"""
Utility Functions
Helper functions for audio processing, visualization, and optimization
"""

import numpy as np
import librosa
import matplotlib.pyplot as plt
import soundfile as sf
from pathlib import Path
from typing import Union, Tuple, Optional
import torch
import warnings
warnings.filterwarnings('ignore')


def normalize_audio(
    audio: np.ndarray,
    target_level: float = -20.0
) -> np.ndarray:
    """
    Normalize audio to target dB level
    
    Args:
        audio: Audio array
        target_level: Target level in dB (default: -20 dB)
    
    Returns:
        Normalized audio
    """
    # Calculate current RMS level
    rms = np.sqrt(np.mean(audio ** 2))
    current_level = 20 * np.log10(rms + 1e-8)
    
    # Calculate gain needed
    gain_db = target_level - current_level
    gain_linear = 10 ** (gain_db / 20)
    
    # Apply gain
    normalized = audio * gain_linear
    
    # Prevent clipping
    normalized = np.clip(normalized, -1.0, 1.0)
    
    return normalized


def trim_silence(
    audio: np.ndarray,
    sr: int,
    top_db: int = 30,
    frame_length: int = 2048,
    hop_length: int = 512
) -> np.ndarray:
    """
    Trim silence from beginning and end of audio
    
    Args:
        audio: Audio array
        sr: Sample rate
        top_db: Threshold in dB below reference to consider as silence
        frame_length: Frame length for analysis
        hop_length: Hop length for analysis
    
    Returns:
        Trimmed audio
    """
    trimmed, _ = librosa.effects.trim(
        audio,
        top_db=top_db,
        frame_length=frame_length,
        hop_length=hop_length
    )
    return trimmed


def split_audio_by_silence(
    audio: np.ndarray,
    sr: int,
    min_silence_len: float = 0.5,
    silence_thresh: int = -40,
    keep_silence: float = 0.1
) -> list:
    """
    Split audio into segments based on silence
    
    Args:
        audio: Audio array
        sr: Sample rate
        min_silence_len: Minimum silence length in seconds
        silence_thresh: Silence threshold in dB
        keep_silence: Amount of silence to keep at edges (seconds)
    
    Returns:
        List of audio segments
    """
    # Convert parameters to samples
    min_silence_samples = int(min_silence_len * sr)
    keep_silence_samples = int(keep_silence * sr)
    
    # Compute energy
    energy = librosa.feature.rms(y=audio, frame_length=2048, hop_length=512)[0]
    energy_db = librosa.amplitude_to_db(energy, ref=np.max)
    
    # Find silent regions
    silent = energy_db < silence_thresh
    
    # Find segment boundaries
    segments = []
    start = 0
    in_silence = False
    silence_start = 0
    
    for i, is_silent in enumerate(silent):
        if is_silent and not in_silence:
            # Start of silence
            silence_start = i
            in_silence = True
        elif not is_silent and in_silence:
            # End of silence
            silence_len = i - silence_start
            if silence_len >= min_silence_samples // 512:  # Account for hop length
                # Split here
                end = max(0, silence_start * 512 - keep_silence_samples)
                if end > start:
                    segments.append(audio[start:end])
                start = min(len(audio), i * 512 + keep_silence_samples)
            in_silence = False
    
    # Add final segment
    if start < len(audio):
        segments.append(audio[start:])
    
    return segments if segments else [audio]


def resample_audio(
    audio: np.ndarray,
    orig_sr: int,
    target_sr: int
) -> np.ndarray:
    """
    Resample audio to target sample rate
    
    Args:
        audio: Audio array
        orig_sr: Original sample rate
        target_sr: Target sample rate
    
    Returns:
        Resampled audio
    """
    if orig_sr == target_sr:
        return audio
    
    resampled = librosa.resample(audio, orig_sr=orig_sr, target_sr=target_sr)
    return resampled


def plot_waveform(
    audio: np.ndarray,
    sr: int,
    title: str = "Waveform",
    figsize: Tuple[int, int] = (12, 4)
) -> plt.Figure:
    """
    Plot audio waveform
    
    Args:
        audio: Audio array
        sr: Sample rate
        title: Plot title
        figsize: Figure size
    
