#include <voe_rtp_rtcp.h>
◆ cumulative_num_packets_lost
uint32_t webrtc::ReportBlock::cumulative_num_packets_lost |
◆ delay_since_last_SR
uint32_t webrtc::ReportBlock::delay_since_last_SR |
◆ extended_highest_sequence_number
uint32_t webrtc::ReportBlock::extended_highest_sequence_number |
◆ fraction_lost
uint8_t webrtc::ReportBlock::fraction_lost |
◆ interarrival_jitter
uint32_t webrtc::ReportBlock::interarrival_jitter |
◆ last_SR_timestamp
uint32_t webrtc::ReportBlock::last_SR_timestamp |
◆ sender_SSRC
uint32_t webrtc::ReportBlock::sender_SSRC |
◆ source_SSRC
uint32_t webrtc::ReportBlock::source_SSRC |
The documentation for this struct was generated from the following file:
- DerivedData/WebKit/Build/Products/Debug/usr/local/include/webrtc/voice_engine/include/voe_rtp_rtcp.h