livekit-plugin-parrot-stt 0.1.0

package/README.md

@@ -0,0 +1,79 @@
+# livekit-plugin-parrot-stt
+
+[Ringg Parrot STT](https://www.ringg.ai/models/speech-to-text/v1) plugin for
+[LiveKit Agents](https://docs.livekit.io/agents/) (Node.js / TypeScript).
+
+It implements LiveKit's streaming STT interface (`stt.STT` + `stt.SpeechStream`)
+by speaking Ringg's WebSocket protocol directly, so it drops straight into a
+`voice.AgentSession` like any first-party plugin.
+
+> Ringg's official SDK (`ringglabs`) is Python-only. This plugin re-implements the
+> same wire protocol in TypeScript.
+
+## Install
+
+```bash
+npm install livekit-plugin-parrot-stt
+# peer deps (provided by your agent app):
+npm install @livekit/agents @livekit/rtc-node
+```
+
+## Usage
+
+```ts
+import { voice } from '@livekit/agents';
+import * as ringg from 'livekit-plugin-parrot-stt';
+
+const session = new voice.AgentSession({
+  stt: new ringg.STT({
+    // reads RINGG_STT_URL / RINGG_STT_API_KEY from the environment by default
+    language: 'en', // 'hi' | 'en'
+  }),
+  // llm, tts, vad, ...
+});
+```
+
+Standalone (no `AgentSession`):
+
+```ts
+import * as ringg from 'livekit-plugin-parrot-stt';
+
+const stt = new ringg.STT({ url: 'wss://.../stt/v1/stream', apiKey: '...', language: 'en' });
+const stream = stt.stream();
+
+// push AudioFrames (Int16 PCM), then close the input:
+for (const frame of frames) stream.pushFrame(frame);
+stream.endInput();
+
+for await (const ev of stream) {
+  if (ev.type === /* FINAL_TRANSCRIPT */ 2) console.log(ev.alternatives?.[0]?.text);
+}
+```
+
+## Options
+
+| Option | Env | Default | Notes |
+| --- | --- | --- | --- |
+| `url` | `RINGG_STT_URL` | `wss://prod-api.ringg.ai/stt/v1/stream` | Full WS URL. Confirm exact path with Ringg. |
+| `apiKey` | `RINGG_STT_API_KEY` | — | Sent inside the `start` JSON frame (not a header). |
+| `language` | `RINGG_STT_LANGUAGE` | `en` | `hi` \| `en` \| … |
+| `sampleRate` | `RINGG_STT_SAMPLE_RATE` | `16000` | LiveKit audio is resampled to this. |
+| `encoding` | `RINGG_STT_ENCODING` | `int16` | `int16` \| `linear16` \| `float32` \| `int32` |
+| `mode` | `RINGG_STT_MODE` | `stream` | `stream` = interim+final; `on_final` = finals only. |
+| `enableCapPunc` | — | `true` | Request capitalisation/punctuation. |
+| `vadTailSilMs` | — | `200` | Server-VAD trailing silence. |
+| `vadConfidence` | — | `0.55` | Server-VAD threshold. |
+| `keepaliveIntervalMs` | — | `30000` | JSON ping keepalive (0 disables). |
+
+## How it maps to LiveKit
+
+- Connect → send JSON `start` (auth = `api_key`) → wait for `ready`.
+- Stream raw **binary PCM16 LE mono** frames (20 ms chunks).
+- Ringg `transcript` events → `INTERIM_TRANSCRIPT` (`is_final:false`) / `FINAL_TRANSCRIPT` (`is_final:true`).
+- `START_OF_SPEECH` / `END_OF_SPEECH` are synthesised around transcript activity.
+- Endpointing uses Ringg's **server-side VAD** (`mode:"stream"`), which is the
+  clean analog of the Pipecat integration through LiveKit's audio-only STT interface.
