import bisect
import functools
import os

from dataclasses import dataclass
from typing import Dict, List, Optional, Tuple

import numpy as np

from faster_whisper.utils import get_assets_path


# The code below is adapted from https://github.com/snakers4/silero-vad.
@dataclass
class VadOptions:
    """VAD options.

    Attributes:
      threshold: Speech threshold. Silero VAD outputs speech probabilities for each audio chunk,
        probabilities ABOVE this value are considered as SPEECH. It is better to tune this
        parameter for each dataset separately, but "lazy" 0.5 is pretty good for most datasets.
      neg_threshold: Silence threshold for determining the end of speech. If a probability is lower
        than neg_threshold, it is always considered silence. Values higher than neg_threshold
        are only considered speech if the previous sample was classified as speech; otherwise,
        they are treated as silence. This parameter helps refine the detection of speech
         transitions, ensuring smoother segment boundaries.
      min_speech_duration_ms: Final speech chunks shorter min_speech_duration_ms are thrown out.
      max_speech_duration_s: Maximum duration of speech chunks in seconds. Chunks longer
        than max_speech_duration_s will be split at the timestamp of the last silence that
        lasts more than 100ms (if any), to prevent aggressive cutting. Otherwise, they will be
        split aggressively just before max_speech_duration_s.
      min_silence_duration_ms: In the end of each speech chunk wait for min_silence_duration_ms
        before separating it
      speech_pad_ms: Final speech chunks are padded by speech_pad_ms each side
    """

    threshold: float = 0.5
    neg_threshold: float = None
    min_speech_duration_ms: int = 0
    max_speech_duration_s: float = float("inf")
    min_silence_duration_ms: int = 2000
    speech_pad_ms: int = 400


def get_speech_timestamps(
    audio: np.ndarray,
    vad_options: Optional[VadOptions] = None,
    sampling_rate: int = 16000,
    **kwargs,
) -> List[dict]:
    """This method is used for splitting long audios into speech chunks using silero VAD.

    Args:
      audio: One dimensional float array.
      vad_options: Options for VAD processing.
      sampling rate: Sampling rate of the audio.
      kwargs: VAD options passed as keyword arguments for backward compatibility.

    Returns:
      List of dicts containing begin and end samples of each speech chunk.
    """
    if vad_options is None:
        vad_options = VadOptions(**kwargs)

    threshold = vad_options.threshold
    neg_threshold = vad_options.neg_threshold
    min_speech_duration_ms = vad_options.min_speech_duration_ms
    max_speech_duration_s = vad_options.max_speech_duration_s
    min_silence_duration_ms = vad_options.min_silence_duration_ms
    window_size_samples = 512
    speech_pad_ms = vad_options.speech_pad_ms
    min_speech_samples = sampling_rate * min_speech_duration_ms / 1000
    speech_pad_samples = sampling_rate * speech_pad_ms / 1000
    max_speech_samples = (
        sampling_rate * max_speech_duration_s
        - window_size_samples
        - 2 * speech_pad_samples
    )
    min_silence_samples = sampling_rate * min_silence_duration_ms / 1000
    min_silence_samples_at_max_speech = sampling_rate * 98 / 1000

    audio_length_samples = len(audio)

    model = get_vad_model()

    padded_audio = np.pad(
        audio, (0, window_size_samples - audio.shape[0] % window_size_samples)
    )
    speech_probs = model(padded_audio)

    triggered = False
    speeches = []
    current_speech = {}
    if neg_threshold is None:
        neg_threshold = max(threshold - 0.15, 0.01)

    # to save potential segment end (and tolerate some silence)
    temp_end = 0
    # to save potential segment limits in case of maximum segment size reached
    prev_end = next_start = 0

    for i, speech_prob in enumerate(speech_probs):
        if (speech_prob >= threshold) and temp_end:
            temp_end = 0
            if next_start < prev_end:
                next_start = window_size_samples * i

        if (speech_prob >= threshold) and not triggered:
            triggered = True
            current_speech["start"] = window_size_samples * i
            continue

        if (
            triggered
            and (window_size_samples * i) - current_speech["start"] > max_speech_samples
        ):
            if prev_end:
                current_speech["end"] = prev_end
                speeches.append(current_speech)
                current_speech = {}
                # previously reached silence (< neg_thres) and is still not speech (< thres)
                if next_start < prev_end:
                    triggered = False
                else:
                    current_speech["start"] = next_start
                prev_end = next_start = temp_end = 0
            else:
                current_speech["end"] = window_size_samples * i
                speeches.append(current_speech)
                current_speech = {}
                prev_end = next_start = temp_end = 0
                triggered = False
                continue

        if (speech_prob < neg_threshold) and triggered:
            if not temp_end:
                temp_end = window_size_samples * i
            # condition to avoid cutting in very short silence
            if (window_size_samples * i) - temp_end > min_silence_samples_at_max_speech:
                prev_end = temp_end
            if (window_size_samples * i) - temp_end < min_silence_samples:
                continue
            else:
                current_speech["end"] = temp_end
                if (
                    current_speech["end"] - current_speech["start"]
                ) > min_speech_samples:
                    speeches.append(current_speech)
                current_speech = {}
                prev_end = next_start = temp_end = 0
                triggered = False
                continue

    if (
        current_speech
        and (audio_length_samples - current_speech["start"]) > min_speech_samples
    ):
        current_speech["end"] = audio_length_samples
        speeches.append(current_speech)

