Medical Mode

import logging
from typing import Type

import assemblyai as aai
from assemblyai.streaming.v3 import (
    BeginEvent,
    StreamingClient,
    StreamingClientOptions,
    StreamingError,
    StreamingEvents,
    StreamingParameters,
    TerminationEvent,
    TurnEvent,
)

api_key = "<YOUR_API_KEY>"

logging.basicConfig(level=logging.INFO)
logger = logging.getLogger(__name__)


def on_begin(self: Type[StreamingClient], event: BeginEvent):
    print(f"Session started: {event.id}")


def on_turn(self: Type[StreamingClient], event: TurnEvent):
    if event.turn_is_formatted:
        print(f"\r{' ' * 100}\r{event.transcript}")
    else:
        print(f"\r{event.transcript}", end='', flush=True)


def on_terminated(self: Type[StreamingClient], event: TerminationEvent):
    print(
        f"Session terminated: {event.audio_duration_seconds} seconds of audio processed"
    )


def on_error(self: Type[StreamingClient], error: StreamingError):
    print(f"Error occurred: {error}")


def main():
    client = StreamingClient(
        StreamingClientOptions(
            api_key=api_key,
            api_host="streaming.assemblyai.com",
        )
    )

    client.on(StreamingEvents.Begin, on_begin)
    client.on(StreamingEvents.Turn, on_turn)
    client.on(StreamingEvents.Termination, on_terminated)
    client.on(StreamingEvents.Error, on_error)

    client.connect(
        StreamingParameters(
            sample_rate=16000,
            speech_model="u3-rt-pro",
            domain="medical-v1",
        )
    )

    try:
        client.stream(
          aai.extras.MicrophoneStream(sample_rate=16000)
        )
    finally:
        client.disconnect(terminate=True)


if __name__ == "__main__":
    main()

const WebSocket = require("ws");
const mic = require("mic");
const querystring = require("querystring");

// --- Configuration ---
const YOUR_API_KEY = "YOUR-API-KEY"; // Replace with your actual API key
const CONNECTION_PARAMS = {
  sample_rate: 16000,
  speech_model: "u3-rt-pro",
  domain: "medical-v1",
};
const API_ENDPOINT_BASE_URL = "wss://streaming.assemblyai.com/v3/ws";
const API_ENDPOINT = `${API_ENDPOINT_BASE_URL}?${querystring.stringify(CONNECTION_PARAMS)}`;

// Audio Configuration
const SAMPLE_RATE = CONNECTION_PARAMS.sample_rate;
const CHANNELS = 1;

// Global variables
let micInstance = null;
let micInputStream = null;
let ws = null;
let stopRequested = false;

// --- Helper functions ---
function formatTimestamp(timestamp) {
  return new Date(timestamp * 1000).toISOString();
}

// --- Main function ---
async function run() {
  console.log("Starting AssemblyAI real-time transcription...");

  ws = new WebSocket(API_ENDPOINT, {
    headers: {
      Authorization: YOUR_API_KEY,
    },
  });

  ws.on("open", () => {
    console.log("WebSocket connection opened.");
    console.log(`Connected to: ${API_ENDPOINT}`);
    startMicrophone();
  });

  ws.on("message", (message) => {
    try {
      const data = JSON.parse(message);
      const msgType = data.type;

      if (msgType === "Begin") {
        const sessionId = data.id;
        const expiresAt = data.expires_at;
        console.log(
          `\nSession began: ID=${sessionId}, ExpiresAt=${formatTimestamp(expiresAt)}`
        );
      } else if (msgType === "Turn") {
        const transcript = data.transcript || "";
        const formatted = data.turn_is_formatted;

        if (formatted) {
          process.stdout.write("\r" + " ".repeat(100) + "\r");
          console.log(transcript);
        } else {
          process.stdout.write(`\r${transcript}`);
        }
      } else if (msgType === "Termination") {
        const audioDuration = data.audio_duration_seconds;
        const sessionDuration = data.session_duration_seconds;
        console.log(
          `\nSession Terminated: Audio Duration=${audioDuration}s, Session Duration=${sessionDuration}s`
        );
      }
    } catch (error) {
      console.error(`\nError handling message: ${error}`);
      console.error(`Message data: ${message}`);
    }
  });

