/external/chromium_org/third_party/webrtc/video_engine/test/libvietest/include/ |
tb_interfaces.h | 41 webrtc::ViERTP_RTCP* rtp_rtcp; member in class:TbInterfaces
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/external/chromium_org/third_party/webrtc/video_engine/ |
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" 330 RtpRtcp* rtp_rtcp = *it++; local 331 rtp_rtcp->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length) [all...] |
vie_channel.cc | 20 #include "webrtc/modules/rtp_rtcp/interface/rtp_receiver.h" 21 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" 185 RtpRtcp* rtp_rtcp = *it; local 186 module_process_thread_.DeRegisterModule(rtp_rtcp); 187 delete rtp_rtcp; 248 RtpRtcp* rtp_rtcp = removed_rtp_rtcp_.front(); local 250 simulcast_rtp_rtcp_.push_back(rtp_rtcp); 251 rtp_rtcp->SetSendingStatus(rtp_rtcp_->Sending()); 252 rtp_rtcp->SetSendingMediaStatus(rtp_rtcp_->SendingMedia()) 268 RtpRtcp* rtp_rtcp = RtpRtcp::CreateRtpRtcp(configuration); local 292 RtpRtcp* rtp_rtcp = simulcast_rtp_rtcp_.back(); local 309 RtpRtcp* rtp_rtcp = *it; local 359 RtpRtcp* rtp_rtcp = simulcast_rtp_rtcp_.back(); local 517 RtpRtcp* rtp_rtcp = *it; local 559 RtpRtcp* rtp_rtcp = *it; local 569 RtpRtcp* rtp_rtcp = *it; local 609 RtpRtcp* rtp_rtcp = *it; local 1106 RtpRtcp* rtp_rtcp = *it; local 1115 RtpRtcp* rtp_rtcp = *it; local 1178 RtpRtcp* rtp_rtcp = *it; local 1200 RtpRtcp* rtp_rtcp = *it; local 1269 RtpRtcp* rtp_rtcp = *it; local 1282 RtpRtcp* rtp_rtcp = *it; local 1297 RtpRtcp* rtp_rtcp = *it; local 1386 RtpRtcp* rtp_rtcp = *it; local 1403 RtpRtcp* ViEChannel::rtp_rtcp() { function in class:webrtc::ViEChannel [all...] |
/external/chromium_org/third_party/webrtc/video_engine/test/auto_test/source/ |
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->SetTMMBRStatus(video_channel, true)) [all...] |
/external/chromium_org/third_party/webrtc/voice_engine/test/cmd_test/ |
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();
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/external/chromium_org/third_party/webrtc/voice_engine/test/android/android_test/jni/ |
android_test.cc | 85 if (!veData1.rtp_rtcp) \ 128 VoERTP_RTCP* rtp_rtcp; member in struct:__anon20151 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"); [all...] |