    for i, speech in enumerate(speeches):
        if i == 0:
            speech["start"] = int(max(0, speech["start"] - speech_pad_samples))
        if i != len(speeches) - 1:
            silence_duration = speeches[i + 1]["start"] - speech["end"]
            if silence_duration < 2 * speech_pad_samples:
                speech["end"] += int(silence_duration // 2)
                speeches[i + 1]["start"] = int(
                    max(0, speeches[i + 1]["start"] - silence_duration // 2)
                )
            else:
                speech["end"] = int(
                    min(audio_length_samples, speech["end"] + speech_pad_samples)
                )
                speeches[i + 1]["start"] = int(
                    max(0, speeches[i + 1]["start"] - speech_pad_samples)
                )
        else:
            speech["end"] = int(
                min(audio_length_samples, speech["end"] + speech_pad_samples)
            )

    return speeches


def collect_chunks(
    audio: np.ndarray,
    chunks: List[dict],
    sampling_rate: int = 16000,
    max_duration: float = float("inf"),
) -> Tuple[List[np.ndarray], List[Dict[str, float]]]:
    """This function merges the chunks of audio into chunks of max_duration (s) length."""
    if not chunks:
        chunk_metadata = {
            "offset": 0,
            "duration": 0,
            "segments": [],
        }
        return [np.array([], dtype=np.float32)], [chunk_metadata]

    audio_chunks = []
    chunks_metadata = []

    current_segments = []
    current_duration = 0
    total_duration = 0
    current_audio = np.array([], dtype=np.float32)

    for chunk in chunks:
        if (
            current_duration + chunk["end"] - chunk["start"]
            > max_duration * sampling_rate
        ):
            audio_chunks.append(current_audio)
            chunk_metadata = {
                "offset": total_duration / sampling_rate,
                "duration": current_duration / sampling_rate,
                "segments": current_segments,
            }
            total_duration += current_duration
            chunks_metadata.append(chunk_metadata)

            current_segments = []

            current_audio = audio[chunk["start"] : chunk["end"]]
            current_duration = chunk["end"] - chunk["start"]
        else:
            current_segments.append(chunk)
            current_audio = np.concatenate(
                (current_audio, audio[chunk["start"] : chunk["end"]])
            )

            current_duration += chunk["end"] - chunk["start"]

    audio_chunks.append(current_audio)

    chunk_metadata = {
        "offset": total_duration / sampling_rate,
        "duration": current_duration / sampling_rate,
        "segments": current_segments,
    }
    chunks_metadata.append(chunk_metadata)
    return audio_chunks, chunks_metadata


class SpeechTimestampsMap:
    """Helper class to restore original speech timestamps."""

    def __init__(self, chunks: List[dict], sampling_rate: int, time_precision: int = 2):
        self.sampling_rate = sampling_rate
        self.time_precision = time_precision
        self.chunk_end_sample = []
        self.total_silence_before = []

        previous_end = 0
        silent_samples = 0

        for chunk in chunks:
            silent_samples += chunk["start"] - previous_end
            previous_end = chunk["end"]

            self.chunk_end_sample.append(chunk["end"] - silent_samples)
            self.total_silence_before.append(silent_samples / sampling_rate)

    def get_original_time(
        self,
        time: float,
        chunk_index: Optional[int] = None,
        is_end: bool = False,
    ) -> float:
        if chunk_index is None:
            chunk_index = self.get_chunk_index(time, is_end)

        total_silence_before = self.total_silence_before[chunk_index]
        return round(total_silence_before + time, self.time_precision)

    def get_chunk_index(self, time: float, is_end: bool = False) -> int:
        sample = int(time * self.sampling_rate)
        if sample in self.chunk_end_sample and is_end:
            return self.chunk_end_sample.index(sample)

        return min(
            bisect.bisect(self.chunk_end_sample, sample),
            len(self.chunk_end_sample) - 1,
        )


@functools.lru_cache
def get_vad_model():
    """Returns the VAD model instance."""
    path = os.path.join(get_assets_path(), "silero_vad_v6.onnx")
    return SileroVADModel(path)


class SileroVADModel:
    def __init__(self, path):
        try:
            import onnxruntime
        except ImportError as e:
            raise RuntimeError(
                "Applying the VAD filter requires the onnxruntime package"
            ) from e

        opts = onnxruntime.SessionOptions()
        opts.inter_op_num_threads = 1
        opts.intra_op_num_threads = 1
        opts.enable_cpu_mem_arena = False
        opts.log_severity_level = 4

        self.session = onnxruntime.InferenceSession(
            path,
            providers=["CPUExecutionProvider"],
            sess_options=opts,
        )

    def __call__(
        self, audio: np.ndarray, num_samples: int = 512, context_size_samples: int = 64
    ):
        assert audio.ndim == 1, "Input should be a 1D array"
        assert (
            audio.shape[0] % num_samples == 0
        ), "Input size should be a multiple of num_samples"

        h = np.zeros((1, 1, 128), dtype="float32")
        c = np.zeros((1, 1, 128), dtype="float32")
        context = np.zeros(
            (1, context_size_samples),
            dtype="float32",
        )

        batched_audio = audio.reshape(-1, num_samples)
        context = batched_audio[..., -context_size_samples:]
        context[-1] = 0
        context = np.roll(context, 1, 0)
        batched_audio = np.concatenate([context, batched_audio], 1)

        batched_audio = batched_audio.reshape(-1, num_samples + context_size_samples)

        encoder_batch_size = 10000
        num_segments = batched_audio.shape[0]
        outputs = []
        for i in range(0, num_segments, encoder_batch_size):
            output, h, c = self.session.run(
                None,
                {"input": batched_audio[i : i + encoder_batch_size], "h": h, "c": c},
            )
            outputs.append(output)

        out = np.concatenate(outputs, axis=0)

        return out