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# Copyright 2025 The HuggingFace Inc. team. All rights reserved.
#
# Licensed under the Apache License, Version 2.0 (the "License");
# you may not use this file except in compliance with the License.
# You may obtain a copy of the License at
#
#     http://www.apache.org/licenses/LICENSE-2.0
#
# Unless required by applicable law or agreed to in writing, software
# distributed under the License is distributed on an "AS IS" BASIS,
# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
# See the License for the specific language governing permissions and
# limitations under the License.

import numpy as np
import torch

from ...audio_utils import mel_filter_bank
from ...feature_extraction_sequence_utils import SequenceFeatureExtractor
from ...feature_extraction_utils import BatchFeature
from ...utils import TensorType, logging
from ...utils.import_utils import requires


logger = logging.get_logger(__name__)


@requires(backends=("torch",))
class VoxtralRealtimeFeatureExtractor(SequenceFeatureExtractor):
    r"""
    Constructs a VOXTRAL_REALTIME feature extractor.

    This feature extractor inherits from [`~feature_extraction_sequence_utils.SequenceFeatureExtractor`] which contains
    most of the main methods. Users should refer to this superclass for more information regarding those methods.

    This class extracts mel-filter bank features from raw speech using a custom numpy implementation of the `Short Time
    Fourier Transform` which should match pytorch's `torch.stft` equivalent.

    Args:
            feature_size (`int`, *optional*, defaults to 128):
                The feature dimension of the extracted features.
            sampling_rate (`int`, *optional*, defaults to 16000):
                The sampling rate at which the audio files should be digitalized expressed in hertz (Hz).
            hop_length (`int`, *optional*, defaults to 160):
                Length of the overlapping windows for the STFT used to obtain the Mel Frequency coefficients.
            n_fft (`int`, *optional*, defaults to 512):
                Size of the Fourier transform.
            win_length (`int`, *optional*, defaults to 400):
                The window length for the STFT computation.
            padding_value (`float`, *optional*, defaults to 0.0):
                Padding value used to pad the audio. Should correspond to silences.
    """

    model_input_names = ["input_features", "attention_mask"]

    def __init__(
        self,
        feature_size=128,
        sampling_rate=16000,
        hop_length=160,
        n_fft=400,
        win_length=400,
        padding_value=0.0,
        global_log_mel_max=1.5,
        **kwargs,
    ):
        super().__init__(feature_size=feature_size, sampling_rate=sampling_rate, padding_value=padding_value, **kwargs)

        self.hop_length = hop_length
        self.n_fft = n_fft
        self.win_length = win_length
        self.mel_filters = mel_filter_bank(
            num_frequency_bins=1 + n_fft // 2,
            num_mel_filters=feature_size,
            min_frequency=0.0,
            max_frequency=8000.0,
            sampling_rate=sampling_rate,
            norm="slaney",
            mel_scale="slaney",
        )
        self.global_log_mel_max = global_log_mel_max

    def _torch_extract_fbank_features(self, waveform, device: str = "cpu", center: bool = True):
        window = torch.hann_window(self.n_fft, device=device)
        stft = torch.stft(waveform, self.n_fft, self.hop_length, window=window, return_complex=True, center=center)
        magnitudes = stft[..., :-1].abs() ** 2

        mel_filters = torch.from_numpy(self.mel_filters).to(device, torch.float32)
        mel_spec = mel_filters.T @ magnitudes

        log_spec = torch.clamp(mel_spec, min=1e-10).log10()
        if self.global_log_mel_max is not None:
            log_spec_max = torch.tensor(
                self.global_log_mel_max,
                device=log_spec.device,
                dtype=log_spec.dtype,
            )
        else:
            log_spec_max = log_spec.max()

        log_spec = torch.maximum(log_spec, log_spec_max - 8.0)
        log_spec = (log_spec + 4.0) / 4.0
        if device != "cpu":
            log_spec = log_spec.detach().cpu()
        return log_spec

    def __call__(
        self,
        raw_speech: np.ndarray | list[float] | list[np.ndarray] | list[list[float]],
        truncation: bool = False,
        pad_to_multiple_of: int | None = None,
        return_tensors: str | TensorType | None = None,
        return_attention_mask: bool | None = None,
        padding: str | None = "longest",
        max_length: int | None = None,
        sampling_rate: int | None = None,
        do_normalize: bool | None = None,
        device: str | None = "cpu",
        return_token_timestamps: bool | None = None,
        center: bool = True,
        **kwargs,
    ) -> BatchFeature:
        """
        Main method to featurize and prepare for the model one or several sequence(s). Implementation uses PyTorch for
        the STFT computation if available, otherwise a slower NumPy based one.

        Args:
            raw_speech (`np.ndarray`, `list[float]`, `list[np.ndarray]`, `list[list[float]]`):
                The sequence or batch of sequences to be padded. Each sequence can be a numpy array, a list of float
                values, a list of numpy arrays or a list of list of float values. Must be mono channel audio, not
                stereo, i.e. single float per timestep.
            truncation (`bool`, *optional*, default to `True`):
                Activates truncation to cut input sequences longer than *max_length* to *max_length*.
            pad_to_multiple_of (`int`, *optional*, defaults to None):
                If set will pad the sequence to a multiple of the provided value.

