rtc 命名空间参考

类
struct	ExternalAudioFrame
struct	ExternalVideoFrame
class	IMicManager
class	IPlayoutManager
class	IRTCAudioFrameObserver
class	IRTCChannel
class	IRTCChannelEventHandler
class	IRTCEngine
class	IRTCEngineEventHandler
class	IRTCVideoFrameObserver
class	IVideoDeviceManager
struct	LocalAudioStats
struct	LocalVideoStats
struct	RemoteAudioStats
struct	RemoteVideoStats
struct	RTC_RESULT
struct	RtcRect
struct	RTCStats
	统计信息 更多...
struct	Stru_RtmpLiveTranscoding
struct	Stru_RtmpRtcImage
struct	Stru_RtmpTranscodingUser
struct	Stru_RtmpUplinesAddr
struct	Stru_RtmpUplinesInfo
struct	VideoProfile
struct	YuvI420VideoFrame

类型定义
typedef void(*	SDK_CB) (RTC_RET_CODE)
	标准的sdk回调函数； 由于系统中有不少是异步调用方式，需要在后续返回结果 更多...
typedef void(*	log_func) (const char *msg)
	log function 更多...
typedef unsigned int	UserId
typedef HWND	WindowIdType
typedef struct rtc::RtcRect	Rect

