| /external/chromium_org/third_party/webrtc/video_engine/test/libvietest/include/ |
| tb_interfaces.h | 41 webrtc::ViERTP_RTCP* rtp_rtcp; member in class:TbInterfaces
|
| /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();
|
| /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...] |
