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Searched refs:rtp_state (Results 1 – 9 of 9) sorted by relevance

/external/webrtc/webrtc/modules/rtp_rtcp/source/
Drtp_rtcp_impl.cc289 const RtpState& rtp_state) { in SetRtpStateForSsrc() argument
291 rtp_sender_.SetRtpState(rtp_state); in SetRtpStateForSsrc()
295 rtp_sender_.SetRtxRtpState(rtp_state); in SetRtpStateForSsrc()
301 bool ModuleRtpRtcpImpl::GetRtpStateForSsrc(uint32_t ssrc, RtpState* rtp_state) { in GetRtpStateForSsrc() argument
303 *rtp_state = rtp_sender_.GetRtpState(); in GetRtpStateForSsrc()
307 *rtp_state = rtp_sender_.GetRtxRtpState(); in GetRtpStateForSsrc()
Drtp_sender.cc1875 void RTPSender::SetRtpState(const RtpState& rtp_state) { in SetRtpState() argument
1876 SetStartTimestamp(rtp_state.start_timestamp, true); in SetRtpState()
1878 sequence_number_ = rtp_state.sequence_number; in SetRtpState()
1880 timestamp_ = rtp_state.timestamp; in SetRtpState()
1881 capture_time_ms_ = rtp_state.capture_time_ms; in SetRtpState()
1882 last_timestamp_time_ms_ = rtp_state.last_timestamp_time_ms; in SetRtpState()
1883 media_has_been_sent_ = rtp_state.media_has_been_sent; in SetRtpState()
1900 void RTPSender::SetRtxRtpState(const RtpState& rtp_state) { in SetRtxRtpState() argument
1902 sequence_number_rtx_ = rtp_state.sequence_number; in SetRtxRtpState()
Drtp_rtcp_impl.h75 bool SetRtpStateForSsrc(uint32_t ssrc, const RtpState& rtp_state) override;
76 bool GetRtpStateForSsrc(uint32_t ssrc, RtpState* rtp_state) override;
Drtp_sender.h309 void SetRtpState(const RtpState& rtp_state);
311 void SetRtxRtpState(const RtpState& rtp_state);
/external/webrtc/webrtc/video/
Dvie_channel.cc717 void ViEChannel::SetRtpStateForSsrc(uint32_t ssrc, const RtpState& rtp_state) { in SetRtpStateForSsrc() argument
720 if (rtp_rtcp->SetRtpStateForSsrc(ssrc, rtp_state)) in SetRtpStateForSsrc()
727 RtpState rtp_state; in GetRtpStateForSsrc() local
729 if (rtp_rtcp->GetRtpStateForSsrc(ssrc, &rtp_state)) in GetRtpStateForSsrc()
730 return rtp_state; in GetRtpStateForSsrc()
733 return rtp_state; in GetRtpStateForSsrc()
Dvie_channel.h137 void SetRtpStateForSsrc(uint32_t ssrc, const RtpState& rtp_state);
/external/webrtc/webrtc/modules/rtp_rtcp/include/
Drtp_rtcp.h206 const RtpState& rtp_state) = 0;
207 virtual bool GetRtpStateForSsrc(uint32_t ssrc, RtpState* rtp_state) = 0;
/external/webrtc/webrtc/modules/rtp_rtcp/mocks/
Dmock_rtp_rtcp.h90 bool(uint32_t ssrc, const RtpState& rtp_state));
91 MOCK_METHOD2(GetRtpStateForSsrc, bool(uint32_t ssrc, RtpState* rtp_state));
/external/webrtc/webrtc/call/
Dcall.cc425 VideoSendStream::RtpStateMap rtp_state = send_stream_impl->GetRtpStates(); in DestroyVideoSendStream() local
427 for (VideoSendStream::RtpStateMap::iterator it = rtp_state.begin(); in DestroyVideoSendStream()
428 it != rtp_state.end(); in DestroyVideoSendStream()