TRTCCloudDef

Namespaces
	liteav

Data Structures
struct	RECT
struct	SIZE
struct	TRTCParams
struct	TRTCVideoEncParam
struct	TRTCNetworkQosParam
struct	TRTCRenderParams
struct	TRTCQualityInfo
struct	TRTCVolumeInfo
struct	TRTCSpeedTestParams
struct	TRTCSpeedTestResult
struct	TRTCVideoFrame
struct	TRTCAudioFrame
struct	TRTCMixUser
struct	TRTCTranscodingConfig
struct	TRTCPublishCDNParam
struct	TRTCAudioRecordingParams
struct	TRTCLocalRecordingParams
struct	TRTCAudioEffectParam
struct	TRTCSwitchRoomConfig
struct	TRTCAudioFrameCallbackFormat
struct	TRTCImageBuffer
struct	TRTCScreenCaptureSourceInfo
class	ITRTCScreenCaptureSourceList
struct	TRTCScreenCaptureProperty
struct	TRTCAudioParallelParams

Macros
#define	__TRTCTYPEDEF_H__
#define	WIN32_LEAN_AND_MEAN
#define	TRTC_API   __declspec(dllexport)
#define	TRTC_API   __declspec(dllimport)
#define	TRTC_API   __attribute__((visibility("default")))
#define	TRTC_API   __attribute__((visibility("default")))
#define	TRTC_API
#define	TARGET_PLATFORM_DESKTOP   ((__APPLE__ && TARGET_OS_MAC && !TARGET_OS_IPHONE) || _WIN32)
#define	TARGET_PLATFORM_PHONE   (__ANDROID__ || (__APPLE__ && TARGET_OS_IOS))
#define	TARGET_PLATFORM_MAC   (__APPLE__ && TARGET_OS_MAC && !TARGET_OS_IPHONE)
#define	TRTCDeviceTypeUnknow   TXMediaDeviceTypeUnknown
#define	TRTCDeviceTypeMic   TXMediaDeviceTypeMic
#define	TRTCDeviceTypeSpeaker   TXMediaDeviceTypeSpeaker
#define	TRTCDeviceTypeCamera   TXMediaDeviceTypeCamera
#define	TRTCDeviceStateAdd   TXMediaDeviceStateAdd
#define	TRTCDeviceStateRemove   TXMediaDeviceStateRemove
#define	TRTCDeviceStateActive   TXMediaDeviceStateActive

Typedefs
typedef HWND	TXView
typedef TXMediaDeviceType	TRTCDeviceType
typedef TXMediaDeviceState	TRTCDeviceState
typedef ITXDeviceCollection	ITRTCDeviceCollection
typedef ITXDeviceInfo	ITRTCDeviceInfo

Enumerations
enum	TRTCVideoResolution
enum	TRTCVideoResolutionMode
enum	TRTCVideoStreamType
enum	TRTCVideoFillMode
enum	TRTCVideoRotation
enum	TRTCBeautyStyle
enum	TRTCVideoPixelFormat
enum	TRTCVideoBufferType
enum	TRTCVideoMirrorType
enum	TRTCSnapshotSourceType
enum	TRTCAppScene
enum	TRTCRoleType
enum	TRTCQosControlMode
enum	TRTCVideoQosPreference
enum	TRTCQuality
enum	TRTCAVStatusType
enum	TRTCAVStatusChangeReason
enum	TRTCAudioQuality
enum	TRTCAudioFrameFormat
enum	TRTCLogLevel
enum	TRTCScreenCaptureSourceType
enum	TRTCTranscodingConfigMode
enum	TRTCLocalRecordType
enum	TRTCMixInputType
enum	TRTCWaterMarkSrcType
enum	TRTCAudioRecordingContent

Detailed Description

Tencent Cloud TRTC Key Type Definition.

Module: TRTC key class definition Description: definitions of enumerated and constant values such as resolution and quality level

---

Data Structure Documentation

liteav::RECT

struct liteav::RECT

Data Fields
int	bottom
int	left
int	right
int	top

liteav::SIZE

struct liteav::SIZE

Data Fields
long	height
long	width

liteav::TRTCParams

struct liteav::TRTCParams

Room entry parameters

As the room entry parameters in the TRTC SDK, these parameters must be correctly set so that the user can successfully enter the audio/video room specified by roomId or strRoomId. For historical reasons, TRTC supports two types of room IDs: roomId and strRoomId. Note: do not mix roomId and strRoomId, because they are not interchangeable. For example, the number 123 and the string 123 are two completely different rooms in TRTC.

Public Member Functions
	TRTCParams ()
Data Fields
uint32_t	sdkAppId
const char *	userId
const char *	userSig
uint32_t	roomId
const char *	strRoomId
TRTCRoleType	role
const char *	streamId
const char *	userDefineRecordId
const char *	privateMapKey
const char *	businessInfo

Constructor & Destructor Documentation

TRTCParams()

TRTCParams ( )

inline

Field Documentation

businessInfo

const char* businessInfo

privateMapKey

const char* privateMapKey

role

TRTCRoleType role

roomId

uint32_t roomId

sdkAppId

uint32_t sdkAppId

streamId

const char* streamId

strRoomId

const char* strRoomId

userDefineRecordId

const char* userDefineRecordId

userId

const char* userId

userSig

const char* userSig

liteav::TRTCVideoEncParam

struct liteav::TRTCVideoEncParam

Video encoding parameters

These settings determine the quality of image viewed by remote users as well as the image quality of recorded video files in the cloud.

Public Member Functions
	TRTCVideoEncParam ()
Data Fields
TRTCVideoResolution	videoResolution
TRTCVideoResolutionMode	resMode
uint32_t	videoFps
uint32_t	videoBitrate
uint32_t	minVideoBitrate
bool	enableAdjustRes

Constructor & Destructor Documentation

TRTCVideoEncParam()

TRTCVideoEncParam ( )

inline

Field Documentation

enableAdjustRes

bool enableAdjustRes

minVideoBitrate

uint32_t minVideoBitrate

resMode

TRTCVideoResolutionMode resMode

videoBitrate

uint32_t videoBitrate

videoFps

uint32_t videoFps

Recommended value: 15 or 20 fps. If the frame rate is lower than 5 fps, there will be obvious lagging; if lower than 10 fps but higher than 5 fps, there will be slight lagging; if higher than 20 fps, the bandwidth will be wasted (the frame rate of movies is generally 24 fps). Note: the front cameras on certain Android phones do not support a capturing frame rate higher than 15 fps. For some Android phones that focus on beautification features, the capturing frame rate of the front cameras may be lower than 10 fps.

videoResolution

TRTCVideoResolution videoResolution

liteav::TRTCNetworkQosParam

struct liteav::TRTCNetworkQosParam

Network QoS control parameter set

Network QoS control parameter. The settings determine the QoS control policy of the SDK in weak network conditions (e.g., whether to "ensure clarity" or "ensure smoothness").