+
+## License
+
+Apache-2.0

package/dist/index.cjs

@@ -0,0 +1,303 @@
+"use strict";
+var __defProp = Object.defineProperty;
+var __getOwnPropDesc = Object.getOwnPropertyDescriptor;
+var __getOwnPropNames = Object.getOwnPropertyNames;
+var __hasOwnProp = Object.prototype.hasOwnProperty;
+var __export = (target, all) => {
+  for (var name in all)
+    __defProp(target, name, { get: all[name], enumerable: true });
+};
+var __copyProps = (to, from, except, desc) => {
+  if (from && typeof from === "object" || typeof from === "function") {
+    for (let key of __getOwnPropNames(from))
+      if (!__hasOwnProp.call(to, key) && key !== except)
+        __defProp(to, key, { get: () => from[key], enumerable: !(desc = __getOwnPropDesc(from, key)) || desc.enumerable });
+  }
+  return to;
+};
+var __toCommonJS = (mod) => __copyProps(__defProp({}, "__esModule", { value: true }), mod);
+
+// src/index.ts
+var index_exports = {};
+__export(index_exports, {
+  STT: () => STT,
+  SpeechStream: () => SpeechStream,
+  buildStartPayload: () => buildStartPayload,
+  defaultRinggSTTOptions: () => defaultRinggSTTOptions
+});
+module.exports = __toCommonJS(index_exports);
+var import_agents2 = require("@livekit/agents");
+
+// src/stt.ts
+var import_agents = require("@livekit/agents");
+var import_ws = require("ws");
+
+// src/models.ts
+var envInt = (name, fallback) => {
+  const v = process.env[name];
+  if (v === void 0 || v === "") return fallback;
+  const n = Number(v);
+  return Number.isFinite(n) ? n : fallback;
+};
+var defaultRinggSTTOptions = {
+  url: process.env.RINGG_STT_URL ?? "wss://prod-api.ringg.ai/stt/v1/stream",
+  apiKey: process.env.RINGG_STT_API_KEY ?? "",
+  language: process.env.RINGG_STT_LANGUAGE ?? "en",
+  sampleRate: envInt("RINGG_STT_SAMPLE_RATE", 16e3),
+  encoding: process.env.RINGG_STT_ENCODING ?? "int16",
+  mode: process.env.RINGG_STT_MODE ?? "stream",
+  enableCapPunc: true,
+  vadTailSilMs: 200,
+  vadConfidence: 0.55,
+  acceptClientVadEvents: false,
+  keepaliveIntervalMs: 3e4,
+  finalizeGraceMs: 1500
+};
+var buildStartPayload = (opts) => ({
+  type: "start",
+  sample_rate: opts.sampleRate,
+  encoding: opts.encoding,
+  language: opts.language,
+  mode: opts.mode,
+  vad_tail_sil_ms: opts.vadTailSilMs,
+  vad_confidence: opts.vadConfidence,
+  enable_cap_punc: opts.enableCapPunc,
+  accept_client_vad_events: opts.acceptClientVadEvents,
+  api_key: opts.apiKey
+});
+
+// src/stt.ts
+var sleep = (ms) => new Promise((r) => setTimeout(r, ms));
+var STT = class extends import_agents.stt.STT {
+  label = "ringg.STT";
+  #opts;
+  constructor(opts = {}) {
+    super({ streaming: true, interimResults: true });
+    this.#opts = { ...defaultRinggSTTOptions, ...opts };
+    if (!this.#opts.apiKey) {
+      throw new Error(
+        "Ringg STT: API key is required \u2014 set $RINGG_STT_API_KEY or pass { apiKey }."
+      );
+    }
+    if (!this.#opts.url) {
+      throw new Error("Ringg STT: url is required \u2014 set $RINGG_STT_URL or pass { url }.");
+    }
+  }
+  get provider() {
+    return "ringg";
+  }
+  get model() {
+    return `ringg-parrot-${this.#opts.language}`;
+  }
+  updateOptions(opts) {
+    this.#opts = { ...this.#opts, ...opts };
+  }
+  async _recognize(_frame) {
+    throw new Error(
+      "Ringg STT plugin is streaming-only; use stream(). (Ringg also exposes a REST /transcriptions endpoint, but it is not wired into this plugin.)"
