# What does this PR do?

This PR fixes the last batch of models that are too big for common tests. After it, only two classes override the common test to skip it:
- timm-backbone
- layoutlmv2
In the first case, it's because there is no timm model small enough to work (I did switch to the smallest resnet though) and in the second detectron2 does not let us configure a smaller backbone (I did try with the constant exposed by they don't seem to have any effect).
Fix last models for common tests that are too big.. # What does this PR do?

This PR fixes the last batch of models that are too big for common tests. After it, only two classes override the common test to skip it:
- timm-backbone
- layoutlmv2
In the first case, it's because there is no timm model small enough to work (I did switch to the smallest resnet though) and in the second detectron2 does not let us configure a smaller backbone (I did try with the constant exposed by they don't seem to have any effect).
# coding=utf-8
# Copyright 2021 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
#
# 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.
"""
Feature extractor class for Speech2Text
"""

from typing import List, Optional, Union

import numpy as np
import torch
import torchaudio.compliance.kaldi as ta_kaldi

from ...feature_extraction_sequence_utils import SequenceFeatureExtractor
from ...feature_extraction_utils import BatchFeature
from ...utils import PaddingStrategy, TensorType, logging


logger = logging.get_logger(__name__)


class Speech2TextFeatureExtractor(SequenceFeatureExtractor):
    r"""
    Constructs a Speech2Text feature extractor.

    This feature extractor inherits from [`Speech2TextFeatureExtractor`] 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 TorchAudio and applies utterance-level cepstral
    mean and variance normalization to the extracted features.

    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).
        num_mel_bins (`int`, defaults to 80):
            Number of Mel-frequency bins.
        padding_value (`float`, defaults to 0.0):
            The value that is used to fill the padding vectors.
        do_ceptral_normalize (`bool`, *optional*, defaults to `True`):
            Whether or not to apply utterance-level cepstral mean and variance normalization to extracted features.
        normalize_means (`bool`, *optional*, defaults to `True`):
            Whether or not to zero-mean normalize the extracted features.
        normalize_vars (`bool`, *optional*, defaults to `True`):
            Whether or not to unit-variance normalize the extracted features.
    """

    model_input_names = ["input_features", "attention_mask"]

    def __init__(
        self,
        feature_size=80,
        sampling_rate=16000,
        num_mel_bins=80,
        padding_value=0.0,
        do_ceptral_normalize=True,
        normalize_means=True,
        normalize_vars=True,
        **kwargs,
    ):
        super().__init__(feature_size=feature_size, sampling_rate=sampling_rate, padding_value=padding_value, **kwargs)
        self.num_mel_bins = num_mel_bins
        self.do_ceptral_normalize = do_ceptral_normalize
        self.normalize_means = normalize_means
        self.normalize_vars = normalize_vars
        self.return_attention_mask = True

    def _extract_fbank_features(
        self,
        waveform: np.ndarray,
    ) -> np.ndarray:
        """
        Get mel-filter bank features using TorchAudio. Note that TorchAudio requires 16-bit signed integers as inputs
        and hence the waveform should not be normalized before feature extraction.
        """
        waveform = waveform * (2**15)  # Kaldi compliance: 16-bit signed integers
        waveform = torch.from_numpy(waveform).unsqueeze(0)
        features = ta_kaldi.fbank(waveform, num_mel_bins=self.num_mel_bins, sample_frequency=self.sampling_rate)
        return features.numpy()

    @staticmethod
    def utterance_cmvn(
        x: np.ndarray,
        input_length: int,
        normalize_means: Optional[bool] = True,
        normalize_vars: Optional[bool] = True,
        padding_value: float = 0.0,
    ) -> np.ndarray:
        # make sure we normalize float32 arrays
        if normalize_means:
            mean = x[:input_length].mean(axis=0)
            x = np.subtract(x, mean)
        if normalize_vars:
            std = x[:input_length].std(axis=0)
            x = np.divide(x, std)

        if input_length < x.shape[0]:
            x[input_length:] = padding_value

        # make sure array is in float32
        x = x.astype(np.float32)

        return x

    def normalize(
        self, input_features: List[np.ndarray], attention_mask: Optional[np.ndarray] = None
    ) -> List[np.ndarray]:
        lengths = attention_mask.sum(-1) if attention_mask is not None else [x.shape[0] for x in input_features]
        return [
            self.utterance_cmvn(x, n, self.normalize_means, self.normalize_vars, self.padding_value)
            for x, n in zip(input_features, lengths)
        ]

    def __call__(
        self,
        raw_speech: Union[np.ndarray, List[float], List[np.ndarray], List[List[float]]],
        padding: Union[bool, str, PaddingStrategy] = False,
        max_length: Optional[int] = None,
        truncation: bool = False,
        pad_to_multiple_of: Optional[int] = None,
        return_tensors: Optional[Union[str, TensorType]] = None,
        sampling_rate: Optional[int] = None,
        return_attention_mask: Optional[bool] = None,
        **kwargs,
    ) -> BatchFeature:
        """
        Main method to featurize and prepare for the model one or several sequence(s).

