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Searched refs:SetRtpState (Results 1 – 11 of 11) sorted by relevance

/external/webrtc/modules/rtp_rtcp/mocks/
Dmock_rtp_rtcp.h66 MOCK_METHOD(void, SetRtpState, (const RtpState& rtp_state), (override));
/external/webrtc/modules/rtp_rtcp/source/
Drtp_sender.h165 void SetRtpState(const RtpState& rtp_state) RTC_LOCKS_EXCLUDED(send_mutex_);
Drtp_rtcp_impl2.h105 void SetRtpState(const RtpState& rtp_state) override;
Drtp_rtcp_interface.h208 virtual void SetRtpState(const RtpState& rtp_state) = 0;
Drtp_rtcp_impl2.cc203 void ModuleRtpRtcpImpl2::SetRtpState(const RtpState& rtp_state) { in SetRtpState() function in webrtc::ModuleRtpRtcpImpl2
204 rtp_sender_->packet_generator.SetRtpState(rtp_state); in SetRtpState()
Drtp_rtcp_impl.h96 void SetRtpState(const RtpState& rtp_state) override;
Drtp_rtcp_impl.cc259 void ModuleRtpRtcpImpl::SetRtpState(const RtpState& rtp_state) { in SetRtpState() function in webrtc::ModuleRtpRtcpImpl
260 rtp_sender_->packet_generator.SetRtpState(rtp_state); in SetRtpState()
Drtp_sender.cc818 void RTPSender::SetRtpState(const RtpState& rtp_state) { in SetRtpState() function in webrtc::RTPSender
Drtp_sender_unittest.cc1717 rtp_sender()->SetRtpState(state); in TEST_P()
/external/webrtc/audio/
Daudio_send_stream.cc242 rtp_rtcp_module_->SetRtpState(*suspended_rtp_state_); in ConfigureStream()
/external/webrtc/call/
Drtp_video_sender.cc643 rtp_rtcp->SetRtpState(it->second); in ConfigureSsrcs()