import argparse
import os
import random
import sys
from typing import Optional, Tuple, Union

import mlx.core as mx
import mlx.nn as nn
import numpy as np
import soundfile as sf
from numpy.lib.stride_tricks import sliding_window_view
from scipy.signal import resample

from .audio_player import AudioPlayer
from .utils import load_model


def load_audio(
    audio_path: str,
    sample_rate: int = 24000,
    length: int = None,
    volume_normalize: bool = False,
    segment_duration: int = None,
) -> mx.array:
    samples, orig_sample_rate = sf.read(audio_path)
    shape = samples.shape

    # Collapse multi channel as mono
    if len(shape) > 1:
        samples = samples.sum(axis=1)
        # Divide summed samples by channel count.
        samples = samples / shape[1]
    if sample_rate != orig_sample_rate:
        print(f"Resampling from {orig_sample_rate} to {sample_rate}")
        duration = samples.shape[0] / orig_sample_rate
        num_samples = int(duration * sample_rate)
        samples = resample(samples, num_samples)

    if segment_duration is not None:
        seg_length = int(sample_rate * segment_duration)
        samples = random_select_audio_segment(samples, seg_length)

    # Audio volume normalize
    if volume_normalize:
        samples = audio_volume_normalize(samples)

    if length is not None:
        assert abs(samples.shape[0] - length) < 1000
        if samples.shape[0] > length:
            samples = samples[:length]
        else:
            samples = np.pad(samples, (0, int(length - samples.shape[0])))

    audio = mx.array(samples, dtype=mx.float32)

    return audio


def audio_volume_normalize(audio: np.ndarray, coeff: float = 0.2) -> np.ndarray:
    """
    Normalize the volume of an audio signal.

    Parameters:
        audio (numpy array): Input audio signal array.
        coeff (float): Target coefficient for normalization, default is 0.2.

    Returns:
        numpy array: The volume-normalized audio signal.
    """
    # Sort the absolute values of the audio signal
    temp = np.sort(np.abs(audio))

    # If the maximum value is less than 0.1, scale the array to have a maximum of 0.1
    if temp[-1] < 0.1:
        scaling_factor = max(
            temp[-1], 1e-3
        )  # Prevent division by zero with a small constant
        audio = audio / scaling_factor * 0.1

    # Filter out values less than 0.01 from temp
    temp = temp[temp > 0.01]
    L = temp.shape[0]  # Length of the filtered array

    # If there are fewer than or equal to 10 significant values, return the audio without further processing
    if L <= 10:
        return audio

    # Compute the average of the top 10% to 1% of values in temp
    volume = np.mean(temp[int(0.9 * L) : int(0.99 * L)])

    # Normalize the audio to the target coefficient level, clamping the scale factor between 0.1 and 10
    audio = audio * np.clip(coeff / volume, a_min=0.1, a_max=10)

    # Ensure the maximum absolute value in the audio does not exceed 1
    max_value = np.max(np.abs(audio))
    if max_value > 1:
        audio = audio / max_value

    return audio


def random_select_audio_segment(audio: np.ndarray, length: int) -> np.ndarray:
    """get an audio segment given the length

    Args:
        audio (np.ndarray):
        length (int): audio length = sampling_rate * duration
    """
    if audio.shape[0] < length:
        audio = np.pad(audio, (0, int(length - audio.shape[0])))
    start_index = random.randint(0, audio.shape[0] - length)
    end_index = int(start_index + length)

    return audio[start_index:end_index]


def detect_speech_boundaries(
    wav: np.ndarray,
    sample_rate: int,
    window_duration: float = 0.1,
    energy_threshold: float = 0.01,
    margin_factor: int = 2,
) -> Tuple[int, int]:
    """Detect the start and end points of speech in an audio signal using RMS energy.

    Args:
        wav: Input audio signal array with values in [-1, 1]
        sample_rate: Audio sample rate in Hz
        window_duration: Duration of detection window in seconds
        energy_threshold: RMS energy threshold for speech detection
        margin_factor: Factor to determine extra margin around detected boundaries

    Returns:
        tuple: (start_index, end_index) of speech segment

    Raises:
        ValueError: If the audio contains only silence
    """
    window_size = int(window_duration * sample_rate)
    margin = margin_factor * window_size
    step_size = window_size // 10

    # Create sliding windows using stride tricks to avoid loops
    windows = sliding_window_view(wav, window_size)[::step_size]

    # Calculate RMS energy for each window
    energy = np.sqrt(np.mean(windows**2, axis=1))
    speech_mask = energy >= energy_threshold

    if not np.any(speech_mask):
        raise ValueError("No speech detected in audio (only silence)")

    start = max(0, np.argmax(speech_mask) * step_size - margin)
    end = min(
        len(wav),
        (len(speech_mask) - 1 - np.argmax(speech_mask[::-1])) * step_size + margin,
    )

    return start, end


def remove_silence_on_both_ends(
    wav: np.ndarray,
    sample_rate: int,
    window_duration: float = 0.1,
    volume_threshold: float = 0.01,
) -> np.ndarray:
    """Remove silence from both ends of an audio signal.

