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Searched refs:suspended_rtp_state (Results 1 – 3 of 3) sorted by relevance

/external/webrtc/audio/
Daudio_send_stream.h69 const absl::optional<RtpState>& suspended_rtp_state,
79 const absl::optional<RtpState>& suspended_rtp_state,
Daudio_send_stream.cc111 const absl::optional<RtpState>& suspended_rtp_state, in AudioSendStream() argument
121 suspended_rtp_state, in AudioSendStream()
145 const absl::optional<RtpState>& suspended_rtp_state, in AudioSendStream() argument
165 suspended_rtp_state_(suspended_rtp_state) { in AudioSendStream()
/external/webrtc/call/
Dcall.cc795 absl::optional<RtpState> suspended_rtp_state; in CreateAudioSendStream() local
799 suspended_rtp_state.emplace(iter->second); in CreateAudioSendStream()
806 call_stats_->AsRtcpRttStats(), suspended_rtp_state, trials()); in CreateAudioSendStream()