枚举
enum	RTCEngineType { RTCEngine_Agora = 1 , RTCEngine_XES = 3 , RTCEngine_XES_ALI = 5 , RTCEngine_TTT = 7 }
enum	RTCLOGTYPE { RTCLOGTYPE_DEBUG = 0 , RTCLOGTYPE_INFO , RTCLOGTYPE_WARN , RTCLOGTYPE_ERROR , RTCLOGTYPE_FATAL }
enum	RTC_RET_CODE {   RTC_SUCCESS = 0 , RTC_FAILED = 1 , RTC_WARN = 2 , RTC_ERR_INVALID_ARGUMENT = 3 , RTC_ERR_NOT_READY = 4 , RTC_ERR_NOT_SUPPORTED = 5 , RTC_ERR_REFUSED = 6 , RTC_ERR_INVALID_TOKEN = 7 , RTC_ERR_TOKEN_EXPIRED = 8 , RTC_ERR_NOT_INITIALIZED = 9 ,   RTC_ERR_NO_PERMISSION = 10 , RTC_ERR_ACCOUNT_DISABLED = 11 , RTC_ERR_SERVER_ERROR = 12 , RTC_ERR_CONNECTION_INTERRUPTED = 13 , RTC_ERR_INVALID_APP_ID = 14 , RTC_ERR_REPEAT = 15 , RTC_ERR_SEND_SEI_ERROR = 16 , RTC_ERR_VIDEO_DEVICE = 23 , RTC_ERR_AUDIO_DEVICE = 24 , RTC_ERR_NO_RECORDING_DEVICE = 26 ,   RTC_ERR_NO_PLAYOUT_DEVICE = 27 , RTC_ERR_START_CAMERA = 50 , RTC_ERR_START_VIDEO_RENDER = 51 , RTC_ERR_ROOM_TIMEOUT = 1050 , RTC_ERR_RESOURCE_UNAVAILABLE = 1051 , RTC_ERR_ROOM_KICKED = 1060 , RTC_ERR_PUBLISH_FAILED = 1080 , RTC_ERR_REPUBLISH_FAILED = 1081 , RTC_ERR_SUBSCRIBE_FAILED = 1082 , RTC_ERR_RESUBSCRIBE_FAILED = 1083 ,   RTC_ERR_ENABLE_VIDEO = 1100 , RTC_ERR_ENABLE_AUDIO = 1120 }
	sdk event handler onError return-code 更多...
enum	RTC_MEDIA_DEVICE_STATE_TYPE { RTC_MEDIA_DEVICE_STATE_ACTIVE = 1 , RTC_MEDIA_DEVICE_STATE_DISABLED = 2 , RTC_MEDIA_DEVICE_STATE_NOT_PRESENT = 4 , RTC_MEDIA_DEVICE_STATE_UNPLUGGED = 8 }
enum	RTC_MEDIA_DEVICE_TYPE { RTC_UNKNOWN_AUDIO_DEVICE = -1 , RTC_AUDIO_PLAYOUT_DEVICE = 0 , RTC_AUDIO_RECORDING_DEVICE = 1 , RTC_VIDEO_RENDER_DEVICE = 2 , RTC_VIDEO_CAPTURE_DEVICE = 3 }
enum	RTC_VIDEO_RENDER_MODE { RTC_VIDEO_RENDER_MODE_HIDDEN = 1 , RTC_VIDEO_RENDER_MODE_FIT = 2 }
enum	CLIENT_ROLE_TYPE { CLIENT_ROLE_BROADCASTER = 1 , CLIENT_ROLE_AUDIENCE = 2 }
enum	STREAM_FALLBACK_OPTIONS { STREAM_FALLBACK_OPTION_DISABLED = 0 , STREAM_FALLBACK_OPTION_VIDEO_STREAM_LOW = 1 , STREAM_FALLBACK_OPTION_AUDIO_ONLY = 2 }
enum	RTC_AUDIO_CAPTURE_SRC_TYPE { AUDIO_SRC_SDK_DATA = 1 , AUDIO_SRC_USER_PCM = 1001 , AUDIO_SRC_USER_AAC = 1002 }
	：音频数据源类型 更多...
enum	RTC_VIDEO_RATIO { RTC_VIDEO_RATIO_4_3 = 0 , RTC_VIDEO_RATIO_16_9 = 1 , RTC_VIDEO_RATIO_9_16 = 2 }
	推流视频比例 更多...
enum	RTC_VIDEO_BITRATE { RTC_VIDEO_BITRATE_200 = 200 , RTC_VIDEO_BITRATE_350 = 350 , RTC_VIDEO_BITRATE_400 = 400 , RTC_VIDEO_BITRATE_600 = 600 , RTC_VIDEO_BITRATE_1000 = 1000 }
enum	RTC_VIDEO_RESOLUTION {   RTC_VIDEO_RESOLUTION_320x240 = 1 , RTC_VIDEO_RESOLUTION_480x360 = 2 , RTC_VIDEO_RESOLUTION_640x480 = 3 , RTC_VIDEO_RESOLUTION_800x600 = 4 , RTC_VIDEO_RESOLUTION_960x720 = 5 , RTC_VIDEO_RESOLUTION_240x320 = 50 , RTC_VIDEO_RESOLUTION_360x480 = 51 , RTC_VIDEO_RESOLUTION_480x640 = 52 , RTC_VIDEO_RESOLUTION_720x960 = 53 , RTC_VIDEO_RESOLUTION_320x180 = 101 ,   RTC_VIDEO_RESOLUTION_480x272 = 102 , RTC_VIDEO_RESOLUTION_640x360 = 103 , RTC_VIDEO_RESOLUTION_960x540 = 104 , RTC_VIDEO_RESOLUTION_1280x720 = 105 , RTC_VIDEO_RESOLUTION_180x320 = 201 , RTC_VIDEO_RESOLUTION_272x480 = 202 , RTC_VIDEO_RESOLUTION_360x640 = 203 , RTC_VIDEO_RESOLUTION_540x960 = 204 , RTC_VIDEO_RESOLUTION_DEFAULT = 1000 }
enum	RTC_VIDEO_ROTATION { RTC_VIDEO_ROTATION_NONE = 1 , RTC_VIDEO_ROTATION_90 = 2 , RTC_VIDEO_ROTATION_180 = 3 , RTC_VIDEO_ROTATION_270 = 4 }
enum	RTC_VIDEO_QUALITY_PARAM_PRESET { RTC_VIDEO_QUALITY_STANDARD_DEFINITION = 1 , RTC_VIDEO_QUALITY_HIGH_DEFINITION = 2 , RTC_VIDEO_QUALITY_SUPER_DEFINITION = 3 , RTC_VIDEO_QUALITY_LINKMIC_MAIN_PUBLISHER = 4 , RTC_VIDEO_QUALITY_LINKMIC_SUB_PUBLISHER = 5 , RTC_VIDEO_QUALITY_REALTIME_VIDEOCHAT = 6 , RTC_VIDEO_QUALITY_STILLIMAGE_DEFINITION = 7 }
	：SDK推流画质预设选项 更多...
enum	RTC_OUTPUT_VIDEO_FORMAT { RTC_OUTPUT_VIDEO_WITHOUT_OUTPUT = 1 , RTC_OUTPUT_VIDEO_FORMAT_YUV420 = 2 , RTC_OUTPUT_VIDEO_FORMAT_RGBA = 3 }
	：设置输出的视频格式 更多...
enum	RTC_VIDEO_BUFFER_TYPE { RTC_VIDEO_BUFFER_TYPE_UNKNOWN = 0 , RTC_VIDEO_BUFFER_TYPE_BUFFER = 1 , RTC_VIDEO_BUFFER_TYPE_TEXTURE = 3 }
enum	RTC_VIDEO_PIXEL_FORMAT { RTC_VIDEO_PIXEL_FORMAT_UNKNOWN = 0 , RTC_VIDEO_PIXEL_FORMAT_I420 = 1 , RTC_VIDEO_PIXEL_FORMAT_TEXTURE_2D = 2 , RTC_VIDEO_PIXEL_FORMAT_BGRA32 = 3 }
enum	RTC_BEAUTY_STYLE { RTC_BEAUTY_STYLE_SMOOTH = 0 , RTC_BEAUTY_STYLE_NATURE = 1 , RTC_BEAUTY_STYLE_BLUR = 2 }
enum	RTC_AUDIO_FRAME_FORMAT { RTC_AUDIO_FRAME_FORMAT_NONE = 0 , RTC_AUDIO_FRAME_FORMAT_PCM }
enum	NETWORK_QUALITY_TYPE { NETWORK_QUALITY_UNKNOWN = 0 , NETWORK_QUALITY_EXCELLENT = 1 , NETWORK_QUALITY_GOOD = 2 , NETWORK_QUALITY_POOR = 3 , NETWORK_QUALITY_BAD = 4 , NETWORK_QUALITY_VBAD = 5 , NETWORK_QUALITY_DOWN = 6 , NETWORK_QUALITY_UNSUPPORTED = 7 , NETWORK_QUALITY_DETECTING = 8 }
enum	RTC_REMOTE_VIDEO_STATE { REMOTE_VIDEO_STATE_STOPPED = 0 , REMOTE_VIDEO_STATE_STARTING = 1 , REMOTE_VIDEO_STATE_DECODING = 2 , REMOTE_VIDEO_STATE_FROZEN = 3 , REMOTE_VIDEO_STATE_FAILED = 4 }
enum	RTC_REMOTE_VIDEO_STATE_REASON { REMOTE_VIDEO_STATE_REASON_INTERNAL = 0 , REMOTE_VIDEO_STATE_REASON_NETWORK_CONGESTION = 1 , REMOTE_VIDEO_STATE_REASON_NETWORK_RECOVERY = 2 , REMOTE_VIDEO_STATE_REASON_LOCAL_MUTED = 3 , REMOTE_VIDEO_STATE_REASON_LOCAL_UNMUTED = 4 , REMOTE_VIDEO_STATE_REASON_REMOTE_MUTED = 5 , REMOTE_VIDEO_STATE_REASON_REMOTE_UNMUTED = 6 , REMOTE_VIDEO_STATE_REASON_REMOTE_OFFLINE = 7 , REMOTE_VIDEO_STATE_REASON_AUDIO_FALLBACK = 8 , REMOTE_VIDEO_STATE_REASON_AUDIO_FALLBACK_RECOVERY = 9 }
