#include <voe_rtp_rtcp.h>
◆ bytesReceived
| size_t webrtc::CallStatistics::bytesReceived |
◆ bytesSent
| size_t webrtc::CallStatistics::bytesSent |
◆ capture_start_ntp_time_ms_
| int64_t webrtc::CallStatistics::capture_start_ntp_time_ms_ |
◆ cumulativeLost
| unsigned int webrtc::CallStatistics::cumulativeLost |
◆ extendedMax
| unsigned int webrtc::CallStatistics::extendedMax |
◆ fractionLost
| unsigned short webrtc::CallStatistics::fractionLost |
◆ jitterSamples
| unsigned int webrtc::CallStatistics::jitterSamples |
◆ packetsReceived
| int webrtc::CallStatistics::packetsReceived |
◆ packetsSent
| int webrtc::CallStatistics::packetsSent |
◆ rttMs
| int64_t webrtc::CallStatistics::rttMs |
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