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Source code for torchaudio.transforms._transforms

# -*- coding: utf-8 -*-

import math
import warnings
from typing import Callable, Optional, Union

import torch
from torch import Tensor
from torch.nn.modules.lazy import LazyModuleMixin
from torch.nn.parameter import UninitializedParameter

from torchaudio import functional as F
from torchaudio.functional.functional import (
    _apply_sinc_resample_kernel,
    _fix_waveform_shape,
    _get_sinc_resample_kernel,
    _stretch_waveform,
)

__all__ = []


[docs]class Spectrogram(torch.nn.Module): r"""Create a spectrogram from a audio signal. .. devices:: CPU CUDA .. properties:: Autograd TorchScript Args: n_fft (int, optional): Size of FFT, creates ``n_fft // 2 + 1`` bins. (Default: ``400``) win_length (int or None, optional): Window size. (Default: ``n_fft``) hop_length (int or None, optional): Length of hop between STFT windows. (Default: ``win_length // 2``) pad (int, optional): Two sided padding of signal. (Default: ``0``) window_fn (Callable[..., Tensor], optional): A function to create a window tensor that is applied/multiplied to each frame/window. (Default: ``torch.hann_window``) power (float or None, optional): Exponent for the magnitude spectrogram, (must be > 0) e.g., 1 for energy, 2 for power, etc. If None, then the complex spectrum is returned instead. (Default: ``2``) normalized (bool or str, optional): Whether to normalize by magnitude after stft. If input is str, choices are ``"window"`` and ``"frame_length"``, if specific normalization type is desirable. ``True`` maps to ``"window"``. (Default: ``False``) wkwargs (dict or None, optional): Arguments for window function. (Default: ``None``) center (bool, optional): whether to pad :attr:`waveform` on both sides so that the :math:`t`-th frame is centered at time :math:`t \times \text{hop\_length}`. (Default: ``True``) pad_mode (string, optional): controls the padding method used when :attr:`center` is ``True``. (Default: ``"reflect"``) onesided (bool, optional): controls whether to return half of results to avoid redundancy (Default: ``True``) return_complex (bool, optional): Deprecated and not used. Example >>> waveform, sample_rate = torchaudio.load("test.wav", normalize=True) >>> transform = torchaudio.transforms.Spectrogram(n_fft=800) >>> spectrogram = transform(waveform) """ __constants__ = ["n_fft", "win_length", "hop_length", "pad", "power", "normalized"] def __init__( self, n_fft: int = 400, win_length: Optional[int] = None, hop_length: Optional[int] = None, pad: int = 0, window_fn: Callable[..., Tensor] = torch.hann_window, power: Optional[float] = 2.0, normalized: Union[bool, str] = False, wkwargs: Optional[dict] = None, center: bool = True, pad_mode: str = "reflect", onesided: bool = True, return_complex: Optional[bool] = None, ) -> None: super(Spectrogram, self).__init__() torch._C._log_api_usage_once("torchaudio.transforms.Spectrogram") self.n_fft = n_fft # number of FFT bins. the returned STFT result will have n_fft // 2 + 1 # number of frequencies due to onesided=True in torch.stft self.win_length = win_length if win_length is not None else n_fft self.hop_length = hop_length if hop_length is not None else self.win_length // 2 window = window_fn(self.win_length) if wkwargs is None else window_fn(self.win_length, **wkwargs) self.register_buffer("window", window) self.pad = pad self.power = power self.normalized = normalized self.center = center self.pad_mode = pad_mode self.onesided = onesided if return_complex is not None: warnings.warn( "`return_complex` argument is now deprecated and is not effective." "`torchaudio.transforms.Spectrogram(power=None)` always returns a tensor with " "complex dtype. Please remove the argument in the function call." )
[docs] def forward(self, waveform: Tensor) -> Tensor: r""" Args: waveform (Tensor): Tensor of audio of dimension (..., time). Returns: Tensor: Dimension (..., freq, time), where freq is ``n_fft // 2 + 1`` where ``n_fft`` is the number of Fourier bins, and time is the number of window hops (n_frame). """ return F.spectrogram( waveform, self.pad, self.window, self.n_fft, self.hop_length, self.win_length, self.power, self.normalized, self.center, self.pad_mode, self.onesided, )
