/external/chromium_org/third_party/webrtc/video_engine/test/libvietest/include/ |
H A D | tb_interfaces.h | 41 webrtc::ViERTP_RTCP* rtp_rtcp; member in class:TbInterfaces
|
/external/chromium_org/third_party/webrtc/video_engine/ |
H A D | vie_remb.cc | 17 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" 38 void VieRemb::AddReceiveChannel(RtpRtcp* rtp_rtcp) { argument 39 assert(rtp_rtcp); 42 if (std::find(receive_modules_.begin(), receive_modules_.end(), rtp_rtcp) != 48 receive_modules_.push_back(rtp_rtcp); 51 void VieRemb::RemoveReceiveChannel(RtpRtcp* rtp_rtcp) { argument 52 assert(rtp_rtcp); 57 if ((*it) == rtp_rtcp) { 64 void VieRemb::AddRembSender(RtpRtcp* rtp_rtcp) { argument 76 RemoveRembSender(RtpRtcp* rtp_rtcp) argument [all...] |
H A D | vie_sync_module.cc | 13 #include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h" 14 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" 28 const RtpRtcp& rtp_rtcp, const RtpReceiver& receiver) { 37 if (0 != rtp_rtcp.RemoteNTP(&ntp_secs, 27 UpdateMeasurements(StreamSynchronization::Measurements* stream, const RtpRtcp& rtp_rtcp, const RtpReceiver& receiver) argument
|
H A D | vie_receiver.cc | 16 #include "webrtc/modules/rtp_rtcp/interface/fec_receiver.h" 17 #include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h" 18 #include "webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h" 19 #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" 20 #include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h" 21 #include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h" 22 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" 331 RtpRtcp* rtp_rtcp = *it++; local 332 rtp_rtcp [all...] |
H A D | vie_channel.cc | 20 #include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h" 21 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" 232 RtpRtcp* rtp_rtcp = *it; local 233 module_process_thread_.DeRegisterModule(rtp_rtcp); 234 delete rtp_rtcp; 311 RtpRtcp* rtp_rtcp = *it; local 313 rtp_rtcp->SetRTCPStatus(rtp_rtcp_->RTCP()); 316 rtp_rtcp->SetStorePacketsStatus(true, nack_history_size_sender_); 318 rtp_rtcp 343 RtpRtcp* rtp_rtcp = simulcast_rtp_rtcp_.back(); local 358 RtpRtcp* rtp_rtcp = *it; local 404 RtpRtcp* rtp_rtcp = simulcast_rtp_rtcp_.back(); local 548 RtpRtcp* rtp_rtcp = *it; local 590 RtpRtcp* rtp_rtcp = *it; local 600 RtpRtcp* rtp_rtcp = *it; local 640 RtpRtcp* rtp_rtcp = *it; local 828 RtpRtcp* rtp_rtcp = GetRtpRtcpModule(simulcast_idx); local 847 RtpRtcp* rtp_rtcp = GetRtpRtcpModule(idx); local 1126 RtpRtcp* rtp_rtcp = *it; local 1135 RtpRtcp* rtp_rtcp = *it; local 1198 RtpRtcp* rtp_rtcp = *it; local 1220 RtpRtcp* rtp_rtcp = *it; local 1288 RtpRtcp* rtp_rtcp = *it; local 1301 RtpRtcp* rtp_rtcp = *it; local 1316 RtpRtcp* rtp_rtcp = *it; local 1405 RtpRtcp* rtp_rtcp = *it; local 1422 RtpRtcp* ViEChannel::rtp_rtcp() { function in class:webrtc::ViEChannel 1557 RtpRtcp* rtp_rtcp = CreateRtpRtcpModule(); local [all...] |
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