    Args:
        wav: Input audio signal array
        sample_rate: Audio sample rate in Hz
        window_duration: Duration of detection window in seconds
        volume_threshold: Amplitude threshold for silence detection

    Returns:
        np.ndarray: Audio signal with silence removed from both ends

    Raises:
        ValueError: If the audio contains only silence
    """
    start, end = detect_speech_boundaries(
        wav, sample_rate, window_duration, volume_threshold
    )
    return wav[start:end]


def hertz_to_mel(pitch: float) -> float:
    """
    Converts a frequency from the Hertz scale to the Mel scale.

    Parameters:
    - pitch: float or ndarray
        Frequency in Hertz.

    Returns:
    - mel: float or ndarray
        Frequency in Mel scale.
    """
    mel = 2595 * np.log10(1 + pitch / 700)
    return mel


def generate_audio(
    text: str,
    model: Optional[Union[str, nn.Module]] = "prince-canuma/Kokoro-82M",
    max_tokens: int = 1200,
    voice: str = "af_heart",
    speed: float = 1.0,
    lang_code: str = "a",
    ref_audio: Optional[str] = None,
    ref_text: Optional[str] = None,
    stt_model: Optional[Union[str, nn.Module]] = "mlx-community/whisper-large-v3-turbo",
    file_prefix: str = "audio",
    audio_format: str = "wav",
    join_audio: bool = False,
    play: bool = False,
    verbose: bool = True,
    temperature: float = 0.7,
    stream: bool = False,
    streaming_interval: float = 2.0,
    **kwargs,
) -> None:
    """
    Generates audio from text using a specified TTS model.

    Parameters:
    - text (str): The input text to be converted to speech.
    - model (str): The TTS model to use.
    - voice (str): The voice style to use.
    - temperature (float): The temperature for the model.
    - speed (float): Playback speed multiplier.
    - lang_code (str): The language code.
    - ref_audio (mx.array): Reference audio you would like to clone the voice from.
    - ref_text (str): Caption for reference audio.
    - stt_model_path (str): A mlx whisper model to use to transcribe.
    - file_prefix (str): The output file path without extension.
    - audio_format (str): Output audio format (e.g., "wav", "flac").
    - join_audio (bool): Whether to join multiple audio files into one.
    - play (bool): Whether to play the generated audio.
    - verbose (bool): Whether to print status messages.
    - model (object): A already loaded model.
    - stt_model (object): A already loaded stt model.
    Returns:
    - None: The function writes the generated audio to a file.
    """
    try:
        play = play or stream

        if model is None:
            raise ValueError("Model path or model instance must be provided.")

        if stt_model is None and (ref_audio and ref_text is None):
            raise ValueError(
                "STT model path or model instance must be provided when ref_text is given."
            )

        if isinstance(model, str):
            # Load model
            model = load_model(model_path=model)

        # Load reference audio for voice matching if specified
        if ref_audio:
            if not os.path.exists(ref_audio):
                raise FileNotFoundError(f"Reference audio file not found: {ref_audio}")

            normalize = False
            if hasattr(model, "model_type") and model.model_type() == "spark":
                normalize = True

            ref_audio = load_audio(
                ref_audio, sample_rate=model.sample_rate, volume_normalize=normalize
            )
            if not ref_text:
                import inspect

                if "ref_text" in inspect.signature(model.generate).parameters:
                    print("Ref_text not found. Transcribing ref_audio...")
                    from mlx_audio.stt.models.whisper import Model as Whisper

                    stt_model = (
                        Whisper.from_pretrained(path_or_hf_repo=stt_model)
                        if isinstance(stt_model, str)
                        else stt_model
                    )
                    ref_text = stt_model.generate(ref_audio).text

                    del stt_model
                    mx.clear_cache()
                    print(f"\033[94mRef_text:\033[0m {ref_text}")