[docs]class InverseSpectrogram(torch.nn.Module): r"""Create an inverse spectrogram to recover an audio signal from a spectrogram. .. devices:: CPU CUDA .. properties:: Autograd TorchScript Args: n_fft (int, optional): Size of FFT, creates ``n_fft // 2 + 1`` bins. (Default: ``400``) win_length (int or None, optional): Window size. (Default: ``n_fft``) hop_length (int or None, optional): Length of hop between STFT windows. (Default: ``win_length // 2``) pad (int, optional): Two sided padding of signal. (Default: ``0``) window_fn (Callable[..., Tensor], optional): A function to create a window tensor that is applied/multiplied to each frame/window. (Default: ``torch.hann_window``) normalized (bool or str, optional): Whether the stft output was normalized by magnitude. If input is str, choices are ``"window"`` and ``"frame_length"``, dependent on normalization mode. ``True`` maps to ``"window"``. (Default: ``False``) wkwargs (dict or None, optional): Arguments for window function. (Default: ``None``) center (bool, optional): whether the signal in spectrogram was padded on both sides so that the :math:`t`-th frame is centered at time :math:`t \times \text{hop\_length}`. (Default: ``True``) pad_mode (string, optional): controls the padding method used when :attr:`center` is ``True``. (Default: ``"reflect"``) onesided (bool, optional): controls whether spectrogram was used to return half of results to avoid redundancy (Default: ``True``) Example >>> batch, freq, time = 2, 257, 100 >>> length = 25344 >>> spectrogram = torch.randn(batch, freq, time, dtype=torch.cdouble) >>> transform = transforms.InverseSpectrogram(n_fft=512) >>> waveform = transform(spectrogram, length) """ __constants__ = ["n_fft", "win_length", "hop_length", "pad", "power", "normalized"] def __init__( self, n_fft: int = 400, win_length: Optional[int] = None, hop_length: Optional[int] = None, pad: int = 0, window_fn: Callable[..., Tensor] = torch.hann_window, normalized: Union[bool, str] = False, wkwargs: Optional[dict] = None, center: bool = True, pad_mode: str = "reflect", onesided: bool = True, ) -> None: super(InverseSpectrogram, self).__init__() self.n_fft = n_fft # number of FFT bins. the returned STFT result will have n_fft // 2 + 1 # number of frequencies due to onesided=True in torch.stft self.win_length = win_length if win_length is not None else n_fft self.hop_length = hop_length if hop_length is not None else self.win_length // 2 window = window_fn(self.win_length) if wkwargs is None else window_fn(self.win_length, **wkwargs) self.register_buffer("window", window) self.pad = pad self.normalized = normalized self.center = center self.pad_mode = pad_mode self.onesided = onesided
[docs] def forward(self, spectrogram: Tensor, length: Optional[int] = None) -> Tensor: r""" Args: spectrogram (Tensor): Complex tensor of audio of dimension (..., freq, time). length (int or None, optional): The output length of the waveform. Returns: Tensor: Dimension (..., time), Least squares estimation of the original signal. """ return F.inverse_spectrogram( spectrogram, length, self.pad, self.window, self.n_fft, self.hop_length, self.win_length, self.normalized, self.center, self.pad_mode, self.onesided, )