+    );
+  }
+  stream(options) {
+    return new SpeechStream(this, this.#opts, options?.connOptions);
+  }
+};
+var SpeechStream = class _SpeechStream extends import_agents.stt.SpeechStream {
+  label = "ringg.SpeechStream";
+  #opts;
+  #logger = (0, import_agents.log)();
+  #speaking = false;
+  #requestId;
+  constructor(sttImpl, opts, connOptions) {
+    super(sttImpl, opts.sampleRate, connOptions);
+    this.#opts = opts;
+  }
+  async run() {
+    const ws = new import_ws.WebSocket(this.#opts.url);
+    ws.binaryType = "nodebuffer";
+    await new Promise((resolve, reject) => {
+      const onErr = (err) => reject(err);
+      ws.once("open", () => {
+        ws.off("error", onErr);
+        resolve();
+      });
+      ws.once("error", onErr);
+    });
+    ws.send(JSON.stringify(buildStartPayload(this.#opts)));
+    await new Promise((resolve, reject) => {
+      const onMsg = (data, isBinary) => {
+        if (isBinary) return;
+        let ev;
+        try {
+          ev = JSON.parse(data.toString());
+        } catch {
+          return;
+        }
+        if (ev.type === "ready") {
+          this.#requestId = ev.request_id;
+          cleanup();
+          resolve();
+        } else if (ev.type === "error") {
+          cleanup();
+          reject(
+            new Error(
+              `Ringg STT: handshake rejected \u2014 ${ev.detail ?? "unknown error"}`
+            )
+          );
+        }
+      };
+      const onClose = (code) => {
+        cleanup();
+        reject(new Error(`Ringg STT: connection closed during handshake (code ${code})`));
+      };
+      const cleanup = () => {
+        ws.off("message", onMsg);
+        ws.off("close", onClose);
+      };
+      ws.on("message", onMsg);
+      ws.once("close", onClose);
+    });
+    this.#logger.child({ url: this.#opts.url, language: this.#opts.language, requestId: this.#requestId }).debug("ringg stt: connected");
+    let closed = false;
+    const closeWs = () => {
+      if (!closed) {
+        closed = true;
+        try {
+          ws.close();
+        } catch {
+        }
+      }
+    };
+    let keepalive;
+    if (this.#opts.keepaliveIntervalMs > 0) {
+      keepalive = setInterval(() => {
+        if (ws.readyState === import_ws.WebSocket.OPEN) {
+          try {
+            ws.send(JSON.stringify({ type: "ping" }));
+          } catch {
+          }
+        }
+      }, this.#opts.keepaliveIntervalMs);
+    }
+    const recvTask = new Promise((resolve, reject) => {
+      ws.on("message", (data, isBinary) => {
+        if (isBinary) return;
+        let ev;
+        try {
+          ev = JSON.parse(data.toString());
+        } catch {
+          return;
+        }
+        this.#handleEvent(ev);
+      });
+      ws.once("close", () => resolve());
+      ws.once("error", (err) => reject(err));
+    });
+    const sendTask = async () => {
+      const samples20ms = Math.floor(this.#opts.sampleRate / 50);
+      const bstream = new import_agents.AudioByteStream(this.#opts.sampleRate, 1, samples20ms);
+      const shipFrames = (frames) => {
+        for (const f of frames) {
+          if (ws.readyState === import_ws.WebSocket.OPEN) {
+            ws.send(new Uint8Array(f.data.buffer, f.data.byteOffset, f.data.byteLength));
+          }
+        }
+      };
+      for (; ; ) {
+        if (closed) break;
+        const { done, value } = await this.input.next();
+        if (done) break;
+        if (value === _SpeechStream.FLUSH_SENTINEL) {
+          shipFrames(bstream.flush());
+        } else {
+          shipFrames(bstream.write(value.data.buffer));
+        }
+      }
+      shipFrames(bstream.flush());
+      if (ws.readyState === import_ws.WebSocket.OPEN) {
+        ws.send(JSON.stringify({ type: "end" }));
+      }
+    };
+    const abortPromise = new Promise((resolve) => {
+      const signal = this.abortController.signal;
+      if (signal.aborted) return resolve();