        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.
            padding (`bool`, `str` or [`~utils.PaddingStrategy`], *optional*, defaults to `True`):
                Select a strategy to pad the returned sequences (according to the model's padding side and padding
                index) among:

                - `True` or `'longest'`: Pad to the longest sequence in the batch (or no padding if only a single
                  sequence if provided).
                - `'max_length'`: Pad to a maximum length specified with the argument `max_length` or to the maximum
                  acceptable input length for the model if that argument is not provided.
                - `False` or `'do_not_pad'` (default): No padding (i.e., can output a batch with sequences of different
                  lengths).
            max_length (`int`, *optional*):
                Maximum length of the returned list and optionally padding length (see above).
            truncation (`bool`):
                Activates truncation to cut input sequences longer than *max_length* to *max_length*.
            pad_to_multiple_of (`int`, *optional*):
                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 Speech2TextTransformer 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.
            padding_value (`float`, defaults to 0.0):
                The value that is used to fill the padding values / vectors.
        """

        if sampling_rate is not None:
            if sampling_rate != self.sampling_rate:
                raise ValueError(
                    f"The model corresponding to this feature extractor: {self} was trained using a sampling rate of"
                    f" {self.sampling_rate}. Please make sure that the provided `raw_speech` input was sampled with"
                    f" {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) 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 = [raw_speech]

        # extract fbank features
        features = [self._extract_fbank_features(waveform) for waveform in raw_speech]

        # convert into correct format for padding
        encoded_inputs = BatchFeature({"input_features": features})

        padded_inputs = self.pad(
            encoded_inputs,
            padding=padding,
            max_length=max_length,
            truncation=truncation,
            pad_to_multiple_of=pad_to_multiple_of,
            return_attention_mask=return_attention_mask,
            **kwargs,
        )

        # make sure list is in array format
        input_features = padded_inputs.get("input_features")
        if isinstance(input_features[0], list):
            padded_inputs["input_features"] = [np.asarray(feature, dtype=np.float32) for feature in input_features]

        attention_mask = padded_inputs.get("attention_mask")
        if attention_mask is not None:
            padded_inputs["attention_mask"] = [np.asarray(array, dtype=np.int32) for array in attention_mask]

        # Utterance-level cepstral mean and variance normalization
        if self.do_ceptral_normalize:
            attention_mask = (
                np.array(attention_mask, dtype=np.int32)
                if self._get_padding_strategies(padding, max_length=max_length) is not PaddingStrategy.DO_NOT_PAD
                else None
            )
            padded_inputs["input_features"] = self.normalize(
                padded_inputs["input_features"], attention_mask=attention_mask
            )

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

        return padded_inputs

# coding=utf-8
# Copyright 2021 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
#
# 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.
"""
Feature extractor class for Speech2Text
"""

from typing import List, Optional, Union

import numpy as np
import torch
import torchaudio.compliance.kaldi as ta_kaldi

from ...feature_extraction_sequence_utils import SequenceFeatureExtractor
from ...feature_extraction_utils import BatchFeature
from ...utils import PaddingStrategy, TensorType, logging


logger = logging.get_logger(__name__)


class Speech2TextFeatureExtractor(SequenceFeatureExtractor):
    r"""
    Constructs a Speech2Text feature extractor.

    This feature extractor inherits from [`Speech2TextFeatureExtractor`] 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 TorchAudio and applies utterance-level cepstral
    mean and variance normalization to the extracted features.

    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).
        num_mel_bins (`int`, defaults to 80):
            Number of Mel-frequency bins.
        padding_value (`float`, defaults to 0.0):
            The value that is used to fill the padding vectors.
        do_ceptral_normalize (`bool`, *optional*, defaults to `True`):
            Whether or not to apply utterance-level cepstral mean and variance normalization to extracted features.
        normalize_means (`bool`, *optional*, defaults to `True`):
            Whether or not to zero-mean normalize the extracted features.
        normalize_vars (`bool`, *optional*, defaults to `True`):
            Whether or not to unit-variance normalize the extracted features.
    """

    model_input_names = ["input_features", "attention_mask"]

    def __init__(
        self,
        feature_size=80,
        sampling_rate=16000,
        num_mel_bins=80,
        padding_value=0.0,
        do_ceptral_normalize=True,
        normalize_means=True,
        normalize_vars=True,
        **kwargs,
    ):
        super().__init__(feature_size=feature_size, sampling_rate=sampling_rate, padding_value=padding_value, **kwargs)
        self.num_mel_bins = num_mel_bins
        self.do_ceptral_normalize = do_ceptral_normalize
        self.normalize_means = normalize_means
        self.normalize_vars = normalize_vars
        self.return_attention_mask = True

    def _extract_fbank_features(
        self,
        waveform: np.ndarray,
    ) -> np.ndarray:
        """
        Get mel-filter bank features using TorchAudio. Note that TorchAudio requires 16-bit signed integers as inputs
        and hence the waveform should not be normalized before feature extraction.
        """
        waveform = waveform * (2**15)  # Kaldi compliance: 16-bit signed integers
        waveform = torch.from_numpy(waveform).unsqueeze(0)
        features = ta_kaldi.fbank(waveform, num_mel_bins=self.num_mel_bins, sample_frequency=self.sampling_rate)
        return features.numpy()