[docs]class GriffinLim(torch.nn.Module): r"""Compute waveform from a linear scale magnitude spectrogram using the Griffin-Lim transformation. .. devices:: CPU CUDA .. properties:: Autograd TorchScript Implementation ported from *librosa* :cite:`brian_mcfee-proc-scipy-2015`, *A fast Griffin-Lim algorithm* :cite:`6701851` and *Signal estimation from modified short-time Fourier transform* :cite:`1172092`. Args: n_fft (int, optional): Size of FFT, creates ``n_fft // 2 + 1`` bins. (Default: ``400``) n_iter (int, optional): Number of iteration for phase recovery process. (Default: ``32``) win_length (int or None, optional): Window size. (Default: ``n_fft``) hop_length (int or None, optional): Length of hop between STFT windows. (Default: ``win_length // 2``) window_fn (Callable[..., Tensor], optional): A function to create a window tensor that is applied/multiplied to each frame/window. (Default: ``torch.hann_window``) power (float, optional): Exponent for the magnitude spectrogram, (must be > 0) e.g., 1 for energy, 2 for power, etc. (Default: ``2``) wkwargs (dict or None, optional): Arguments for window function. (Default: ``None``) momentum (float, optional): The momentum parameter for fast Griffin-Lim. Setting this to 0 recovers the original Griffin-Lim method. Values near 1 can lead to faster convergence, but above 1 may not converge. (Default: ``0.99``) length (int, optional): Array length of the expected output. (Default: ``None``) rand_init (bool, optional): Initializes phase randomly if True and to zero otherwise. (Default: ``True``) Example >>> batch, freq, time = 2, 257, 100 >>> spectrogram = torch.randn(batch, freq, time) >>> transform = transforms.GriffinLim(n_fft=512) >>> waveform = transform(spectrogram) """ __constants__ = ["n_fft", "n_iter", "win_length", "hop_length", "power", "length", "momentum", "rand_init"] def __init__( self, n_fft: int = 400, n_iter: int = 32, win_length: Optional[int] = None, hop_length: Optional[int] = None, window_fn: Callable[..., Tensor] = torch.hann_window, power: float = 2.0, wkwargs: Optional[dict] = None, momentum: float = 0.99, length: Optional[int] = None, rand_init: bool = True, ) -> None: super(GriffinLim, self).__init__() if not (0 <= momentum < 1): raise ValueError("momentum must be in the range [0, 1). Found: {}".format(momentum)) self.n_fft = n_fft self.n_iter = n_iter self.win_length = win_length if win_length is not None else n_fft self.hop_length = hop_length if hop_length is not None else self.win_length // 2 window = window_fn(self.win_length) if wkwargs is None else window_fn(self.win_length, **wkwargs) self.register_buffer("window", window) self.length = length self.power = power self.momentum = momentum self.rand_init = rand_init
[docs] def forward(self, specgram: Tensor) -> Tensor: r""" Args: specgram (Tensor): A magnitude-only STFT spectrogram of dimension (..., freq, frames) where freq is ``n_fft // 2 + 1``. Returns: Tensor: waveform of (..., time), where time equals the ``length`` parameter if given. """ return F.griffinlim( specgram, self.window, self.n_fft, self.hop_length, self.win_length, self.power, self.n_iter, self.momentum, self.length, self.rand_init, )
[docs]class AmplitudeToDB(torch.nn.Module): r"""Turn a tensor from the power/amplitude scale to the decibel scale. .. devices:: CPU CUDA .. properties:: Autograd TorchScript This output depends on the maximum value in the input tensor, and so may return different values for an audio clip split into snippets vs. a a full clip. Args: stype (str, optional): scale of input tensor (``"power"`` or ``"magnitude"``). The power being the elementwise square of the magnitude. (Default: ``"power"``) top_db (float or None, optional): minimum negative cut-off in decibels. A reasonable number is 80. (Default: ``None``) Example >>> waveform, sample_rate = torchaudio.load("test.wav", normalize=True) >>> transform = transforms.AmplitudeToDB(stype="amplitude", top_db=80) >>> waveform_db = transform(waveform) """ __constants__ = ["multiplier", "amin", "ref_value", "db_multiplier"] def __init__(self, stype: str = "power", top_db: Optional[float] = None) -> None: super(AmplitudeToDB, self).__init__() self.stype = stype if top_db is not None and top_db < 0: raise ValueError("top_db must be positive value") self.top_db = top_db self.multiplier = 10.0 if stype == "power" else 20.0 self.amin = 1e-10 self.ref_value = 1.0 self.db_multiplier = math.log10(max(self.amin, self.ref_value))