Public Member Functions
	TRTCNetworkQosParam ()
Data Fields
TRTCVideoQosPreference	preference
TRTCQosControlMode	controlMode

Constructor & Destructor Documentation

TRTCNetworkQosParam()

TRTCNetworkQosParam ( )

inline

Field Documentation

controlMode

TRTCQosControlMode controlMode

preference

TRTCVideoQosPreference preference

liteav::TRTCRenderParams

struct liteav::TRTCRenderParams

Rendering parameters of video image

You can use these parameters to control the video image rotation angle, fill mode, and mirror mode.

Public Member Functions
	TRTCRenderParams ()
Data Fields
TRTCVideoRotation	rotation
TRTCVideoFillMode	fillMode
TRTCVideoMirrorType	mirrorType

Constructor & Destructor Documentation

TRTCRenderParams()

TRTCRenderParams ( )

inline

Field Documentation

fillMode

TRTCVideoFillMode fillMode

mirrorType

TRTCVideoMirrorType mirrorType

rotation

TRTCVideoRotation rotation

liteav::TRTCQualityInfo

struct liteav::TRTCQualityInfo

Network quality

This indicates the quality of the network. You can use it to display the network quality of each user on the UI.

Public Member Functions
	TRTCQualityInfo ()
Data Fields
const char *	userId
TRTCQuality	quality

Constructor & Destructor Documentation

TRTCQualityInfo()

TRTCQualityInfo ( )

inline

Field Documentation

quality

TRTCQuality quality

Network quality.

userId

const char* userId

User ID.

liteav::TRTCVolumeInfo

struct liteav::TRTCVolumeInfo

Volume

This indicates the audio volume value. You can use it to display the volume of each user in the UI.

Public Member Functions
	TRTCVolumeInfo ()
Data Fields
const char *	userId
uint32_t	volume

Constructor & Destructor Documentation

TRTCVolumeInfo()

TRTCVolumeInfo ( )

inline

Field Documentation

userId

const char* userId

userId of the speaker. An empty value indicates the local user.

volume

uint32_t volume

Volume of the speaker. Value range: 0–100.

liteav::TRTCSpeedTestParams

struct liteav::TRTCSpeedTestParams

Network speed testing parameters

You can test the network speed through the startSpeedTest: interface before the user enters the room (this API cannot be called during a call).

Public Member Functions
	TRTCSpeedTestParams ()
Data Fields
int	sdkAppId
const char *	userId
const char *	userSig
int	expectedUpBandwidth
int	expectedDownBandwidth

Constructor & Destructor Documentation

TRTCSpeedTestParams()

TRTCSpeedTestParams ( )

inline

Field Documentation

expectedDownBandwidth

int expectedDownBandwidth

Expected downstream bandwidth (kbps, value range: 10 to 5000, no downlink bandwidth test when it is 0).

expectedUpBandwidth

int expectedUpBandwidth

Expected upstream bandwidth (kbps, value range: 10 to 5000, no uplink bandwidth test when it is 0).

sdkAppId

int sdkAppId

Application identification, please refer to the relevant instructions in TRTCParams.

userId

const char* userId

User identification, please refer to the relevant instructions in TRTCParams.

userSig

const char* userSig

User signature, please refer to the relevant instructions in TRTCParams.

liteav::TRTCSpeedTestResult

struct liteav::TRTCSpeedTestResult

Network speed test result

The startSpeedTest: API can be used to test the network speed before a user enters a room (this API cannot be called during a call).

Public Member Functions
	TRTCSpeedTestResult ()
Data Fields
bool	success
const char *	errMsg
const char *	ip
TRTCQuality	quality
float	upLostRate
float	downLostRate
int	rtt
int	availableUpBandwidth
int	availableDownBandwidth

Constructor & Destructor Documentation

TRTCSpeedTestResult()

TRTCSpeedTestResult ( )

inline

Field Documentation

availableDownBandwidth

int availableDownBandwidth

Downstream bandwidth (in kbps, -1: invalid value).

availableUpBandwidth

int availableUpBandwidth

Upstream bandwidth (in kbps, -1: invalid value).

downLostRate

float downLostRate

Downstream packet loss rate between 0 and 1.0. For example, 0.2 indicates that 2 data packets may be lost in every 10 packets received from the server.

errMsg

const char* errMsg

Error message for network speed test.

ip

const char* ip

Server IP address.

quality

TRTCQuality quality

Network quality, which is tested and calculated based on the internal evaluation algorithm. For more information, please see TRTCQuality.

rtt

int rtt

Delay in milliseconds, which is the round-trip time between the current device and TRTC server. The smaller the value, the better. The normal value range is 10–100 ms.

success

bool success

Whether the network speed test is successful.

upLostRate

float upLostRate

Upstream packet loss rate between 0 and 1.0. For example, 0.3 indicates that 3 data packets may be lost in every 10 packets sent to the server.

liteav::TRTCVideoFrame

struct liteav::TRTCVideoFrame

Video frame information

TRTCVideoFrame is used to describe the raw data of a frame of the video image, which is the image data before frame encoding or after frame decoding.

Data Fields
uint32_t	height
uint64_t	timestamp
uint32_t	width	Field description:** video pixel format TRTCVideoPixelFormat videoFormat; /**Field description:** video data structure type TRTCVideoBufferType bufferType; /**Field description:** video data when bufferType is TRTCVideoBufferType_Buffer, which carries the memory data blocks for the C++ layer. char *data; /**Field description:** video texture ID, i.e., video data when bufferType is TRTCVideoBufferType_Texture, which carries the texture data used for OpenGL rendering. int textureId; /**Field description:** video data length in bytes. For I420, length = width * height * 3 / 2; for BGRA32, length = width * height * 4. uint32_t length; /****Field description:** video width Recommended value: please enter the width of the video data passed in.

liteav::TRTCAudioFrame

struct liteav::TRTCAudioFrame

Audio frame data

liteav::TRTCMixUser

struct liteav::TRTCMixUser

Description information of each video image in On-Cloud MixTranscoding

TRTCMixUser is used to specify the location, size, layer, and stream type of each video image in On-Cloud MixTranscoding.