enum	RTC_LOCAL_VIDEO_STREAM_STATE { LOCAL_VIDEO_STREAM_STATE_STOPPED = 0 , LOCAL_VIDEO_STREAM_STATE_CAPTURING = 1 , LOCAL_VIDEO_STREAM_STATE_ENCODING = 2 , LOCAL_VIDEO_STREAM_STATE_FAILED = 3 }
enum	RTC_LOCAL_VIDEO_STREAM_ERROR { LOCAL_VIDEO_STREAM_ERROR_OK = 0 , LOCAL_VIDEO_STREAM_ERROR_FAILURE = 1 , LOCAL_VIDEO_STREAM_ERROR_DEVICE_NO_PERMISSION = 2 , LOCAL_VIDEO_STREAM_ERROR_DEVICE_BUSY = 3 , LOCAL_VIDEO_STREAM_ERROR_CAPTURE_FAILURE = 4 , LOCAL_VIDEO_STREAM_ERROR_ENCODE_FAILURE = 5 , LOCAL_VIDEO_STREAM_ERROR_DEVICE_NOT_FOUND = 6 }
enum	RTMP_STREAM_PUBLISH_STATE { RTMP_STREAM_PUBLISH_STATE_IDLE = 0 , RTMP_STREAM_PUBLISH_STATE_CONNECTING = 1 , RTMP_STREAM_PUBLISH_STATE_RUNNING = 2 , RTMP_STREAM_PUBLISH_STATE_RECOVERING = 3 , RTMP_STREAM_PUBLISH_STATE_FAILURE = 4 }
enum	RTMP_STREAM_PUBLISH_ERROR {   RTMP_STREAM_PUBLISH_ERROR_OK = 0 , RTMP_STREAM_PUBLISH_ERROR_INVALID_ARGUMENT = 1 , RTMP_STREAM_PUBLISH_ERROR_ENCRYPTED_STREAM_NOT_ALLOWED = 2 , RTMP_STREAM_PUBLISH_ERROR_CONNECTION_TIMEOUT = 3 , RTMP_STREAM_PUBLISH_ERROR_INTERNAL_SERVER_ERROR = 4 , RTMP_STREAM_PUBLISH_ERROR_RTMP_SERVER_ERROR = 5 , RTMP_STREAM_PUBLISH_ERROR_TOO_OFTEN = 6 , RTMP_STREAM_PUBLISH_ERROR_REACH_LIMIT = 7 , RTMP_STREAM_PUBLISH_ERROR_NOT_AUTHORIZED = 8 , RTMP_STREAM_PUBLISH_ERROR_STREAM_NOT_FOUND = 9 ,   RTMP_STREAM_PUBLISH_ERROR_FORMAT_NOT_SUPPORTED = 10 }
enum	REMOTE_AUDIO_STATE { REMOTE_AUDIO_STATE_STOPPED = 0 , REMOTE_AUDIO_STATE_STARTING = 1 , REMOTE_AUDIO_STATE_DECODING = 2 , REMOTE_AUDIO_STATE_FROZEN = 3 , REMOTE_AUDIO_STATE_FAILED = 4 }
enum	REMOTE_AUDIO_STATE_REASON { REMOTE_AUDIO_REASON_INTERNAL = 0 , REMOTE_AUDIO_REASON_NETWORK_CONGESTION = 1 , REMOTE_AUDIO_REASON_NETWORK_RECOVERY = 2 , REMOTE_AUDIO_REASON_LOCAL_MUTED = 3 , REMOTE_AUDIO_REASON_LOCAL_UNMUTED = 4 , REMOTE_AUDIO_REASON_REMOTE_MUTED = 5 , REMOTE_AUDIO_REASON_REMOTE_UNMUTED = 6 , REMOTE_AUDIO_REASON_REMOTE_OFFLINE = 7 }
enum	CONNECTION_CHANGED_REASON_TYPE {   CONNECTION_CHANGED_CONNECTING = 0 , CONNECTION_CHANGED_JOIN_SUCCESS = 1 , CONNECTION_CHANGED_INTERRUPTED = 2 , CONNECTION_CHANGED_BANNED_BY_SERVER = 3 , CONNECTION_CHANGED_JOIN_FAILED = 4 , CONNECTION_CHANGED_LEAVE_CHANNEL = 5 , CONNECTION_CHANGED_INVALID_APP_ID = 6 , CONNECTION_CHANGED_INVALID_CHANNEL_NAME = 7 , CONNECTION_CHANGED_INVALID_TOKEN = 8 , CONNECTION_CHANGED_TOKEN_EXPIRED = 9 ,   CONNECTION_CHANGED_REJECTED_BY_SERVER = 10 , CONNECTION_CHANGED_SETTING_PROXY_SERVER = 11 , CONNECTION_CHANGED_RENEW_TOKEN = 12 , CONNECTION_CHANGED_CLIENT_IP_ADDRESS_CHANGED = 13 , CONNECTION_CHANGED_KEEP_ALIVE_TIMEOUT = 14 , CONNECTION_CHANGED_VIDEO_SWITCH_SERVER = 15 , CONNECTION_CHANGED_AUDIO_SWITCH_SERVER = 16 }
enum	RTC_CONNECTION_STATE_TYPE { CONNECTION_STATE_DISCONNECTED = 1 , CONNECTION_STATE_CONNECTING = 2 , CONNECTION_STATE_CONNECTED = 3 , CONNECTION_STATE_RECONNECTING = 4 , CONNECTION_STATE_FAILED = 5 }
enum	VIDEO_BUFFERING_STATE { VIDEO_BUFFERING_STATE_BUFFERING_START = 0 , VIDEO_BUFFERING_STATE_BUFFERING_END = 1 }
enum	REMOTE_VIDEO_STREAM_TYPE { REMOTE_VIDEO_STREAM_HIGH = 0 , REMOTE_VIDEO_STREAM_LOW = 1 , REMOTE_VIDO_STREAM_AUDIO = 2 }
enum	BUSI_USER_ROLE { BUSI_USER_ROLE_STUDENT = 0 , BUSI_USER_ROLE_TEACHER = 1 , BUSI_USER_ROLE_TUTOR = 2 }
enum	RAW_AUDIO_FRAME_OP_MODE_TYPE { RAW_AUDIO_FRAME_OP_MODE_READ_ONLY = 0 , RAW_AUDIO_FRAME_OP_MODE_WRITE_ONLY = 1 , RAW_AUDIO_FRAME_OP_MODE_READ_WRITE = 2 }
enum	SUBSCRIPTION_STREAM_TYPE { SUBSCRIBE_VIDEO_HIGH , SUBSCRIBE_VIDEO_LOW , SUBSCRIBE_AUDIO_ONLY }
enum	PRIORITY_TYPE { PRIORITY_HIGH = 50 , PRIORITY_NORMAL = 100 }
enum	RTCFeature { FeatureAudioBeauty }
enum	CHANNEL_MEDIA_RELAY_STATE { RELAY_STATE_IDLE = 0 , RELAY_STATE_CONNECTING = 1 , RELAY_STATE_RUNNING = 2 , RELAY_STATE_FAILURE = 3 }
enum	CHANNEL_MEDIA_RELAY_ERROR {   RELAY_OK = 0 , RELAY_ERROR_SERVER_ERROR_RESPONSE = 1 , RELAY_ERROR_SERVER_NO_RESPONSE = 2 , RELAY_ERROR_NO_RESOURCE_AVAILABLE = 3 , RELAY_ERROR_FAILED_JOIN_SRC = 4 , RELAY_ERROR_FAILED_JOIN_DEST = 5 , RELAY_ERROR_FAILED_PACKET_RECEIVED_FROM_SRC = 6 , RELAY_ERROR_FAILED_PACKET_SENT_TO_DEST = 7 , RELAY_ERROR_SERVER_CONNECTION_LOST = 8 , RELAY_ERROR_INTERNAL_ERROR = 9 ,   RELAY_ERROR_SRC_TOKEN_EXPIRED = 10 , RELAY_ERROR_DEST_TOKEN_EXPIRED = 11 }
enum	CHANNEL_MEDIA_RELAY_EVENT {   RELAY_EVENT_NETWORK_DISCONNECTED = 0 , RELAY_EVENT_NETWORK_CONNECTED = 1 , RELAY_EVENT_PACKET_JOINED_SRC_CHANNEL = 2 , RELAY_EVENT_PACKET_JOINED_DEST_CHANNEL = 3 , RELAY_EVENT_PACKET_SENT_TO_DEST_CHANNEL = 4 , RELAY_EVENT_PACKET_RECEIVED_VIDEO_FROM_SRC = 5 , RELAY_EVENT_PACKET_RECEIVED_AUDIO_FROM_SRC = 6 , RELAY_EVENT_PACKET_UPDATE_DEST_CHANNEL = 7 , RELAY_EVENT_PACKET_UPDATE_DEST_CHANNEL_REFUSED = 8 , RELAY_EVENT_PACKET_UPDATE_DEST_CHANNEL_NOT_CHANGE = 9 ,   RELAY_EVENT_PACKET_UPDATE_DEST_CHANNEL_IS_NULL = 10 , RELAY_EVENT_VIDEO_PROFILE_UPDATE = 11 }
enum	RTC_AUDIO_PROCESS_MODE { RTC_AUDIO_PROCESS_MODE_NORMAL = 0 , RTC_AUDIO_PROCESS_MODE_FASTPROCESSING = 1 , RTC_AUDIO_PROCESS_MODE_FASTRECORDING = 2 }
enum	STREAM_SUBSCRIBE_STATE { SUB_STATE_IDLE = 0 , SUB_STATE_NO_SUBSCRIBED = 1 , SUB_STATE_SUBSCRIBING = 2 , SUB_STATE_SUBSCRIBED = 3 }
enum	STREAM_PUBLISH_STATE { PUB_STATE_IDLE = 0 , PUB_STATE_NO_PUBLISHED = 1 , PUB_STATE_PUBLISHING = 2 , PUB_STATE_PUBLISHED = 3 }

