RTC Engine
  • 개요
  • Web
    • Demo 실행
    • SDK 빠른 실행
    • 기본 기능
      • 화면 공유
      • 라이브 방송
      • 미디어 장치
      • 볼륨
      • 인코딩 구성 파일 설정
      • 네트워크 품질 검사
      • 검출 능력
    • 고급 기능
      • AI 노이즈 캔슬링 활성화
      • 오디오 믹싱 활성화
      • 워터마크 활성화
      • Enable Virtual Background
      • 뷰티 및 특수효과 활성화
      • 데이터 메시지
      • 사용자 정의 수집 및 렌더링
    • 인터페이스 목록
    • 배포 설명
    • 플랫폼 지원
    • Web 관련 자주 받는 질문
      • 다중 사용자 비디오 통화 최적화
      • 자동 재생 제한 처리
      • 방화벽 제한 대응
      • 기타
  • Android
    • Run Sample Code
    • 통합
      • 2. SDK 가져오기
      • 3. 룸 들어가기
      • 4. 오디오/비디오 스트림 구독
      • 5. 오디오/비디오 스트림 배포
      • 6. 룸 퇴출
      • 7. 네트워크 품질 검사
      • 8. 화면 공유 활성화
      • 9. 비디오 품질 설정
      • 10. 비디오 회전
    • 네트워크 품질 테스트
    • 사용자 정의 수집 및 렌더링
    • 사용자 정의 오디오 수집 및 재생
    • 고객 인터페이스
      • 개요
      • TRTCCloud
      • TRTCStatistics
      • TRTCCloudListener
      • TXAudioEffectManager
      • TXBeautyManager
      • TXDeviceManager
      • 유형 정의
      • 인터페이스 폐기
      • 에러코드
    • 솔루션
      • 실시간 합창 (TUIKaraoke)
        • 빠른 통합
        • 실시 절차
        • 노래 동기화
        • 가사 동기화
        • 음성 동기화
        • 혼합 스트림 솔루션
        • TRTCKaraoke 관련 인터페이스
        • 자주 받는 질문들
    • 배포 설명
  • iOS
    • Run Sample Code
    • 통합
      • 2. SDK 가져오기
      • 3. 룸 들어가기
      • 4. 오디오/비디오 스트림 구독
      • 5. 오디오/비디오 스트림 배포
      • 6. 룸 퇴출
      • 7. 네트워크 품질 검사
      • 8. 화면 공유 활성화
      • 9. 비디오 품질 설정
      • 10. 비디오 회전
    • 네트워크 품질 테스트
    • 사용자 정의 수집 및 렌더링
    • 사용자 정의 오디오 수집 및 재생
    • 고객 인터페이스
      • 개요
      • TRTCCloud
      • TRTCCloudDelegate
      • TRTCStatistics
      • TXAudioEffectManager
      • TXBeautyManager
      • TXDeviceManager
      • 유형 정의
      • 인터페이스 폐기
      • ErrorCode
    • 솔루션
      • 빠른 통합
      • 실시 절차
      • 노래 동기화
      • 가사 동기화
      • 음성 동기화
      • 혼합 스트림 솔루션
      • TRTCKaraoke 관련 인터페이스
      • 자주 받는 질문들
    • 배포 설명
  • macOS
    • Run Sample Code
    • 통합
      • 2. SDK 가져오기
      • 3. 룸 들어가기
      • 4. 오디오/비디오 스트림 구독
      • 5. 오디오/비디오 스트림 배포
      • 6. 룸 퇴출
      • 7. 네트워크 품질 검사
      • 8. 화면 공유 활성화
      • 9. 컴퓨터 소리 공유