  ws.on("error", (error) => {
    console.error(`\nWebSocket Error: ${error}`);
    cleanup();
  });

  ws.on("close", (code, reason) => {
    console.log(`\nWebSocket Disconnected: Status=${code}, Msg=${reason}`);
    cleanup();
  });

  setupTerminationHandlers();
}

function startMicrophone() {
  try {
    micInstance = mic({
      rate: SAMPLE_RATE.toString(),
      channels: CHANNELS.toString(),
      debug: false,
      exitOnSilence: 6,
    });

    micInputStream = micInstance.getAudioStream();

    micInputStream.on("data", (data) => {
      if (ws && ws.readyState === WebSocket.OPEN && !stopRequested) {
        ws.send(data);
      }
    });

    micInputStream.on("error", (err) => {
      console.error(`Microphone Error: ${err}`);
      cleanup();
    });

    micInstance.start();
    console.log("Microphone stream opened successfully.");
    console.log("Speak into your microphone. Press Ctrl+C to stop.");
  } catch (error) {
    console.error(`Error opening microphone stream: ${error}`);
    cleanup();
  }
}

function cleanup() {
  stopRequested = true;

  if (micInstance) {
    try {
      micInstance.stop();
    } catch (error) {
      console.error(`Error stopping microphone: ${error}`);
    }
    micInstance = null;
  }

  if (ws && [WebSocket.OPEN, WebSocket.CONNECTING].includes(ws.readyState)) {
    try {
      if (ws.readyState === WebSocket.OPEN) {
        const terminateMessage = { type: "Terminate" };
        console.log(
          `Sending termination message: ${JSON.stringify(terminateMessage)}`
        );
        ws.send(JSON.stringify(terminateMessage));
      }
      ws.close();
    } catch (error) {
      console.error(`Error closing WebSocket: ${error}`);
    }
    ws = null;
  }

  console.log("Cleanup complete.");
}

function setupTerminationHandlers() {
  process.on("SIGINT", () => {
    console.log("\nCtrl+C received. Stopping...");
    cleanup();
    setTimeout(() => process.exit(0), 1000);
  });

  process.on("SIGTERM", () => {
    console.log("\nTermination signal received. Stopping...");
    cleanup();
    setTimeout(() => process.exit(0), 1000);
  });

  process.on("uncaughtException", (error) => {
    console.error(`\nUncaught exception: ${error}`);
    cleanup();
    setTimeout(() => process.exit(1), 1000);
  });
}

// Start the application
run();

import { Readable } from "stream";
import { AssemblyAI } from "assemblyai";
import recorder from "node-record-lpcm16";

const run = async () => {
  const client = new AssemblyAI({
    apiKey: "<YOUR_API_KEY>",
  });

  const transcriber = client.streaming.transcriber({
    sampleRate: 16_000,
    speechModel: "u3-rt-pro",
    domain: "medical-v1",
  });

  transcriber.on("open", ({ id }) => {
    console.log(`Session opened with ID: ${id}`);
  });

  transcriber.on("error", (error) => {
    console.error("Error:", error);
  });

  transcriber.on("close", (code, reason) =>
    console.log("Session closed:", code, reason)
  );

  transcriber.on("turn", (turn) => {
    if (turn.turn_is_formatted) {
      process.stdout.write("\r" + " ".repeat(100) + "\r");
      console.log(turn.transcript);
    } else {
      process.stdout.write(`\r${turn.transcript}`);
    }
  });

  try {
    console.log("Connecting to streaming transcript service");

    await transcriber.connect();

    console.log("Starting recording");

    const recording = recorder.record({
      channels: 1,
      sampleRate: 16_000,
      audioType: "wav", // Linear PCM
    });

    Readable.toWeb(recording.stream()).pipeTo(transcriber.stream());

    // Stop recording and close connection using Ctrl-C.

    process.on("SIGINT", async function () {
      console.log();
      console.log("Stopping recording");
      recording.stop();

      console.log("Closing streaming transcript connection");
      await transcriber.close();

      process.exit();
    });
  } catch (error) {
    console.error(error);
  }
};

run();