    @staticmethod
    def utterance_cmvn(
        x: np.ndarray,
        input_length: int,
        normalize_means: Optional[bool] = True,
        normalize_vars: Optional[bool] = True,
        padding_value: float = 0.0,
    ) -> np.ndarray:
        # make sure we normalize float32 arrays
        if normalize_means:
            mean = x[:input_length].mean(axis=0)
            x = np.subtract(x, mean)
        if normalize_vars:
            std = x[:input_length].std(axis=0)
            x = np.divide(x, std)

        if input_length < x.shape[0]:
            x[input_length:] = padding_value

        # make sure array is in float32
        x = x.astype(np.float32)

        return x

    def normalize(
        self, input_features: List[np.ndarray], attention_mask: Optional[np.ndarray] = None
    ) -> List[np.ndarray]:
        lengths = attention_mask.sum(-1) if attention_mask is not None else [x.shape[0] for x in input_features]
        return [
            self.utterance_cmvn(x, n, self.normalize_means, self.normalize_vars, self.padding_value)
            for x, n in zip(input_features, lengths)
        ]

    def __call__(
        self,
        raw_speech: Union[np.ndarray, List[float], List[np.ndarray], List[List[float]]],
        padding: Union[bool, str, PaddingStrategy] = False,
        max_length: Optional[int] = None,
        truncation: bool = False,
        pad_to_multiple_of: Optional[int] = None,
        return_tensors: Optional[Union[str, TensorType]] = None,
        sampling_rate: Optional[int] = None,
        return_attention_mask: Optional[bool] = None,
        **kwargs,
    ) -> BatchFeature:
        """
        Main method to featurize and prepare for the model one or several sequence(s).

        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.
            padding (`bool`, `str` or [`~utils.PaddingStrategy`], *optional*, defaults to `True`):
                Select a strategy to pad the returned sequences (according to the model's padding side and padding
                index) among:

                - `True` or `'longest'`: Pad to the longest sequence in the batch (or no padding if only a single
                  sequence if provided).
                - `'max_length'`: Pad to a maximum length specified with the argument `max_length` or to the maximum
                  acceptable input length for the model if that argument is not provided.
                - `False` or `'do_not_pad'` (default): No padding (i.e., can output a batch with sequences of different
                  lengths).
            max_length (`int`, *optional*):
                Maximum length of the returned list and optionally padding length (see above).
            truncation (`bool`):
                Activates truncation to cut input sequences longer than *max_length* to *max_length*.
            pad_to_multiple_of (`int`, *optional*):
                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 Speech2TextTransformer 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.
            padding_value (`float`, defaults to 0.0):
                The value that is used to fill the padding values / vectors.
        """

        if sampling_rate is not None:
            if sampling_rate != self.sampling_rate:
                raise ValueError(
                    f"The model corresponding to this feature extractor: {self} was trained using a sampling rate of"
                    f" {self.sampling_rate}. Please make sure that the provided `raw_speech` input was sampled with"
                    f" {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) 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 = [raw_speech]

        # extract fbank features
        features = [self._extract_fbank_features(waveform) for waveform in raw_speech]

        # convert into correct format for padding
        encoded_inputs = BatchFeature({"input_features": features})

        padded_inputs = self.pad(
            encoded_inputs,
            padding=padding,
            max_length=max_length,
            truncation=truncation,
            pad_to_multiple_of=pad_to_multiple_of,
            return_attention_mask=return_attention_mask,
            **kwargs,
        )

        # make sure list is in array format
        input_features = padded_inputs.get("input_features")
        if isinstance(input_features[0], list):
            padded_inputs["input_features"] = [np.asarray(feature, dtype=np.float32) for feature in input_features]

        attention_mask = padded_inputs.get("attention_mask")
        if attention_mask is not None:
            padded_inputs["attention_mask"] = [np.asarray(array, dtype=np.int32) for array in attention_mask]

        # Utterance-level cepstral mean and variance normalization
        if self.do_ceptral_normalize:
            attention_mask = (
                np.array(attention_mask, dtype=np.int32)
                if self._get_padding_strategies(padding, max_length=max_length) is not PaddingStrategy.DO_NOT_PAD
                else None
            )
            padded_inputs["input_features"] = self.normalize(
                padded_inputs["input_features"], attention_mask=attention_mask
            )

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

        return padded_inputs
