Streaming Migration Guide: Gladia to AssemblyAI

This guide walks through the process of migrating from Gladia to AssemblyAI for transcribing streaming audio.

Get Started

Before we begin, make sure you have an AssemblyAI account and an API key. You can sign up for a free account and get your API key from your AssemblyAI dashboard.

Side-By-Side Code Comparison

Below is a side-by-side comparison of a basic snippet to transcribe live audio by Gladia and AssemblyAI using a microphone:

Gladia
AssemblyAI

import asyncio
import base64
import json
import signal
from datetime import time
import pyaudio
import requests
from websockets.asyncio.client import connect
from websockets.exceptions import ConnectionClosedOK

# Constants
GLADIA_API_KEY = "<YOUR_GLADIA_API_KEY>"
GLADIA_API_URL = "https://api.gladia.io"

# Audio configuration
CHANNELS = 1
FORMAT = pyaudio.paInt16
FRAMES_PER_BUFFER = 3200
SAMPLE_RATE = 16_000

async def main():
    # Initialize the session
    config = {
        "encoding": "wav/pcm",
        "sample_rate": SAMPLE_RATE,
        "bit_depth": 16,
        "channels": CHANNELS,
        "language_config": {
            "languages": ["en"],
        },
    }

    # Start the live session
    response = requests.post(
        f"{GLADIA_API_URL}/v2/live",
        headers={"X-Gladia-Key": GLADIA_API_KEY},
        json=config,
        timeout=3,
    )

    if not response.ok:
        print(f"{response.status_code}: {response.text or response.reason}")
        exit(response.status_code)

    session_data = response.json()

    # Connect to the websocket
    async with connect(session_data["url"]) as websocket:
        print("\n################ Begin session ################\n")

        # Handle Ctrl+C to stop recording
        loop = asyncio.get_running_loop()
        loop.add_signal_handler(
            signal.SIGINT,
            lambda: loop.create_task(stop_recording(websocket)),
        )

        # Create tasks for sending audio and receiving transcripts
        send_audio_task = asyncio.create_task(send_audio(websocket))
        receive_transcript_task = asyncio.create_task(receive_transcript(websocket))

        await asyncio.wait([send_audio_task, receive_transcript_task])

async def stop_recording(websocket):
    print(">>>>> Ending the recording…")
    await websocket.send(json.dumps({"type": "stop_recording"}))
    await asyncio.sleep(0)

async def send_audio(websocket):
    # Initialize PyAudio
    p = pyaudio.PyAudio()

    # Open audio stream
    stream = p.open(
        format=FORMAT,
        channels=CHANNELS,
        rate=SAMPLE_RATE,
        input=True,
        frames_per_buffer=FRAMES_PER_BUFFER,
    )

    # Send audio chunks
    while True:
        data = stream.read(FRAMES_PER_BUFFER)
        data = base64.b64encode(data).decode("utf-8")
        json_data = json.dumps({"type": "audio_chunk", "data": {"chunk": str(data)}})
        try:
            await websocket.send(json_data)
            await asyncio.sleep(0.1)  # Send audio every 100ms
        except ConnectionClosedOK:
            return

async def receive_transcript(websocket):
    # Process incoming messages
    async for message in websocket:
        content = json.loads(message)

        # Print transcripts
        if content["type"] == "transcript" and content["data"]["is_final"]:
            text = content["data"]["utterance"]["text"].strip()
            print(f"Final: {text}")

        # Print final results
        if content["type"] == "post_final_transcript":
            print("\n################ End of session ################\n")
            print(json.dumps(content, indent=2, ensure_ascii=False))

if __name__ == "__main__":
    asyncio.run(main())

import pyaudio
import websocket
import json
import threading
import time
from urllib.parse import urlencode
from datetime import datetime

# --- Configuration ---
YOUR_API_KEY = "YOUR-API-KEY"  # Replace with your actual API key

CONNECTION_PARAMS = {
    "sample_rate": 16000,
    "speech_model": "universal-3-5-pro",
}
API_ENDPOINT_BASE_URL = "wss://streaming.assemblyai.com/v3/ws"
API_ENDPOINT = f"{API_ENDPOINT_BASE_URL}?{urlencode(CONNECTION_PARAMS)}"

# Audio Configuration
FRAMES_PER_BUFFER = 800  # 50ms of audio (0.05s * 16000Hz)
SAMPLE_RATE = CONNECTION_PARAMS["sample_rate"]
CHANNELS = 1
FORMAT = pyaudio.paInt16