    Returns:
        Matplotlib figure
    """
    fig, ax = plt.subplots(figsize=figsize)
    
    time = np.arange(len(audio)) / sr
    ax.plot(time, audio, linewidth=0.5)
    ax.set_xlabel("Time (s)")
    ax.set_ylabel("Amplitude")
    ax.set_title(title)
    ax.grid(True, alpha=0.3)
    
    plt.tight_layout()
    return fig


def plot_spectrogram(
    audio: np.ndarray,
    sr: int,
    title: str = "Spectrogram",
    figsize: Tuple[int, int] = (12, 6)
) -> plt.Figure:
    """
    Plot audio spectrogram
    
    Args:
        audio: Audio array
        sr: Sample rate
        title: Plot title
        figsize: Figure size
    
    Returns:
        Matplotlib figure
    """
    fig, ax = plt.subplots(figsize=figsize)
    
    # Compute spectrogram
    D = librosa.amplitude_to_db(
        np.abs(librosa.stft(audio)),
        ref=np.max
    )
    
    # Plot
    img = librosa.display.specshow(
        D,
        sr=sr,
        x_axis='time',
        y_axis='hz',
        ax=ax,
        cmap='viridis'
    )
    
    ax.set_title(title)
    fig.colorbar(img, ax=ax, format='%+2.0f dB')
    
    plt.tight_layout()
    return fig


def plot_mel_spectrogram(
    audio: np.ndarray,
    sr: int,
    n_mels: int = 80,
    title: str = "Mel Spectrogram",
    figsize: Tuple[int, int] = (12, 6)
) -> plt.Figure:
    """
    Plot mel spectrogram
    
    Args:
        audio: Audio array
        sr: Sample rate
        n_mels: Number of mel bands
        title: Plot title
        figsize: Figure size
    
    Returns:
        Matplotlib figure
    """
    fig, ax = plt.subplots(figsize=figsize)
    
    # Compute mel spectrogram
    mel_spec = librosa.feature.melspectrogram(
        y=audio,
        sr=sr,
        n_mels=n_mels
    )
    mel_spec_db = librosa.amplitude_to_db(mel_spec, ref=np.max)
    
    # Plot
    img = librosa.display.specshow(
        mel_spec_db,
        sr=sr,
        x_axis='time',
        y_axis='mel',
        ax=ax,
        cmap='viridis'
    )
    
    ax.set_title(title)
    fig.colorbar(img, ax=ax, format='%+2.0f dB')
    
    plt.tight_layout()
    return fig


def compute_audio_metrics(
    audio: np.ndarray,
    sr: int
) -> dict:
    """
    Compute comprehensive audio metrics
    
    Args:
        audio: Audio array
        sr: Sample rate
    
    Returns:
        Dict of audio metrics
    """
    metrics = {}
    
    # Duration
    metrics["duration_seconds"] = len(audio) / sr
    
    # RMS Energy
    rms = np.sqrt(np.mean(audio ** 2))
    metrics["rms_energy"] = float(rms)
    metrics["rms_db"] = float(20 * np.log10(rms + 1e-8))
    
    # Peak amplitude
    metrics["peak_amplitude"] = float(np.max(np.abs(audio)))
    
    # Dynamic range
    metrics["dynamic_range_db"] = float(
        20 * np.log10((np.max(np.abs(audio)) + 1e-8) / (np.mean(np.abs(audio)) + 1e-8))
    )
    
    # Zero crossing rate
    zcr = librosa.feature.zero_crossing_rate(audio)
    metrics["zero_crossing_rate"] = float(np.mean(zcr))
    
    # Spectral features
    spectral_centroid = librosa.feature.spectral_centroid(y=audio, sr=sr)
    metrics["spectral_centroid_hz"] = float(np.mean(spectral_centroid))
    
    spectral_bandwidth = librosa.feature.spectral_bandwidth(y=audio, sr=sr)
    metrics["spectral_bandwidth_hz"] = float(np.mean(spectral_bandwidth))
    
    spectral_rolloff = librosa.feature.spectral_rolloff(y=audio, sr=sr)
    metrics["spectral_rolloff_hz"] = float(np.mean(spectral_rolloff))
    