H A D | remote_ntp_time_estimator.cc | 11 #include "webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h" 13 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" 30 RtpRtcp* rtp_rtcp) { 31 assert(rtp_rtcp); 33 rtp_rtcp->RTT(ssrc, &rtt, NULL, NULL, NULL); 42 if (0 != rtp_rtcp->RemoteNTP(&ntp_secs, 29 UpdateRtcpTimestamp(uint32_t ssrc, RtpRtcp* rtp_rtcp) argument
|
H A D | remote_ntp_time_estimator_unittest.cc | 14 #include "webrtc/modules/rtp_rtcp/interface/remote_ntp_time_estimator.h" 15 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h" 58 void UpdateRtcpTimestamp(MockRtpRtcp* rtp_rtcp, bool expected_result) { argument 59 if (rtp_rtcp) { 60 EXPECT_CALL(*rtp_rtcp, RTT(_, _, _, _, _)) 65 estimator_.UpdateRtcpTimestamp(kTestSsrc, rtp_rtcp));
|
H A D | nack_rtx_unittest.cc | 18 #include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h" 19 #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" 20 #include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h" 21 #include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h" 22 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" 23 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" 55 TestRtpFeedback(RtpRtcp* rtp_rtcp) : rtp_rtcp_(rtp_rtcp) {} argument
|
/external/chromium_org/third_party/webrtc/video/ |
H A D | receive_statistics_proxy.cc | 21 ViERTP_RTCP* rtp_rtcp, 27 rtp_rtcp_(rtp_rtcp), 19 ReceiveStatisticsProxy(uint32_t ssrc, Clock* clock, ViERTP_RTCP* rtp_rtcp, ViECodec* codec, int channel) argument
|
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/test/testAPI/ |
H A D | test_api_rtcp.cc | 17 #include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h" 18 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" 19 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" 20 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h" 21 #include "webrtc/modules/rtp_rtcp/test/testAPI/test_api.h" 75 TestRtpFeedback(RtpRtcp* rtp_rtcp) : rtp_rtcp_(rtp_rtcp) {} argument
|
/external/chromium_org/third_party/webrtc/voice_engine/test/cmd_test/ |
H A D | voe_cmd_test.cc | 59 VoERTP_RTCP* rtp_rtcp = NULL; variable 133 rtp_rtcp = VoERTP_RTCP::GetInterface(m_voe); 194 if (rtp_rtcp) 195 rtp_rtcp->Release();
|
/external/chromium_org/third_party/webrtc/video_engine/test/auto_test/source/ |
H A D | vie_autotest_codec.cc | 149 webrtc::ViERTP_RTCP* rtp_rtcp = interfaces.rtp_rtcp; local 155 EXPECT_EQ(0, rtp_rtcp->SetRTCPStatus( 158 EXPECT_EQ(0, rtp_rtcp->SetKeyFrameRequestMethod( 160 EXPECT_EQ(0, rtp_rtcp->SetTMMBRStatus(video_channel, true)); 303 webrtc::ViERTP_RTCP* rtp_rtcp = interfaces.rtp_rtcp; local 312 EXPECT_EQ(0, rtp_rtcp->SetRTCPStatus( 314 EXPECT_EQ(0, rtp_rtcp->SetKeyFrameRequestMethod( 316 EXPECT_EQ(0, rtp_rtcp [all...] |
/external/chromium_org/third_party/webrtc/voice_engine/test/android/android_test/jni/ |
H A D | android_test.cc | 85 if (!veData1.rtp_rtcp) \ 128 VoERTP_RTCP* rtp_rtcp; member in struct:__anon16247 713 /* if (veData1.rtp_rtcp->SetREDStatus(channel, 1) != 0) 758 /* if (veData1.rtp_rtcp->SetREDStatus(channel, 0) != 0) 1145 if (veData1.rtp_rtcp->SetREDStatus(0, enable, -1) != 0) 1240 veData.rtp_rtcp = VoERTP_RTCP::GetInterface(veData.ve); 1241 if (!veData.rtp_rtcp) 1244 "Get rtp_rtcp sub-API failed"); 1364 if (veData.rtp_rtcp) 1366 if (0 != veData.rtp_rtcp [all...] |