# Global variables for audio stream and websocket
audio = None
stream = None
ws_app = None
audio_thread = None
stop_event = threading.Event()  # To signal the audio thread to stop

# --- WebSocket Event Handlers ---


def on_open(ws):
    """Called when the WebSocket connection is established."""
    print("WebSocket connection opened.")
    print(f"Connected to: {API_ENDPOINT}")

    # Start sending audio data in a separate thread
    def stream_audio():
        global stream
        print("Starting audio streaming...")
        while not stop_event.is_set():
            try:
                audio_data = stream.read(FRAMES_PER_BUFFER, exception_on_overflow=False)
                # Send audio data as binary message
                ws.send(audio_data, websocket.ABNF.OPCODE_BINARY)
            except Exception as e:
                print(f"Error streaming audio: {e}")
                # If stream read fails, likely means it's closed, stop the loop
                break
        print("Audio streaming stopped.")

    global audio_thread
    audio_thread = threading.Thread(target=stream_audio)
    audio_thread.daemon = (
        True  # Allow main thread to exit even if this thread is running
    )
    audio_thread.start()

def on_message(ws, message):
    try:
        data = json.loads(message)
        msg_type = data.get('type')

        if msg_type == "Begin":
            session_id = data.get('id')
            expires_at = data.get('expires_at')
            print(f"\nSession began: ID={session_id}, ExpiresAt={datetime.fromtimestamp(expires_at)}")
        elif msg_type == "Turn":
            transcript = data.get('transcript', '')
            end_of_turn = data.get('end_of_turn', False)

            # Print final end-of-turn transcript
            if end_of_turn:
                print('\r' + ' ' * 80 + '\r', end='')
                print(transcript)
            else:
                print(f"\r{transcript}", end='')
        elif msg_type == "Termination":
            audio_duration = data.get('audio_duration_seconds', 0)
            session_duration = data.get('session_duration_seconds', 0)
            print(f"\nSession Terminated: Audio Duration={audio_duration}s, Session Duration={session_duration}s")
    except json.JSONDecodeError as e:
        print(f"Error decoding message: {e}")
    except Exception as e:
        print(f"Error handling message: {e}")

def on_error(ws, error):
    """Called when a WebSocket error occurs."""
    print(f"\nWebSocket Error: {error}")
    # Attempt to signal stop on error
    stop_event.set()


def on_close(ws, close_status_code, close_msg):
    """Called when the WebSocket connection is closed."""
    print(f"\nWebSocket Disconnected: Status={close_status_code}, Msg={close_msg}")
    # Ensure audio resources are released
    global stream, audio
    stop_event.set()  # Signal audio thread just in case it's still running

    if stream:
        if stream.is_active():
            stream.stop_stream()
        stream.close()
        stream = None
    if audio:
        audio.terminate()
        audio = None
    # Try to join the audio thread to ensure clean exit
    if audio_thread and audio_thread.is_alive():
        audio_thread.join(timeout=1.0)


# --- Main Execution ---
def run():
    global audio, stream, ws_app

    # Initialize PyAudio
    audio = pyaudio.PyAudio()

    # Open microphone stream
    try:
        stream = audio.open(
            input=True,
            frames_per_buffer=FRAMES_PER_BUFFER,
            channels=CHANNELS,
            format=FORMAT,
            rate=SAMPLE_RATE,
        )
        print("Microphone stream opened successfully.")
        print("Speak into your microphone. Press Ctrl+C to stop.")
    except Exception as e:
        print(f"Error opening microphone stream: {e}")
        if audio:
            audio.terminate()
        return  # Exit if microphone cannot be opened

    # Create WebSocketApp
    ws_app = websocket.WebSocketApp(
        API_ENDPOINT,
        header={"Authorization": YOUR_API_KEY},
        on_open=on_open,
        on_message=on_message,
        on_error=on_error,
        on_close=on_close,
    )

    # Run WebSocketApp in a separate thread to allow main thread to catch KeyboardInterrupt
    ws_thread = threading.Thread(target=ws_app.run_forever)
    ws_thread.daemon = True
    ws_thread.start()

    try:
        # Keep main thread alive until interrupted
        while ws_thread.is_alive():
            time.sleep(0.1)
    except KeyboardInterrupt:
        print("\nCtrl+C received. Stopping...")
        stop_event.set()  # Signal audio thread to stop

