Decided on Parakeet ONNX Runtime. Works pretty great. Realtime voice chat possible now. UX lacking.
This commit is contained in:
235
stt-parakeet/client/mic_stream.py
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235
stt-parakeet/client/mic_stream.py
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"""
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Microphone streaming client for ASR WebSocket server
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"""
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import asyncio
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import websockets
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import sounddevice as sd
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import numpy as np
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import json
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import logging
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import queue
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from typing import Optional
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logging.basicConfig(
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level=logging.INFO,
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format='%(asctime)s - %(name)s - %(levelname)s - %(message)s'
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)
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logger = logging.getLogger(__name__)
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class MicrophoneStreamClient:
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"""
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Client for streaming microphone audio to ASR WebSocket server.
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"""
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def __init__(
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self,
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server_url: str = "ws://localhost:8766",
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sample_rate: int = 16000,
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channels: int = 1,
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chunk_duration: float = 0.1, # seconds
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device: Optional[int] = None,
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):
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"""
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Initialize microphone streaming client.
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Args:
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server_url: WebSocket server URL
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sample_rate: Audio sample rate (16000 Hz recommended)
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channels: Number of audio channels (1 for mono)
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chunk_duration: Duration of each audio chunk in seconds
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device: Optional audio input device index
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"""
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self.server_url = server_url
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self.sample_rate = sample_rate
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self.channels = channels
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self.chunk_duration = chunk_duration
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self.chunk_samples = int(sample_rate * chunk_duration)
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self.device = device
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self.audio_queue = queue.Queue()
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self.is_recording = False
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self.websocket = None
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logger.info(f"Microphone client initialized")
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logger.info(f"Server URL: {server_url}")
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logger.info(f"Sample rate: {sample_rate} Hz")
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logger.info(f"Chunk duration: {chunk_duration}s ({self.chunk_samples} samples)")
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def audio_callback(self, indata, frames, time_info, status):
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"""
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Callback for sounddevice stream.
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Args:
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indata: Input audio data
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frames: Number of frames
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time_info: Timing information
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status: Status flags
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"""
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if status:
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logger.warning(f"Audio callback status: {status}")
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# Convert to int16 and put in queue
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audio_data = (indata[:, 0] * 32767).astype(np.int16)
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self.audio_queue.put(audio_data.tobytes())
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async def send_audio(self):
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"""
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Coroutine to send audio from queue to WebSocket.
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"""
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while self.is_recording:
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try:
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# Get audio data from queue (non-blocking)
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audio_bytes = self.audio_queue.get_nowait()
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if self.websocket:
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await self.websocket.send(audio_bytes)
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except queue.Empty:
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# No audio data available, wait a bit
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await asyncio.sleep(0.01)
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except Exception as e:
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logger.error(f"Error sending audio: {e}")
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break
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async def receive_transcripts(self):
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"""
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Coroutine to receive transcripts from WebSocket.
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"""
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while self.is_recording:
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try:
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if self.websocket:
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message = await asyncio.wait_for(
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self.websocket.recv(),
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timeout=0.1
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)
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try:
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data = json.loads(message)
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if data.get("type") == "transcript":
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text = data.get("text", "")
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is_final = data.get("is_final", False)
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if is_final:
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logger.info(f"[FINAL] {text}")
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else:
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logger.info(f"[PARTIAL] {text}")
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elif data.get("type") == "info":
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logger.info(f"Server: {data.get('message')}")
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elif data.get("type") == "error":
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logger.error(f"Server error: {data.get('message')}")
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except json.JSONDecodeError:
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logger.warning(f"Invalid JSON response: {message}")
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except asyncio.TimeoutError:
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continue
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except Exception as e:
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logger.error(f"Error receiving transcript: {e}")
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break
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async def stream_audio(self):
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"""
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Main coroutine to stream audio to server.
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"""
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try:
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async with websockets.connect(self.server_url) as websocket:
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self.websocket = websocket
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logger.info(f"Connected to server: {self.server_url}")
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self.is_recording = True
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# Start audio stream
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with sd.InputStream(
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samplerate=self.sample_rate,
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channels=self.channels,
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dtype=np.float32,
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blocksize=self.chunk_samples,
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device=self.device,
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callback=self.audio_callback,
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):
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logger.info("Recording started. Press Ctrl+C to stop.")
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# Run send and receive coroutines concurrently
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await asyncio.gather(
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self.send_audio(),
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self.receive_transcripts(),
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)
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except websockets.exceptions.WebSocketException as e:
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logger.error(f"WebSocket error: {e}")
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except KeyboardInterrupt:
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logger.info("Stopped by user")
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finally:
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self.is_recording = False
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# Send final command
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if self.websocket:
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try:
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await self.websocket.send(json.dumps({"type": "final"}))
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await asyncio.sleep(0.5) # Wait for final response
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except:
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pass
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self.websocket = None
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logger.info("Disconnected from server")
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def run(self):
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"""
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Run the client (blocking).
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"""
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try:
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asyncio.run(self.stream_audio())
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except KeyboardInterrupt:
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logger.info("Client stopped by user")
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def list_audio_devices():
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"""
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List available audio input devices.
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"""
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print("\nAvailable audio input devices:")
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print("-" * 80)
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devices = sd.query_devices()
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for i, device in enumerate(devices):
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if device['max_input_channels'] > 0:
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print(f"[{i}] {device['name']}")
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print(f" Channels: {device['max_input_channels']}")
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print(f" Sample rate: {device['default_samplerate']} Hz")
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print("-" * 80)
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def main():
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"""
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Main entry point for the microphone client.
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"""
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import argparse
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parser = argparse.ArgumentParser(description="Microphone Streaming Client")
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parser.add_argument("--url", default="ws://localhost:8766", help="WebSocket server URL")
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parser.add_argument("--sample-rate", type=int, default=16000, help="Audio sample rate")
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parser.add_argument("--device", type=int, default=None, help="Audio input device index")
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parser.add_argument("--list-devices", action="store_true", help="List audio devices and exit")
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parser.add_argument("--chunk-duration", type=float, default=0.1, help="Audio chunk duration (seconds)")
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args = parser.parse_args()
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if args.list_devices:
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list_audio_devices()
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return
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client = MicrophoneStreamClient(
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server_url=args.url,
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sample_rate=args.sample_rate,
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device=args.device,
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chunk_duration=args.chunk_duration,
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)
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client.run()
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if __name__ == "__main__":
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main()
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