Upload feature extractor

preprocessor_config.json

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+{
+  "auto_map": {
+    "AutoFeatureExtractor": "processing_bestrq_conformer.ModifiedWhisperFeatureExtractor"
+  },
+  "chunk_length": 30,
+  "feature_extractor_type": "ModifiedWhisperFeatureExtractor",
+  "feature_size": 80,
+  "hop_length": 160,
+  "n_fft": 400,
+  "n_samples": 480000,
+  "nb_max_frames": 3000,
+  "padding_side": "right",
+  "padding_value": 0.0,
+  "return_attention_mask": false,
+  "sampling_rate": 16000
+}

processing_bestrq_conformer.py

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+# 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
+#
+# 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 MERaLiON-SpeechEncoder, modified from original WhisperFeatureExtractor
+"""
+
+from typing import List, Optional, Union
+
+import numpy as np
+
+from transformers import is_torch_available
+from transformers.audio_utils import mel_filter_bank, spectrogram, window_function
+from transformers.feature_extraction_sequence_utils import SequenceFeatureExtractor
+from transformers.feature_extraction_utils import BatchFeature
+from transformers.utils import TensorType, logging
+
+
+if is_torch_available():
+    import torch
+
+logger = logging.get_logger(__name__)
+
+
+class ModifiedWhisperFeatureExtractor(SequenceFeatureExtractor):
+    r"""
+    Constructs a modified 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.
+
+    Differences from WhisperFeatureExtractor:
+    - mel_filter_bank
+        - norm: "slaney" -> None
+        - mel_scale: "slaney" -> "htk"
+    - still uses log scaling and clamp but removes additional min-max/mean normalization
+
+    Args:
+        feature_size (`int`, *optional*, defaults to 80):
+            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.
+        chunk_length (`int`, *optional*, defaults to 30):
+            The maximum number of chunks of `sampling_rate` samples used to trim and pad longer or shorter audio
+            sequences.
+        n_fft (`int`, *optional*, 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_values"]
+
+    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=None,
+            mel_scale="htk",
+        )
+
+    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_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 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`, *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`):
+                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_values": raw_speech})
+
+        # convert into correct format for padding
+
+        padded_inputs = self.pad(  #whisper pads first then transform, while we do the reverse
+            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_values"] = self.zero_mean_unit_var_norm(
+                padded_inputs["input_values"],
+                attention_mask=padded_inputs["attention_mask"],
+                padding_value=self.padding_value,
+            )
+            padded_inputs["input_values"] = np.stack(padded_inputs["input_values"], axis=0)
+
+        # make sure list is in array format
+        input_values = padded_inputs.get("input_values").transpose(2, 0, 1)
+
+        extract_fbank_features = (
+            self._torch_extract_fbank_features if is_torch_available() else self._np_extract_fbank_features
+        )
+        input_values = extract_fbank_features(input_values[0], device)
+
+        if isinstance(input_values[0], List):
+            padded_inputs["input_values"] = [np.asarray(feature, dtype=np.float32) for feature in input_values]
+
+        else:
+            padded_inputs["input_values"] = input_values
+
+        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