        # Send termination message to the server
        if ws_app and ws_app.sock and ws_app.sock.connected:
            try:
                terminate_message = {"type": "Terminate"}
                print(f"Sending termination message: {json.dumps(terminate_message)}")
                ws_app.send(json.dumps(terminate_message))
                # Give a moment for messages to process before forceful close
                time.sleep(5)
            except Exception as e:
                print(f"Error sending termination message: {e}")

        # Close the WebSocket connection (will trigger on_close)
        if ws_app:
            ws_app.close()

        # Wait for WebSocket thread to finish
        ws_thread.join(timeout=2.0)

    except Exception as e:
        print(f"\nAn unexpected error occurred: {e}")
        stop_event.set()
        if ws_app:
            ws_app.close()
        ws_thread.join(timeout=2.0)

    finally:
        # Final cleanup (already handled in on_close, but good as a fallback)
        if stream and stream.is_active():
            stream.stop_stream()
        if stream:
            stream.close()
        if audio:
            audio.terminate()
        print("Cleanup complete. Exiting.")


if __name__ == "__main__":
    run()

Authentication

Gladia
AssemblyAI

import asyncio
import base64
import json
import signal
from datetime import time
import pyaudio
import requests
from websockets.asyncio.client import connect
from websockets.exceptions import ConnectionClosedOK

GLADIA_API_KEY = "<YOUR_GLADIA_API_KEY>"

import pyaudio
import websocket
import json
import threading
import time
from urllib.parse import urlencode
from datetime import datetime

YOUR_API_KEY = "YOUR-API-KEY"

Protect Your API KeyFor improved security, store your API key as an environment variable.

Connection Parameters & Microphone Setup

Gladia
AssemblyAI

GLADIA_API_URL = "https://api.gladia.io"

CHANNELS = 1
FORMAT = pyaudio.paInt16
FRAMES_PER_BUFFER = 3200
SAMPLE_RATE = 16_000

config = {
    "encoding": "wav/pcm",
    "sample_rate": SAMPLE_RATE,
    "bit_depth": 16,
    "channels": CHANNELS,
    "language_config": {
        "languages": ["en"]
    },
}

CONNECTION_PARAMS = {
    "sample_rate": 16000,
    "speech_model": "universal-3-5-pro",
}
API_ENDPOINT_BASE_URL = "wss://streaming.assemblyai.com/v3/ws"
API_ENDPOINT = f"{API_ENDPOINT_BASE_URL}?{urlencode(CONNECTION_PARAMS)}"

# Audio Configuration
FRAMES_PER_BUFFER = 800  # 50ms of audio (0.05s * 16000Hz)
SAMPLE_RATE = CONNECTION_PARAMS["sample_rate"]
CHANNELS = 1
FORMAT = pyaudio.paInt16

# Global variables for audio stream and websocket
audio = None
stream = None
ws_app = None
audio_thread = None
stop_event = threading.Event()  # To signal the audio thread to stop

Helpful information about our streaming model:

Universal-3.5 Pro Model — Connect to wss://streaming.assemblyai.com/v3/ws with speech_model=universal-3-5-pro to use our latest, highest-accuracy streaming model — Universal-3.5 Pro.
Built-in Formatting — Universal-3.5 Pro always returns formatted transcripts with smart punctuation & casing. No extra parameter is needed.
Partial Transcripts — AssemblyAI streams interim results automatically. Universal-3.5 Pro emits partials during periods of silence, with at most one partial per silence period.

Open Microphone Stream & Create WebSocket

Gladia
AssemblyAI

# Start the live session
response = requests.post(
    f"{GLADIA_API_URL}/v2/live",
    headers={"X-Gladia-Key": GLADIA_API_KEY},
    json=config,
    timeout=3,
)

if not response.ok:
    print(f"{response.status_code}: {response.text or response.reason}")
    exit(response.status_code)

session_data = response.json()

async def send_audio(websocket):
    # Initialize PyAudio
    p = pyaudio.PyAudio()

    # Open audio stream
    stream = p.open(
        format=FORMAT,
        channels=CHANNELS,
        rate=SAMPLE_RATE,
        input=True,
        frames_per_buffer=FRAMES_PER_BUFFER,
    )

    # Send audio chunks
    while True:
        data = stream.read(FRAMES_PER_BUFFER)
        data = base64.b64encode(data).decode("utf-8")
        json_data = json.dumps({"type": "audio_chunk", "data": {"chunk": str(data)}})
        try:
            await websocket.send(json_data)
            await asyncio.sleep(0.1)  # Send audio every 100ms
        except ConnectionClosedOK:
            return

global audio, stream, ws_app
# Initialize PyAudio
audio = pyaudio.PyAudio()