函数
RTCSDKDLL_API VideoProfile *	getVideoProfile ()
RTCSDKDLL_API bool	getVideoResolution (RTC_VIDEO_RESOLUTION profile, int &width, int &height)

变量
const unsigned	MAX_DEVICE_ID_LENGTH = 512
const unsigned	MAX_DEVICE_COUNT = 16

类型定义说明

SDK_CB

typedef void(* rtc::SDK_CB) (RTC_RET_CODE)

标准的sdk回调函数； 由于系统中有不少是异步调用方式，需要在后续返回结果

log_func

typedef void(* rtc::log_func) (const char *msg)

log function

UserId

typedef unsigned int rtc::UserId

WindowIdType

typedef HWND rtc::WindowIdType

Rect

typedef struct rtc::RtcRect rtc::Rect

Definition of the rectangular region.

枚举类型说明

RTCEngineType

enum rtc::RTCEngineType

枚举值
RTCEngine_Agora
RTCEngine_XES
RTCEngine_XES_ALI
RTCEngine_TTT

RTCLOGTYPE

enum rtc::RTCLOGTYPE

枚举值
RTCLOGTYPE_DEBUG
RTCLOGTYPE_INFO
RTCLOGTYPE_WARN
RTCLOGTYPE_ERROR
RTCLOGTYPE_FATAL

RTC_RET_CODE

enum rtc::RTC_RET_CODE

sdk event handler onError return-code

枚举值
RTC_SUCCESS
RTC_FAILED	RTC_OK success
RTC_WARN	RTC_ERR_FAILED
RTC_ERR_INVALID_ARGUMENT	基本的错误信息，如果用到，需要说明场景 other warning
RTC_ERR_NOT_READY
RTC_ERR_NOT_SUPPORTED
RTC_ERR_REFUSED
RTC_ERR_INVALID_TOKEN
RTC_ERR_TOKEN_EXPIRED	invalid token
RTC_ERR_NOT_INITIALIZED	token is expired, maybe handle this error event when join room
RTC_ERR_NO_PERMISSION
RTC_ERR_ACCOUNT_DISABLED
RTC_ERR_SERVER_ERROR
RTC_ERR_CONNECTION_INTERRUPTED
RTC_ERR_INVALID_APP_ID	connection is lost
RTC_ERR_REPEAT
RTC_ERR_SEND_SEI_ERROR	case 1. join room repeat;
RTC_ERR_VIDEO_DEVICE
RTC_ERR_AUDIO_DEVICE	video related error
RTC_ERR_NO_RECORDING_DEVICE	audio related error
RTC_ERR_NO_PLAYOUT_DEVICE	no mic
RTC_ERR_START_CAMERA	no speaker
RTC_ERR_START_VIDEO_RENDER	cann't start camera(maybe occupyed)
RTC_ERR_ROOM_TIMEOUT	can not display video
RTC_ERR_RESOURCE_UNAVAILABLE	timeout to join room, maybe user network is poor, cannot connect to room server.
RTC_ERR_ROOM_KICKED	no enough resource for user, for example, no room resource.
RTC_ERR_PUBLISH_FAILED	user is kicked off from this room
RTC_ERR_REPUBLISH_FAILED	fail to publish stream
RTC_ERR_SUBSCRIBE_FAILED
RTC_ERR_RESUBSCRIBE_FAILED
RTC_ERR_ENABLE_VIDEO
RTC_ERR_ENABLE_AUDIO

RTC_MEDIA_DEVICE_STATE_TYPE

enum rtc::RTC_MEDIA_DEVICE_STATE_TYPE

Media device state.

枚举值
RTC_MEDIA_DEVICE_STATE_ACTIVE	1: The device is active. 使用中
RTC_MEDIA_DEVICE_STATE_DISABLED	2: The device is disabled. 被禁用
RTC_MEDIA_DEVICE_STATE_NOT_PRESENT	4: The device is not present. 一开始就没有
RTC_MEDIA_DEVICE_STATE_UNPLUGGED	8: The device is unplugged. 被拔出

RTC_MEDIA_DEVICE_TYPE

enum rtc::RTC_MEDIA_DEVICE_TYPE

Media device type.

枚举值
RTC_UNKNOWN_AUDIO_DEVICE	-1: Unknown audio device type. 未知
RTC_AUDIO_PLAYOUT_DEVICE	0: Audio playback device. 播放器
RTC_AUDIO_RECORDING_DEVICE	1: Audio recording device. 麦克风
RTC_VIDEO_RENDER_DEVICE	2: Video renderer. 视频显示设备
RTC_VIDEO_CAPTURE_DEVICE	3: Video capturer. 视频捕捉设备（摄像头）

RTC_VIDEO_RENDER_MODE

enum rtc::RTC_VIDEO_RENDER_MODE

枚举值
RTC_VIDEO_RENDER_MODE_HIDDEN
RTC_VIDEO_RENDER_MODE_FIT

CLIENT_ROLE_TYPE

enum rtc::CLIENT_ROLE_TYPE

This is an enum class

枚举值
CLIENT_ROLE_BROADCASTER	1: Host
CLIENT_ROLE_AUDIENCE	2: Audience

STREAM_FALLBACK_OPTIONS

enum rtc::STREAM_FALLBACK_OPTIONS

Stream fallback options.

枚举值
STREAM_FALLBACK_OPTION_DISABLED	0: No fallback behavior for the local/remote video stream when the uplink/downlink network conditions are poor.
STREAM_FALLBACK_OPTION_VIDEO_STREAM_LOW	1: Under poor downlink network conditions, the remote video stream, to which you subscribe, falls back to the low-stream (low resolution and low bitrate) video.
STREAM_FALLBACK_OPTION_AUDIO_ONLY	2: Under poor uplink network conditions, the locally published video stream falls back to audio only.