        # Load AudioPlayer
        player = AudioPlayer(sample_rate=model.sample_rate) if play else None

        print(
            f"\033[94mText:\033[0m {text}\n"
            f"\033[94mVoice:\033[0m {voice}\n"
            f"\033[94mSpeed:\033[0m {speed}x\n"
            f"\033[94mLanguage:\033[0m {lang_code}"
        )

        results = model.generate(
            text=text,
            voice=voice,
            speed=speed,
            lang_code=lang_code,
            ref_audio=ref_audio,
            ref_text=ref_text,
            temperature=temperature,
            max_tokens=max_tokens,
            verbose=verbose,
            stream=stream,
            streaming_interval=streaming_interval,
            **kwargs,
        )

        audio_list = []
        file_name = f"{file_prefix}.{audio_format}"
        for i, result in enumerate(results):
            if play:
                player.queue_audio(result.audio)

            if join_audio:
                audio_list.append(result.audio)
            elif not stream:
                file_name = f"{file_prefix}_{i:03d}.{audio_format}"
                sf.write(file_name, result.audio, result.sample_rate)
                print(f"✅ Audio successfully generated and saving as: {file_name}")

            if verbose:

                print("==========")
                print(f"Duration:              {result.audio_duration}")
                print(
                    f"Samples/sec:           {result.audio_samples['samples-per-sec']:.1f}"
                )
                print(
                    f"Prompt:                {result.token_count} tokens, {result.prompt['tokens-per-sec']:.1f} tokens-per-sec"
                )
                print(
                    f"Audio:                 {result.audio_samples['samples']} samples, {result.audio_samples['samples-per-sec']:.1f} samples-per-sec"
                )
                print(f"Real-time factor:      {result.real_time_factor:.2f}x")
                print(f"Processing time:       {result.processing_time_seconds:.2f}s")
                print(f"Peak memory usage:     {result.peak_memory_usage:.2f}GB")

        if join_audio and not stream:
            if verbose:
                print(f"Joining {len(audio_list)} audio files")
            audio = mx.concatenate(audio_list, axis=0)
            sf.write(
                f"{file_prefix}.{audio_format}",
                audio,
                model.sample_rate,
            )
            if verbose:
                print(f"✅ Audio successfully generated and saving as: {file_name}")

        if play:
            player.wait_for_drain()
            player.stop()

    except ImportError as e:
        print(f"Import error: {e}")
        print(
            "This might be due to incorrect Python path. Check your project structure."
        )
    except Exception as e:
        print(f"Error loading model: {e}")
        import traceback

        traceback.print_exc()


def parse_args():
    parser = argparse.ArgumentParser(description="Generate audio from text using TTS.")
    parser.add_argument(
        "--model",
        type=str,
        default="mlx-community/Kokoro-82M-bf16",
        help="Path or repo id of the model",
    )
    parser.add_argument(
        "--max_tokens",
        type=int,
        default=1200,
        help="Maximum number of tokens to generate",
    )
    parser.add_argument(
        "--text",
        type=str,
        default=None,
        help="Text to generate (leave blank to input via stdin)",
    )
    parser.add_argument("--voice", type=str, default=None, help="Voice name")
    parser.add_argument("--speed", type=float, default=1.0, help="Speed of the audio")
    parser.add_argument(
        "--gender", type=str, default="male", help="Gender of the voice [male, female]"
    )
    parser.add_argument("--pitch", type=float, default=1.0, help="Pitch of the voice")
    parser.add_argument("--lang_code", type=str, default="a", help="Language code")
    parser.add_argument(
        "--file_prefix", type=str, default="audio", help="Output file name prefix"
    )
    parser.add_argument("--verbose", action="store_true", help="Print verbose output")
    parser.add_argument(
        "--join_audio", action="store_true", help="Join all audio files into one"
    )
    parser.add_argument("--play", action="store_true", help="Play the output audio")
    parser.add_argument(
        "--audio_format", type=str, default="wav", help="Output audio format"
    )
    parser.add_argument(
        "--ref_audio", type=str, default=None, help="Path to reference audio"
    )
    parser.add_argument(
        "--ref_text", type=str, default=None, help="Caption for reference audio"
    )
    parser.add_argument(
        "--stt_model",
        type=str,
        default="mlx-community/whisper-large-v3-turbo",
        help="STT model to use to transcribe reference audio",
    )
    parser.add_argument(
        "--temperature", type=float, default=0.7, help="Temperature for the model"
    )
    parser.add_argument("--top_p", type=float, default=0.9, help="Top-p for the model")
    parser.add_argument("--top_k", type=int, default=50, help="Top-k for the model")
    parser.add_argument(
        "--repetition_penalty",
        type=float,
        default=1.1,
        help="Repetition penalty for the model",
    )
    parser.add_argument(
        "--stream",
        action="store_true",
        help="Stream the audio as segments instead of saving to a file",
    )
    parser.add_argument(
        "--streaming_interval",
        type=float,
        default=2.0,
        help="The time interval in seconds for streaming segments",
    )

    args = parser.parse_args()

    if args.text is None:
        if not sys.stdin.isatty():
            args.text = sys.stdin.read().strip()
        else:
            print("Please enter the text to generate:")
            args.text = input("> ").strip()

    return args


def main():
    args = parse_args()
    generate_audio(**vars(args))


if __name__ == "__main__":
    main()