    # Clipping detection
    clipping_ratio = np.sum(np.abs(audio) > 0.99) / len(audio)
    metrics["clipping_ratio"] = float(clipping_ratio)
    metrics["is_clipped"] = clipping_ratio > 0.01
    
    return metrics


def get_gpu_memory_info() -> dict:
    """
    Get GPU memory information
    
    Returns:
        Dict with GPU memory stats
    """
    if not torch.cuda.is_available():
        return {"available": False}
    
    info = {
        "available": True,
        "device_name": torch.cuda.get_device_name(0),
        "total_gb": torch.cuda.get_device_properties(0).total_memory / 1e9,
        "allocated_gb": torch.cuda.memory_allocated(0) / 1e9,
        "reserved_gb": torch.cuda.memory_reserved(0) / 1e9,
        "free_gb": (torch.cuda.get_device_properties(0).total_memory - torch.cuda.memory_allocated(0)) / 1e9
    }
    
    return info


def optimize_for_inference(model: torch.nn.Module) -> torch.nn.Module:
    """
    Optimize model for inference
    
    Args:
        model: PyTorch model
    
    Returns:
        Optimized model
    """
    model.eval()
    
    # Disable gradient computation
    for param in model.parameters():
        param.requires_grad = False
    
    # Try to compile (PyTorch 2.0+)
    try:
        if hasattr(torch, 'compile'):
            model = torch.compile(model, mode='reduce-overhead')
            print("✓ Model compiled with torch.compile")
    except Exception as e:
        print(f"⚠️  Could not compile model: {e}")
    
    return model


def save_audio_with_metadata(
    audio: np.ndarray,
    output_path: Union[str, Path],
    sr: int,
    metadata: Optional[dict] = None
):
    """
    Save audio with metadata
    
    Args:
        audio: Audio array
        output_path: Output file path
        sr: Sample rate
        metadata: Optional metadata dict
    """
    output_path = Path(output_path)
    output_path.parent.mkdir(parents=True, exist_ok=True)
    
    # Save audio
    sf.write(str(output_path), audio, sr)
    
    # Save metadata if provided
    if metadata:
        metadata_path = output_path.with_suffix('.json')
        import json
        with open(metadata_path, 'w') as f:
            json.dump(metadata, f, indent=2)


def benchmark_inference(
    func,
    *args,
    n_runs: int = 10,
    warmup: int = 2,
    **kwargs
) -> dict:
    """
    Benchmark inference speed
    
    Args:
        func: Function to benchmark
        *args: Function arguments
        n_runs: Number of runs
        warmup: Number of warmup runs
        **kwargs: Function keyword arguments
    
    Returns:
        Dict with benchmark results
    """
    import time
    
    # Warmup
    for _ in range(warmup):
        func(*args, **kwargs)
    
    # Benchmark
    times = []
    for _ in range(n_runs):
        if torch.cuda.is_available():
            torch.cuda.synchronize()
        
        start = time.time()
        func(*args, **kwargs)
        
        if torch.cuda.is_available():
            torch.cuda.synchronize()
        
        end = time.time()
        times.append(end - start)
    
    results = {
        "mean_time": np.mean(times),
        "std_time": np.std(times),
        "min_time": np.min(times),
        "max_time": np.max(times),
        "n_runs": n_runs
    }
    
    return results


def main():
    """Demo utility functions"""
    print("=" * 60)
    print("Utility Functions Demo")
    print("=" * 60)
    
    print("\n📦 Available utilities:")
    print("   - Audio normalization")
    print("   - Silence trimming and splitting")
    print("   - Resampling")
    print("   - Waveform and spectrogram plotting")
    print("   - Audio metrics computation")
    print("   - GPU memory monitoring")
    print("   - Inference optimization")
    print("   - Benchmarking")
    
    # Show GPU info
    gpu_info = get_gpu_memory_info()
    if gpu_info["available"]:
        print(f"\n🎮 GPU Information:")
        print(f"   Device: {gpu_info['device_name']}")
        print(f"   Total: {gpu_info['total_gb']:.2f} GB")
        print(f"   Free: {gpu_info['free_gb']:.2f} GB")
    else:
        print("\n⚠️  No GPU available")
    
    print("\n" + "=" * 60)


if __name__ == "__main__":
    main()