import mlx_whisper
import numpy as np
import sounddevice as sd
import queue
import sys
import torch
from silero_vad import load_silero_vad, get_speech_timestamps

# Parameters
MODEL_PATH = "mlx-community/whisper-small.en-mlx" # MLX optimized small model
CHANNELS = 1
SAMPLERATE = 16000
BLOCK_SIZE = 512 # Silero VAD prefers 512, 1024, or 1536
VAD_THRESHOLD = 0.5 
BUFFER_LIMIT = SAMPLERATE * 30 
MIN_SILENCE_DURATION_MS = 500 

audio_queue = queue.Queue()

def callback(indata, frames, time, status):
    if status:
        print(status, file=sys.stderr)
    audio_queue.put(indata.copy())

def main():
    print(f"Loading MLX-optimized Whisper model '{MODEL_PATH}'...")
    # mlx-whisper uses the same model names or Hugging Face paths
    
    print("Loading Silero VAD model...")
    vad_model = load_silero_vad()
    
    print("Models loaded.")

    print("\nStarting live transcription (MLX + VAD)... (Press Ctrl+C to stop)")
    
    audio_buffer = []
    speech_started = False

    try:
        with sd.InputStream(samplerate=SAMPLERATE, channels=CHANNELS, callback=callback, blocksize=BLOCK_SIZE):
            while True:
                while not audio_queue.empty():
                    data = audio_queue.get()
                    audio_buffer.append(data.flatten())

                if len(audio_buffer) > 0:
                    current_audio = np.concatenate(audio_buffer)
                    audio_tensor = torch.from_numpy(current_audio)
                    
                    speech_timestamps = get_speech_timestamps(
                        audio_tensor, 
                        vad_model, 
                        sampling_rate=SAMPLERATE,
                        threshold=VAD_THRESHOLD,
                        min_silence_duration_ms=MIN_SILENCE_DURATION_MS
                    )

                    if len(speech_timestamps) > 0:
                        speech_started = True
                        last_end = speech_timestamps[-1]['end']
                        buffer_len_samples = len(current_audio)
                        
                        if (buffer_len_samples - last_end) > (SAMPLERATE * MIN_SILENCE_DURATION_MS / 1000) or buffer_len_samples > BUFFER_LIMIT:
                            
                            # Transcribe with MLX
                            result = mlx_whisper.transcribe(current_audio, path_or_hf_repo=MODEL_PATH)
                            text = result['text'].strip()
                            
                            if text:
                                print(f"Transcription: {text}")
                            
                            audio_buffer = []
                            speech_started = False
                    
                    elif not speech_started and len(current_audio) > SAMPLERATE * 2:
                        audio_buffer = []

    except KeyboardInterrupt:
        print("\nStopped by user.")
    except Exception as e:
        print(f"\nError: {e}")

if __name__ == "__main__":
    main()