Added mic_test and mic_query scripts
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Ali
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scripts/mic_query.py
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scripts/mic_query.py
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# /// script
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# requires-python = ">=3.12"
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# dependencies = [
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# "msgpack",
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# "numpy",
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# "sounddevice",
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# "websockets",
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# ]
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# ///
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import argparse
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import asyncio
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import json
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import msgpack
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import queue
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import struct
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import time
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import threading
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import numpy as np
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import sounddevice as sd
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import websockets
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# Desired audio properties
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TARGET_SAMPLE_RATE = 24000
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TARGET_CHANNELS = 1 # Mono
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HEADERS = {"kyutai-api-key": "open_token"}
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all_text = []
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transcript = []
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finished = False
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audio_queue = queue.Queue()
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recording = True
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def audio_callback(indata, frames, time, status):
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"""Callback function for sounddevice to capture audio."""
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if status:
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print(f"Audio callback status: {status}")
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# Convert to float32 and flatten to mono
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audio_data = indata[:, 0].astype(np.float32)
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audio_queue.put(audio_data.copy())
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async def receive_messages(websocket):
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"""Receive and process messages from the WebSocket server."""
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global all_text
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global transcript
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global finished
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try:
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async for message in websocket:
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data = msgpack.unpackb(message, raw=False)
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if data["type"] == "Step":
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continue
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print("received:", data)
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if data["type"] == "Word":
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all_text.append(data["text"])
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transcript.append({
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"speaker": "SPEAKER_00",
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"text": data["text"],
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"timestamp": [data["start_time"], data["start_time"]],
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})
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# Print words in real-time
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print(f"Word: {data['text']}")
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if data["type"] == "EndWord":
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if len(transcript) > 0:
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transcript[-1]["timestamp"][1] = data["stop_time"]
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if data["type"] == "Marker":
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print("Received marker, stopping stream.")
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break
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except websockets.ConnectionClosed:
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print("Connection closed while receiving messages.")
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finished = True
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async def send_messages(websocket, rtf: float):
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"""Send audio data from microphone to WebSocket server."""
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global finished
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global recording
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try:
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# Start with a second of silence
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chunk = {"type": "Audio", "pcm": [0.0] * 24000}
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msg = msgpack.packb(chunk, use_bin_type=True, use_single_float=True)
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await websocket.send(msg)
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chunk_size = 1920 # Send data in chunks (80ms at 24kHz)
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while recording and not finished:
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try:
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# Get audio data from queue with timeout
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audio_data = audio_queue.get(timeout=0.1)
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# Process audio in chunks
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for i in range(0, len(audio_data), chunk_size):
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if not recording:
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break
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chunk_data = audio_data[i:i + chunk_size]
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# Pad with zeros if chunk is smaller than expected
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if len(chunk_data) < chunk_size:
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chunk_data = np.pad(chunk_data, (0, chunk_size - len(chunk_data)), 'constant')
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chunk = {"type": "Audio", "pcm": [float(x) for x in chunk_data]}
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msg = msgpack.packb(chunk, use_bin_type=True, use_single_float=True)
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await websocket.send(msg)
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# Small delay to avoid overwhelming the server
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await asyncio.sleep(0.001)
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except queue.Empty:
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# No audio data available, continue
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continue
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# Send final silence and marker
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chunk = {"type": "Audio", "pcm": [0.0] * 1920 * 5}
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msg = msgpack.packb(chunk, use_bin_type=True, use_single_float=True)
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await websocket.send(msg)
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msg = msgpack.packb({"type": "Marker", "id": 0}, use_bin_type=True, use_single_float=True)
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await websocket.send(msg)
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# Send additional silence chunks
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for _ in range(35):
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chunk = {"type": "Audio", "pcm": [0.0] * 1920}
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msg = msgpack.packb(chunk, use_bin_type=True, use_single_float=True)
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await websocket.send(msg)
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# Keep connection alive
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while not finished:
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await asyncio.sleep(1.0)
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await websocket.ping()
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except websockets.ConnectionClosed:
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print("Connection closed while sending messages.")
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def start_recording():
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"""Start recording audio from microphone."""
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global recording
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print("Starting microphone recording...")
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print("Press Ctrl+C to stop recording")
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# Start audio stream
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with sd.InputStream(
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samplerate=TARGET_SAMPLE_RATE,
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channels=TARGET_CHANNELS,
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dtype='float32',
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callback=audio_callback,
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blocksize=1920 # 80ms blocks
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):
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try:
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while recording:
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time.sleep(0.1)
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except KeyboardInterrupt:
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print("\nStopping recording...")
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recording = False
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async def stream_audio(url: str, rtf: float):
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"""Stream audio data to a WebSocket server."""
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global recording
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# Start recording in a separate thread
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recording_thread = threading.Thread(target=start_recording)
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recording_thread.daemon = True
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recording_thread.start()
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try:
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async with websockets.connect(url, additional_headers=HEADERS) as websocket:
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send_task = asyncio.create_task(send_messages(websocket, rtf))
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receive_task = asyncio.create_task(receive_messages(websocket))
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await asyncio.gather(send_task, receive_task)
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except KeyboardInterrupt:
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print("\nInterrupted by user")
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recording = False
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finally:
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recording = False
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print("Exiting...")
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if __name__ == "__main__":
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parser = argparse.ArgumentParser(description="Real-time microphone transcription")
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parser.add_argument("--transcript", help="Output transcript file (JSON)")
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parser.add_argument(
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"--url",
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help="The URL of the server to which to send the audio",
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default="ws://5.9.97.57:8080",
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)
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parser.add_argument("--rtf", type=float, default=1.01, help="Real-time factor")
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parser.add_argument("--list-devices", action="store_true", help="List available audio devices")
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parser.add_argument("--device", type=int, help="Input device ID (use --list-devices to see options)")
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args = parser.parse_args()
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if args.list_devices:
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print("Available audio devices:")
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print(sd.query_devices())
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exit(0)
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if args.device is not None:
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sd.default.device[0] = args.device # Set input device
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url = f"{args.url}/api/asr-streaming"
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try:
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asyncio.run(stream_audio(url, args.rtf))
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print("\nFinal transcript:")
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print(" ".join(all_text))
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if args.transcript is not None:
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with open(args.transcript, "w") as fobj:
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json.dump({"transcript": transcript}, fobj, indent=4)
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print(f"Transcript saved to {args.transcript}")
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except KeyboardInterrupt:
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print("\nProgram interrupted by user")
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except Exception as e:
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print(f"Error: {e}")
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finally:
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recording = False
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9
scripts/mic_test.py
Normal file
9
scripts/mic_test.py
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@ -0,0 +1,9 @@
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import sounddevice as sd
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import numpy as np
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print("Recording 3 seconds...")
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audio = sd.rec(3 * 24000, samplerate=24000, channels=1, dtype='float32')
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sd.wait()
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print("Playing back...")
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sd.play(audio, samplerate=24000)
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sd.wait()
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