TRTCStatistics

Constructor

new TRTCStatistics(upLoss, downLoss, appCpu, systemCpu, rtt, receivedBytes, sentBytes, localStatisticsArray, localStatisticsArraySize, remoteStatisticsArray, remoteStatisticsArraySize, gatewayRtt, systemMemoryInMB, systemMemoryUsageInMB, appMemoryUsageInMB)

Parameters:

Name	Type	Description
`upLoss`	Number	`required` upstream packet loss rate (%) from the SDK to cloud. The smaller the value, the better. If `upLoss` is `0%`, the upstream network quality is very good, and the data packets uploaded to the cloud are basically not lost. If `upLoss` is `30%`, 30% of the audio/video data packets sent to the cloud by the SDK are lost on the transfer linkage.
`downLoss`	Number	`required` downstream packet loss rate (%) from cloud to the SDK. The smaller the value, the better. If `downLoss` is `0%`, the downstream network quality is very good, and the data packets received from the cloud are basically not lost. If `downLoss` is `30%`, 30% of the audio/video data packets sent to the SDK by the cloud are lost on the transfer linkage.
`appCpu`	Number	`required` CPU utilization (%) of the current application
`systemCpu`	Number	`required` CPU utilization (%) of the current system
`rtt`	Number	`required` round-trip delay (ms) from the SDK to cloud. This value represents the total time it takes to send a network packet from the SDK to the cloud and then send a network packet back from the cloud to the SDK, i.e., the total time it takes for a network packet to go through the linkage of "SDK -> cloud -> SDK". The smaller the value, the better. If `rtt` is below 50 ms, it means a short audio/video call delay; if `rtt` is above 200 ms, it means a long audio/video call delay. It should be explained that `rtt` represents the total time spent on the linkage of "SDK -> cloud -> SDK"; therefore, there is no need to distinguish between `upRtt` and `downRtt`.
`receivedBytes`	Number	`required` total number of received bytes (including signaling data and audio/video data)
`sentBytes`	Number	`required` total number of sent bytes (including signaling data and audio/video data)
`localStatisticsArray`	Array.<TRTCLocalStatistics>	`required` local audio/video statistics. As there may be three local audio/video streams (i.e., HD big image, smooth small image, and substream image), the local audio/video statistics are an array.
`localStatisticsArraySize`	Number	`required` `localStatisticsArray` array size
`remoteStatisticsArray`	Array.<TRTCRemoteStatistics>	`required` remote audio/video statistics. As there may be multiple concurrent remote users, and each of them may have multiple concurrent audio/video streams (i.e., HD big image, smooth small image, and substream image), the remote audio/video statistics are an array.
`remoteStatisticsArraySize`	Number	`required` `remoteStatisticsArray` array size
`gatewayRtt`	Number	`required` round-trip delay (ms) from the SDK to client gatewaty, unit: ms. This value represents the total time taken for a network packet to be sent from the SDK to the local router gateway, and then for the gateway to send the packet back to the SDK, completing the "SDK>gateway>SDK" round trip. The smaller this value, the better: if gatewayRtt is less than 50ms, it indicates low audio and video call latency; if gatewayRtt is greater than 200ms, it indicates high audio and video call latency.
`systemMemoryInMB`	Number	`required` System memory size, unit: MB.
`systemMemoryUsageInMB`	Number	`required` System memory size in usage, unit: MB.
`appMemoryUsageInMB`	Number	`required` Application memory size, unit: MB.