+      signal.addEventListener("abort", () => resolve(), { once: true });
+    });
+    const finiteInputPath = (async () => {
+      await sendTask();
+      await sleep(this.#opts.finalizeGraceMs);
+    })();
+    try {
+      await Promise.race([abortPromise, recvTask, finiteInputPath]);
+    } finally {
+      if (keepalive) clearInterval(keepalive);
+      closeWs();
+      await recvTask.catch(() => {
+      });
+    }
+  }
+  #handleEvent(ev) {
+    switch (ev.type) {
+      case "transcript": {
+        const t = ev;
+        const text = (t.transcription ?? "").trim();
+        if (!text) return;
+        if (!this.#speaking) {
+          this.#speaking = true;
+          this.queue.put({ type: import_agents.stt.SpeechEventType.START_OF_SPEECH });
+        }
+        const data = {
+          language: (0, import_agents.asLanguageCode)(t.language ?? this.#opts.language),
+          text,
+          startTime: 0,
+          endTime: 0,
+          confidence: 1
+        };
+        if (t.is_final) {
+          this.queue.put({
+            type: import_agents.stt.SpeechEventType.FINAL_TRANSCRIPT,
+            alternatives: [data],
+            requestId: this.#requestId
+          });
+          if (this.#speaking) {
+            this.#speaking = false;
+            this.queue.put({ type: import_agents.stt.SpeechEventType.END_OF_SPEECH });
+          }
+        } else {
+          this.queue.put({
+            type: import_agents.stt.SpeechEventType.INTERIM_TRANSCRIPT,
+            alternatives: [data],
+            requestId: this.#requestId
+          });
+        }
+        break;
+      }
+      case "error": {
+        this.#logger.child({ event: ev }).error("ringg stt: server error");
+        break;
+      }
+      // ready (duplicate), ack, pong, and unknown types are ignored
+      default:
+        break;
+    }
+  }
+};
+
+// src/index.ts
+var RinggPlugin = class extends import_agents2.Plugin {
+  constructor() {
+    super({
+      title: "ringg",
+      version: "0.1.0",
+      package: "livekit-plugin-parrot-stt"
+    });
+  }
+};
+import_agents2.Plugin.registerPlugin(new RinggPlugin());
+// Annotate the CommonJS export names for ESM import in node:
+0 && (module.exports = {
+  STT,
+  SpeechStream,
+  buildStartPayload,
+  defaultRinggSTTOptions
+});
+//# sourceMappingURL=index.cjs.map

package/dist/index.cjs.map

@@ -0,0 +1 @@
+{"version":3,"sources":["../src/index.ts","../src/stt.ts","../src/models.ts"],"sourcesContent":["// livekit-plugin-parrot-stt — Ringg Parrot STT for LiveKit Agents (Node.js)\n\nimport { Plugin } from '@livekit/agents';\n\nexport * from './stt.js';\nexport * from './models.js';\n\nclass RinggPlugin extends Plugin {\n  constructor() {\n    super({\n      title: 'ringg',\n      version: '0.1.0',\n      package: 'livekit-plugin-parrot-stt',\n    });\n  }\n}\n\nPlugin.registerPlugin(new RinggPlugin());\n","// Ringg Parrot STT plugin for LiveKit Agents.\n//\n// Implements the streaming STT interface (`stt.STT` + `stt.SpeechStream`) by\n// speaking the Ringg WebSocket protocol directly (there is no JS SDK).\n\nimport {\n  type APIConnectOptions,\n  type AudioBuffer,\n  AudioByteStream,\n  asLanguageCode,\n  log,\n  stt,\n} from '@livekit/agents';\nimport type { AudioFrame } from '@livekit/rtc-node';\nimport { type RawData, WebSocket } from 'ws';\nimport {\n  type RinggSTTOptions,\n  type RinggServerEvent,\n  type RinggTranscriptEvent,\n  buildStartPayload,\n  defaultRinggSTTOptions,\n} from './models.js';\n\nconst sleep = (ms: number): Promise<void> => new Promise((r) => setTimeout(r, ms));\n\nexport class STT extends stt.STT {\n  label = 'ringg.STT';\n  #opts: RinggSTTOptions;\n\n  constructor(opts: Partial<RinggSTTOptions> = {}) {\n    super({ streaming: true, interimResults: true });\n    this.#opts = { ...defaultRinggSTTOptions, ...opts };\n    if (!this.