Public Member Functions
	TRTCMixUser ()
Data Fields
bool	pureAudio
TRTCMixInputType	inputType
uint32_t	renderMode
const char *	image

Constructor & Destructor Documentation

TRTCMixUser()

TRTCMixUser ( )

inline

Field Documentation

image

const char* image

inputType

TRTCMixInputType inputType

pureAudio

bool pureAudio

Field description:** user ID const char *userId;

/**Field description:** ID of the room where this audio/video stream is located (an empty value indicates the local room ID) const char *roomId;

/**Field description:** specify the coordinate area of this video image in px RECT rect;

/**Field description:** specify the level of this video image (value range: 1–15; the value must be unique) int zOrder;

/**Field description:** specify whether this video image is the primary stream image (TRTCVideoStreamTypeBig) or substream image (TRTCVideoStreamTypeSub). TRTCVideoStreamType streamType;

/****Field description:** specify whether this stream mixes audio only Recommended value: default value: false Note: this field has been disused. We recommend you use the new field inputType introduced in v8.5.

renderMode

uint32_t renderMode

liteav::TRTCTranscodingConfig

struct liteav::TRTCTranscodingConfig

Layout and transcoding parameters of On-Cloud MixTranscoding

These parameters are used to specify the layout position information of each video image and the encoding parameters of mixtranscoding during On-Cloud MixTranscoding.

Public Member Functions
	TRTCTranscodingConfig ()
Data Fields
TRTCTranscodingConfigMode	mode
uint32_t	appId
uint32_t	bizId
uint32_t	videoWidth
uint32_t	videoHeight
uint32_t	videoBitrate
uint32_t	videoFramerate
uint32_t	videoGOP
uint32_t	backgroundColor
const char *	backgroundImage
uint32_t	audioSampleRate
uint32_t	audioBitrate
uint32_t	audioChannels
uint32_t	audioCodec
TRTCMixUser *	mixUsersArray
const char *	streamId

Constructor & Destructor Documentation

TRTCTranscodingConfig()

TRTCTranscodingConfig ( )

inline

Field Documentation

appId

uint32_t appId

audioBitrate

uint32_t audioBitrate

audioChannels

uint32_t audioChannels

audioCodec

uint32_t audioCodec

audioSampleRate

uint32_t audioSampleRate

backgroundColor

uint32_t backgroundColor

backgroundImage

const char* backgroundImage

bizId

uint32_t bizId

mixUsersArray

TRTCMixUser* mixUsersArray

mode

TRTCTranscodingConfigMode mode

streamId

const char* streamId

Field description:** number of elements in the mixUsersArray array uint32_t mixUsersArraySize;

/****Field description:** ID of the live stream output to CDN Recommended value: default value: null, that is, the audio/video streams in the room will be mixed into the audio/video stream of the caller of this API.

• If you don't set this parameter, the SDK will execute the default logic, that is, it will mix the multiple audio/video streams in the room into the audio/video stream of the caller of this API, i.e., A + B => A.
• If you set this parameter, the SDK will mix the audio/video streams in the room into the live stream you specify, i.e., A + B => C (C is the streamId you specify).

videoBitrate

uint32_t videoBitrate

videoFramerate

uint32_t videoFramerate

videoGOP

uint32_t videoGOP

videoHeight

uint32_t videoHeight

videoWidth

uint32_t videoWidth

liteav::TRTCPublishCDNParam

struct liteav::TRTCPublishCDNParam

Push parameters required to be set when publishing audio/video streams to non-Tencent Cloud CDN

TRTC's backend service supports publishing audio/video streams to third-party live CDN service providers through the standard RTMP protocol. If you use the Tencent Cloud CSS CDN service, you don't need to care about this parameter; instead, just use the startPublish API.

Public Member Functions
	TRTCPublishCDNParam ()
Data Fields
uint32_t	appId
uint32_t	bizId
const char *	url
const char *	streamId

Constructor & Destructor Documentation

TRTCPublishCDNParam()

TRTCPublishCDNParam ( )

inline

Field Documentation

appId

uint32_t appId

bizId

uint32_t bizId

streamId

const char* streamId

url

const char* url

liteav::TRTCAudioRecordingParams

struct liteav::TRTCAudioRecordingParams

Local audio file recording parameters

This parameter is used to specify the recording parameters in the audio recording API startAudioRecording.

Public Member Functions
	TRTCAudioRecordingParams ()
Data Fields
const char *	filePath
TRTCAudioRecordingContent	recordingContent

Constructor & Destructor Documentation

TRTCAudioRecordingParams()

TRTCAudioRecordingParams ( )

inline

Field Documentation

filePath

const char* filePath

recordingContent

TRTCAudioRecordingContent recordingContent

liteav::TRTCLocalRecordingParams

struct liteav::TRTCLocalRecordingParams

Local media file recording parameters

This parameter is used to specify the recording parameters in the local media file recording API startLocalRecording. The startLocalRecording API is an enhanced version of the startAudioRecording API. The former can record video files, while the latter can only record audio files.

Data Fields
const char *	filePath

liteav::TRTCAudioEffectParam

struct liteav::TRTCAudioEffectParam

Sound effect parameter (disused)

"Sound effects" in TRTC refer to some short audio files (usually only a few seconds), such as "applause" and "laughter". This parameter is used to specify the path and number of playback times of a sound effect file (short audio file) in the sound effect playback API TRTCCloud#playAudioEffect on legacy versions. After v7.3, the sound effect API has been replaced by a new TXAudioEffectManager#startPlayMusic API. When you specify the TXAudioMusicParam parameter of startPlayMusic, if isShortFile is set to true, the file is a "sound effect" file.

Public Member Functions
	TRTCAudioEffectParam (const int _effectId, const char *_path)
Data Fields
int	effectId
int	loopCount
bool	publish
int	volume

Constructor & Destructor Documentation

TRTCAudioEffectParam()

TRTCAudioEffectParam ( const int  _effectId, const char *  _path  )

inline

Field Documentation

effectId

int effectId

loopCount

int loopCount

Field description:** sound effect file path. Supported file formats include AAC, MP3, and M4A. const char *path;

/****Field description:** number of times the sound effect is looped Valid values: 0 or any positive integer. 0 (default) indicates that the sound effect is played once, 1 twice, and so on.

publish

bool publish

volume

int volume

liteav::TRTCSwitchRoomConfig

struct liteav::TRTCSwitchRoomConfig

Room switch parameter

This parameter is used for the room switch API switchRoom, which can quickly switch a user from one room to another.

Public Member Functions
	TRTCSwitchRoomConfig ()
Data Fields
uint32_t	roomId
const char *	strRoomId
const char *	userSig
const char *	privateMapKey

Constructor & Destructor Documentation

TRTCSwitchRoomConfig()

TRTCSwitchRoomConfig ( )

inline

Field Documentation

privateMapKey

const char* privateMapKey

roomId

uint32_t roomId

strRoomId

const char* strRoomId

userSig

const char* userSig

liteav::TRTCAudioFrameCallbackFormat

struct liteav::TRTCAudioFrameCallbackFormat

Format parameter of custom audio callback

This parameter is used to set the relevant format (including sample rate and number of channels) of the audio data called back by the SDK in the APIs related to custom audio callback.

