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Searched refs:SetSendingMediaStatus (Results 1 – 18 of 18) sorted by relevance

/external/webrtc/audio/voip/
Daudio_egress.cc60 rtp_rtcp_->SetSendingMediaStatus(true); in StartSend()
64 rtp_rtcp_->SetSendingMediaStatus(false); in StopSend()
Daudio_channel.cc56 rtp_rtcp_->SetSendingMediaStatus(false); in AudioChannel()
/external/webrtc/audio/
Dchannel_send.cc499 rtp_rtcp_->SetSendingMediaStatus(false); in ChannelSend()
535 rtp_rtcp_->SetSendingMediaStatus(true); in StartSend()
565 rtp_rtcp_->SetSendingMediaStatus(false); in StopSend()
Dchannel_receive.cc511 rtp_rtcp_->SetSendingMediaStatus(false); in ChannelReceive()
/external/webrtc/audio/voip/test/
Daudio_ingress_unittest.cc51 rtp_rtcp_->SetSendingMediaStatus(false); in AudioIngressTest()
Daudio_egress_unittest.cc41 rtp_rtcp->SetSendingMediaStatus(false); in CreateRtpStack()
/external/webrtc/modules/rtp_rtcp/source/
Drtp_sender.h54 void SetSendingMediaStatus(bool enabled) RTC_LOCKS_EXCLUDED(send_mutex_);
Drtp_rtcp_interface.h258 virtual void SetSendingMediaStatus(bool sending) = 0;
Drtp_rtcp_impl2.h135 void SetSendingMediaStatus(bool sending) override;
Drtp_rtcp_impl2.cc291 void ModuleRtpRtcpImpl2::SetSendingMediaStatus(const bool sending) { in SetSendingMediaStatus() function in webrtc::ModuleRtpRtcpImpl2
293 rtp_sender_->packet_generator.SetSendingMediaStatus(sending); in SetSendingMediaStatus()
Drtp_rtcp_impl.h126 void SetSendingMediaStatus(bool sending) override;
Drtp_rtcp_impl.cc343 void ModuleRtpRtcpImpl::SetSendingMediaStatus(const bool sending) { in SetSendingMediaStatus() function in webrtc::ModuleRtpRtcpImpl
345 rtp_sender_->packet_generator.SetSendingMediaStatus(sending); in SetSendingMediaStatus()
Drtp_rtcp_impl_unittest.cc177 sender_.impl_->SetSendingMediaStatus(true); in SetUp()
195 receiver_.impl_->SetSendingMediaStatus(false); in SetUp()
Drtp_rtcp_impl2_unittest.cc182 sender_.impl_->SetSendingMediaStatus(true); in SetUp()
200 receiver_.impl_->SetSendingMediaStatus(false); in SetUp()
Drtp_sender_unittest.cc533 rtp_sender()->SetSendingMediaStatus(false); in TEST_P()
2674 rtp_sender()->SetSendingMediaStatus(sending_media); in TEST_P()
2706 rtp_sender()->SetSendingMediaStatus(true); in TEST_P()
2770 rtp_sender()->SetSendingMediaStatus(true); in TEST_P()
2799 rtp_sender()->SetSendingMediaStatus(true); in TEST_P()
2820 rtp_sender()->SetSendingMediaStatus(false); in TEST_P()
Drtp_sender.cc629 void RTPSender::SetSendingMediaStatus(bool enabled) { in SetSendingMediaStatus() function in webrtc::RTPSender
/external/webrtc/modules/rtp_rtcp/mocks/
Dmock_rtp_rtcp.h81 MOCK_METHOD(void, SetSendingMediaStatus, (bool sending), (override));
/external/webrtc/call/
Drtp_video_sender.cc262 rtp_rtcp->SetSendingMediaStatus(false); in CreateRtpStreamSenders()
493 rtp_streams_[i].rtp_rtcp->SetSendingMediaStatus(active_modules[i]); in SetActiveModulesLocked()