Searched refs:GetRtpState (Results 1 – 19 of 19) sorted by relevance
50 virtual absl::optional<RtpState> GetRtpState() = 0;
60 absl::optional<RtpState> GetRtpState() override { return absl::nullopt; } in GetRtpState() function
196 absl::optional<RtpState> FlexfecSender::GetRtpState() { in GetRtpState() function in webrtc::FlexfecSender
166 RtpState GetRtpState() const RTC_LOCKS_EXCLUDED(send_mutex_);
210 virtual RtpState GetRtpState() const = 0;
107 RtpState GetRtpState() const override;
212 RtpState ModuleRtpRtcpImpl2::GetRtpState() const { in GetRtpState() function in webrtc::ModuleRtpRtcpImpl2213 RtpState state = rtp_sender_->packet_generator.GetRtpState(); in GetRtpState()
98 RtpState GetRtpState() const override;
268 RtpState ModuleRtpRtcpImpl::GetRtpState() const { in GetRtpState() function in webrtc::ModuleRtpRtcpImpl269 RtpState state = rtp_sender_->packet_generator.GetRtpState(); in GetRtpState()
329 RtpState updated_rtp_state = sender.GetRtpState().value(); in TEST()
830 RtpState RTPSender::GetRtpState() const { in GetRtpState() function in webrtc::RTPSender
1714 RtpState state = rtp_sender()->GetRtpState(); in TEST_P()
73 absl::optional<RtpState> GetRtpState() override;
102 RtpState GetRtpState() const;
582 RtpState AudioSendStream::GetRtpState() const { in GetRtpState() function in webrtc::internal::AudioSendStream583 return rtp_rtcp_module_->GetRtpState(); in GetRtpState()
68 MOCK_METHOD(RtpState, GetRtpState, (), (const, override));
704 rtp_states[ssrc] = rtp_streams_[i].rtp_rtcp->GetRtpState(); in GetRtpStates()710 rtp_streams_[i].fec_generator->GetRtpState(); in GetRtpStates()
332 static_cast<internal::AudioSendStream*>(stream)->GetRtpState(); in TEST()
806 suspended_audio_send_ssrcs_[ssrc] = audio_send_stream->GetRtpState(); in DestroyAudioSendStream()