# coding=utf-8
# Copyright 2022 The HuggingFace Inc. team.
#
# 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
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# Unless required by applicable law or agreed to in writing, software
# distributed under the License is distributed on an "AS IS" BASIS,
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"""
Feature extractor class for Whisper
"""
from typing import List, Optional, Union

import numpy as np

from ... import is_torch_available
from ...audio_utils import mel_filter_bank, spectrogram, window_function
from ...feature_extraction_sequence_utils import SequenceFeatureExtractor
from ...feature_extraction_utils import BatchFeature
from ...utils import TensorType, logging


if is_torch_available():
    import torch

logger = logging.get_logger(__name__)


class WhisperFeatureExtractor(SequenceFeatureExtractor):
    r"""
    Constructs a Whisper 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`, defaults to 80):
            The feature dimension of the extracted features.
        sampling_rate (`int`, defaults to 16000):
            The sampling rate at which the audio files should be digitalized expressed in hertz (Hz).
        hop_length (`int`, defaults to 160):
            Length of the overlaping windows for the STFT used to obtain the Mel Frequency coefficients.
        chunk_length (`int`, defaults to 30):
            The maximum number of chuncks of `sampling_rate` samples used to trim and pad longer or shorter audio
            sequences.
        n_fft (`int`, defaults to 400):
            Size of the Fourier transform.
        padding_value (`float`, *optional*, defaults to 0.0):
            Padding value used to pad the audio. Should correspond to silences.
    """

    model_input_names = ["input_features"]

    def __init__(
        self,
        feature_size=80,
        sampling_rate=16000,
        hop_length=160,
        chunk_length=30,
        n_fft=400,
        padding_value=0.0,
        return_attention_mask=False,  # pad inputs to max length with silence token (zero) and no attention mask
        **kwargs,
    ):
        super().__init__(
            feature_size=feature_size,
            sampling_rate=sampling_rate,
            padding_value=padding_value,
            return_attention_mask=return_attention_mask,
            **kwargs,
        )
        self.n_fft = n_fft
        self.hop_length = hop_length
        self.chunk_length = chunk_length
        self.n_samples = chunk_length * sampling_rate
        self.nb_max_frames = self.n_samples // hop_length
        self.sampling_rate = sampling_rate
        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",
        )

    def _np_extract_fbank_features(self, waveform_batch: np.array, device: str) -> np.ndarray:
        """
        Compute the log-mel spectrogram of the provided audio, gives similar results to Whisper's original torch
        implementation with 1e-5 tolerance.
        """
        if device != "cpu":
            raise ValueError(
                f"Got device `{device}` for feature extraction, but feature extraction on CUDA accelerator "
                "devices requires torch, which is not installed. Either set `device='cpu'`, or "
                "install torch according to the official instructions: https://pytorch.org/get-started/locally/"
            )
        log_spec_batch = []
        for waveform in waveform_batch:
            log_spec = spectrogram(
                waveform,
                window_function(self.n_fft, "hann"),
                frame_length=self.n_fft,
                hop_length=self.hop_length,
                power=2.0,
                mel_filters=self.mel_filters,
                log_mel="log10",
            )
            log_spec = log_spec[:, :-1]
            log_spec = np.maximum(log_spec, log_spec.max() - 8.0)
            log_spec = (log_spec + 4.0) / 4.0
            log_spec_batch.append(log_spec)
        log_spec_batch = np.array(log_spec_batch)
        return log_spec_batch

    def _torch_extract_fbank_features(self, waveform: np.array, device: str = "cpu") -> np.ndarray:
        """
        Compute the log-mel spectrogram of the audio using PyTorch's GPU-accelerated STFT implementation with batching,
        yielding results similar to cpu computing with 1e-5 tolerance.
        """
        waveform = torch.from_numpy(waveform).type(torch.float32)

        window = torch.hann_window(self.n_fft)
        if device != "cpu":
            waveform = waveform.to(device)
            window = window.to(device)
        stft = torch.stft(waveform, self.n_fft, self.hop_length, window=window, return_complex=True)
        magnitudes = stft[..., :-1].abs() ** 2

        mel_filters = torch.from_numpy(self.mel_filters).type(torch.float32)
        if device != "cpu":
            mel_filters = mel_filters.to(device)
        mel_spec = mel_filters.T @ magnitudes

        log_spec = torch.clamp(mel_spec, min=1e-10).log10()
        if waveform.dim() == 2:
            max_val = log_spec.max(dim=2, keepdim=True)[0].max(dim=1, keepdim=True)[0]
            log_spec = torch.maximum(log_spec, max_val - 8.0)
        else:
            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.numpy()