      • 10. 비디오 품질 설정
      • 11. 비디오 회전
    • 하드웨어 장치 테스트
    • 네트워크 품질 테스트
    • 사용자 정의 수집 및 렌더링
    • 사용자 정의 오디오 수집 및 재생
    • 고객 인터페이스
      • 개요
      • TRTCCloud
      • TRTCCloudDelegate
      • TRTCStatistics
      • TXAudioEffectManager
      • TXBeautyManager
      • TXDeviceManager
      • 유형 정의
      • 인터페이스 폐기
      • ErrorCode
      • 배포 설명
    • 배포 설명
  • Windows
    • Run Sample Code
    • 통합
      • 2. SDK 가져오기
      • 3. 룸 들어가기
      • 4. 오디오/비디오 스트림 구독
      • 5. 오디오/비디오 스트림 배포
      • 6. 룸 퇴출
      • 7. 네트워크 품질 검사
      • 8. 화면 공유 활성화
      • 9. 비디오 품질 설정
      • 10. 비디오 회전
    • 하드웨어 장치 테스트
    • 네트워크 품질 테스트
    • 사용자 정의 수집 및 렌더링
    • 사용자 정의 오디오 수집 및 재생
    • 고객 인터페이스
      • 개요
      • ITRTCCloud
      • ITRTCStatistics
      • TRTCCloudCallback
      • ITXAudioEffectManager
      • ITXDeviceManager
      • 유형 정의
      • 인터페이스 폐기
      • 에러코드
    • 배포 설명
  • Electron
    • 통합
      • 1. 인터페이스 예시
      • 2. SDK 가져오기
      • 3. 룸 들어가기
      • 4. 오디오/비디오 스트림 구독
      • 5. 오디오/비디오 스트림 배포
      • 6. 룸 퇴출
      • 7. 네트워크 품질 검사
      • 8. 화면 공유 활성화
      • 9. 컴퓨터 소리 공유
      • 10. 비디오 품질 설정
      • 11. 비디오 회전
    • 고객 인터페이스
      • 개요
      • 에러코드
  • Flutter
    • 통합
      • 1. 인터페이스 예시
      • 2. SDK 가져오기
      • 3. 룸 들어가기
      • 4. 오디오/비디오 스트림 구독
      • 5. 오디오/비디오 스트림 배포
      • 6. 룸 퇴출
      • 7. 네트워크 품질 검사
      • 8. 화면 공유 활성화
      • 9. 컴퓨터 소리 공유
      • 10. 비디오 품질 설정
      • 11. 비디오 회전
    • 고객 인터페이스
      • 개요
      • 에러코드
  • Unity
    • 통합
      • 1. 인터페이스 예시
      • 2. SDK 가져오기
    • 고객 인터페이스
      • 개요
      • 에러코드
  • QT
    • 통합
      • 1. SDK 가져오기
  • 개요
    • 개요
  • 개념
  • Features
  • 성능 통계
  • 가격
    • 무료시간
    • RTC-Engine 패키지
    • 구독 패키지 기간 과금 설명
    • 종량제
      • 오디오 및 비디오 시간 과금 설명
      • 클라우드 녹화 과금 설명
      • 믹스 트랜스코딩 및 CDN 중계 과금 설명
  • 자주 받는 질문들
    • 자주 받는 질문들
    • 마이그레이션 가이드
      • Twilio Video에서 TRTC로의 마이그레이션
      • 과금
      • Features
      • UserSig
      • 방화벽 제한
      • 설치 패키지 축소 방법
      • TRTCCalling Web 관련
      • 오디오/비디오 품질
      • 기타
RTC Engine
이 페이지는 현재 영어로만 제공되며 한국어 버전은 곧 제공될 예정입니다. 기다려 주셔서 감사드립니다.

