> ## 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.

# RNN noise suppression plugin

> Enable AI noise suppression on a microphone track with @seastart/srtc-plugin-rnnoise: install, host the three static assets in one directory, attach with setProcessor and basePath, options, and trade-offs vs. the browser's native noiseSuppression.

### Overview

`@seastart/srtc-plugin-rnnoise` is an AI noise suppression plugin based on [RNNoise](https://github.com/xiph/rnnoise) (AudioWorklet + WebAssembly) that implements the SDK's [`TrackProcessor`](/en/rtc/web/advanced/audio-processor) interface. Compared with the browser's native `noiseSuppression`, RNNoise suppresses both stationary and non-stationary noise (keyboard, fans, ambient noise, etc.) more strongly.

***

### Installation

```bash theme={null}
npm i @seastart/srtc-web-sdk @seastart/srtc-plugin-rnnoise
```

***

### Host the static assets

The plugin depends on three static assets (in the package's `assets/` directory):

* `rnnoise-runtime.js`
* `rnnoise-processor.js`
* `rnnoise-processor.wasm`

<Warning>
  `rnnoise-runtime.js` locates the worklet and wasm in the same directory via `document.currentScript.src`, so these three files **must be placed in the same directory**, and `basePath` must point to it.
</Warning>

Copy the assets to your static directory, for example `public/rnnoise/`:

```bash theme={null}
cp node_modules/@seastart/srtc-plugin-rnnoise/assets/* public/rnnoise/
```

> In Vite projects you can also copy them automatically at build time with `vite-plugin-static-copy`.

***

### Usage

```typescript theme={null}
import { SRTC } from "@seastart/srtc-web-sdk";
import { RnnoiseProcessor } from "@seastart/srtc-plugin-rnnoise";

const srtc = new SRTC();

// 1. Create the microphone track and start capture
const mic = srtc.createLocalMicTrack();
await mic.startCapture({ deviceId });

// 2. Enable noise suppression (basePath points to the directory hosting the assets from the previous step)
await mic.setProcessor(new RnnoiseProcessor({ basePath: "/rnnoise/" }));

// 3. Publish
await srtc.publishLocalTrack(mic);
```

To turn off noise suppression and restore the original microphone track:

```typescript theme={null}
await mic.removeProcessor();
```

***

### Options

| Option | Type | Default | Description |
| - | - | - | - |
| `basePath` | `string` | `"/"` | Directory containing the static assets; must end with `/` |

***

### Trade-offs vs. native `noiseSuppression`

| | RNNoise plugin | Native `noiseSuppression` |
| - | - | - |
| Noise suppression | Strong, suits complex noise | Moderate |
| Extra cost | Uses CPU, loads \~125 KB of wasm | Almost none |
| Deployment | Requires hosting static assets | None |

> For low-end devices or scenarios with little noise, keep using the native `noiseSuppression` of `createLocalMicTrack`; for strong noise and high call-quality requirements, use this plugin.

***

### Notes

* Supports audio tracks only.
* Must run over HTTPS (or localhost); `AudioContext` may need a user gesture before it can `resume`.
* The sample rate is fixed at 48 kHz.
* After switching the microphone device (`changeDeviceId`) or calling `startCapture` again, noise suppression resumes automatically; you don't need to call `setProcessor` again.
* Can be chained with other processors; see [Audio/video processors](/en/rtc/web/advanced/audio-processor#chaining-multiple-processors).