#opts.apiKey) {\n      throw new Error(\n        'Ringg STT: API key is required — set $RINGG_STT_API_KEY or pass { apiKey }.',\n      );\n    }\n    if (!this.#opts.url) {\n      throw new Error('Ringg STT: url is required — set $RINGG_STT_URL or pass { url }.');\n    }\n  }\n\n  get provider(): string {\n    return 'ringg';\n  }\n\n  get model(): string {\n    return `ringg-parrot-${this.#opts.language}`;\n  }\n\n  updateOptions(opts: Partial<RinggSTTOptions>): void {\n    this.#opts = { ...this.#opts, ...opts };\n  }\n\n  async _recognize(_frame: AudioBuffer): Promise<stt.SpeechEvent> {\n    throw new Error(\n      'Ringg STT plugin is streaming-only; use stream(). ' +\n        '(Ringg also exposes a REST /transcriptions endpoint, but it is not wired into this plugin.)',\n    );\n  }\n\n  stream(options?: { connOptions?: APIConnectOptions }): stt.SpeechStream {\n    return new SpeechStream(this, this.#opts, options?.connOptions);\n  }\n}\n\nexport class SpeechStream extends stt.SpeechStream {\n  label = 'ringg.SpeechStream';\n  #opts: RinggSTTOptions;\n  #logger = log();\n  #speaking = false;\n  #requestId?: string;\n\n  constructor(sttImpl: STT, opts: RinggSTTOptions, connOptions?: APIConnectOptions) {\n    // Passing sampleRate as `neededSampleRate` makes the base class resample\n    // incoming (typically 48kHz) LiveKit frames down to Ringg's rate for us.\n    super(sttImpl, opts.sampleRate, connOptions);\n    this.#opts = opts;\n  }\n\n  protected async run(): Promise<void> {\n    const ws = new WebSocket(this.#opts.url);\n    ws.binaryType = 'nodebuffer';\n\n    // 1) wait for the socket to open\n    await new Promise<void>((resolve, reject) => {\n      const onErr = (err: Error) => reject(err);\n      ws.once('open', () => {\n        ws.off('error', onErr);\n        resolve();\n      });\n      ws.once('error', onErr);\n    });\n\n    // 2) send the `start` handshake and wait for the `ready` reply\n    ws.send(JSON.stringify(buildStartPayload(this.#opts)));\n    await new Promise<void>((resolve, reject) => {\n      const onMsg = (data: RawData, isBinary: boolean) => {\n        if (isBinary) return;\n        let ev: RinggServerEvent;\n        try {\n          ev = JSON.parse(data.toString());\n        } catch {\n          return;\n        }\n        if (ev.type === 'ready') {\n          this.#requestId = (ev as { request_id?: string }).request_id;\n          cleanup();\n          resolve();\n        } else if (ev.type === 'error') {\n          cleanup();\n          reject(\n            new Error(\n              `Ringg STT: handshake rejected — ${(ev as { detail?: string }).detail ?? 'unknown error'}`,\n            ),\n          );\n        }\n      };\n      const onClose = (code: number) => {\n        cleanup();\n        reject(new Error(`Ringg STT: connection closed during handshake (code ${code})`));\n      };\n      const cleanup = () => {\n        ws.off('message', onMsg);\n        ws.off('close', onClose);\n      };\n      ws.on('message', onMsg);\n      ws.once('close', onClose);\n    });\n\n    this.#logger\n      .child({ url: this.#opts.url, language: this.#opts.language, requestId: this.#requestId })\n      .debug('ringg stt: connected');\n\n    let closed = false;\n    const closeWs = () => {\n      if (!closed) {\n        closed = true;\n        try {\n          ws.close();\n        } catch {\n          /* ignore */\n        }\n      }\n    };\n\n    // keepalive: application-level JSON ping (Ringg replies with `pong`)\n    let keepalive: ReturnType<typeof setInterval> | undefined;\n    if (this.#opts.keepaliveIntervalMs > 0) {\n      keepalive = setInterval(() => {\n        if (ws.readyState === WebSocket.OPEN) {\n          try {\n            ws.send(JSON.stringify({ type: 'ping' }));\n          } catch {\n            /* ignore */\n          }\n        }\n      }, this.