| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/interface/ |
| 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/source/ |
| rtp_sender.cc | [all...] |
| rtp_rtcp_impl.cc | 341 const RtpState& rtp_state) {
343 rtp_sender_.SetRtpState(rtp_state);
347 rtp_sender_.SetRtxRtpState(rtp_state);
353 child_modules_[i]->SetRtpStateForSsrc(ssrc, rtp_state);
357 bool ModuleRtpRtcpImpl::GetRtpStateForSsrc(uint32_t ssrc, RtpState* rtp_state) {
359 *rtp_state = rtp_sender_.GetRtpState();
364 *rtp_state = rtp_sender_.GetRtxRtpState();
370 if (child_modules_[i]->GetRtpStateForSsrc(ssrc, rtp_state))
[all...] |
| 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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| 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/ |
| call.cc | 246 VideoSendStream::RtpStateMap rtp_state = send_stream_impl->GetRtpStates(); local
248 for (VideoSendStream::RtpStateMap::iterator it = rtp_state.begin();
249 it != rtp_state.end();
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| /external/chromium_org/third_party/webrtc/video_engine/ |
| vie_rtp_rtcp_impl.h | 51 const RtpState& rtp_state) OVERRIDE;
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| vie_channel.cc | 923 void ViEChannel::SetRtpStateForSsrc(uint32_t ssrc, const RtpState& rtp_state) {
925 default_rtp_rtcp_->SetRtpStateForSsrc(ssrc, rtp_state);
931 RtpState rtp_state; local
932 if (!default_rtp_rtcp_->GetRtpStateForSsrc(ssrc, &rtp_state)) {
935 return rtp_state;
[all...] |
| vie_channel.h | 156 void SetRtpStateForSsrc(uint32_t ssrc, const RtpState& rtp_state);
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| vie_rtp_rtcp_impl.cc | 261 const RtpState& rtp_state) {
271 vie_channel->SetRtpStateForSsrc(ssrc, rtp_state);
[all...] |
| /external/chromium_org/third_party/webrtc/modules/rtp_rtcp/mocks/ |
| 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/include/ |
| vie_rtp_rtcp.h | 159 const RtpState& rtp_state) {}
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