Searched refs:rtp_state (Results 1 – 12 of 12) sorted by relevance
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
| D | rtp_sender.cc | 1688 void RTPSender::SetRtpState(const RtpState& rtp_state) { in SetRtpState() argument
1689 SetStartTimestamp(rtp_state.start_timestamp, true); in SetRtpState()
1691 sequence_number_ = rtp_state.sequence_number; in SetRtpState()
1693 timestamp_ = rtp_state.timestamp; in SetRtpState()
1694 capture_time_ms_ = rtp_state.capture_time_ms; in SetRtpState()
1695 last_timestamp_time_ms_ = rtp_state.last_timestamp_time_ms; in SetRtpState()
1696 media_has_been_sent_ = rtp_state.media_has_been_sent; in SetRtpState()
1713 void RTPSender::SetRtxRtpState(const RtpState& rtp_state) { in SetRtxRtpState() argument
1715 sequence_number_rtx_ = rtp_state.sequence_number; in SetRtxRtpState()
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| D | rtp_rtcp_impl.cc | 341 const RtpState& rtp_state) { in SetRtpStateForSsrc() argument
343 rtp_sender_.SetRtpState(rtp_state); in SetRtpStateForSsrc()
347 rtp_sender_.SetRtxRtpState(rtp_state); in SetRtpStateForSsrc()
353 child_modules_[i]->SetRtpStateForSsrc(ssrc, rtp_state); in SetRtpStateForSsrc()
357 bool ModuleRtpRtcpImpl::GetRtpStateForSsrc(uint32_t ssrc, RtpState* rtp_state) { in GetRtpStateForSsrc() argument
359 *rtp_state = rtp_sender_.GetRtpState(); in GetRtpStateForSsrc()
364 *rtp_state = rtp_sender_.GetRtxRtpState(); in GetRtpStateForSsrc()
370 if (child_modules_[i]->GetRtpStateForSsrc(ssrc, rtp_state)) in GetRtpStateForSsrc()
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| D | rtp_rtcp_impl.h | 77 const RtpState& rtp_state) OVERRIDE;
78 virtual bool GetRtpStateForSsrc(uint32_t ssrc, RtpState* rtp_state) OVERRIDE;
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| D | rtp_sender.h | 281 void SetRtpState(const RtpState& rtp_state);
283 void SetRtxRtpState(const RtpState& rtp_state);
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| /external/chromium_org/third_party/webrtc/video/ |
| D | call.cc | 246 VideoSendStream::RtpStateMap rtp_state = send_stream_impl->GetRtpStates(); in DestroyVideoSendStream() local
248 for (VideoSendStream::RtpStateMap::iterator it = rtp_state.begin(); in DestroyVideoSendStream()
249 it != rtp_state.end(); in DestroyVideoSendStream()
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| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/interface/ |
| D | rtp_rtcp.h | 210 const RtpState& rtp_state) = 0;
211 virtual bool GetRtpStateForSsrc(uint32_t ssrc, RtpState* rtp_state) = 0;
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| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/mocks/ |
| D | mock_rtp_rtcp.h | 89 void(uint32_t ssrc, const RtpState& rtp_state));
90 MOCK_METHOD2(GetRtpStateForSsrc, bool(uint32_t ssrc, RtpState* rtp_state));
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| /external/chromium_org/third_party/webrtc/video_engine/ |
| D | vie_rtp_rtcp_impl.h | 51 const RtpState& rtp_state) OVERRIDE;
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| D | vie_channel.cc | 923 void ViEChannel::SetRtpStateForSsrc(uint32_t ssrc, const RtpState& rtp_state) { in SetRtpStateForSsrc() argument
925 default_rtp_rtcp_->SetRtpStateForSsrc(ssrc, rtp_state); in SetRtpStateForSsrc()
931 RtpState rtp_state; in GetRtpStateForSsrc() local
932 if (!default_rtp_rtcp_->GetRtpStateForSsrc(ssrc, &rtp_state)) { in GetRtpStateForSsrc()
935 return rtp_state; in GetRtpStateForSsrc()
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| D | vie_rtp_rtcp_impl.cc | 261 const RtpState& rtp_state) { in SetRtpStateForSsrc() argument
271 vie_channel->SetRtpStateForSsrc(ssrc, rtp_state); in SetRtpStateForSsrc()
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| D | vie_channel.h | 156 void SetRtpStateForSsrc(uint32_t ssrc, const RtpState& rtp_state);
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| /external/chromium_org/third_party/webrtc/video_engine/include/ |
| D | vie_rtp_rtcp.h | 159 const RtpState& rtp_state) {} in SetRtpStateForSsrc() argument
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