#opts.keepaliveIntervalMs);\n    }\n\n    // 3) receive loop — resolves when the server closes the socket\n    const recvTask = new Promise<void>((resolve, reject) => {\n      ws.on('message', (data: RawData, isBinary: boolean) => {\n        if (isBinary) return; // Ringg only sends JSON text frames\n        let ev: RinggServerEvent;\n        try {\n          ev = JSON.parse(data.toString());\n        } catch {\n          return;\n        }\n        this.#handleEvent(ev);\n      });\n      ws.once('close', () => resolve());\n      ws.once('error', (err) => reject(err));\n    });\n\n    // 4) send loop — reads resampled frames from `this.input`, ships raw PCM16\n    const sendTask = async () => {\n      const samples20ms = Math.floor(this.#opts.sampleRate / 50); // 20ms chunks\n      const bstream = new AudioByteStream(this.#opts.sampleRate, 1, samples20ms);\n      const shipFrames = (frames: AudioFrame[]) => {\n        for (const f of frames) {\n          if (ws.readyState === WebSocket.OPEN) {\n            // send exactly this frame's bytes (raw PCM16 LE), regardless of view offset\n            ws.send(new Uint8Array(f.data.buffer, f.data.byteOffset, f.data.byteLength));\n          }\n        }\n      };\n\n      for (;;) {\n        if (closed) break;\n        const { done, value } = await this.input.next();\n        if (done) break;\n        if (value === SpeechStream.FLUSH_SENTINEL) {\n          shipFrames(bstream.flush());\n        } else {\n          shipFrames(bstream.write((value as AudioFrame).data.buffer));\n        }\n      }\n\n      // input exhausted: flush remaining audio and ask the server to finalize\n      shipFrames(bstream.flush());\n      if (ws.readyState === WebSocket.OPEN) {\n        ws.send(JSON.stringify({ type: 'end' }));\n      }\n    };\n\n    // abort (session close / cancel) → stop promptly\n    const abortPromise = new Promise<void>((resolve) => {\n      const signal = this.abortController.signal;\n      if (signal.aborted) return resolve();\n      signal.addEventListener('abort', () => resolve(), { once: true });\n    });\n\n    // finite input (a file): after the send loop finishes, wait a grace period\n    // for trailing final transcripts, then let the socket close.\n    const finiteInputPath = (async () => {\n      await sendTask();\n      await sleep(this.#opts.finalizeGraceMs);\n    })();\n\n    try {\n      await Promise.race([abortPromise, recvTask, finiteInputPath]);\n    } finally {\n      if (keepalive) clearInterval(keepalive);\n      closeWs();\n      // let recvTask settle so its listeners are cleaned up\n      await recvTask.catch(() => {});\n    }\n  }\n\n  #handleEvent(ev: RinggServerEvent): void {\n    switch (ev.type) {\n      case 'transcript': {\n        const t = ev as RinggTranscriptEvent;\n        const text = (t.transcription ?? '').trim();\n        if (!text) return;\n\n        if (!this.#speaking) {\n          this.#speaking = true;\n          this.queue.put({ type: stt.SpeechEventType.START_OF_SPEECH });\n        }\n\n        const data: stt.SpeechData = {\n          language: asLanguageCode(t.language ?? this.#opts.language),\n          text,\n          startTime: 0,\n          endTime: 0,\n          confidence: 1.0,\n        };\n\n        if (t.is_final) {\n          this.queue.put({\n            type: stt.SpeechEventType.FINAL_TRANSCRIPT,\n            alternatives: [data],\n            requestId: this.#requestId,\n          });\n          if (this.#speaking) {\n            this.#speaking = false;\n            this.queue.put({ type: stt.SpeechEventType.END_OF_SPEECH });\n          }\n        } else {\n          this.queue.put({\n            type: stt.SpeechEventType.INTERIM_TRANSCRIPT,\n            alternatives: [data],\n            requestId: this.#requestId,\n          });\n        }\n        break;\n      }\n      case 'error': {\n        this.#logger.child({ event: ev }).error('ringg stt: server error');\n        break;\n      }\n      // ready (duplicate), ack, pong, and unknown types are ignored\n      default:\n        break;\n    }\n  }\n}\n","// Types and defaults for the Ringg Parrot STT plugin.