Searched defs:rtp_rtcp (Results 1 - 13 of 13) sorted by relevance

/external/chromium_org/third_party/webrtc/video_engine/test/libvietest/include/
H A Dtb_interfaces.h41 webrtc::ViERTP_RTCP* rtp_rtcp; member in class:TbInterfaces
/external/chromium_org/third_party/webrtc/video_engine/
H A Dvie_remb.cc17 #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 Dvie_sync_module.cc13 #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 Dvie_receiver.cc16 #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 Dvie_channel.cc20 #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 Dremote_ntp_time_estimator.cc11 #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 Dremote_ntp_time_estimator_unittest.cc14 #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 Dnack_rtx_unittest.cc18 #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 Dreceive_statistics_proxy.cc21 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 Dtest_api_rtcp.cc17 #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 Dvoe_cmd_test.cc59 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 Dvie_autotest_codec.cc149 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 Dandroid_test.cc85 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...]

Completed in 476 milliseconds