TRTCStatisic

Namespaces
	liteav

Data Structures
struct	TRTCLocalStatistics
struct	TRTCRemoteStatistics
struct	TRTCStatistics

Macros
#define	__TRTCSTATISTIC_H__

Detailed Description

Tencent Cloud TRTC : audio, video and network related statistical indicators.

Module: TRTC audio/video metrics (read-only) Function: the TRTC SDK reports to you the current real-time audio/video metrics (frame rate, bitrate, lag, etc.) once every two seconds

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Data Structure Documentation

liteav::TRTCLocalStatistics

struct liteav::TRTCLocalStatistics

Public Member Functions
	TRTCLocalStatistics ()
Data Fields
uint32_t	audioCaptureState

Constructor & Destructor Documentation

TRTCLocalStatistics()

TRTCLocalStatistics ( )

inline

Field Documentation

audioCaptureState

uint32_t audioCaptureState

Field description:** local video width in px uint32_t width;

/**Field description:** local video height in px uint32_t height;

/**Field description:** local video frame rate in fps, i.e., how many video frames there are per second uint32_t frameRate;

/**Field description:** remote video bitrate in Kbps, i.e., how much video data is generated per second uint32_t videoBitrate;

/**Field description:** remote audio sample rate (Hz) uint32_t audioSampleRate;

/**Field description:** local audio bitrate in Kbps, i.e., how much audio data is generated per second uint32_t audioBitrate;

/**Field description:** video stream type (HD big image | smooth small image | substream image) TRTCVideoStreamType streamType;

/****Field description:**Audio equipment collection status( 0：Normal；1：Long silence detected；2：Broken sound detected；3：Abnormal intermittent sound detected;)

liteav::TRTCRemoteStatistics

struct liteav::TRTCRemoteStatistics

Data Fields
uint32_t	audioBlockRate	Field description:** cumulative audio playback lag duration (ms) uint32_t audioTotalBlockTime; /****Field description:** audio playback lag rate (%) Audio playback lag rate (audioBlockRate) = cumulative audio playback lag duration (audioTotalBlockTime)/total audio playback duration
uint32_t	audioPacketLoss	Field description:** user ID const char* userId; /****Field description:** total packet loss rate (%) of the audio stream audioPacketLoss represents the packet loss rate eventually calculated on the audience side after the audio/video stream goes through the complete transfer linkage of "anchor -> cloud -> audience". The smaller the audioPacketLoss, the better. The packet loss rate of 0 indicates that all data of the audio stream has entirely reached the audience. If downLoss is 0 but audioPacketLoss isn't, there is no packet loss on the linkage of "cloud -> audience" for the audiostream, but there are unrecoverable packet losses on the linkage of "anchor -> cloud".
uint32_t	finalLoss
uint32_t	jitterBufferDelay	Field description:** remote video width in px uint32_t width; /**Field description:** remote video height in px uint32_t height; /**Field description:** remote video frame rate (fps) uint32_t frameRate; /**Field description:** remote video bitrate (Kbps) uint32_t videoBitrate; /**Field description:** local audio sample rate (Hz) uint32_t audioSampleRate; /**Field description:** local audio bitrate (Kbps) uint32_t audioBitrate; /****Field description:** playback delay (ms) In order to avoid audio/video lags caused by network jitters and network packet disorders, TRTC maintains a playback buffer on the playback side to organize the received network data packets. The size of the buffer is adaptively adjusted according to the current network quality and converted to the length of time in milliseconds, i.e., jitterBufferDelay.
uint32_t	point2PointDelay
uint32_t	videoBlockRate	Field description:** cumulative video playback lag duration (ms) uint32_t videoTotalBlockTime; /****Field description:** video playback lag rate (%) Video playback lag rate (videoBlockRate) = cumulative video playback lag duration (videoTotalBlockTime)/total video playback duration
uint32_t	videoPacketLoss

liteav::TRTCStatistics

struct liteav::TRTCStatistics

Data Fields
uint32_t	downLoss
uint32_t	gatewayRtt
TRTCLocalStatistics *	localStatisticsArray	Field description:** total number of sent bytes (including signaling data and audio/video data) uint32_t sentBytes; /**Field description:** total number of received bytes (including signaling data and audio/video data) uint32_t receivedBytes; /****Field description:** 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.
TRTCRemoteStatistics *	remoteStatisticsArray	Field description:** localStatisticsArray array size uint32_t localStatisticsArraySize; /****Field description:** 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.
uint32_t	rtt
uint32_t	upLoss	Field description:** CPU utilization (%) of the current application uint32_t appCpu; /**Field description:** CPU utilization (%) of the current system uint32_t systemCpu; /****Field description:** 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.

Macro Definition Documentation

__TRTCSTATISTIC_H__

#define __TRTCSTATISTIC_H__