\n//\n// The wire protocol is the one implemented by the Python `ringglabs` SDK\n// (ringglabs.stt) and used by the Pipecat `RinggSTTService`:\n//   - connect to a WebSocket\n//   - send a JSON `start` frame (auth = `api_key` inside this frame)\n//   - stream raw binary PCM16 (little-endian, mono) audio frames\n//   - receive JSON `transcript` events discriminated by `is_final`\n\nexport type RinggEncoding = 'int16' | 'linear16' | 'float32' | 'int32';\nexport type RinggMode = 'stream' | 'on_final';\n// hi | en are the documented values; keep it open for forward-compat.\nexport type RinggLanguage = 'hi' | 'en' | (string & {});\n\n/** Resolved options for {@link STT}. */\nexport interface RinggSTTOptions {\n  /** Full WebSocket URL, e.g. `wss://prod-api.ringg.ai/stt/v1/stream`. */\n  url: string;\n  /** Ringg API key. 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package/dist/index.d.cts

@@ -0,0 +1,89 @@
+import { stt, AudioBuffer, APIConnectOptions } from '@livekit/agents';
+
+type RinggEncoding = 'int16' | 'linear16' | 'float32' | 'int32';
+type RinggMode = 'stream' | 'on_final';
+type RinggLanguage = 'hi' | 'en' | (string & {});
+/** Resolved options for {@link STT}. */
+interface RinggSTTOptions {
+    /** Full WebSocket URL, e.g. `wss://prod-api.ringg.ai/stt/v1/stream`. */
+    url: string;
+    /** Ringg API key. Sent inside the `start` JSON frame (not a header). */
+    apiKey: string;
+    /** Language code sent to the server (default `en`). */
+    language: RinggLanguage;
+    /** Audio sample rate the plugin sends to Ringg. LiveKit frames are resampled to this. */
+    sampleRate: number;
+    /** Audio encoding declared to the server. LiveKit gives PCM16, so `int16`. */
+    encoding: RinggEncoding;
+    /** `stream` = interim + final; `on_final` = finals only. */
+    mode: RinggMode;
+    /** Ask the server for capitalisation/punctuation. */
+    enableCapPunc: boolean;
+    /** Server-VAD trailing silence (ms) before finalizing a segment. */
+    vadTailSilMs: number;
+    /** Server-VAD confidence threshold (0..1). */
+    vadConfidence: number;
+    /** If true, the client is expected to send speaking-boundary events (advanced). */
+    acceptClientVadEvents: boolean;
+    /** Application-level keepalive ping interval (ms). 0 disables. */
+    keepaliveIntervalMs: number;
+    /** After the input ends, how long to keep reading for trailing finals (ms). */
+    finalizeGraceMs: number;
+}
+declare const defaultRinggSTTOptions: RinggSTTOptions;
+interface RinggReadyEvent {
+    type: 'ready';
+    request_id: string;
+    [k: string]: unknown;
+}
+interface RinggTranscriptEvent {
+    type: 'transcript';
+    transcription: string;
+    is_final: boolean;
+    language?: string;
+    request_id?: string;
+    segment_idx?: number;
+    [k: string]: unknown;
+}
+interface RinggErrorEvent {
+    type: 'error';
+    detail: string;
+    code?: string;
+    status_code?: number;
+}
+interface RinggAckEvent {
+    type: 'ack';
+    ack_for: string;
+    state: string;
+}
+interface RinggPongEvent {
+    type: 'pong';
+    timestamp: number;
+}
+type RinggServerEvent = RinggReadyEvent | RinggTranscriptEvent | RinggErrorEvent | RinggAckEvent | RinggPongEvent | {
+    type: string;
+    [k: string]: unknown;
+};
+/** Build the JSON `start` handshake payload. */
+declare const buildStartPayload: (opts: RinggSTTOptions) => Record<string, unknown>;
+
+declare class STT extends stt.STT {
+    #private;
+    label: string;
+    constructor(opts?: Partial<RinggSTTOptions>);
+    get provider(): string;
+    get model(): string;
+    updateOptions(opts: Partial<RinggSTTOptions>): void;
+    _recognize(_frame: AudioBuffer): Promise<stt.SpeechEvent>;
+    stream(options?: {
+        connOptions?: APIConnectOptions;
+    }): stt.SpeechStream;
+}