                This is especially useful to enable the use of Tensor Cores on NVIDIA hardware with compute capability
                `>= 7.5` (Volta), or on TPUs which benefit from having sequence lengths be a multiple of 128.
            return_attention_mask (`bool`, *optional*):
                Whether to return the attention mask. If left to the default, will return the attention mask according
                to the specific feature_extractor's default.

                [What are attention masks?](../glossary#attention-mask)

                <Tip>

                For Parakeet models, `attention_mask` should always be passed for batched inference, to avoid subtle
                bugs.

                </Tip>

            return_tensors (`str` or [`~utils.TensorType`], *optional*):
                If set, will return tensors instead of list of python integers. Acceptable values are:

                - `'tf'`: Return TensorFlow `tf.constant` objects.
                - `'pt'`: Return PyTorch `torch.Tensor` objects.
                - `'np'`: Return Numpy `np.ndarray` objects.
            sampling_rate (`int`, *optional*):
                The sampling rate at which the `raw_speech` input was sampled. It is strongly recommended to pass
                `sampling_rate` at the forward call to prevent silent errors and allow automatic speech recognition
                pipeline.
            padding_value (`float`, *optional*, defaults to 0.0):
                The value that is used to fill the padding values / vectors.
            do_normalize (`bool`, *optional*, defaults to `False`):
                Whether or not to zero-mean unit-variance normalize the input. Normalizing can help to significantly
                improve the performance of the model.
            device (`str`, *optional*, defaults to `'cpu'`):
                Specifies the device for computation of the log-mel spectrogram of audio signals in the
                `_torch_extract_fbank_features` method. (e.g., "cpu", "cuda")
            return_token_timestamps (`bool`, *optional*, defaults to `None`):
                Deprecated. Use `return_attention_mask` instead from which the number of frames can be inferred.

                Whether or not to return the number of frames of the input raw_speech.
                These num_frames can be used by the model to compute word level timestamps.
            center (`bool`, *optional*, defaults to `True`):
                Whether to use centering for the STFT computation.
        """
        if sampling_rate is not None:
            if sampling_rate != self.sampling_rate:
                raise ValueError(
                    f"The model corresponding to this feature extractor: {self.__class__.__name__} was trained using a"
                    f" sampling rate of {self.sampling_rate}. Please make sure that the provided `raw_speech` input"
                    f" was sampled with {self.sampling_rate} and not {sampling_rate}."
                )
        else:
            logger.warning(
                f"It is strongly recommended to pass the `sampling_rate` argument to `{self.__class__.__name__}()`. "
                "Failing to do so can result in silent errors that might be hard to debug."
            )

        # Convert to torch tensor
        if isinstance(raw_speech, np.ndarray):
            raw_speech = torch.tensor(raw_speech)
        elif isinstance(raw_speech, (list, tuple)):
            if isinstance(raw_speech[0], (list, np.ndarray)):
                raw_speech = [torch.tensor(speech) for speech in raw_speech]
            else:  # list[float]
                raw_speech = torch.tensor(raw_speech)

        is_batched_torch = isinstance(raw_speech, torch.Tensor) and len(raw_speech.shape) > 1
        if is_batched_torch and len(raw_speech.shape) > 2:
            logger.warning(
                f"Only mono-channel audio is supported for input to {self.__class__.__name__}. "
                "We will take the mean of the channels to convert to mono."
            )
            raw_speech = raw_speech.mean(-1)

        is_batched_sequence = isinstance(raw_speech, (list, tuple))
        if is_batched_sequence:
            for speech in raw_speech:
                if len(speech.shape) > 1:
                    logger.warning(
                        f"Only mono-channel audio is supported for input to {self.__class__.__name__}. "
                        "We will take the mean of the channels to convert to mono."
                    )
                    speech = speech.mean(-1)

        if is_batched_torch or is_batched_sequence:
            raw_speech = [speech[:, None].to(torch.float32) for speech in raw_speech]
        else:
            raw_speech = [raw_speech[:, None].to(torch.float32)]

        batched_speech = BatchFeature({"input_features": raw_speech})
        padded_inputs = self.pad(
            batched_speech,
            padding=padding,
            max_length=max_length,
            truncation=truncation,
            pad_to_multiple_of=pad_to_multiple_of,
            return_attention_mask=return_attention_mask,
            return_tensors="pt",
        )
        input_features = padded_inputs.input_features.squeeze(-1)
        input_features = self._torch_extract_fbank_features(input_features, device, center)
        data = {
            "input_features": input_features.to(torch.float32),
        }

        if return_attention_mask:
            attention_mask = padded_inputs.attention_mask[:, self.win_length - 1 :: self.hop_length]
            data["attention_mask"] = attention_mask.to(torch.bool)

        return BatchFeature(data=data, tensor_type=return_tensors)


__all__ = ["VoxtralRealtimeFeatureExtractor"]