Public Member Functions
	TRTCAudioFrameCallbackFormat ()
Data Fields
int	sampleRate
int	channel
int	samplesPerCall

Constructor & Destructor Documentation

TRTCAudioFrameCallbackFormat()

TRTCAudioFrameCallbackFormat ( )

inline

Field Documentation

channel

int channel

sampleRate

int sampleRate

samplesPerCall

int samplesPerCall

liteav::TRTCImageBuffer

struct liteav::TRTCImageBuffer

Screen sharing target information (for desktop systems only)

When users share the screen, they can choose to share the entire desktop or only the window of a certain program. TRTCScreenCaptureSourceInfo is used to describe the information of the target to be shared, including ID, name, and thumbnail. The fields in this structure are read-only.

Public Member Functions
	TRTCImageBuffer ()
Data Fields
const char *	buffer
uint32_t	length
uint32_t	width
uint32_t	height

Constructor & Destructor Documentation

TRTCImageBuffer()

TRTCImageBuffer ( )

inline

Field Documentation

buffer

const char* buffer

image content in BGRA format

height

uint32_t height

image height

length

uint32_t length

buffer size

width

uint32_t width

image width

liteav::TRTCScreenCaptureSourceInfo

struct liteav::TRTCScreenCaptureSourceInfo

liteav::ITRTCScreenCaptureSourceList

class liteav::ITRTCScreenCaptureSourceList

List of sharable screens and windows

This structure is equivalent to std::vector<TRTCScreenCaptureSourceInfo>, which is used to solve the binary compatibility problem of different versions of STL containers.

Public Member Functions
virtual uint32_t	getCount ()=0
virtual TRTCScreenCaptureSourceInfo	getSourceInfo (uint32_t index)=0
virtual void	release ()=0
Protected Member Functions
virtual	~ITRTCScreenCaptureSourceList ()

Constructor & Destructor Documentation

~ITRTCScreenCaptureSourceList()

virtual ~ITRTCScreenCaptureSourceList ( )

inlineprotectedvirtual

Member Function Documentation

getCount()

virtual uint32_t getCount ( )

pure virtual

Size of this list.

getSourceInfo()

virtual TRTCScreenCaptureSourceInfo getSourceInfo ( uint32_t  index )

pure virtual

Get element(TRTCScreenCaptureSourceInfo) by index.

release()

virtual void release ( )

pure virtual

Don't use delete!!!

liteav::TRTCScreenCaptureProperty

struct liteav::TRTCScreenCaptureProperty

Advanced control parameter of screen sharing

This parameter is used in the screen sharing-related API selectScreenCaptureTarget to set a series of advanced control parameters when specifying the sharing target. For example, whether to capture the cursor, whether to capture the subwindow, and whether to draw a frame around the shared target.

Data Fields
bool	enableHighPerformance	Field description:** whether to capture the cursor while capturing the target content. Default value: true. bool enableCaptureMouse; /**Field description:** whether to highlight the window being shared (i.e., drawing a frame around the shared target). Default value: true. bool enableHighLight; /****Field description:** whether to enable the high performance mode (which will take effect only during screen sharing). Default value: true. Note: the screen capturing performance is the best after this mode is enabled, but the anti-blocking ability will be lost. If you enable enableHighLight and enableHighPerformance at the same time, remote users will see the highlighted frame.

liteav::TRTCAudioParallelParams

struct liteav::TRTCAudioParallelParams

parameter of the parallel strategy of remote audio streams

This parameter is used to set the parallel strategy of remote audio streams.

Public Member Functions
	TRTCAudioParallelParams ()
Data Fields
uint32_t	maxCount
char **	includeUsers
uint32_t	includeUsersCount

Constructor & Destructor Documentation

TRTCAudioParallelParams()

TRTCAudioParallelParams ( )

inline

Field Documentation

includeUsers

char** includeUsers

includeUsersCount

uint32_t includeUsersCount

maxCount

uint32_t maxCount

Macro Definition Documentation

__TRTCTYPEDEF_H__

#define __TRTCTYPEDEF_H__

TARGET_PLATFORM_DESKTOP

#define TARGET_PLATFORM_DESKTOP   ((__APPLE__ && TARGET_OS_MAC && !TARGET_OS_IPHONE) || _WIN32)

TARGET_PLATFORM_MAC

#define TARGET_PLATFORM_MAC   (__APPLE__ && TARGET_OS_MAC && !TARGET_OS_IPHONE)

TARGET_PLATFORM_PHONE

#define TARGET_PLATFORM_PHONE   (__ANDROID__ || (__APPLE__ && TARGET_OS_IOS))

TRTC_API [1/5]

#define TRTC_API   __declspec(dllexport)

TRTC_API [2/5]

#define TRTC_API   __declspec(dllimport)

TRTC_API [3/5]

#define TRTC_API   __attribute__((visibility("default")))

TRTC_API [4/5]

#define TRTC_API   __attribute__((visibility("default")))

TRTC_API [5/5]

#define TRTC_API

TRTCDeviceStateActive

#define TRTCDeviceStateActive   TXMediaDeviceStateActive

TRTCDeviceStateAdd

#define TRTCDeviceStateAdd   TXMediaDeviceStateAdd

TRTCDeviceStateRemove

#define TRTCDeviceStateRemove   TXMediaDeviceStateRemove

TRTCDeviceTypeCamera

#define TRTCDeviceTypeCamera   TXMediaDeviceTypeCamera

TRTCDeviceTypeMic

#define TRTCDeviceTypeMic   TXMediaDeviceTypeMic

TRTCDeviceTypeSpeaker

#define TRTCDeviceTypeSpeaker   TXMediaDeviceTypeSpeaker

TRTCDeviceTypeUnknow

#define TRTCDeviceTypeUnknow   TXMediaDeviceTypeUnknown

WIN32_LEAN_AND_MEAN

#define WIN32_LEAN_AND_MEAN

Typedef Documentation

ITRTCDeviceCollection

typedef ITXDeviceCollection ITRTCDeviceCollection

ITRTCDeviceInfo

typedef ITXDeviceInfo ITRTCDeviceInfo

TRTCDeviceState

typedef TXMediaDeviceState TRTCDeviceState

Device operation

This enumerated value is used to notify the status change of the local device onDeviceChange.

TRTCDeviceType

typedef TXMediaDeviceType TRTCDeviceType

Device type (for desktop platforms only)

This enumerated value is used to define three types of audio/video devices, namely, camera, mic, and speaker, so that the same device management API can control the three different types of devices. Starting from v8.0, TRTC redefines TXMediaDeviceType in TXDeviceManager to replace TRTCMediaDeviceType on legacy versions. Only the definition of TRTCMediaDeviceType is retained here for compatibility with customer code on legacy versions.

TXView

typedef void * TXView

[VIEW] Rendering control that renders the video image

There are many APIs in TRTC that need to manipulate the video image, for which you should specify the video rendering control. The SDK provides rendering controls for different terminal platforms. As the all-platform C++ APIs need to use a unified parameter type, you should uniformly convert the rendering controls into pointers in TXView type when calling these APIs:

• iOS: you can use the UIView object as the rendering control. When calling the C++ APIs, please pass in the pointer to the UIView object (which needs to be forcibly converted to the void* type).
• macOS: you can use the NSView object as the rendering control. When calling the C++ APIs, please pass in the pointer to the NSView object (which needs to be forcibly converted to the void* type).
• Android: when calling the C++ APIs, please pass in the jobject pointer to the TXCloudVideoView object (which needs to be forcibly converted to the void* type).
• Windows: you can use the window handle HWND as the rendering control. When calling the C++ APIs, you need to forcibly convert the HWND to void* type.

Code sample 1. Use the all-platform C++ APIs under QT

QWidget *videoView;
// The relevant code for setting the videoView is omitted here...
getTRTCShareInstance()->startLocalPreview(reinterpret_cast<TXView>(videoView->winId()));

Code sample 2. Call the all-platform C++ APIs through JNI on Android

native void nativeStartLocalPreview(String userId, int streamType, TXCloudVideoView view);
//...
Java_com_example_test_MainActivity_nativeStartRemoteView(JNIEnv *env, jobject thiz, jstring user_id, jint stream_type, jobject view) {
    const char *user_id_chars = env->GetStringUTFChars(user_id, nullptr);
    trtc_cloud->startRemoteView(user_id_chars, (trtc::TRTCVideoStreamType)stream_type, view);
    env->ReleaseStringUTFChars(user_id, user_id_chars);
}

Enumeration Type Documentation

TRTCAppScene

enum TRTCAppScene

Use cases

TRTC features targeted optimizations for common audio/video application scenarios to meet the differentiated requirements in various verticals. The main scenarios can be divided into the following two categories:

• Live streaming scenario (LIVE): including LIVE (audio + video) and VoiceChatRoom (pure audio). In the live streaming scenario, users are divided into two roles: "anchor" and "audience". A single room can sustain up to 100,000 concurrent online users. This is suitable for live streaming to a large audience.
• Real-Time scenario (RTC): including VideoCall (audio + video) and AudioCall (pure audio). In the real-time scenario, there is no role difference between users, but a single room can sustain only up to 300 concurrent online users. This is suitable for small-scale real-time communication.

Enumerator
TRTCAppSceneVideoCall	In the video call scenario, 720p and 1080p HD image quality is supported. A single room can sustain up to 300 concurrent online users, and up to 50 of them can speak simultaneously. Use cases: [one-to-one video call], [video conferencing with up to 300 participants], [online medical diagnosis], [small class], [video interview], etc.
TRTCAppSceneLIVE	In the interactive video live streaming scenario, mic can be turned on/off smoothly without waiting for switchover, and the anchor latency is as low as less than 300 ms. Live streaming to hundreds of thousands of concurrent users in the audience role is supported with the playback latency down to 1,000 ms. Use cases: [low-latency interactive live streaming], [big class], [anchor competition], [video dating room], [online interactive classroom], [remote training], [large-scale conferencing], etc. Note In this scenario, you must use the role field in TRTCParams to specify the role of the current user.
TRTCAppSceneAudioCall	Audio call scenario, where the SPEECH sound quality is used by default. A single room can sustain up to 300 concurrent online users, and up to 50 of them can speak simultaneously. Use cases: [one-to-one audio call], [audio conferencing with up to 300 participants], [audio chat], [online Werewolf], etc.
TRTCAppSceneVoiceChatRoom	In the interactive audio live streaming scenario, mic can be turned on/off smoothly without waiting for switchover, and the anchor latency is as low as less than 300 ms. Live streaming to hundreds of thousands of concurrent users in the audience role is supported with the playback latency down to 1,000 ms. Use cases: [audio club], [online karaoke room], [music live room], [FM radio], etc. Note In this scenario, you must use the role field in TRTCParams to specify the role of the current user.

TRTCAudioFrameFormat

enum TRTCAudioFrameFormat

Audio frame content format

Enumerator
TRTCAudioFrameFormatNone	None.
TRTCAudioFrameFormatPCM	Audio data in PCM format.

TRTCAudioQuality

enum TRTCAudioQuality

Sound quality

TRTC provides three well-tuned modes to meet the differentiated requirements for sound quality in various verticals:

• Speech mode (Speech): it is suitable for application scenarios that focus on human communication. In this mode, the audio transfer is more resistant, and TRTC uses various voice processing technologies to ensure the optimal smoothness even in weak network environments.
• Music mode (Music): it is suitable for scenarios with demanding requirements for music. In this mode, the amount of transferred audio data is very large, and TRTC uses various technologies to ensure that the high-fidelity details of music signals can be restored in each frequency band.
• Default mode (Default): it is between Speech and Music. In this mode, the reproduction of music is better than that in Speech mode, and the amount of transferred data is much lower than that in Music mode; therefore, this mode has good adaptability to various scenarios.

Enumerator
TRTCAudioQualitySpeech	Speech mode: sample rate: 16 kHz; mono channel; bitrate: 16 Kbps. This mode has the best resistance among all modes and is suitable for audio call scenarios, such as online meeting and audio call.
TRTCAudioQualityDefault	Default mode: sample rate: 48 kHz; mono channel; bitrate: 50 Kbps. This mode is between the speech mode and the music mode as the default mode in the SDK and is recommended.
TRTCAudioQualityMusic	Music mode: sample rate: 48 kHz; full-band stereo; bitrate: 128 Kbps. This mode is suitable for scenarios where Hi-Fi music transfer is required, such as online karaoke and music live streaming.

TRTCAudioRecordingContent

enum TRTCAudioRecordingContent

Audio recording content type

This enumerated type is used in the audio recording API startAudioRecording to specify the content of the recorded audio.

Enumerator
TRTCAudioRecordingContentAll	Record both local and remote audio.
TRTCAudioRecordingContentLocal	Record local audio only.
TRTCAudioRecordingContentRemote	Record remote audio only.

TRTCAVStatusChangeReason

enum TRTCAVStatusChangeReason

Reasons for playback status changes

This enumerated type is used in the video status changed API onRemoteVideoStatusUpdated to specify the reason for the current video status change.

Enumerator
TRTCAVStatusChangeReasonInternal	Default value.
TRTCAVStatusChangeReasonBufferingBegin	The stream enters the "Loading" state due to network congestion.
TRTCAVStatusChangeReasonBufferingEnd	The stream enters the "Playing" state after network recovery.
TRTCAVStatusChangeReasonLocalStarted	As a start-related API was directly called locally, the stream enters the "Playing" state.
TRTCAVStatusChangeReasonLocalStopped	As a stop-related API was directly called locally, the stream enters the "Stopped" state.
TRTCAVStatusChangeReasonRemoteStarted	As the remote user started (or resumed) publishing the video stream, the stream enters the "Loading" or "Playing" state.
TRTCAVStatusChangeReasonRemoteStopped	As the remote user stopped (or paused) publishing the video stream, the stream enters the "Stopped" state.

TRTCAVStatusType

enum TRTCAVStatusType

Audio/Video playback status

This enumerated type is used in the video status changed API onRemoteVideoStatusUpdated to specify the current video status.

Enumerator
TRTCAVStatusStopped	Stopped.
TRTCAVStatusPlaying	Playing.
TRTCAVStatusLoading	Loading.

TRTCBeautyStyle

enum TRTCBeautyStyle

Beauty (skin smoothing) filter algorithm

TRTC has multiple built-in skin smoothing algorithms. You can select the one most suitable for your product.

Enumerator
TRTCBeautyStyleSmooth	Smooth style, which uses a more radical algorithm for more obvious effect and is suitable for show live streaming.
TRTCBeautyStyleNature	Natural style, which retains more facial details for more natural effect and is suitable for most live streaming use cases.

TRTCLocalRecordType

enum TRTCLocalRecordType

Media recording type

This enumerated type is used in the local media recording API startLocalRecording to specify whether to record audio/video files or pure audio files.

Enumerator
TRTCLocalRecordType_Audio	Record audio only.
TRTCLocalRecordType_Video	Record video only.
TRTCLocalRecordType_Both	Record both audio and video.

TRTCLogLevel

enum TRTCLogLevel

Log level

Different log levels indicate different levels of details and number of logs. We recommend you set the log level to TRTCLogLevelInfo generally.

Enumerator
TRTCLogLevelVerbose	Output logs at all levels.
TRTCLogLevelDebug	Output logs at the DEBUG, INFO, WARNING, ERROR, and FATAL levels.
TRTCLogLevelInfo	Output logs at the INFO, WARNING, ERROR, and FATAL levels.
TRTCLogLevelWarn	Output logs at the WARNING, ERROR, and FATAL levels.
TRTCLogLevelError	Output logs at the ERROR and FATAL levels.
TRTCLogLevelFatal	Output logs at the FATAL level.
TRTCLogLevelNone	Do not output any SDK logs.

TRTCMixInputType

enum TRTCMixInputType

Stream mix input type

Enumerator
TRTCMixInputTypeUndefined	Default. Considering the compatibility with older versions, if you specify the inputType as Undefined, the SDK will determine the stream mix input type according to the value of the pureAudio parameter
TRTCMixInputTypeAudioVideo	Mix both audio and video.
TRTCMixInputTypePureVideo	Mix video only.
TRTCMixInputTypePureAudio	Mix audio only.
TRTCMixInputTypeWatermark	Mix watermark In this case, you don't need to specify the userId parameter, but you need to specify the image parameter. It is recommended to use png format.

TRTCQosControlMode

enum TRTCQosControlMode

QoS control mode (disused)

Enumerator
TRTCQosControlModeClient	Client-based control, which is for internal debugging of SDK and shall not be used by users.
TRTCQosControlModeServer	On-cloud control, which is the default and recommended mode.

TRTCQuality

enum TRTCQuality

Network quality

TRTC evaluates the current network quality once every two seconds. The evaluation results are divided into six levels: Excellent indicates the best, and Down indicates the worst.

Enumerator
TRTCQuality_Unknown	Undefined.
TRTCQuality_Excellent	The current network is excellent.
TRTCQuality_Good	The current network is good.
TRTCQuality_Poor	The current network is fair.
TRTCQuality_Bad	The current network is bad.
TRTCQuality_Vbad	The current network is very bad.
TRTCQuality_Down	The current network cannot meet the minimum requirements of TRTC.

TRTCRoleType

enum TRTCRoleType

Role

Role is applicable only to live streaming scenarios (TRTCAppSceneLIVE and TRTCAppSceneVoiceChatRoom). Users are divided into two roles:

• Anchor, who can publish their audio/video streams. There is a limit on the number of anchors. Up to 50 anchors are allowed to publish streams at the same time in one room.
• Audience, who can only listen to or watch audio/video streams of anchors in the room. If they want to publish their streams, they need to switch to the "anchor" role first through switchRole. One room can sustain up to 100,000 concurrent online users in the audience role.

Enumerator
TRTCRoleAnchor	An anchor can publish their audio/video streams. There is a limit on the number of anchors. Up to 50 anchors are allowed to publish streams at the same time in one room.
TRTCRoleAudience	Audience can only listen to or watch audio/video streams of anchors in the room. If they want to publish their streams, they need to switch to the "anchor" role first through switchRole. One room can sustain up to 100,000 concurrent online users in the audience role.

TRTCScreenCaptureSourceType

enum TRTCScreenCaptureSourceType

Screen sharing target type (for desktops only)

Enumerator
TRTCScreenCaptureSourceTypeUnknown	Undefined.
TRTCScreenCaptureSourceTypeWindow	The screen sharing target is the window of an application.
TRTCScreenCaptureSourceTypeScreen	The screen sharing target is the entire screen.
TRTCScreenCaptureSourceTypeCustom	The screen sharing target is a user-defined data source.

TRTCSnapshotSourceType

enum TRTCSnapshotSourceType

Data source of local video screenshot

The SDK can take screenshots from the following two data sources and save them as local files:

• Video stream: the SDK screencaptures the native video content from the video stream. The screenshots are not controlled by the display of the rendering control.
• Rendering layer: the SDK screencaptures the displayed video content from the rendering control, which can achieve the effect of WYSIWYG, but if the display area is too small, the screenshots will also be very small.

Enumerator
TRTCSnapshotSourceTypeStream	The SDK screencaptures the native video content from the video stream. The screenshots are not controlled by the display of the rendering control.
TRTCSnapshotSourceTypeView	The SDK screencaptures the displayed video content from the rendering control, which can achieve the effect of WYSIWYG, but if the display area is too small, the screenshots will also be very small.

TRTCTranscodingConfigMode

enum TRTCTranscodingConfigMode

Layout mode of On-Cloud MixTranscoding

TRTC's On-Cloud MixTranscoding service can mix multiple audio/video streams in the room into one stream. Therefore, you need to specify the layout scheme of the video images. The following layout modes are provided:

Enumerator
TRTCTranscodingConfigMode_Unknown	Undefined.
TRTCTranscodingConfigMode_Manual	Manual layout mode In this mode, you need to specify the precise position of each video image. This mode has the highest degree of freedom, but its ease of use is the worst: You need to enter all the parameters in TRTCTranscodingConfig, including the position coordinates of each video image (TRTCMixUser). You need to listen on the onUserVideoAvailable() and onUserAudioAvailable() event callbacks in TRTCCloudDelegate and constantly adjust the mixUsers parameter according to the audio/video status of each user with mic on in the current room.
TRTCTranscodingConfigMode_Template_PureAudio	Pure audio mode This mode is suitable for pure audio scenarios such as audio call (AudioCall) and audio chat room (VoiceChatRoom). You only need to set it once through the setMixTranscodingConfig() API after room entry, and then the SDK will automatically mix the audio of all mic-on users in the room into the current user's live stream. You don't need to set the mixUsers parameter in TRTCTranscodingConfig; instead, you only need to set the audioSampleRate, audioBitrate and audioChannels parameters.
TRTCTranscodingConfigMode_Template_PresetLayout	Preset layout mode This is the most popular layout mode, because it allows you to set the position of each video image in advance through placeholders, and then the SDK automatically adjusts it dynamically according to the number of video images in the room. In this mode, you still need to set the mixUsers parameter, but you can set userId as a "placeholder". Placeholder values include: "$PLACE_HOLDER_REMOTE$": image of remote user. Multiple images can be set. "$PLACE_HOLDER_LOCAL_MAIN$": local camera image. Only one image can be set. "$PLACE_HOLDER_LOCAL_SUB$": local screen sharing image. Only one image can be set. In this mode, you don't need to listen on the onUserVideoAvailable() and onUserAudioAvailable() callbacks in TRTCCloudDelegate to make real-time adjustments. Instead, you only need to call setMixTranscodingConfig() once after successful room entry. Then, the SDK will automatically populate the placeholders you set with real userId values.
TRTCTranscodingConfigMode_Template_ScreenSharing	Screen sharing mode This mode is suitable for screen sharing-based use cases such as online education and supported only by the SDKs for Windows and macOS. In this mode, the SDK will first build a canvas according to the target resolution you set (through the videoWidth and videoHeight parameters). Before the teacher enables screen sharing, the SDK will scale up the teacher's camera image and draw it onto the canvas. After the teacher enables screen sharing, the SDK will draw the video image shared on the screen onto the same canvas. The purpose of this layout mode is to ensure consistency in the output resolution of the mixtranscoding module and avoid problems with blurred screen during course replay and webpage playback (web players don't support adjustable resolution). Meanwhile, the audio of mic-on students will be mixed into the teacher's audio/video stream by default. br> Video content is primarily the shared screen in teaching mode, and it is a waste of bandwidth to transfer camera image and screen image at the same time. Therefore, the recommended practice is to directly draw the camera image onto the current screen through the setLocalVideoRenderCallback API. In this mode, you don't need to set the mixUsers parameter in TRTCTranscodingConfig, and the SDK will not mix students' images so as not to interfere with the screen sharing effect. br> You can set width x height in TRTCTranscodingConfig to 0 px x 0 px, and the SDK will automatically calculate a suitable resolution based on the aspect ratio of the user's current screen. If the teacher's current screen width is less than or equal to 1920 px, the SDK will use the actual resolution of the teacher's current screen. If the teacher's current screen width is greater than 1920 px, the SDK will select one of the three resolutions of 1920x1080 (16:9), 1920x1200 (16:10), and 1920x1440 (4:3) according to the current screen aspect ratio.

TRTCVideoBufferType

enum TRTCVideoBufferType

Video data transfer method

For custom capturing and rendering features, you need to use the following enumerated values to specify the method of transferring video data:

• Method 1. This method uses memory buffer to transfer video data. It is efficient on iOS but inefficient on Android. It is the only method supported on Windows currently.
• Method 2. This method uses texture to transfer video data. It is efficient on both iOS and Android but is not supported on Windows. To use this method, you should have a general familiarity with OpenGL programming.

Enumerator
TRTCVideoBufferType_Unknown	Undefined transfer method.
TRTCVideoBufferType_Buffer	Use memory buffer to transfer video data. iOS: PixelBuffer; Android: Direct Buffer for JNI layer; Windows: memory data block.
TRTCVideoBufferType_Texture	Use texture to transfer video data.

TRTCVideoFillMode

enum TRTCVideoFillMode

Video image fill mode

If the aspect ratio of the video display area is not equal to that of the video image, you need to specify the fill mode:

Enumerator
TRTCVideoFillMode_Fill	Fill mode: the video image will be centered and scaled to fill the entire display area, where parts that exceed the area will be cropped. The displayed image may be incomplete in this mode.
TRTCVideoFillMode_Fit	Fit mode: the video image will be scaled based on its long side to fit the display area, where the short side will be filled with black bars. The displayed image is complete in this mode, but there may be black bars.

TRTCVideoMirrorType

enum TRTCVideoMirrorType

Video mirror type

Video mirroring refers to the left-to-right flipping of the video image, especially for the local camera preview image. After mirroring is enabled, it can bring anchors a familiar "look into the mirror" experience.

Enumerator
TRTCVideoMirrorType_Auto	Auto mode: mirror the front camera's image but not the rear camera's image (for mobile devices only).
TRTCVideoMirrorType_Enable	Mirror the images of both the front and rear cameras.
TRTCVideoMirrorType_Disable	Disable mirroring for both the front and rear cameras.

TRTCVideoPixelFormat

enum TRTCVideoPixelFormat

Video pixel format

TRTC provides custom video capturing and rendering features.

• For the custom capturing feature, you can use the following enumerated values to describe the pixel format of the video you capture.
• For the custom rendering feature, you can specify the pixel format of the video you expect the SDK to call back.

Enumerator
TRTCVideoPixelFormat_Unknown	Undefined format.
TRTCVideoPixelFormat_I420	YUV420P (I420) format.
TRTCVideoPixelFormat_Texture_2D	OpenGL 2D texture format.
TRTCVideoPixelFormat_BGRA32	BGRA32 format.
TRTCVideoPixelFormat_RGBA32	RGBA format.

TRTCVideoQosPreference

enum TRTCVideoQosPreference

Image quality preference

TRTC has two control modes in weak network environments: "ensuring clarity" and "ensuring smoothness". Both modes will give priority to the transfer of audio data.

Enumerator
TRTCVideoQosPreferenceSmooth	Ensuring smoothness: in this mode, when the current network is unable to transfer a clear and smooth video image, the smoothness of the image will be given priority, but there will be blurs.
TRTCVideoQosPreferenceClear	Ensuring clarity (default value): in this mode, when the current network is unable to transfer a clear and smooth video image, the clarity of the image will be given priority, but there will be lags.

TRTCVideoResolution

enum TRTCVideoResolution

Video resolution

Here, only the landscape resolution (e.g., 640x360) is defined. If the portrait resolution (e.g., 360x640) needs to be used, Portrait must be selected for TRTCVideoResolutionMode.

Enumerator
TRTCVideoResolution_120_120	Aspect ratio: 1:1; resolution: 120x120; recommended bitrate (VideoCall): 80 Kbps; recommended bitrate (LIVE): 120 Kbps.
TRTCVideoResolution_160_160	Aspect ratio: 1:1; resolution: 160x160; recommended bitrate (VideoCall): 100 Kbps; recommended bitrate (LIVE): 150 Kbps.
TRTCVideoResolution_270_270	Aspect ratio: 1:1; resolution: 270x270; recommended bitrate (VideoCall): 200 Kbps; recommended bitrate (LIVE): 300 Kbps.
TRTCVideoResolution_480_480	Aspect ratio: 1:1; resolution: 480x480; recommended bitrate (VideoCall): 350 Kbps; recommended bitrate (LIVE): 500 Kbps.
TRTCVideoResolution_160_120	Aspect ratio: 4:3; resolution: 160x120; recommended bitrate (VideoCall): 100 Kbps; recommended bitrate (LIVE): 150 Kbps.
TRTCVideoResolution_240_180	Aspect ratio: 4:3; resolution: 240x180; recommended bitrate (VideoCall): 150 Kbps; recommended bitrate (LIVE): 250 Kbps.
TRTCVideoResolution_280_210	Aspect ratio: 4:3; resolution: 280x210; recommended bitrate (VideoCall): 200 Kbps; recommended bitrate (LIVE): 300 Kbps.
TRTCVideoResolution_320_240	Aspect ratio: 4:3; resolution: 320x240; recommended bitrate (VideoCall): 250 Kbps; recommended bitrate (LIVE): 375 Kbps.
TRTCVideoResolution_400_300	Aspect ratio: 4:3; resolution: 400x300; recommended bitrate (VideoCall): 300 Kbps; recommended bitrate (LIVE): 450 Kbps.
TRTCVideoResolution_480_360	Aspect ratio: 4:3; resolution: 480x360; recommended bitrate (VideoCall): 400 Kbps; recommended bitrate (LIVE): 600 Kbps.
TRTCVideoResolution_640_480	Aspect ratio: 4:3; resolution: 640x480; recommended bitrate (VideoCall): 600 Kbps; recommended bitrate (LIVE): 900 Kbps.
TRTCVideoResolution_960_720	Aspect ratio: 4:3; resolution: 960x720; recommended bitrate (VideoCall): 1000 Kbps; recommended bitrate (LIVE): 1500 Kbps.
TRTCVideoResolution_160_90	Aspect ratio: 16:9; resolution: 160x90; recommended bitrate (VideoCall): 150 Kbps; recommended bitrate (LIVE): 250 Kbps.
TRTCVideoResolution_256_144	Aspect ratio: 16:9; resolution: 256x144; recommended bitrate (VideoCall): 200 Kbps; recommended bitrate (LIVE): 300 Kbps.
TRTCVideoResolution_320_180	Aspect ratio: 16:9; resolution: 320x180; recommended bitrate (VideoCall): 250 Kbps; recommended bitrate (LIVE): 400 Kbps.
TRTCVideoResolution_480_270	Aspect ratio: 16:9; resolution: 480x270; recommended bitrate (VideoCall): 350 Kbps; recommended bitrate (LIVE): 550 Kbps.
TRTCVideoResolution_640_360	Aspect ratio: 16:9; resolution: 640x360; recommended bitrate (VideoCall): 500 Kbps; recommended bitrate (LIVE): 900 Kbps.
TRTCVideoResolution_960_540	Aspect ratio: 16:9; resolution: 960x540; recommended bitrate (VideoCall): 850 Kbps; recommended bitrate (LIVE): 1300 Kbps.
TRTCVideoResolution_1280_720	Aspect ratio: 16:9; resolution: 1280x720; recommended bitrate (VideoCall): 1200 Kbps; recommended bitrate (LIVE): 1800 Kbps.
TRTCVideoResolution_1920_1080	Aspect ratio: 16:9; resolution: 1920x1080; recommended bitrate (VideoCall): 2000 Kbps; recommended bitrate (LIVE): 3000 Kbps.

TRTCVideoResolutionMode

enum TRTCVideoResolutionMode

Video aspect ratio mode

Only the landscape resolution (e.g., 640x360) is defined in TRTCVideoResolution. If the portrait resolution (e.g., 360x640) needs to be used, Portrait must be selected for TRTCVideoResolutionMode.

Enumerator
TRTCVideoResolutionModeLandscape	Landscape resolution, such as TRTCVideoResolution_640_360 + TRTCVideoResolutionModeLandscape = 640x360.
TRTCVideoResolutionModePortrait	Portrait resolution, such as TRTCVideoResolution_640_360 + TRTCVideoResolutionModePortrait = 360x640.

TRTCVideoRotation

enum TRTCVideoRotation

Video image rotation direction

TRTC provides rotation angle setting APIs for local and remote images. The following rotation angles are all clockwise.

Enumerator
TRTCVideoRotation0	No rotation.
TRTCVideoRotation90	Clockwise rotation by 90 degrees.
TRTCVideoRotation180	Clockwise rotation by 180 degrees.
TRTCVideoRotation270	Clockwise rotation by 270 degrees.

TRTCVideoStreamType

enum TRTCVideoStreamType

Video stream type

TRTC provides three different video streams, including:

• HD big image: it is generally used to transfer video data from the camera.
• Smooth small image: it has the same content as the big image, but with lower resolution and bitrate and thus lower definition.
• Substream image: it is generally used for screen sharing. Only one user in the room is allowed to publish the substream video image at any time, while other users must wait for this user to close the substream before they can publish their own substream.

  Attention

  The SDK does not support enabling the smooth small image alone, which must be enabled together with the big image. It will automatically set the resolution and bitrate of the small image.

Enumerator
TRTCVideoStreamTypeBig	HD big image: it is generally used to transfer video data from the camera.
TRTCVideoStreamTypeSmall	Smooth small image: it has the same content as the big image, but with lower resolution and bitrate and thus lower definition.
TRTCVideoStreamTypeSub	Substream image: it is generally used for screen sharing. Only one user in the room is allowed to publish the substream video image at any time, while other users must wait for this user to close the substream before they can publish their own substream.

TRTCWaterMarkSrcType

enum TRTCWaterMarkSrcType

Watermark image source type

Enumerator
TRTCWaterMarkSrcTypeFile	Path of the image file, which can be in BMP, GIF, JPEG, PNG, TIFF, Exif, WMF, or EMF format.
TRTCWaterMarkSrcTypeBGRA32	Memory block in BGRA32 format.
TRTCWaterMarkSrcTypeRGBA32	Memory block in RGBA32 format.