+declare class SpeechStream extends stt.SpeechStream {
+    #private;
+    label: string;
+    constructor(sttImpl: STT, opts: RinggSTTOptions, connOptions?: APIConnectOptions);
+    protected run(): Promise<void>;
+}
+
+export { type RinggAckEvent, type RinggEncoding, type RinggErrorEvent, type RinggLanguage, type RinggMode, type RinggPongEvent, type RinggReadyEvent, type RinggSTTOptions, type RinggServerEvent, type RinggTranscriptEvent, STT, SpeechStream, buildStartPayload, defaultRinggSTTOptions };

package/dist/index.d.ts

@@ -0,0 +1,89 @@
+import { stt, AudioBuffer, APIConnectOptions } from '@livekit/agents';
+
+type RinggEncoding = 'int16' | 'linear16' | 'float32' | 'int32';
+type RinggMode = 'stream' | 'on_final';
+type RinggLanguage = 'hi' | 'en' | (string & {});
+/** Resolved options for {@link STT}. */
+interface RinggSTTOptions {
+    /** Full WebSocket URL, e.g. `wss://prod-api.ringg.ai/stt/v1/stream`. */
+    url: string;
+    /** Ringg API key. Sent inside the `start` JSON frame (not a header). */
+    apiKey: string;
+    /** Language code sent to the server (default `en`). */
+    language: RinggLanguage;
+    /** Audio sample rate the plugin sends to Ringg. LiveKit frames are resampled to this. */
+    sampleRate: number;
+    /** Audio encoding declared to the server. LiveKit gives PCM16, so `int16`. */
+    encoding: RinggEncoding;
+    /** `stream` = interim + final; `on_final` = finals only. */
+    mode: RinggMode;
+    /** Ask the server for capitalisation/punctuation. */
+    enableCapPunc: boolean;
+    /** Server-VAD trailing silence (ms) before finalizing a segment. */
+    vadTailSilMs: number;
+    /** Server-VAD confidence threshold (0..1). */
+    vadConfidence: number;
+    /** If true, the client is expected to send speaking-boundary events (advanced). */
+    acceptClientVadEvents: boolean;
+    /** Application-level keepalive ping interval (ms). 0 disables. */
+    keepaliveIntervalMs: number;
+    /** After the input ends, how long to keep reading for trailing finals (ms). */
+    finalizeGraceMs: number;
+}
+declare const defaultRinggSTTOptions: RinggSTTOptions;
+interface RinggReadyEvent {
+    type: 'ready';
+    request_id: string;
+    [k: string]: unknown;
+}
+interface RinggTranscriptEvent {
+    type: 'transcript';
+    transcription: string;
+    is_final: boolean;
+    language?: string;
+    request_id?: string;
+    segment_idx?: number;
+    [k: string]: unknown;
+}
+interface RinggErrorEvent {
+    type: 'error';
+    detail: string;
+    code?: string;
+    status_code?: number;
+}
+interface RinggAckEvent {
+    type: 'ack';
+    ack_for: string;
+    state: string;
+}
+interface RinggPongEvent {
+    type: 'pong';
+    timestamp: number;
+}
+type RinggServerEvent = RinggReadyEvent | RinggTranscriptEvent | RinggErrorEvent | RinggAckEvent | RinggPongEvent | {
+    type: string;
+    [k: string]: unknown;
+};
+/** Build the JSON `start` handshake payload. */
+declare const buildStartPayload: (opts: RinggSTTOptions) => Record<string, unknown>;
+
+declare class STT extends stt.STT {
+    #private;
+    label: string;
+    constructor(opts?: Partial<RinggSTTOptions>);
+    get provider(): string;
+    get model(): string;
+    updateOptions(opts: Partial<RinggSTTOptions>): void;
+    _recognize(_frame: AudioBuffer): Promise<stt.SpeechEvent>;
+    stream(options?: {
+        connOptions?: APIConnectOptions;
+    }): stt.SpeechStream;
+}
+declare class SpeechStream extends stt.SpeechStream {
+    #private;
+    label: string;
+    constructor(sttImpl: STT, opts: RinggSTTOptions, connOptions?: APIConnectOptions);
+    protected run(): Promise<void>;
+}
+
+export { type RinggAckEvent, type RinggEncoding, type RinggErrorEvent, type RinggLanguage, type RinggMode, type RinggPongEvent, type RinggReadyEvent, type RinggSTTOptions, type RinggServerEvent, type RinggTranscriptEvent, STT, SpeechStream, buildStartPayload, defaultRinggSTTOptions };