    @staticmethod
    # Copied from transformers.models.wav2vec2.feature_extraction_wav2vec2.Wav2Vec2FeatureExtractor.zero_mean_unit_var_norm
    def zero_mean_unit_var_norm(
        input_values: List[np.ndarray], attention_mask: List[np.ndarray], padding_value: float = 0.0
    ) -> List[np.ndarray]:
        """
        Every array in the list is normalized to have zero mean and unit variance
        """
        if attention_mask is not None:
            attention_mask = np.array(attention_mask, np.int32)
            normed_input_values = []

            for vector, length in zip(input_values, attention_mask.sum(-1)):
                normed_slice = (vector - vector[:length].mean()) / np.sqrt(vector[:length].var() + 1e-7)
                if length < normed_slice.shape[0]:
                    normed_slice[length:] = padding_value

                normed_input_values.append(normed_slice)
        else:
            normed_input_values = [(x - x.mean()) / np.sqrt(x.var() + 1e-7) for x in input_values]

        return normed_input_values

    def __call__(
        self,
        raw_speech: Union[np.ndarray, List[float], List[np.ndarray], List[List[float]]],
        truncation: bool = True,
        pad_to_multiple_of: Optional[int] = None,
        return_tensors: Optional[Union[str, TensorType]] = None,
        return_attention_mask: Optional[bool] = None,
        padding: Optional[str] = "max_length",
        max_length: Optional[int] = None,
        sampling_rate: Optional[int] = None,
        do_normalize: Optional[bool] = None,
        device: Optional[str] = "cpu",
        return_token_timestamps: Optional[bool] = None,
        **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 Whisper 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`, 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`):
                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.
        """

        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(
                "It is strongly recommended to pass the `sampling_rate` argument to this function. "
                "Failing to do so can result in silent errors that might be hard to debug."
            )

        is_batched_numpy = isinstance(raw_speech, np.ndarray) and len(raw_speech.shape) > 1
        if is_batched_numpy and len(raw_speech.shape) > 2:
            raise ValueError(f"Only mono-channel audio is supported for input to {self}")
        is_batched = is_batched_numpy or (
            isinstance(raw_speech, (list, tuple)) and (isinstance(raw_speech[0], (np.ndarray, tuple, list)))
        )

        if is_batched:
            raw_speech = [np.asarray([speech], dtype=np.float32).T for speech in raw_speech]
        elif not is_batched and not isinstance(raw_speech, np.ndarray):
            raw_speech = np.asarray(raw_speech, dtype=np.float32)
        elif isinstance(raw_speech, np.ndarray) and raw_speech.dtype is np.dtype(np.float64):
            raw_speech = raw_speech.astype(np.float32)

        # always return batch
        if not is_batched:
            raw_speech = [np.asarray([raw_speech]).T]

        batched_speech = BatchFeature({"input_features": raw_speech})

        # convert into correct format for padding

        padded_inputs = self.pad(
            batched_speech,
            padding=padding,
            max_length=max_length if max_length else self.n_samples,
            truncation=truncation,
            pad_to_multiple_of=pad_to_multiple_of,
            return_attention_mask=return_attention_mask or do_normalize,
        )

        # zero-mean and unit-variance normalization
        if do_normalize:
            padded_inputs["input_features"] = self.zero_mean_unit_var_norm(
                padded_inputs["input_features"],
                attention_mask=padded_inputs["attention_mask"],
                padding_value=self.padding_value,
            )
            padded_inputs["input_features"] = np.stack(padded_inputs["input_features"], axis=0)

        # make sure list is in array format
        input_features = padded_inputs.get("input_features").transpose(2, 0, 1)

        extract_fbank_features = (
            self._torch_extract_fbank_features if is_torch_available() else self._np_extract_fbank_features
        )
        input_features = extract_fbank_features(input_features[0], device)

        if isinstance(input_features[0], List):
            padded_inputs["input_features"] = [np.asarray(feature, dtype=np.float32) for feature in input_features]

        else:
            padded_inputs["input_features"] = input_features

        if return_attention_mask:
            # rescale from sample (48000) to feature (3000)
            padded_inputs["attention_mask"] = padded_inputs["attention_mask"][:, :: self.hop_length]

        if return_token_timestamps is not None:
            padded_inputs["num_frames"] = [len(raw_speech_i) // self.hop_length for raw_speech_i in raw_speech]

        if return_tensors is not None:
            padded_inputs = padded_inputs.convert_to_tensors(return_tensors)

        return padded_inputs
