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Upload feature extractor

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README.md ADDED
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+ ---
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+ library_name: transformers
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+ tags: []
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+ ## Model Details
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+ Users (both direct and downstream) should be made aware of the risks, biases and limitations of the model. More information needed for further recommendations.
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+ ## How to Get Started with the Model
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+ Use the code below to get started with the model.
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+ [More Information Needed]
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+ ## Training Details
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+ ### Training Data
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+ ### Training Procedure
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+ <!-- This relates heavily to the Technical Specifications. Content here should link to that section when it is relevant to the training procedure. -->
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+
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+ #### Preprocessing [optional]
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+ [More Information Needed]
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+ #### Training Hyperparameters
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+ - **Training regime:** [More Information Needed] <!--fp32, fp16 mixed precision, bf16 mixed precision, bf16 non-mixed precision, fp16 non-mixed precision, fp8 mixed precision -->
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+ <!-- This section provides information about throughput, start/end time, checkpoint size if relevant, etc. -->
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+ [More Information Needed]
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+ #### Metrics
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+ ### Results
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+ #### Summary
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+ Carbon emissions can be estimated using the [Machine Learning Impact calculator](https://mlco2.github.io/impact#compute) presented in [Lacoste et al. (2019)](https://arxiv.org/abs/1910.09700).
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+ ## Technical Specifications [optional]
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+ ### Model Architecture and Objective
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+ ### Compute Infrastructure
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feature_extraction_gramt_ambisonics.py ADDED
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+ from typing import Optional, Union
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+ import numpy as np
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+
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+ from transformers import SequenceFeatureExtractor
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+ from transformers import BatchFeature
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+ from transformers.utils import TensorType
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+
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+ import torch
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+ from torch import Tensor
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+ from typing import Callable
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+ from torchaudio.transforms import Spectrogram, MelScale
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+
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+ class FeatureExtractor(torch.nn.Module):
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+
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+ def __init__(
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+ self,
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+ sample_rate: int = 32000,
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+ n_fft: int = 400,
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+ win_length: Optional[int] = None,
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+ hop_length: Optional[int] = None,
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+ f_min: float = 0.0,
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+ f_max: Optional[float] = None,
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+ pad: int = 0,
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+ n_mels: int = 128,
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+ window_fn: Callable[..., Tensor] = torch.hann_window,
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+ power: float = None,
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+ normalized: bool = False,
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+ wkwargs: Optional[dict] = None,
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+ center: bool = True,
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+ pad_mode: str = "reflect",
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+ onesided: Optional[bool] = None,
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+ norm: Optional[str] = None,
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+ mel_scale: str = "htk",
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+ ) -> None:
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+ super().__init__()
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+
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+ self.sample_rate = sample_rate
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+ self.power = power
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+ self.n_fft = n_fft
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+ self.win_length = win_length if win_length is not None else n_fft
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+ self.hop_length = hop_length if hop_length is not None else self.win_length // 2
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+ self.pad = pad
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+ self.power = power
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+ self.normalized = normalized
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+ self.n_mels = n_mels # number of mel frequency bins
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+ self.f_max = f_max
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+ self.f_min = f_min
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+ self.eps = 1e-6
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+ self.spectrogram = Spectrogram(
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+ n_fft=self.n_fft,
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+ win_length=self.win_length,
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+ hop_length=self.hop_length,
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+ pad=self.pad,
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+ window_fn=window_fn,
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+ power=None,
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+ normalized=self.normalized,
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+ wkwargs=wkwargs,
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+ center=center,
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+ pad_mode=pad_mode,
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+ onesided=True,
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+ )
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+ self.mel_scale = MelScale(
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+ self.n_mels, self.sample_rate, self.f_min, self.f_max, self.n_fft // 2 + 1, norm, mel_scale
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+ )
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+ self.processed_spec = None
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+
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+
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+ def _get_foa_intensity_vectors(self, linear_spectra):
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+ """
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+ Convert FOA (First Order Ambisonic) linear spectra to intensity vectors.
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+
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+ Args:
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+ linear_spectra: Complex tensor of shape (batch, freq_bins, 4)
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+ where the 4 channels are [W, X, Y, Z]
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+
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+ Returns:
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+ foa_iv: Tensor of shape (batch, nb_mel_bins * 3)
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+ """
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+
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+ # Extract W channel (omnidirectional component)
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+ W = linear_spectra[: , [0], ...]
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+ XYZ = linear_spectra[:, 1:, ...]
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+
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+ # Compute intensity vectors using complex conjugate
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+ # I = 2 * Re(conj(W) * [X, Y, Z])
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+ I = 2 * torch.real(torch.conj(W) * XYZ)
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+
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+ # Compute energy with epsilon for numerical stability
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+ # E = eps + |W|^2 + (|X|^2 + |Y|^2 + |Z|^2)/3
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+ W_power = torch.squeeze(torch.abs(W) ** 2)
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+ xyz_power = torch.sum(torch.abs(XYZ) ** 2, dim=1)
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+ E = self.eps + W_power + xyz_power
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+
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+ # Normalize intensity vectors
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+
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+ I_norm = I / E.unsqueeze(dim = 1)
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+
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+ foa_iv = self.mel_scale(I_norm)
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+
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+ return foa_iv
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+
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+ def forward(self, audio):
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+ spec = self.spectrogram(audio)
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+ power_spec = torch.abs(spec)**self.power
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+ mel_spec = torch.log(self.mel_scale(power_spec) + self.eps)
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+ foa_aiv = self._get_foa_intensity_vectors(spec)
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+ return torch.cat([mel_spec, foa_aiv], dim = 1)
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+
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+
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+ class AmbisonicsFeatureExtractor(SequenceFeatureExtractor):
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+ r"""
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+ Constructs a Audio Spectrogram Transformer (AST) feature extractor.
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+
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+ This feature extractor inherits from [`~feature_extraction_sequence_utils.SequenceFeatureExtractor`] which contains
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+ most of the main methods. Users should refer to this superclass for more information regarding those methods.
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+
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+ This class extracts mel-filter bank features from raw speech using TorchAudio if installed or using numpy
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+ otherwise, pads/truncates them to a fixed length and normalizes them using a mean and standard deviation.
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+
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+ Args:
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+ feature_size (`int`, *optional*, defaults to 1):
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+ The feature dimension of the extracted features.
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+ sampling_rate (`int`, *optional*, defaults to 16000):
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+ The sampling rate at which the audio files should be digitalized expressed in hertz (Hz).
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+ num_mel_bins (`int`, *optional*, defaults to 128):
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+ Number of Mel-frequency bins.
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+ max_length (`int`, *optional*, defaults to 1024):
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+ Maximum length to which to pad/truncate the extracted features
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+
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+ """
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+ in_channels = 4
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+ feature_extractor_type = "gram-ambisonics"
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+
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+ def __init__(
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+ self,
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+ feature_size=1,
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+ sampling_rate=32000,
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+ num_mel_bins=128,
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+ padding_value=0.0,
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+ **kwargs,
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+ ):
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+ super().__init__(feature_size=feature_size, sampling_rate=sampling_rate, padding_value=padding_value, **kwargs)
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+ self.num_mel_bins = num_mel_bins
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+
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+ def _extract_fbank_features(
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+ self,
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+ waveform: np.ndarray,
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+ ) -> np.ndarray:
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+ """
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+ Get mel-filter bank features using TorchAudio. Note that TorchAudio requires 16-bit signed integers as inputs
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+ and hence the waveform should not be normalized before feature extraction.
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+ """
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+ melspec = FeatureExtractor(
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+ sample_rate=self.sampling_rate,
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+ n_fft=1024,
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+ win_length=1024,
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+ hop_length=self.sampling_rate // 100,
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+ f_min=50,
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+ f_max=self.sampling_rate // 2,
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+ n_mels=self.num_mel_bins,
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+ power=2.0,
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+ )
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+
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+ waveform = torch.tensor(waveform.clone().detach())
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+ # If waveform has two channels, but the channel information is not the first dimension, transpose.
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+ if (waveform.ndim == 2) and (waveform.shape[0] > 100):
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+ waveform = waveform.transpose(1, 0)
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+ if waveform.ndim == 1:
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+ waveform = waveform.unsqueeze(0)
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+
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+ # Handle stereo/mono channels consistently
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+ if waveform.shape[0] == 1:
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+ waveform = torch.cat([waveform, waveform, waveform, waveform], dim = 0).unsqueeze(0)
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+ log_mel = melspec(waveform).transpose(3, 2)[0]
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+ elif waveform.shape[0] == 2:
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+ waveform = waveform[0]
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+ waveform = torch.cat([waveform, waveform, waveform, waveform], dim = 0).unsqueeze(0)
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+ log_mel = melspec(waveform).transpose(3, 2)[0]
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+ return log_mel
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+ elif waveform.shape[0] == 4:
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+ log_mel = melspec(waveform.unsqueeze(0)).transpose(3, 2)[0]
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+ return log_mel
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+ else:
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+ raise Exception("Unknowm channel count")
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+
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+
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+ def _normalize_audio(self, audio_data, target_dBFS=-14.0):
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+ rms = torch.sqrt(torch.mean(audio_data**2)) # Calculate the RMS of the audio
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+ if rms == 0: # Avoid division by zero in case of a completely silent audio
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+ return audio_data
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+ current_dBFS = 20 * torch.log10(rms) # Convert RMS to dBFS
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+ gain_dB = target_dBFS - current_dBFS # Calculate the required gain in dB
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+ gain_linear = 10 ** (gain_dB / 20) # Convert gain from dB to linear scale
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+ normalized_audio = audio_data * gain_linear # Apply the gain to the audio data
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+ return normalized_audio
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+
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+ def __call__(
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+ self,
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+ raw_speech: Union[np.ndarray, list[float], list[np.ndarray], list[list[float]]],
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+ sampling_rate: Optional[int] = None,
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+ return_tensors: Optional[Union[str, TensorType]] = None,
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+ **kwargs,
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+ ) -> BatchFeature:
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+ """
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+ Main method to featurize and prepare for the model one or several sequence(s).
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+
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+ Args:
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+ raw_speech (`np.ndarray`, `list[float]`, `list[np.ndarray]`, `list[list[float]]`):
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+ The sequence or batch of sequences to be padded. Each sequence can be a numpy array, a list of float
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+ values, a list of numpy arrays or a list of list of float values.
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+
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+ sampling_rate (`int`, *optional*):
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+ The sampling rate at which the `raw_speech` input was sampled. It is strongly recommended to pass
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+ `sampling_rate` at the forward call to prevent silent errors.
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+ return_tensors (`str` or [`~utils.TensorType`], *optional*):
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+ If set, will return tensors instead of list of python integers. Acceptable values are:
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+
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+ - `'tf'`: Return TensorFlow `tf.constant` objects.
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+ - `'pt'`: Return PyTorch `torch.Tensor` objects.
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+ - `'np'`: Return Numpy `np.ndarray` objects.
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+ """
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+
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+ if sampling_rate is not None:
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+ if sampling_rate != self.sampling_rate:
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+ raise ValueError(
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+ f"The model corresponding to this feature extractor: {self} was trained using a sampling rate of"
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+ f" {self.sampling_rate}. Please make sure that the provided `raw_speech` input was sampled with"
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+ f" {self.sampling_rate} and not {sampling_rate}."
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+ )
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+
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+ # extract fbank features and pad/truncate to max_length
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+ features = [self._extract_fbank_features(waveform) for waveform in raw_speech]
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+ features = torch.nn.utils.rnn.pad_sequence(features, batch_first=True)
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+ inputs = BatchFeature({"input_values": features})
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+ return inputs
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+
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+
preprocessor_config.json ADDED
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+ {
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+ "auto_map": {
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+ "AutoFeatureExtractor": "feature_extraction_gramt_ambisonics.AmbisonicsFeatureExtractor"
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+ },
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+ "feature_extractor_type": "AmbisonicsFeatureExtractor",
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+ "feature_size": 1,
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+ "num_mel_bins": 128,
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+ "padding_side": "right",
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+ "padding_value": 0.0,
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+ "return_attention_mask": true,
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+ "sampling_rate": 32000
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+ }