#include <rtp_rtcp_defines.h>
◆ RtpState() [1/2]
webrtc::RtpState::RtpState |
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◆ RtpState() [2/2]
webrtc::RtpState::RtpState |
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◆ capture_time_ms
int64_t webrtc::RtpState::capture_time_ms |
◆ last_timestamp_time_ms
int64_t webrtc::RtpState::last_timestamp_time_ms |
◆ media_has_been_sent
bool webrtc::RtpState::media_has_been_sent |
◆ sequence_number
uint16_t webrtc::RtpState::sequence_number |
◆ start_timestamp
uint32_t webrtc::RtpState::start_timestamp |
◆ timestamp
The documentation for this struct was generated from the following file:
- DerivedData/WebKit/Build/Products/Debug/usr/local/include/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h