RTC_AUDIO_CAPTURE_SRC_TYPE

enum rtc::RTC_AUDIO_CAPTURE_SRC_TYPE

：音频数据源类型

枚举值
AUDIO_SRC_SDK_DATA
AUDIO_SRC_USER_PCM
AUDIO_SRC_USER_AAC

RTC_VIDEO_RATIO

enum rtc::RTC_VIDEO_RATIO

推流视频比例

枚举值
RTC_VIDEO_RATIO_4_3
RTC_VIDEO_RATIO_16_9
RTC_VIDEO_RATIO_9_16

RTC_VIDEO_BITRATE

enum rtc::RTC_VIDEO_BITRATE

枚举值
RTC_VIDEO_BITRATE_200
RTC_VIDEO_BITRATE_350
RTC_VIDEO_BITRATE_400
RTC_VIDEO_BITRATE_600
RTC_VIDEO_BITRATE_1000

RTC_VIDEO_RESOLUTION

enum rtc::RTC_VIDEO_RESOLUTION

枚举值
RTC_VIDEO_RESOLUTION_320x240
RTC_VIDEO_RESOLUTION_480x360
RTC_VIDEO_RESOLUTION_640x480
RTC_VIDEO_RESOLUTION_800x600
RTC_VIDEO_RESOLUTION_960x720
RTC_VIDEO_RESOLUTION_240x320
RTC_VIDEO_RESOLUTION_360x480
RTC_VIDEO_RESOLUTION_480x640
RTC_VIDEO_RESOLUTION_720x960
RTC_VIDEO_RESOLUTION_320x180
RTC_VIDEO_RESOLUTION_480x272
RTC_VIDEO_RESOLUTION_640x360
RTC_VIDEO_RESOLUTION_960x540
RTC_VIDEO_RESOLUTION_1280x720
RTC_VIDEO_RESOLUTION_180x320
RTC_VIDEO_RESOLUTION_272x480
RTC_VIDEO_RESOLUTION_360x640
RTC_VIDEO_RESOLUTION_540x960
RTC_VIDEO_RESOLUTION_DEFAULT

RTC_VIDEO_ROTATION

enum rtc::RTC_VIDEO_ROTATION

枚举值
RTC_VIDEO_ROTATION_NONE
RTC_VIDEO_ROTATION_90
RTC_VIDEO_ROTATION_180
RTC_VIDEO_ROTATION_270

RTC_VIDEO_QUALITY_PARAM_PRESET

enum rtc::RTC_VIDEO_QUALITY_PARAM_PRESET

：SDK推流画质预设选项

\\\\\\\\\\\\\\\\\ 推流的画面质量预设选项 \\\\\\\\\\\\\\\\\

• 1.2.0 版本开始引入推流画质接口 setVideoQualityParamPreset 用于傻瓜化的选择推流画质效果。
• LivePush::setVideoQualityParamPreset 内部通过预先配置不同的视频参数调整到对应的质量
• 目前支持的几种画质选项如下：

（1）标清 - 采用 640 * 360 级别分辨率，码率会在 400kbps - 800kbps 之间自适应，如果主播的网络条件不理想， 直播的画质会偏模糊，但总体卡顿率不会太高。 Android平台下这一档我们会选择采用软编码，软编码虽然更加耗电，但在运动画面的表现要优于硬编码。

（2）高清 - 采用 960 * 540 级别分辨率，码率会锁定在 1200kbps，如果主播的网络条件不理想，直播画质不会有变化， 但这段时间内会出现频繁的卡顿和跳帧。 两个平台下，这一档我们都会采用硬编码。

（3）超清 - 采用 1280 * 720 级别分辨率，码率会锁定在 1500kbps，对主播的上行带宽要求比较高，适合观看端是大屏的业务场景。

（4）大主播 - 顾名思义，连麦中大主播使用，因为是观众的主画面，追求清晰一些的效果，所以分辨率会优先选择 960 * 540。

（5）小主播 - 顾名思义，连麦中小主播使用，因为是小画面，画面追求流畅，分辨率采用 480 * 320， 码率 350kbps 固定。

（6）实时音视频通话 - 时延低，小画面，画面追求流畅，分辨率采用 480 * 320， 码率 350kbps 固定。

（7）静态画质场景 - 画面变动小，能保证非常低的码率下，视频相对清晰，如截屏正在播放的ppt，1080p，400kb左右码率。 【特别说明】

1. 使用 setVideoQualityParamPreset 之后，依然可以使用setVideoBitrate、setVideoFPS、setAutoAdjustStrategy等设置自定义画质，以最后一次的设置为准。

枚举值
RTC_VIDEO_QUALITY_STANDARD_DEFINITION
RTC_VIDEO_QUALITY_HIGH_DEFINITION
RTC_VIDEO_QUALITY_SUPER_DEFINITION
RTC_VIDEO_QUALITY_LINKMIC_MAIN_PUBLISHER
RTC_VIDEO_QUALITY_LINKMIC_SUB_PUBLISHER
RTC_VIDEO_QUALITY_REALTIME_VIDEOCHAT
RTC_VIDEO_QUALITY_STILLIMAGE_DEFINITION

RTC_OUTPUT_VIDEO_FORMAT

enum rtc::RTC_OUTPUT_VIDEO_FORMAT

：设置输出的视频格式

枚举值
RTC_OUTPUT_VIDEO_WITHOUT_OUTPUT
RTC_OUTPUT_VIDEO_FORMAT_YUV420
RTC_OUTPUT_VIDEO_FORMAT_RGBA

RTC_VIDEO_BUFFER_TYPE

enum rtc::RTC_VIDEO_BUFFER_TYPE

视频数据结构类型

枚举值
RTC_VIDEO_BUFFER_TYPE_UNKNOWN
RTC_VIDEO_BUFFER_TYPE_BUFFER	二进制BUFFER类型
RTC_VIDEO_BUFFER_TYPE_TEXTURE	纹理类型

RTC_VIDEO_PIXEL_FORMAT

enum rtc::RTC_VIDEO_PIXEL_FORMAT

视频帧的格式

枚举值
RTC_VIDEO_PIXEL_FORMAT_UNKNOWN
RTC_VIDEO_PIXEL_FORMAT_I420	I420
RTC_VIDEO_PIXEL_FORMAT_TEXTURE_2D	OPENGL 2D 纹理
RTC_VIDEO_PIXEL_FORMAT_BGRA32	BGRA32

RTC_BEAUTY_STYLE

enum rtc::RTC_BEAUTY_STYLE

枚举值
RTC_BEAUTY_STYLE_SMOOTH
RTC_BEAUTY_STYLE_NATURE
RTC_BEAUTY_STYLE_BLUR

RTC_AUDIO_FRAME_FORMAT

enum rtc::RTC_AUDIO_FRAME_FORMAT

音频帧的格式

枚举值
RTC_AUDIO_FRAME_FORMAT_NONE
RTC_AUDIO_FRAME_FORMAT_PCM	PCM

NETWORK_QUALITY_TYPE

enum rtc::NETWORK_QUALITY_TYPE

枚举值
NETWORK_QUALITY_UNKNOWN
NETWORK_QUALITY_EXCELLENT
NETWORK_QUALITY_GOOD
NETWORK_QUALITY_POOR
NETWORK_QUALITY_BAD
NETWORK_QUALITY_VBAD
NETWORK_QUALITY_DOWN
NETWORK_QUALITY_UNSUPPORTED
NETWORK_QUALITY_DETECTING