ITRTCStatistics

Copyright (c) 2021 Tencent. All rights reserved.

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

ITRTCStatistics

StructType

FuncList
DESC
Local audio/video metrics
Remote audio/video metrics
Network and performance metrics

TRTCLocalStatistics

TRTCLocalStatistics

Local audio/video metrics

EnumType
DESC
audioBitrate
Field description: local audio bitrate in Kbps, i.e., how much audio data is generated per second
audioCaptureState
Field description:Audio equipment collection status(
0:Normal;1:Long silence detected;2:Broken sound detected;3:Abnormal intermittent sound detected;)
audioSampleRate
Field description: local audio sample rate (Hz)
frameRate
Field description: local video frame rate in fps, i.e., how many video frames there are per second
height
Field description: local video height in px
streamType
Field description: video stream type (HD big image | smooth small image | substream image)
videoBitrate
Field description: local video bitrate in Kbps, i.e., how much video data is generated per second
width
Field description: local video width in px

TRTCRemoteStatistics

TRTCRemoteStatistics

Remote audio/video metrics

EnumType
DESC
audioBitrate
Field description: local audio bitrate (Kbps)
audioBlockRate
Field description: audio playback lag rate (%)
Audio playback lag rate (audioBlockRate) = cumulative audio playback lag duration (audioTotalBlockTime)/total audio playback duration
audioPacketLoss
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".
audioSampleRate
Field description: local audio sample rate (Hz)
audioTotalBlockTime
Field description: cumulative audio playback lag duration (ms)
finalLoss
Field description: total packet loss rate (%) of the audio/video stream
Deprecated, please use audioPacketLoss and videoPacketLoss instead.
frameRate
Field description: remote video frame rate (fps)
height
Field description: remote video height in px
jitterBufferDelay
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 .
point2PointDelay
Field description: end-to-end delay (ms)
point2PointDelay represents the delay of "anchor -> cloud -> audience". To be more precise, it represents the delay of the entire linkage of "collection -> encoding -> network transfer -> receiving -> buffering -> decoding -> playback".
point2PointDelay works only if both the local and remote SDKs are on version 8.5 or above. If the remote SDK is on a version below 8.5, this value will always be 0 and thus meaningless.
remoteNetworkRTT
Field description: 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 remoteNetworkRTT is below 50 ms, it means a short audio/video call delay; if remoteNetworkRTT is above 200 ms, it means a long audio/video call delay.
It should be explained that remoteNetworkRTT represents the total time spent on the linkage of "SDK -> cloud -> SDK"; therefore, there is no need to distinguish between remoteNetworkUpRTT and remoteNetworkDownRTT .
remoteNetworkUplinkLoss
Field description: upstream packet loss rate (%) from the SDK to cloud
The smaller the value, the better. If remoteNetworkUplinkLoss is 0% , the upstream network quality is very good, and the data packets uploaded to the cloud are basically not lost.
If remoteNetworkUplinkLoss is 30% , 30% of the audio/video data packets sent to the cloud by the SDK are lost on the transfer linkage.
streamType
Field description: video stream type (HD big image | smooth small image | substream image)
userId
Field description: user ID
videoBitrate
Field description: remote video bitrate (Kbps)
videoBlockRate
Field description: video playback lag rate (%)
Video playback lag rate (videoBlockRate) = cumulative video playback lag duration (videoTotalBlockTime)/total video playback duration
videoPacketLoss
Field description: total packet loss rate (%) of the video stream
videoPacketLoss 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 videoPacketLoss , the better. The packet loss rate of 0 indicates that all data of the video stream has entirely reached the audience.
If downLoss is 0 but videoPacketLoss isn't, there is no packet loss on the linkage of "cloud -> audience" for the video stream, but there are unrecoverable packet losses on the linkage of "anchor -> cloud".
videoTotalBlockTime
Field description: cumulative video playback lag duration (ms)
width
Field description: remote video width in px

TRTCStatistics

TRTCStatistics

Network and performance metrics

EnumType
DESC
appCpu
Field description: CPU utilization (%) of the current application, Android 8.0 and above systems are not supported
appMemoryUsageInMB
Field description: Memory usage size (MB) of current application
downLoss
Field description: 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.
gatewayRtt
Field description: round-trip delay (ms) from the SDK to gateway
This value represents the total time it takes to send a network packet from the SDK to the gateway and then send a network packet back from the gateway to the SDK, i.e., the total time it takes for a network packet to go through the linkage of "SDK -> gateway -> SDK".
The smaller the value, the better. If gatewayRtt is below 50 ms, it means a short audio/video call delay; if gatewayRtt is above 200 ms, it means a long audio/video call delay.
It should be explained that gatewayRtt is invalid for cellular network.
localStatisticsArray
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.
localStatisticsArraySize
Field description: localStatisticsArray array size
receivedBytes
Field description: total number of received bytes (including signaling data and audio/video data)
remoteStatisticsArray
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.
remoteStatisticsArraySize
Field description: remoteStatisticsArray array size
rtt
Field description: 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 .
sentBytes
Field description: total number of sent bytes (including signaling data and audio/video data)
systemCpu
Field description: CPU utilization (%) of the current system, Android 8.0 and above systems are not supported
systemMemoryInMB
Field description: Memory size (MB) of current system
systemMemoryUsageInMB
Field description: Memory usage size (MB) of current system , iOS and MAC are not supported
upLoss
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.