport_.last_sent_packet() in TEST_P() [all …]
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D | nack_rtx_unittest.cc | 123 transport_(kTestRtxSsrc), in RtpRtcpRtxNackTest() 135 configuration.outgoing_transport = &transport_; in SetUp() 158 transport_.SetSendModule(rtp_rtcp_module_.get()); in SetUp() 159 media_receiver_ = transport_.stream_receiver_controller_.CreateReceiver( in SetUp() 197 transport_.expected_sequence_numbers_); in ExpectedPacketsReceived() 201 rtx_receiver_ = transport_.stream_receiver_controller_.CreateReceiver( in RunRtxTest() 204 transport_.DropEveryNthPacket(loss); in RunRtxTest() 231 RtxLoopBackTransport transport_; member in webrtc::RtpRtcpRtxNackTest 254 transport_.DropConsecutivePackets(2, kNumPacketsToDrop); in TEST_F() 270 EXPECT_FALSE(transport_.expected_sequence_numbers_.empty()); in TEST_F() [all …]
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D | rtp_rtcp_impl_unittest.cc | 105 transport_.SimulateNetworkDelay(kOneWayNetworkDelayMs, clock); in RtpRtcpModule() 112 SendTransport transport_; member in webrtc::__anonc3f987650111::RtpRtcpModule 132 int RtpSent() { return transport_.rtp_packets_sent_; } in RtpSent() 134 return transport_.last_rtp_header_.sequenceNumber; in LastRtpSequenceNumber() 137 return transport_.last_nack_list_; in LastNackListSent() 149 config.outgoing_transport = &transport_; in CreateModuleImpl() 197 sender_.transport_.SetRtpRtcpModule(receiver_.impl_.get()); in SetUp() 198 receiver_.transport_.SetRtpRtcpModule(sender_.impl_.get()); in SetUp() 344 sender_.transport_.SimulateNetworkDelay(0, &clock_); in TEST_F() 345 receiver_.transport_.SimulateNetworkDelay(0, &clock_); in TEST_F() [all …]
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D | rtp_rtcp_impl2_unittest.cc | 110 transport_.SimulateNetworkDelay(kOneWayNetworkDelayMs, time_controller); in RtpRtcpModule() 117 SendTransport transport_; member in webrtc::__anon350570d70111::RtpRtcpModule 137 int RtpSent() { return transport_.rtp_packets_sent_; } in RtpSent() 139 return transport_.last_rtp_header_.sequenceNumber; in LastRtpSequenceNumber() 142 return transport_.last_nack_list_; in LastNackListSent() 154 config.outgoing_transport = &transport_; in CreateModuleImpl() 202 sender_.transport_.SetRtpRtcpModule(receiver_.impl_.get()); in SetUp() 203 receiver_.transport_.SetRtpRtcpModule(sender_.impl_.get()); in SetUp() 356 sender_.transport_.SimulateNetworkDelay(0, &time_controller_); in TEST_F() 357 receiver_.transport_.SimulateNetworkDelay(0, &time_controller_); in TEST_F() [all …]
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D | rtcp_sender.cc | 57 : transport_(transport), event_log_(event_log) {} in PacketContainer() 66 if (transport_->SendRtcp(packet.data(), packet.size())) { in SendPackets() 77 Transport* transport_; member in webrtc::__anonc19026600111::PacketContainer 158 transport_(config.outgoing_transport), in RTCPSender() 183 RTC_DCHECK(transport_ != nullptr); in RTCPSender() 701 PacketContainer container(transport_, event_log_); in SendCompoundRTCP() 723 PacketContainer container(transport_, event_log_); in SendCompoundRTCPLocked() 1014 if (transport_->SendRtcp(packet.data(), packet.size())) { in SendCombinedRtcpPacket()
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/external/webrtc/video/ |
D | transport_adapter.cc | 19 : transport_(transport), enabled_(false) { in TransportAdapter() 31 return transport_->SendRtp(packet, length, options); in SendRtp() 38 return transport_->SendRtcp(packet, length); in SendRtcp()
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/external/webrtc/media/sctp/ |
D | sctp_transport.cc | 448 transport_(transport), in SctpTransport() 470 transport_ = nullptr; in ~SctpTransport() 476 transport_ = transport; in SetDtlsTransport() 688 if (!transport_) { in ConnectTransportSignals() 691 transport_->SignalWritableState.connect(this, in ConnectTransportSignals() 693 transport_->SignalReadPacket.connect(this, &SctpTransport::OnPacketRead); in ConnectTransportSignals() 694 transport_->SignalClosed.connect(this, &SctpTransport::OnClosed); in ConnectTransportSignals() 699 if (!transport_) { in DisconnectTransportSignals() 702 transport_->SignalWritableState.disconnect(this); in DisconnectTransportSignals() 703 transport_->SignalReadPacket.disconnect(this); in DisconnectTransportSignals() [all …]
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/external/webrtc/audio/voip/test/ |
D | audio_channel_unittest.cc | 53 &transport_, kLocalSsrc, task_queue_factory_.get(), in SetUp() 82 NiceMock<MockTransport> transport_; member in webrtc::__anonc02a49680111::AudioChannelTest 102 EXPECT_CALL(transport_, SendRtp).WillOnce(Invoke(loop_rtp)); in TEST_F() 131 EXPECT_CALL(transport_, SendRtp).WillOnce(Invoke(loop_rtp)); in TEST_F()
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D | audio_ingress_unittest.cc | 47 rtp_config.outgoing_transport = &transport_; in AudioIngressTest() 96 NiceMock<MockTransport> transport_; member in webrtc::__anonc756ecca0111::AudioIngressTest 119 EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(handle_rtp)); in TEST_F() 149 EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(handle_rtp)); in TEST_F() 172 EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(handle_rtp)); in TEST_F()
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D | audio_egress_unittest.cc | 59 rtp_rtcp_ = CreateRtpStack(&fake_clock_, &transport_, kRemoteSsrc); in AudioEgressTest() 102 NiceMock<MockTransport> transport_; member in webrtc::__anon7c5499780111::AudioEgressTest 129 EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(rtp_sent)); in TEST_F() 167 EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(rtp_sent)); in TEST_F() 201 EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(rtp_sent)); in TEST_F() 277 EXPECT_CALL(transport_, SendRtp).WillRepeatedly(Invoke(rtp_sent)); in TEST_F()
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D | voip_core_unittest.cc | 47 NiceMock<MockTransport> transport_; member in webrtc::__anonf018ce240111::VoipCoreTest 62 auto channel = voip_core_->CreateChannel(&transport_, 0xdeadc0de); in TEST_F() 83 auto channel = voip_core_->CreateChannel(&transport_, 0xdeadc0de); in TEST_F()
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/external/webrtc/call/ |
D | fake_network_pipe.cc | 47 transport_(transport) {} in NetworkPacket() 57 transport_(o.transport_) {} in NetworkPacket() 69 transport_ = o.transport_; in operator =()
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/external/webrtc/test/scenario/ |
D | audio_stream.cc | 134 sender_->transport_->packet_overhead().bytes()); in SendAudioStream() 228 send_stream_(sender, config, encoder_factory, sender->transport_.get()), in AudioStreamPair() 233 receiver->transport_.get()) {} in AudioStreamPair()
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