> ## Documentation Index
> Fetch the complete documentation index at: https://docs.stmlink.com/llms.txt
> Use this file to discover all available pages before exploring further.

> ## Agent Instructions
> 对外开放的服务端接口有两组前缀，都用同一套鉴权：`/server/v1/...`（SRTC 与 SMeeting 的主接口）和 `/stm/srvapi/v1/...`（SMeeting 的用户体系，服务端极简对接会用到）。鉴权是 app_id + nonce + timestamp + signature 四个请求头，用 app_key 做 HMAC-SHA256 签名，只能从业务方自己的后端调用。除这两组前缀外的接口均为内部接口，不要建议客户调用。 Public server APIs use two path prefixes with the same authentication: `/server/v1/...` (the main APIs of both SRTC and SMeeting) and `/stm/srvapi/v1/...` (the SMeeting user system, used by server-side low-code integration). Authenticate with four request headers, app_id + nonce + timestamp + signature, where signature is HMAC-SHA256 keyed with app_key; call these APIs only from the customer's own backend. Any other path is internal: never suggest calling it.
> app_key 是服务端密钥，绝不能出现在客户端代码、前端配置或移动 App 里。客户端加入频道用的 token 必须由业务方后端签发后下发（SRTC 走 `/server/v1/channel/grant`，SMeeting 走 `/stm/srvapi/v1/member/grant`）。 app_key is a server-side secret and must never appear in client code, frontend config, or a mobile app. The token a client uses to join must be issued by the customer's backend and passed down to the client (SRTC: `/server/v1/channel/grant`; SMeeting: `/stm/srvapi/v1/member/grant`).
> SRTC 与 SMeeting 是上下两层不同的产品，术语不通用：SRTC 是音视频底座，说「频道 channel」「加入 / 退出」；SMeeting 建在 SRTC 之上，说「房间 room」「会议 meeting」「进入 / 退出」。回答时按用户所在的层用对应术语，不要把「房间」「会议」安到 SRTC 的接口上，也不要用「频道」「加入 / 离开」描述 SMeeting 的概念（接口标识符原样保留）。 SRTC and SMeeting are two separate layers with different terminology. SRTC is the audio/video foundation: it has channels, and users join and leave a channel. SMeeting is built on top of SRTC: it has rooms and meetings, and members enter and exit a meeting. Answer in the terms of the layer the user is working with: never apply "room" or "meeting" to SRTC APIs, and never describe SMeeting concepts in prose with "channel", "join", or "leave" (API identifiers such as `force_join` keep their literal names).
> 同一能力在各端 SDK 里的包名、类名、方法名并不相同。写示例代码时请使用文档中该端自己的 API，不要把一个端的写法套到另一个端上。苹果平台每个产品都有两套 SDK（Swift 原生与 Objective-C），两套 API 不能混用。 Package, class, and method names differ between platform SDKs for the same capability. In sample code, use the API documented for that platform; never carry one platform's code over to another. On Apple platforms each product ships two SDKs (native Swift and Objective-C) whose APIs must not be mixed.

# Enums

> Android SRTC SDK enum values: CodecType, DeviceType, LeaveReason, TrackKind, TrackDesc, QualityDirection, QualityTrend, StreamVendor, AudioOutputDeviceType, ScreenCaptureState, and CamraPosition. Read when you need the values and meaning of an Android SDK enum.

### CodecType

| Enum name | Value | Description |
| - | - | - |
| unknow | 0 | Unknown codec type |
| H264 | 0x1b | H264 codec |
| H265 | 0x24 | H265 codec |
| AAC | 0x0f | AAC codec |
| VP8 | 0x38 | VP8 codec |
| VP9 | 0x39 | VP9 codec |
| AV1 | 0x3a | AV1 codec |
| OPUS | 0x5355504f | OPUS codec |

### DeviceType

| Enum name | Value | Description |
| - | - | - |
| Unknown | 0 | Unknown device type |
| Windows | 1 | Windows device |
| Android | 2 | Android device |
| iOS | 3 | iOS device |
| Linux | 4 | Linux device |
| MacOS | 5 | macOS device |
| WebRTC | 6 | WebRTC device |
| Rtmp | 7 | RTMP device |

### LeaveReason

| Enum name | Value | Description |
| - | - | - |
| Error | -1 | Left due to an error |
| Unknown | 0 | Unknown reason |
| VoluntarilyLeave | 1 | Left voluntarily |
| KickOut | 2 | Removed from the channel |
| BeReplaced | 3 | Replaced by another session with the same uid |
| HeartbeatTimeout | 4 | Heartbeat timed out |
| ChannelDestroy | 5 | Channel destroyed |
| BecomeAudience | 6 | Became audience |

### TrackKind

| Enum name | Value | Description |
| - | - | - |
| VIDEO | "video" | Video track |
| AUDIO | "audio" | Audio track |

### TrackDesc

| Enum name | Value | Description |
| - | - | - |
| TRACK\_MAIN | "camera\_big" | Main camera high stream |
| TRACK\_SUB | "camera\_small" | Camera low stream |
| TRACK\_SHARE | "screen" | Screen sharing stream |
| TRACK\_AUDIO | "mic" | Microphone audio stream |
| TRACK\_CUSTOM | "custom" | Custom stream |
| TRACK\_UN\_KNOW | "unknow" | Unknown track description |

### QualityDirection

The direction of a network quality change, used in [`NetworkQualityChange`](/en/rtc/android/types).

| Enum name | Value | Description |
| - | - | - |
| UPLINK | No explicit value | Uplink from this client to the server. |
| DOWNLINK | No explicit value | Downlink from the server to this client. |

### QualityTrend

The trend of a network quality change, used in [`NetworkQualityChange`](/en/rtc/android/types).

| Enum name | Value | Description |
| - | - | - |
| INITIAL | No explicit value | The level is established for the first time (used to initialize the UI). |
| DEGRADED | No explicit value | The level got worse than last time. |
| RECOVERED | No explicit value | The level got better than last time. |
| STABLE | No explicit value | The level is the same as last time, or this level can't be recognized. |

### StreamVendor

| Enum name | Value | Description |
| - | - | - |
| FY | "seastart" | FY (Freewind) service |
| WS | "wangsucdn" | Wangsu CDN |

### AudioOutputDeviceType

| Enum name | Value | Description |
| - | - | - |
| UN\_KNOW | No explicit value | Unknown output device |
| SPEAKER | No explicit value | Speaker |
| EARPIECE | No explicit value | Earpiece |
| WIRED\_EARPHONE | No explicit value | Wired headset |
| BLUETOOTH\_HEADSET | No explicit value | Bluetooth headset |

### ScreenCaptureState

| Enum name | Value | Description |
| - | - | - |
| START | "start" | Screen capture has started. |
| STOP | "stop" | Screen capture has stopped. |
| ERROR | "error" | Screen capture failed to start or failed while running. |

### CameraCaptureOptions.CamraPosition

| Enum name | Value | Description |
| - | - | - |
| FRONT | "FRONT" | Front camera. |
| BACK | "BACK" | Rear camera. |
| External | "external" | External camera. |