RTC_REMOTE_VIDEO_STATE

enum rtc::RTC_REMOTE_VIDEO_STATE

枚举值
REMOTE_VIDEO_STATE_STOPPED	␈␈0: The remote video is in the default state, probably due to REMOTE_VIDEO_STATE_REASON_LOCAL_MUTED (3), REMOTE_VIDEO_STATE_REASON_REMOTE_MUTED (5), or REMOTE_VIDEO_STATE_REASON_REMOTE_OFFLINE (7).
REMOTE_VIDEO_STATE_STARTING	1: The first remote video packet is received.
REMOTE_VIDEO_STATE_DECODING	2: The remote video stream is decoded and plays normally, probably due to REMOTE_VIDEO_STATE_REASON_NETWORK_RECOVERY (2), REMOTE_VIDEO_STATE_REASON_LOCAL_UNMUTED (4), REMOTE_VIDEO_STATE_REASON_REMOTE_UNMUTED (6), or REMOTE_VIDEO_STATE_REASON_AUDIO_FALLBACK_RECOVERY (9).
REMOTE_VIDEO_STATE_FROZEN	3: The remote video is frozen, probably due to REMOTE_VIDEO_STATE_REASON_NETWORK_CONGESTION (1) or REMOTE_VIDEO_STATE_REASON_AUDIO_FALLBACK (8).
REMOTE_VIDEO_STATE_FAILED	4: The remote video fails to start, probably due to REMOTE_VIDEO_STATE_REASON_INTERNAL (0).

RTC_REMOTE_VIDEO_STATE_REASON

enum rtc::RTC_REMOTE_VIDEO_STATE_REASON

枚举值
REMOTE_VIDEO_STATE_REASON_INTERNAL	0: Internal reasons.
REMOTE_VIDEO_STATE_REASON_NETWORK_CONGESTION	1: Network congestion.
REMOTE_VIDEO_STATE_REASON_NETWORK_RECOVERY	2: Network recovery.
REMOTE_VIDEO_STATE_REASON_LOCAL_MUTED	3: The local user stops receiving the remote video stream or disables the video module.
REMOTE_VIDEO_STATE_REASON_LOCAL_UNMUTED	4: The local user resumes receiving the remote video stream or enables the video module.
REMOTE_VIDEO_STATE_REASON_REMOTE_MUTED	5: The remote user stops sending the video stream or disables the video module.
REMOTE_VIDEO_STATE_REASON_REMOTE_UNMUTED	6: The remote user resumes sending the video stream or enables the video module.
REMOTE_VIDEO_STATE_REASON_REMOTE_OFFLINE	7: The remote user leaves the channel.
REMOTE_VIDEO_STATE_REASON_AUDIO_FALLBACK	8: The remote media stream falls back to the audio-only stream due to poor network conditions.
REMOTE_VIDEO_STATE_REASON_AUDIO_FALLBACK_RECOVERY	9: The remote media stream switches back to the video stream after the network conditions improve.

RTC_LOCAL_VIDEO_STREAM_STATE

enum rtc::RTC_LOCAL_VIDEO_STREAM_STATE

Local video state types

枚举值
LOCAL_VIDEO_STREAM_STATE_STOPPED	Initial state
LOCAL_VIDEO_STREAM_STATE_CAPTURING	The capturer starts successfully.
LOCAL_VIDEO_STREAM_STATE_ENCODING	The first video frame is successfully encoded.
LOCAL_VIDEO_STREAM_STATE_FAILED	The local video fails to start.

RTC_LOCAL_VIDEO_STREAM_ERROR

enum rtc::RTC_LOCAL_VIDEO_STREAM_ERROR

Local video state error codes

枚举值
LOCAL_VIDEO_STREAM_ERROR_OK	The local video is normal.
LOCAL_VIDEO_STREAM_ERROR_FAILURE	No specified reason for the local video failure.
LOCAL_VIDEO_STREAM_ERROR_DEVICE_NO_PERMISSION	No permission to use the local video device.
LOCAL_VIDEO_STREAM_ERROR_DEVICE_BUSY	The local video capturer is in use.
LOCAL_VIDEO_STREAM_ERROR_CAPTURE_FAILURE	The local video capture fails. Check whether the capturer is working properly.
LOCAL_VIDEO_STREAM_ERROR_ENCODE_FAILURE	The local video encoding fails.
LOCAL_VIDEO_STREAM_ERROR_DEVICE_NOT_FOUND	No camera device.

RTMP_STREAM_PUBLISH_STATE

enum rtc::RTMP_STREAM_PUBLISH_STATE

枚举值
RTMP_STREAM_PUBLISH_STATE_IDLE	The RTMP streaming has not started or has ended. This state is also triggered after you remove an RTMP address from the CDN by calling removePublishStreamUrl.
RTMP_STREAM_PUBLISH_STATE_CONNECTING	The SDK is connecting to Agora's streaming server and the RTMP server. This state is triggered after you call the addPublishStreamUrl method.
RTMP_STREAM_PUBLISH_STATE_RUNNING	The RTMP streaming publishes. The SDK successfully publishes the RTMP streaming and returns this state.
RTMP_STREAM_PUBLISH_STATE_RECOVERING	The RTMP streaming is recovering. When exceptions occur to the CDN, or the streaming is interrupted, the SDK tries to resume RTMP streaming and returns this state. - If the SDK successfully resumes the streaming, #RTMP_STREAM_PUBLISH_STATE_RUNNING (2) returns. - If the streaming does not resume within 60 seconds or server errors occur, #RTMP_STREAM_PUBLISH_STATE_FAILURE (4) returns. You can also reconnect to the server by calling the \ref IRtcEngine::removePublishStreamUrl "removePublishStreamUrl" and \ref IRtcEngine::addPublishStreamUrl "addPublishStreamUrl" methods.
RTMP_STREAM_PUBLISH_STATE_FAILURE	The RTMP streaming fails. See the errCode parameter for the detailed error information. You can also call the addPublishStreamUrl method to publish the RTMP streaming again.

RTMP_STREAM_PUBLISH_ERROR

enum rtc::RTMP_STREAM_PUBLISH_ERROR

枚举值
RTMP_STREAM_PUBLISH_ERROR_OK	The RTMP streaming publishes successfully.
RTMP_STREAM_PUBLISH_ERROR_INVALID_ARGUMENT	Invalid argument used. If, for example, you do not call the setLiveTranscoding method to configure the LiveTranscoding parameters before calling the addPublishStreamUrl method, the SDK returns this error. Check whether you set the parameters in the setLiveTranscoding method properly.
RTMP_STREAM_PUBLISH_ERROR_ENCRYPTED_STREAM_NOT_ALLOWED	The RTMP streaming is encrypted and cannot be published.
RTMP_STREAM_PUBLISH_ERROR_CONNECTION_TIMEOUT	Timeout for the RTMP streaming. Call the addPublishStreamUrl method to publish the streaming again.
RTMP_STREAM_PUBLISH_ERROR_INTERNAL_SERVER_ERROR	An error occurs in Agora's streaming server. Call the addPublishStreamUrl method to publish the streaming again.
RTMP_STREAM_PUBLISH_ERROR_RTMP_SERVER_ERROR	An error occurs in the RTMP server.
RTMP_STREAM_PUBLISH_ERROR_TOO_OFTEN	The RTMP streaming publishes too frequently.
RTMP_STREAM_PUBLISH_ERROR_REACH_LIMIT	The host publishes more than 10 URLs. Delete the unnecessary URLs before adding new ones.
RTMP_STREAM_PUBLISH_ERROR_NOT_AUTHORIZED	The host manipulates other hosts' URLs. Check your app logic.
RTMP_STREAM_PUBLISH_ERROR_STREAM_NOT_FOUND	Agora's server fails to find the RTMP streaming.
RTMP_STREAM_PUBLISH_ERROR_FORMAT_NOT_SUPPORTED	The format of the RTMP streaming URL is not supported. Check whether the URL format is correct.

REMOTE_AUDIO_STATE

enum rtc::REMOTE_AUDIO_STATE

枚举值
REMOTE_AUDIO_STATE_STOPPED
REMOTE_AUDIO_STATE_STARTING
REMOTE_AUDIO_STATE_DECODING
REMOTE_AUDIO_STATE_FROZEN
REMOTE_AUDIO_STATE_FAILED

REMOTE_AUDIO_STATE_REASON

enum rtc::REMOTE_AUDIO_STATE_REASON

枚举值
REMOTE_AUDIO_REASON_INTERNAL	0: Internal reasons.
REMOTE_AUDIO_REASON_NETWORK_CONGESTION	1: Network congestion.
REMOTE_AUDIO_REASON_NETWORK_RECOVERY	2: Network recovery.
REMOTE_AUDIO_REASON_LOCAL_MUTED	3: The local user stops receiving the remote audio stream or disables the audio module.
REMOTE_AUDIO_REASON_LOCAL_UNMUTED	4: The local user resumes receiving the remote audio stream or enables the audio module.
REMOTE_AUDIO_REASON_REMOTE_MUTED	5: The remote user stops sending the audio stream or disables the audio module.
REMOTE_AUDIO_REASON_REMOTE_UNMUTED	6: The remote user resumes sending the audio stream or enables the audio module.
REMOTE_AUDIO_REASON_REMOTE_OFFLINE	7: The remote user leaves the channel.

CONNECTION_CHANGED_REASON_TYPE

enum rtc::CONNECTION_CHANGED_REASON_TYPE

Reasons for a connection state change.

枚举值
CONNECTION_CHANGED_CONNECTING	0: The SDK is connecting to Agora's edge server.
CONNECTION_CHANGED_JOIN_SUCCESS	1: The SDK has joined the channel successfully.
CONNECTION_CHANGED_INTERRUPTED	2: The connection between the SDK and Agora's edge server is interrupted.
CONNECTION_CHANGED_BANNED_BY_SERVER	3: The connection between the SDK and Agora's edge server is banned by Agora's edge server.
CONNECTION_CHANGED_JOIN_FAILED	4: The SDK fails to join the channel for more than 20 minutes and stops reconnecting to the channel.
CONNECTION_CHANGED_LEAVE_CHANNEL	5: The SDK has left the channel.
CONNECTION_CHANGED_INVALID_APP_ID	6: The connection failed since Appid is not valid.
CONNECTION_CHANGED_INVALID_CHANNEL_NAME	7: The connection failed since channel name is not valid.
CONNECTION_CHANGED_INVALID_TOKEN	8: The connection failed since token is not valid, possibly because: - The App Certificate for the project is enabled in Dashboard, but you do not use Token when joining the channel. If you enable the App Certificate, you must use a token to join the channel. - The uid that you specify in the \ref agora::rtc::IRtcEngine::joinChannel "joinChannel" method is different from the uid that you pass for generating the token.
CONNECTION_CHANGED_TOKEN_EXPIRED	9: The connection failed since token is expired.
CONNECTION_CHANGED_REJECTED_BY_SERVER	10: The connection is rejected by server.
CONNECTION_CHANGED_SETTING_PROXY_SERVER	11: The connection changed to reconnecting since SDK has set a proxy server.
CONNECTION_CHANGED_RENEW_TOKEN	12: When SDK is in connection failed, the renew token operation will make it connecting.
CONNECTION_CHANGED_CLIENT_IP_ADDRESS_CHANGED	13: The IP Address of SDK client has changed. i.e., Network type or IP/Port changed by network operator might change client IP address.
CONNECTION_CHANGED_KEEP_ALIVE_TIMEOUT	14: Timeout for the keep-alive of the connection between the SDK and Agora's edge server. The connection state changes to CONNECTION_STATE_RECONNECTING(4).
CONNECTION_CHANGED_VIDEO_SWITCH_SERVER
CONNECTION_CHANGED_AUDIO_SWITCH_SERVER

RTC_CONNECTION_STATE_TYPE

enum rtc::RTC_CONNECTION_STATE_TYPE

Connection states.

枚举值
CONNECTION_STATE_DISCONNECTED	1: The SDK is disconnected from server.
CONNECTION_STATE_CONNECTING	2: The SDK is connecting to server.
CONNECTION_STATE_CONNECTED	3: The SDK is connected to server and has joined a room. You can now publish or subscribe to a media stream in the room.
CONNECTION_STATE_RECONNECTING	4: The SDK keeps rejoining the channel after being disconnected from a joined room because of network issues.
CONNECTION_STATE_FAILED	5: The SDK fails to connect to server or join the room.

VIDEO_BUFFERING_STATE

enum rtc::VIDEO_BUFFERING_STATE

枚举值
VIDEO_BUFFERING_STATE_BUFFERING_START
VIDEO_BUFFERING_STATE_BUFFERING_END

REMOTE_VIDEO_STREAM_TYPE

enum rtc::REMOTE_VIDEO_STREAM_TYPE

Remote video stream types.

枚举值
REMOTE_VIDEO_STREAM_HIGH	0: High-stream video.
REMOTE_VIDEO_STREAM_LOW	1: Low-stream video.
REMOTE_VIDO_STREAM_AUDIO	2: Auto-stream video.