# Open microphone stream
try:
    stream = audio.open(
        input=True,
        frames_per_buffer=FRAMES_PER_BUFFER,
        channels=CHANNELS,
        format=FORMAT,
        rate=SAMPLE_RATE,
    )
    print("Microphone stream opened successfully.")
    print("Speak into your microphone. Press Ctrl+C to stop.")
except Exception as e:
    print(f"Error opening microphone stream: {e}")
    if audio:
        audio.terminate()
    return  # Exit if microphone cannot be opened

# Create WebSocketApp
ws_app = websocket.WebSocketApp(
    API_ENDPOINT,
    header={"Authorization": YOUR_API_KEY},
    on_open=on_open,
    on_message=on_message,
    on_error=on_error,
    on_close=on_close,
)
# Run WebSocketApp in a separate thread to allow main thread to catch KeyboardInterrupt
ws_thread = threading.Thread(target=ws_app.run_forever)
ws_thread.daemon = True
ws_thread.start()

Websocket Error HandlingCapture and log any errors emitted by the WebSocket connection to streamline troubleshooting and maintain smooth operation.

def on_error(ws, error):
    """Called when a WebSocket error occurs."""
    print(f'\nWebSocket Error: {error}')
    # Attempt to signal stop on error
    stop_event.set()

Open WebSocket

Gladia
AssemblyAI

# Connect to Websocket
async with connect(session_data["url"]) as websocket:
    print("\n################ Begin session ################\n")
    
    # Create tasks for sending audio and receiving transcripts
    send_audio_task = asyncio.create_task(send_audio(websocket))
    receive_transcript_task = asyncio.create_task(receive_transcript(websocket))
    
    await asyncio.wait([send_audio_task, receive_transcript_task])

def on_open(ws):
    """Called when the WebSocket connection is established."""
    print("WebSocket connection opened.")
    print(f"Connected to: {API_ENDPOINT}")

    # Start sending audio data in a separate thread
    def stream_audio():
        global stream
        print("Starting audio streaming...")
        while not stop_event.is_set():
            try:
                audio_data = stream.read(FRAMES_PER_BUFFER, exception_on_overflow=False)
                # Send audio data as binary message
                ws.send(audio_data, websocket.ABNF.OPCODE_BINARY)
            except Exception as e:
                print(f"Error streaming audio: {e}")
                # If stream read fails, likely means it's closed, stop the loop
                break
        print("Audio streaming stopped.")

    global audio_thread
    audio_thread = threading.Thread(target=stream_audio)
    audio_thread.daemon = (
        True  # Allow main thread to exit even if this thread is running
    )
    audio_thread.start()

Receive Messsages from WebSocket

Gladia
AssemblyAI

async def receive_transcript(websocket):
    # Process incoming messages
    async for message in websocket:
        content = json.loads(message)
        
        # Print transcripts
        if content["type"] == "transcript" and content["data"]["is_final"]:
            text = content["data"]["utterance"]["text"].strip()
            print(f"Final: {text}")
            
        # Print final results
        if content["type"] == "post_final_transcript":
            print("\n################ End of session ################\n")
            print(json.dumps(content, indent=2, ensure_ascii=False))

def on_message(ws, message):
    try:
        data = json.loads(message)
        msg_type = data.get('type')

        if msg_type == "Begin":
            session_id = data.get('id')
            expires_at = data.get('expires_at')
            print(f"\nSession began: ID={session_id}, ExpiresAt={datetime.fromtimestamp(expires_at)}")
        elif msg_type == "Turn":
            transcript = data.get('transcript', '')
            end_of_turn = data.get('end_of_turn', False)

            # Print final end-of-turn transcript
            if end_of_turn:
                print('\r' + ' ' * 80 + '\r', end='')
                print(transcript)
            else:
                print(f"\r{transcript}", end='')
        elif msg_type == "Termination":
            audio_duration = data.get('audio_duration_seconds', 0)
            session_duration = data.get('session_duration_seconds', 0)
            print(f"\nSession Terminated: Audio Duration={audio_duration}s, Session Duration={session_duration}s")
    except json.JSONDecodeError as e:
        print(f"Error decoding message: {e}")
    except Exception as e:
        print(f"Error handling message: {e}")

Helpful information about AssemblyAI’s message payloads:

Clear Message Types – Instead of checking is_final, you’ll receive explicit "Begin", "Turn", and "Termination" events, making your logic simpler and more readable.
Session Metadata Up-Front – The first "Begin" message delivers a session_id and expiry timestamp so you can immediately log or surface these for tracing or billing.
End-of-Turn Detection – Each "Turn" object includes an end_of_turn boolean. When end_of_turn is true, the transcript is a final, formatted result. When false, it is a partial transcript. Universal-3.5 Pro always returns formatted transcripts with smart punctuation & casing built in.

