#include <rtp_rtcp.h>
◆ Configuration() [1/2]
webrtc::RtpRtcp::Configuration::Configuration |
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◆ Configuration() [2/2]
webrtc::RtpRtcp::Configuration::Configuration |
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◆ audio
bool webrtc::RtpRtcp::Configuration::audio = false |
◆ bandwidth_callback
◆ bitrate_allocation_observer
◆ clock
Clock * webrtc::RtpRtcp::Configuration::clock = nullptr |
◆ event_log
RtcEventLog * webrtc::RtpRtcp::Configuration::event_log = nullptr |
◆ flexfec_sender
FlexfecSender * webrtc::RtpRtcp::Configuration::flexfec_sender = nullptr |
◆ intra_frame_callback
◆ outgoing_transport
Transport * webrtc::RtpRtcp::Configuration::outgoing_transport = nullptr |
◆ overhead_observer
◆ paced_sender
◆ receive_statistics
◆ receiver_only
bool webrtc::RtpRtcp::Configuration::receiver_only = false |
◆ remote_bitrate_estimator
◆ retransmission_rate_limiter
RateLimiter * webrtc::RtpRtcp::Configuration::retransmission_rate_limiter = nullptr |
◆ rtcp_packet_type_counter_observer
◆ rtt_stats
RtcpRttStats * webrtc::RtpRtcp::Configuration::rtt_stats = nullptr |
◆ send_bitrate_observer
◆ send_frame_count_observer
◆ send_packet_observer
◆ send_side_delay_observer
◆ transport_feedback_callback
◆ transport_sequence_number_allocator
The documentation for this struct was generated from the following files:
- DerivedData/WebKit/Build/Products/Debug/usr/local/include/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
- Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc