TRTCCloudDef

Data Structures
class	TRTCParams
class	TRTCVideoEncParam
class	TRTCNetworkQosParam
class	TRTCRenderParams
class	TRTCQualityInfo
class	TRTCVolumeInfo
class	TRTCSpeedTestParams
class	TRTCSpeedTestResult
class	TRTCVideoFrame
class	TRTCAudioFrame
class	TRTCMixUser
class	TRTCTranscodingConfig
class	TRTCPublishCDNParam
class	TRTCAudioRecordingParams
class	TRTCLocalRecordingParams
class	TRTCAudioEffectParam
class	TRTCSwitchRoomConfig
class	TRTCAudioFrameDelegateFormat
class	TRTCScreenCaptureSourceInfo
class	TRTCAudioParallelParams

Typedefs
typedef UIView	TXView
typedef UIImage	TXImage
typedef UIEdgeInsets	TXEdgeInsets
typedef TXMediaDeviceInfo	TRTCMediaDeviceInfo

Enumerations
enum	TRTCVideoResolution : NSInteger
enum	TRTCVideoResolutionMode : NSInteger
enum	TRTCVideoStreamType : NSInteger
enum	TRTCVideoFillMode : NSInteger
enum	TRTCVideoRotation : NSInteger
enum	TRTCBeautyStyle : NSInteger
enum	TRTCVideoPixelFormat : NSInteger
enum	TRTCVideoBufferType : NSInteger
enum	TRTCVideoMirrorType : NSUInteger
enum	TRTCLocalVideoMirrorType : NSUInteger
enum	TRTCSnapshotSourceType : NSUInteger
enum	TRTCAppScene : NSInteger
enum	TRTCRoleType : NSInteger
enum	TRTCQosControlMode : NSInteger
enum	TRTCVideoQosPreference : NSInteger
enum	TRTCQuality : NSInteger
enum	TRTCAVStatusType : NSUInteger
enum	TRTCAVStatusChangeReason : NSUInteger
enum	TRTCAudioSampleRate : NSInteger
enum	TRTCAudioQuality : NSInteger
enum	TRTCAudioRoute : NSInteger
enum	TRTCReverbType : NSInteger
enum	TRTCVoiceChangerType : NSInteger
enum	TRTCSystemVolumeType : NSInteger
enum	TRTCLogLevel : NSInteger
enum	TRTCGSensorMode : NSInteger
enum	TRTCScreenCaptureSourceType : NSInteger
enum	TRTCTranscodingConfigMode : NSInteger
enum	TRTCRecordType : NSUInteger
enum	TRTCMixInputType : NSUInteger
enum	TRTCMediaDeviceType : NSInteger
enum	TRTCAudioRecordingContent : NSUInteger

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

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Data Structure Documentation

TRTCParams

class 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.

Inheritance diagram for TRTCParams:

Properties
UInt32	sdkAppId
NSString *	userId
NSString *	userSig
UInt32	roomId
NSString *	strRoomId
TRTCRoleType	role
NSString *	streamId
NSString *	userDefineRecordId
NSString *	privateMapKey
NSString *	bussInfo

Property Documentation

bussInfo

- (NSString*) bussInfo

readwritenonatomiccopy

privateMapKey

- (NSString*) privateMapKey

readwritenonatomiccopy

role

- (TRTCRoleType) role

readwritenonatomicassign

roomId

- (UInt32) roomId

readwritenonatomicassign

sdkAppId

- (UInt32) sdkAppId

readwritenonatomicassign

streamId

- (NSString*) streamId

readwritenonatomiccopy

strRoomId

- (NSString*) strRoomId

readwritenonatomiccopy

userDefineRecordId

- (NSString*) userDefineRecordId

readwritenonatomiccopy

userId

- (NSString*) userId

readwritenonatomiccopy

userSig

- (NSString*) userSig

readwritenonatomiccopy

TRTCVideoEncParam

class 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.

Inheritance diagram for TRTCVideoEncParam:

Properties
TRTCVideoResolution	videoResolution
TRTCVideoResolutionMode	resMode
int	videoFps
int	videoBitrate
int	minVideoBitrate
BOOL	enableAdjustRes

Property Documentation

enableAdjustRes

- (BOOL) enableAdjustRes

readwritenonatomicassign

minVideoBitrate

- (int) minVideoBitrate

readwritenonatomicassign

resMode

- (TRTCVideoResolutionMode) resMode

readwritenonatomicassign

videoBitrate

- (int) videoBitrate

readwritenonatomicassign

videoFps

- (int) videoFps

readwritenonatomicassign

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

readwritenonatomicassign

TRTCNetworkQosParam

class 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").

Inheritance diagram for TRTCNetworkQosParam:

Properties
TRTCVideoQosPreference	preference
TRTCQosControlMode	controlMode

Property Documentation

controlMode

- (TRTCQosControlMode) controlMode

readwritenonatomicassign

preference

- (TRTCVideoQosPreference) preference

readwritenonatomicassign

TRTCRenderParams

class TRTCRenderParams

Rendering parameters of video image

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

Inheritance diagram for TRTCRenderParams:

Properties
TRTCVideoRotation	rotation
TRTCVideoFillMode	fillMode
TRTCVideoMirrorType	mirrorType

Property Documentation

fillMode

- (TRTCVideoFillMode) fillMode

readwritenonatomicassign

mirrorType

- (TRTCVideoMirrorType) mirrorType

readwritenonatomicassign

rotation

- (TRTCVideoRotation) rotation

readwritenonatomicassign

TRTCQualityInfo

class 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.

Inheritance diagram for TRTCQualityInfo:

Properties
NSString *	userId
TRTCQuality	quality

Property Documentation

quality

- (TRTCQuality) quality

readwritenonatomicassign

Network quality.

userId

- (NSString*) userId

readwritenonatomiccopy

User ID.

TRTCVolumeInfo

class TRTCVolumeInfo

Volume

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

Inheritance diagram for TRTCVolumeInfo:

Properties
NSString *	userId
NSUInteger	volume

Property Documentation

userId

- (NSString*) userId

readwritenonatomiccopy

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

volume

- (NSUInteger) volume

readwritenonatomicassign

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

TRTCSpeedTestParams

class 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).

Inheritance diagram for TRTCSpeedTestParams:

Properties
uint32_t	sdkAppId
NSString *	userId
NSString *	userSig
NSInteger	expectedUpBandwidth
NSInteger	expectedDownBandwidth

Property Documentation

expectedDownBandwidth

- (NSInteger) expectedDownBandwidth

readwritenonatomicassign

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

expectedUpBandwidth

- (NSInteger) expectedUpBandwidth

readwritenonatomicassign

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

sdkAppId

- (uint32_t) sdkAppId

readwritenonatomicassign

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

userId

- (NSString*) userId

readwritenonatomiccopy

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

userSig

- (NSString*) userSig

readwritenonatomiccopy

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

TRTCSpeedTestResult

class 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).

Inheritance diagram for TRTCSpeedTestResult:

Properties
BOOL	success
NSString *	errMsg
NSString *	ip
TRTCQuality	quality
float	upLostRate
float	downLostRate
uint32_t	rtt
NSInteger	availableUpBandwidth
NSInteger	availableDownBandwidth

Property Documentation

availableDownBandwidth

- (NSInteger) availableDownBandwidth

readwritenonatomicassign

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

availableUpBandwidth

- (NSInteger) availableUpBandwidth

readwritenonatomicassign

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

downLostRate

- (float) downLostRate

readwritenonatomicassign

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

- (NSString*) errMsg

readwritenonatomiccopy

Error message for network speed test.

ip

- (NSString*) ip

readwritenonatomiccopy

Server IP address.

quality

- (TRTCQuality) quality

readwritenonatomicassign

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

rtt

- (uint32_t) rtt

readwritenonatomicassign

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

readwritenonatomicassign

Whether the network speed test is successful.

upLostRate

- (float) upLostRate

readwritenonatomicassign

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.

TRTCVideoFrame

class 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.

Inheritance diagram for TRTCVideoFrame:

Properties
uint32_t	width
uint32_t	height
uint64_t	timestamp

Property Documentation

height

- (uint32_t) height

readwritenonatomicassign

timestamp

- (uint64_t) timestamp

readwritenonatomicassign

width

- (uint32_t) width

readwritenonatomicassign

TRTCAudioFrame

class TRTCAudioFrame

Audio frame data

Inheritance diagram for TRTCAudioFrame:

TRTCMixUser

class 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.

Inheritance diagram for TRTCMixUser:

Properties
BOOL	pureAudio
TRTCMixInputType	inputType
int	renderMode
NSString *	image

Property Documentation

image

- (NSString*) image

readwritenonatomiccopy

inputType

- (TRTCMixInputType) inputType

readwritenonatomicassign

pureAudio

- (BOOL) pureAudio

readwritenonatomicassign

renderMode

- (int) renderMode

readwritenonatomicassign

TRTCTranscodingConfig

class 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.

Inheritance diagram for TRTCTranscodingConfig:

Properties
TRTCTranscodingConfigMode	mode
int	appId
int	bizId
int	videoWidth
int	videoHeight
int	videoBitrate
int	videoFramerate
int	videoGOP
int	backgroundColor
NSString *	backgroundImage
int	audioSampleRate
int	audioBitrate
int	audioChannels
int	audioCodec
NSArray< TRTCMixUser * > *	mixUsers
NSString *	streamId

Property Documentation

appId

- (int) appId

readwritenonatomicassign

audioBitrate

- (int) audioBitrate

readwritenonatomicassign

audioChannels

- (int) audioChannels

readwritenonatomicassign

audioCodec

- (int) audioCodec

readwritenonatomicassign

audioSampleRate

- (int) audioSampleRate

readwritenonatomicassign

backgroundColor

- (int) backgroundColor

readwritenonatomicassign

backgroundImage

- (NSString*) backgroundImage

readwritenonatomiccopy

bizId

- (int) bizId

readwritenonatomicassign

mixUsers

- (NSArray<TRTCMixUser *>*) mixUsers

readwritenonatomiccopy

mode

- (TRTCTranscodingConfigMode) mode

readwritenonatomicassign

streamId

- (NSString*) streamId

readwritenonatomiccopy

videoBitrate

- (int) videoBitrate

readwritenonatomicassign

videoFramerate

- (int) videoFramerate

readwritenonatomicassign

videoGOP

- (int) videoGOP

readwritenonatomicassign

videoHeight

- (int) videoHeight

readwritenonatomicassign

videoWidth

- (int) videoWidth

readwritenonatomicassign

TRTCPublishCDNParam

class 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.

Inheritance diagram for TRTCPublishCDNParam:

Properties
int	appId
int	bizId
NSString *	url
NSString *	streamId

Property Documentation

appId

- (int) appId

readwritenonatomicassign

bizId

- (int) bizId

readwritenonatomicassign

streamId

- (NSString*) streamId

readwritenonatomicstrong

url

- (NSString*) url

readwritenonatomicstrong

TRTCAudioRecordingParams

class TRTCAudioRecordingParams

Local audio file recording parameters

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

Inheritance diagram for TRTCAudioRecordingParams:

Properties
NSString *	filePath
TRTCAudioRecordingContent	recordingContent

Property Documentation

filePath

- (NSString*) filePath

readwritenonatomicstrong

recordingContent

- (TRTCAudioRecordingContent) recordingContent

readwritenonatomicassign

TRTCLocalRecordingParams

class 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.

Inheritance diagram for TRTCLocalRecordingParams:

Properties
NSString *	filePath

Property Documentation

filePath

- (NSString*) filePath

readwritenonatomiccopy

TRTCAudioEffectParam

class 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 YES, the file is a "sound effect" file.

Inheritance diagram for TRTCAudioEffectParam:

Instance Methods
(_Nonnull instancetype)	- init
(_Nonnull instancetype)	- initWith:path:
Class Methods
(_Nonnull instancetype)	+ new
Properties
int	effectId
int	loopCount
BOOL	publish
int	volume

Method Documentation

init()

- (_Nonnull instancetype) init

initWith:path:()

- (_Nonnull instancetype) initWith:		(int)	effectId
path:		(NSString *_Nonnull)	path

new()

+ (_Nonnull instancetype) new

Property Documentation

effectId

- (int) effectId

readwritenonatomicassign

loopCount

- (int) loopCount

readwritenonatomicassign

publish

- (BOOL) publish

readwritenonatomicassign

volume

- (int) volume

readwritenonatomicassign

TRTCSwitchRoomConfig

class 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.

Inheritance diagram for TRTCSwitchRoomConfig:

Properties
UInt32	roomId
NSString *	strRoomId
NSString *	userSig
NSString *	privateMapKey

Property Documentation

privateMapKey

- (NSString*) privateMapKey

readwritenonatomiccopy

roomId

- (UInt32) roomId

readwritenonatomicassign

strRoomId

- (NSString*) strRoomId

readwritenonatomiccopy

userSig

- (NSString*) userSig

readwritenonatomiccopy

TRTCAudioFrameDelegateFormat

class TRTCAudioFrameDelegateFormat

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.

Inheritance diagram for TRTCAudioFrameDelegateFormat:

Properties
TRTCAudioSampleRate	sampleRate
int	channels
int	samplesPerCall

Property Documentation

channels

- (int) channels

readwritenonatomicassign

sampleRate

- (TRTCAudioSampleRate) sampleRate

readwritenonatomicassign

samplesPerCall

- (int) samplesPerCall

readwritenonatomicassign

TRTCScreenCaptureSourceInfo

class TRTCScreenCaptureSourceInfo

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.

Inheritance diagram for TRTCScreenCaptureSourceInfo:

TRTCAudioParallelParams

class TRTCAudioParallelParams

parameter of the parallel strategy of remote audio streams

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

Inheritance diagram for TRTCAudioParallelParams:

Properties
UInt32	maxCount
NSArray< NSString * > *	includeUsers

Property Documentation

includeUsers

- (NSArray<NSString *>*) includeUsers

readwritenonatomicstrong

maxCount

- (UInt32) maxCount

readwritenonatomicassign

Typedef Documentation

TRTCMediaDeviceInfo

typedef TXMediaDeviceInfo TRTCMediaDeviceInfo

TXEdgeInsets

typedef NSEdgeInsets TXEdgeInsets

TXImage

typedef NSImage TXImage

TXView

typedef NSView 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.

• On iOS, you can directly use UIView as the video rendering control, and the SDK will draw the video image on the UIView you provide.
• On macOS, you can directly use NSView as the video rendering control, and the SDK will draw the video image on the NSView you provide. Below is the sample code:

  UIView *videoView = [[UIView alloc] initWithFrame:CGRectMake(0, 0, 360, 640)];
  [self.view addSubview:videoView];
  [trtcCloud startLocalPreview:YES view:_localView];
  

Enumeration Type Documentation

TRTCAppScene

enum TRTCAppScene : NSInteger

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.

TRTCAudioQuality

enum TRTCAudioQuality : NSInteger

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 : NSUInteger

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.

TRTCAudioRoute

enum TRTCAudioRoute : NSInteger

Audio route (i.e., audio playback mode)

"Audio route" determines whether the sound is played back from the speaker or receiver of a mobile device; therefore, this API is applicable only to mobile devices such as phones. Generally, a phone has two speakers: one is the receiver at the top, and the other is the stereo speaker at the bottom.

• If the audio route is set to the receiver, the volume is relatively low, and the sound can be heard clearly only when the phone is put near the ear. This mode has a high level of privacy and is suitable for answering calls.
• If the audio route is set to the speaker, the volume is relatively high, so there is no need to put the phone near the ear. Therefore, this mode can implement the "hands-free" feature.

Enumerator
TRTCAudioModeSpeakerphone	Speakerphone: the speaker at the bottom is used for playback (hands-free). With relatively high volume, it is used to play music out loud.
TRTCAudioModeEarpiece	Earpiece: the receiver at the top is used for playback. With relatively low volume, it is suitable for call scenarios that require privacy.

TRTCAudioSampleRate

enum TRTCAudioSampleRate : NSInteger

Audio sample rate

The audio sample rate is used to measure the audio fidelity. A higher sample rate indicates higher fidelity. If there is music in the use case, TRTCAudioSampleRate48000 is recommended.

Enumerator
TRTCAudioSampleRate16000	16 kHz sample rate
TRTCAudioSampleRate32000	32 kHz sample rate
TRTCAudioSampleRate44100	44.1 kHz sample rate
TRTCAudioSampleRate48000	48 kHz sample rate

TRTCAVStatusChangeReason

enum TRTCAVStatusChangeReason : NSUInteger

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 : NSUInteger

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 : NSInteger

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.
TRTCBeautyStylePitu	Pitu style, which is provided by YouTu Lab. Its skin smoothing effect is between the smooth style and the natural style, that is, it retains more skin details than the smooth style and has a higher skin smoothing degree than the natural style.

TRTCGSensorMode

enum TRTCGSensorMode : NSInteger

G-sensor switch (for mobile devices only)

Enumerator
TRTCGSensorMode_Disable	Do not adapt to G-sensor orientation This mode is the default value for desktop platforms. In this mode, the video image published by the current user is not affected by the change of the G-sensor orientation.
TRTCGSensorMode_UIAutoLayout	Adapt to G-sensor orientation This mode is the default value on mobile platforms. In this mode, the video image published by the current user is adjusted according to the G-sensor orientation, while the orientation of the local preview image remains unchanged. One of the adaptation modes currently supported by the SDK is as follows: when the phone or tablet is upside down, in order to ensure that the screen orientation seen by the remote user is normal, the SDK will automatically rotate the published video image by 180 degrees. If the UI layer of your application has enabled G-sensor adaption, we recommend you use the UIFixLayout mode.
TRTCGSensorMode_UIFixLayout	Adapt to G-sensor orientation In this mode, the video image published by the current user is adjusted according to the G-sensor orientation, and the local preview image will also be rotated accordingly. One of the features currently supported is as follows: when the phone or tablet is upside down, in order to ensure that the screen orientation seen by the remote user is normal, the SDK will automatically rotate the published video image by 180 degrees. If the UI layer of your application doesn't support G-sensor adaption, but you want the video image in the SDK to adapt to the G-sensor orientation, we recommend you use the UIFixLayout mode.

TRTCLocalVideoMirrorType

enum TRTCLocalVideoMirrorType : NSUInteger

Old version of TRTCVideoMirrorType, reserved for compatibility with older interface.

Enumerator
TRTCLocalVideoMirrorType_Auto
TRTCLocalVideoMirrorType_Enable
TRTCLocalVideoMirrorType_Disable

TRTCLogLevel

enum TRTCLogLevel : NSInteger

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.

TRTCMediaDeviceType

enum TRTCMediaDeviceType : NSInteger

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.

Enumerator
TRTCMediaDeviceTypeUnknown	undefined device type
TRTCMediaDeviceTypeAudioInput	microphone
TRTCMediaDeviceTypeAudioOutput	speaker
TRTCMediaDeviceTypeVideoCamera	camera
TRTCMediaDeviceTypeVideoWindow	windows(for screen share)
TRTCMediaDeviceTypeVideoScreen	screen (for screen share)

TRTCMixInputType

enum TRTCMixInputType : NSUInteger

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 : NSInteger

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 : NSInteger

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.

TRTCRecordType

enum TRTCRecordType : NSUInteger

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
TRTCRecordTypeAudio	Record audio only.
TRTCRecordTypeVideo	Record video only.
TRTCRecordTypeBoth	Record both audio and video.

TRTCReverbType

enum TRTCReverbType : NSInteger

Audio reverb mode

This enumerated value is used to set the audio reverb mode in the live streaming scenario and is often used in show live streaming.

Enumerator
TRTCReverbType_0	Disable reverb.
TRTCReverbType_1	KTV.
TRTCReverbType_2	Small room.
TRTCReverbType_3	Hall.
TRTCReverbType_4	Deep.
TRTCReverbType_5	Resonant.
TRTCReverbType_6	Metallic.
TRTCReverbType_7	Husky.

TRTCRoleType

enum TRTCRoleType : NSInteger

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 : NSInteger

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.

TRTCSnapshotSourceType

enum TRTCSnapshotSourceType : NSUInteger

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.

TRTCSystemVolumeType

enum TRTCSystemVolumeType : NSInteger

System volume type (only for mobile devices)

Smartphones usually have two types of system volume: call volume and media volume.

• Call volume is designed for call scenarios. It comes with acoustic echo cancellation (AEC) and supports audio capturing by Bluetooth earphones, but its sound quality is average. If you cannot turn the volume down to 0 (i.e., mute the phone) using the volume buttons, then your phone is using call volume.
• Media volume is designed for media scenarios such as music playback. AEC does not work when media volume is used, and Bluetooth earphones cannot be used for audio capturing. However, media volume delivers better music listening experience. If you are able to mute your phone using the volume buttons, then your phone is using media volume. The SDK offers three system volume control modes: auto, call volume, and media volume.

Enumerator
TRTCSystemVolumeTypeAuto	Auto: In the auto mode, call volume is used for anchors, and media volume for audience. This mode is suitable for live streaming scenarios. If the scenario you select during enterRoom is TRTCAppSceneLIVE or TRTCAppSceneVoiceChatRoom, the SDK will automatically use this mode.
TRTCSystemVolumeTypeMedia	Media volume: In this mode, media volume is used in all scenarios. It is rarely used, mainly suitable for music scenarios with demanding requirements on audio quality. Use this mode if most of your users use peripheral devices such as audio cards. Otherwise, it is not recommended.
TRTCSystemVolumeTypeVOIP	Call volume: In this mode, the audio module does not change its work mode when users switch between anchors and audience, enabling seamless mic on/off. This mode is suitable for scenarios where users need to switch frequently between anchors and audience. If the scenario you select during enterRoom is TRTCAppSceneVideoCall or TRTCAppSceneAudioCall, the SDK will automatically use this mode.

TRTCTranscodingConfigMode

enum TRTCTranscodingConfigMode : NSInteger

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 : NSInteger

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_PixelBuffer	Use memory buffer to transfer video data. iOS: PixelBuffer; Android: Direct Buffer for JNI layer; Windows: memory data block.
TRTCVideoBufferType_NSData	Use memory buffer to transfer video data. iOS: more compact memory block in NSData type after additional processing; Android: byte[] for Java layer. This transfer method has a lower efficiency than other methods.
TRTCVideoBufferType_Texture	Use texture to transfer video data.

TRTCVideoFillMode

enum TRTCVideoFillMode : NSInteger

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 : NSUInteger

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

TRTCVideoPixelFormat

enum TRTCVideoPixelFormat : NSInteger

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_32BGRA	BGRA32 format.
TRTCVideoPixelFormat_NV12	YUV420SP (NV12) format.

TRTCVideoQosPreference

enum TRTCVideoQosPreference : NSInteger

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 : NSInteger

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 : NSInteger

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 : NSInteger

Video image rotation direction

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

Enumerator
TRTCVideoRotation_0	No rotation.
TRTCVideoRotation_90	Clockwise rotation by 90 degrees.
TRTCVideoRotation_180	Clockwise rotation by 180 degrees.
TRTCVideoRotation_270	Clockwise rotation by 270 degrees.

TRTCVideoStreamType

enum TRTCVideoStreamType : NSInteger

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.

TRTCVoiceChangerType

enum TRTCVoiceChangerType : NSInteger

Voice changing type

This enumerated value is used to set the voice changing mode in the live streaming scenario and is often used in show live streaming.

Enumerator
TRTCVoiceChangerType_0	Disable voice changing.
TRTCVoiceChangerType_1	Child.
TRTCVoiceChangerType_2	Girl.
TRTCVoiceChangerType_3	Middle-Aged man.
TRTCVoiceChangerType_4	Heavy metal.
TRTCVoiceChangerType_5	Nasal.
TRTCVoiceChangerType_6	Punk.
TRTCVoiceChangerType_7	Trapped beast.
TRTCVoiceChangerType_8	Otaku.
TRTCVoiceChangerType_9	Electronic.
TRTCVoiceChangerType_10	Robot.
TRTCVoiceChangerType_11	Ethereal.