Close the WebSocket

Gladia
AssemblyAI

async def stop_recording(websocket):
    print(">>>>> Ending the recording…")
    await websocket.send(json.dumps({"type": "stop_recording"}))
    await asyncio.sleep(0)

def on_close(ws, close_status_code, close_msg):
    """Called when the WebSocket connection is closed."""
    print(f"\nWebSocket Disconnected: Status={close_status_code}, Msg={close_msg}")
    # Ensure audio resources are released
    global stream, audio
    stop_event.set()  # Signal audio thread just in case it's still running

    if stream:
        if stream.is_active():
            stream.stop_stream()
        stream.close()
        stream = None
    if audio:
        audio.terminate()
        audio = None
    # Try to join the audio thread to ensure clean exit
    if audio_thread and audio_thread.is_alive():
        audio_thread.join(timeout=1.0)

Helpful information about AssemblyAI’s WebSocket Closure:

Connection Diagnostics - If the socket closes unexpectedly, AssemblyAI supplies both a status code and a reason message (close_status_code, close_msg), so you know immediately whether the server timed out, refused authentication, or encountered a different error.

Session Shutdown

Gladia
AssemblyAI

async with connect(session_data["url"]) as websocket:
    ...
    # Handle Ctrl+C to stop recording
    loop = asyncio.get_running_loop()
    loop.add_signal_handler(
        signal.SIGINT,
        lambda: loop.create_task(stop_recording(websocket)),
    )

try:
    # Keep main thread alive until interrupted
    while ws_thread.is_alive():
        time.sleep(0.1)
except KeyboardInterrupt:
    print("\nCtrl+C received. Stopping...")
    stop_event.set()  # Signal audio thread to stop

    # Send termination message to the server
    if ws_app and ws_app.sock and ws_app.sock.connected:
        try:
            terminate_message = {"type": "Terminate"}
            print(f"Sending termination message: {json.dumps(terminate_message)}")
            ws_app.send(json.dumps(terminate_message))
            # Give a moment for messages to process before forceful close
            time.sleep(5)
        except Exception as e:
            print(f"Error sending termination message: {e}")

    # Close the WebSocket connection (will trigger on_close)
    if ws_app:
        ws_app.close()

    # Wait for WebSocket thread to finish
    ws_thread.join(timeout=2.0)

except Exception as e:
    print(f"\nAn unexpected error occurred: {e}")
    stop_event.set()
    if ws_app:
        ws_app.close()
    ws_thread.join(timeout=2.0)

finally:
    # Final cleanup (already handled in on_close, but good as a fallback)
    if stream and stream.is_active():
        stream.stop_stream()
    if stream:
        stream.close()
    if audio:
        audio.terminate()
    print("Cleanup complete. Exiting.")

Helpful information to know about AssemblyAI’s shutdown:

JSON Payload Difference - When closing the stream with AssemblyAI, your JSON payload will be { "type": "Terminate" } instead of { "type": "stop_recording" }.
No Metadata Race Condition - AssemblyAI provides session info at “Begin” and doesn’t append extra data at shutdown, making the exit faster and less error-prone.

Resources

For additional information about using AssemblyAI’s Streaming Speech-To-Text API you can also refer to:

Getting started

Features

API reference

Advanced

Integrations

Guides

Streaming Migration Guide: Gladia to AssemblyAI

Get Started

Side-By-Side Code Comparison

Authentication

Connection Parameters & Microphone Setup

Open Microphone Stream & Create WebSocket

Open WebSocket

Receive Messsages from WebSocket

Close the WebSocket

Session Shutdown

Resources

​Get Started

​Side-By-Side Code Comparison

​Authentication

​Connection Parameters & Microphone Setup

​Open Microphone Stream & Create WebSocket

​Open WebSocket

​Receive Messsages from WebSocket

​Close the WebSocket

​Session Shutdown

​Resources

Get Started

Side-By-Side Code Comparison

Authentication

Connection Parameters & Microphone Setup

Open Microphone Stream & Create WebSocket

Open WebSocket

Receive Messsages from WebSocket

Close the WebSocket

Session Shutdown

Resources