BUSI_USER_ROLE

enum rtc::BUSI_USER_ROLE

枚举值
BUSI_USER_ROLE_STUDENT
BUSI_USER_ROLE_TEACHER
BUSI_USER_ROLE_TUTOR

RAW_AUDIO_FRAME_OP_MODE_TYPE

enum rtc::RAW_AUDIO_FRAME_OP_MODE_TYPE

Use modes of the onRecordAudioFrame callback.

枚举值
RAW_AUDIO_FRAME_OP_MODE_READ_ONLY	0: Read-only mode: Users only read the AudioFrame data without modifying anything. For example, when users acquire the data with the Agora SDK, then push the RTMP streams.
RAW_AUDIO_FRAME_OP_MODE_WRITE_ONLY	1: Write-only mode: Users replace the AudioFrame data with their own data and pass the data to the SDK for encoding. For example, when users acquire the data.
RAW_AUDIO_FRAME_OP_MODE_READ_WRITE	2: Read and write mode: Users read the data from AudioFrame, modify it, and then play it. For example, when users have their own sound-effect processing module and perform some voice pre-processing, such as a voice change.

SUBSCRIPTION_STREAM_TYPE

enum rtc::SUBSCRIPTION_STREAM_TYPE

Subscribing Remote media stream types

枚举值
SUBSCRIBE_VIDEO_HIGH	0: Subscribing High-stream video.
SUBSCRIBE_VIDEO_LOW	1: Subscribing Low-stream video.
SUBSCRIBE_AUDIO_ONLY	2: Subscribing audio-stream.

PRIORITY_TYPE

enum rtc::PRIORITY_TYPE

The priority of the remote user.

枚举值
PRIORITY_HIGH	50: The user's priority is high.
PRIORITY_NORMAL	100: (Default) The user's priority is normal.

RTCFeature

enum rtc::RTCFeature

枚举值
FeatureAudioBeauty

CHANNEL_MEDIA_RELAY_STATE

enum rtc::CHANNEL_MEDIA_RELAY_STATE

枚举值
RELAY_STATE_IDLE	0: The SDK is initializing.
RELAY_STATE_CONNECTING	1: The SDK tries to relay the media stream to the destination channel.
RELAY_STATE_RUNNING	2: The SDK successfully relays the media stream to the destination channel.
RELAY_STATE_FAILURE	3: A failure occurs. See the details in code.

CHANNEL_MEDIA_RELAY_ERROR

enum rtc::CHANNEL_MEDIA_RELAY_ERROR

枚举值
RELAY_OK	0: The state is normal.
RELAY_ERROR_SERVER_ERROR_RESPONSE	1: An error occurs in the server response.
RELAY_ERROR_SERVER_NO_RESPONSE	2: No server response. You can call the leaveChannel method to leave the channel.
RELAY_ERROR_NO_RESOURCE_AVAILABLE	3: The SDK fails to access the service, probably due to limited resources of the server.
RELAY_ERROR_FAILED_JOIN_SRC	4: The server fails to join the source channel.
RELAY_ERROR_FAILED_JOIN_DEST	5: The server fails to join the destination channel.
RELAY_ERROR_FAILED_PACKET_RECEIVED_FROM_SRC	6: The server fails to receive the data from the source channel.
RELAY_ERROR_FAILED_PACKET_SENT_TO_DEST	7: The source channel fails to transmit data.
RELAY_ERROR_SERVER_CONNECTION_LOST	8: The SDK disconnects from the server due to poor network connections. You can call the leaveChannel method to leave the channel.
RELAY_ERROR_INTERNAL_ERROR	9: An internal error occurs in the server.
RELAY_ERROR_SRC_TOKEN_EXPIRED	10: The token of the source channel has expired.
RELAY_ERROR_DEST_TOKEN_EXPIRED	11: The token of the destination channel has expired.

CHANNEL_MEDIA_RELAY_EVENT

enum rtc::CHANNEL_MEDIA_RELAY_EVENT

枚举值
RELAY_EVENT_NETWORK_DISCONNECTED	0: The user disconnects from the server due to poor network connections.
RELAY_EVENT_NETWORK_CONNECTED	1: The network reconnects.
RELAY_EVENT_PACKET_JOINED_SRC_CHANNEL	2: The user joins the source channel.
RELAY_EVENT_PACKET_JOINED_DEST_CHANNEL	3: The user joins the destination channel.
RELAY_EVENT_PACKET_SENT_TO_DEST_CHANNEL	4: The SDK starts relaying the media stream to the destination channel.
RELAY_EVENT_PACKET_RECEIVED_VIDEO_FROM_SRC	5: The server receives the video stream from the source channel.
RELAY_EVENT_PACKET_RECEIVED_AUDIO_FROM_SRC	6: The server receives the audio stream from the source channel.
RELAY_EVENT_PACKET_UPDATE_DEST_CHANNEL	7: The destination channel is updated.
RELAY_EVENT_PACKET_UPDATE_DEST_CHANNEL_REFUSED	8: The destination channel update fails due to internal reasons.
RELAY_EVENT_PACKET_UPDATE_DEST_CHANNEL_NOT_CHANGE	9: The destination channel does not change, which means that the destination channel fails to be updated.
RELAY_EVENT_PACKET_UPDATE_DEST_CHANNEL_IS_NULL	10: The destination channel name is NULL.
RELAY_EVENT_VIDEO_PROFILE_UPDATE	11: The video profile is sent to the server.

RTC_AUDIO_PROCESS_MODE

enum rtc::RTC_AUDIO_PROCESS_MODE

枚举值
RTC_AUDIO_PROCESS_MODE_NORMAL
RTC_AUDIO_PROCESS_MODE_FASTPROCESSING
RTC_AUDIO_PROCESS_MODE_FASTRECORDING

STREAM_SUBSCRIBE_STATE

enum rtc::STREAM_SUBSCRIBE_STATE

枚举值
SUB_STATE_IDLE
SUB_STATE_NO_SUBSCRIBED
SUB_STATE_SUBSCRIBING
SUB_STATE_SUBSCRIBED

STREAM_PUBLISH_STATE

enum rtc::STREAM_PUBLISH_STATE

枚举值
PUB_STATE_IDLE
PUB_STATE_NO_PUBLISHED
PUB_STATE_PUBLISHING
PUB_STATE_PUBLISHED

函数说明

getVideoProfile()

RTCSDKDLL_API VideoProfile * rtc::getVideoProfile

(

)

获取视频分辨率配置表

返回

返回视频分辨率配置表， 以各项数据为0表示结束，外部不能释放这个配置表

getVideoResolution()

RTCSDKDLL_API bool rtc::getVideoResolution	(	RTC_VIDEO_RESOLUTION	profile,
		int &	width,
		int &	height
	)

获取视频配置数据

参数

profile	配置类型
width	视频宽度
height	视频高度

返回

RTCSDKDLL_API 详见：RTCSDKDLL_API

变量说明

MAX_DEVICE_ID_LENGTH

const unsigned rtc::MAX_DEVICE_ID_LENGTH = 512

MAX_DEVICE_COUNT

const unsigned rtc::MAX_DEVICE_COUNT = 16