[docs] def forward(self, x: Tensor) -> Tensor: r"""Numerically stable implementation from Librosa. https://librosa.org/doc/latest/generated/librosa.amplitude_to_db.html Args: x (Tensor): Input tensor before being converted to decibel scale. Returns: Tensor: Output tensor in decibel scale. """ return F.amplitude_to_DB(x, self.multiplier, self.amin, self.db_multiplier, self.top_db)
[docs]class MelScale(torch.nn.Module): r"""Turn a normal STFT into a mel frequency STFT with triangular filter banks. .. devices:: CPU CUDA .. properties:: Autograd TorchScript Args: n_mels (int, optional): Number of mel filterbanks. (Default: ``128``) sample_rate (int, optional): Sample rate of audio signal. (Default: ``16000``) f_min (float, optional): Minimum frequency. (Default: ``0.``) f_max (float or None, optional): Maximum frequency. (Default: ``sample_rate // 2``) n_stft (int, optional): Number of bins in STFT. See ``n_fft`` in :class:`Spectrogram`. (Default: ``201``) norm (str or None, optional): If ``"slaney"``, divide the triangular mel weights by the width of the mel band (area normalization). (Default: ``None``) mel_scale (str, optional): Scale to use: ``htk`` or ``slaney``. (Default: ``htk``) Example >>> waveform, sample_rate = torchaudio.load("test.wav", normalize=True) >>> spectrogram_transform = transforms.Spectrogram(n_fft=1024) >>> spectrogram = spectrogram_transform(waveform) >>> melscale_transform = transforms.MelScale(sample_rate=sample_rate, n_stft=1024 // 2 + 1) >>> melscale_spectrogram = melscale_transform(spectrogram) See also: :py:func:`torchaudio.functional.melscale_fbanks` - The function used to generate the filter banks. """ __constants__ = ["n_mels", "sample_rate", "f_min", "f_max"] def __init__( self, n_mels: int = 128, sample_rate: int = 16000, f_min: float = 0.0, f_max: Optional[float] = None, n_stft: int = 201, norm: Optional[str] = None, mel_scale: str = "htk", ) -> None: super(MelScale, self).__init__() self.n_mels = n_mels self.sample_rate = sample_rate self.f_max = f_max if f_max is not None else float(sample_rate // 2) self.f_min = f_min self.norm = norm self.mel_scale = mel_scale if f_min > self.f_max: raise ValueError("Require f_min: {} <= f_max: {}".format(f_min, self.f_max)) fb = F.melscale_fbanks(n_stft, self.f_min, self.f_max, self.n_mels, self.sample_rate, self.norm, self.mel_scale) self.register_buffer("fb", fb)
[docs] def forward(self, specgram: Tensor) -> Tensor: r""" Args: specgram (Tensor): A spectrogram STFT of dimension (..., freq, time). Returns: Tensor: Mel frequency spectrogram of size (..., ``n_mels``, time). """ # (..., time, freq) dot (freq, n_mels) -> (..., n_mels, time) mel_specgram = torch.matmul(specgram.transpose(-1, -2), self.fb).transpose(-1, -2) return mel_specgram
class InverseMelScale(torch.nn.Module): r"""Estimate a STFT in normal frequency domain from mel frequency domain. .. devices:: CPU CUDA It minimizes the euclidian norm between the input mel-spectrogram and the product between the estimated spectrogram and the filter banks using SGD. Args: n_stft (int): Number of bins in STFT. See ``n_fft`` in :class:`Spectrogram`. n_mels (int, optional): Number of mel filterbanks. (Default: ``128``) sample_rate (int, optional): Sample rate of audio signal. (Default: ``16000``) f_min (float, optional): Minimum frequency. (Default: ``0.``) f_max (float or None, optional): Maximum frequency. (Default: ``sample_rate // 2``) max_iter (int, optional): Maximum number of optimization iterations. (Default: ``100000``) tolerance_loss (float, optional): Value of loss to stop optimization at. (Default: ``1e-5``) tolerance_change (float, optional): Difference in losses to stop optimization at. (Default: ``1e-8``) sgdargs (dict or None, optional): Arguments for the SGD optimizer. (Default: ``None``) norm (str or None, optional): If "slaney", divide the triangular mel weights by the width of the mel band (area normalization). (Default: ``None``) mel_scale (str, optional): Scale to use: ``htk`` or ``slaney``. (Default: ``htk``) Example >>> waveform, sample_rate = torchaudio.load("test.wav", normalize=True) >>> mel_spectrogram_transform = transforms.MelSpectrogram(sample_rate, n_fft=1024) >>> mel_spectrogram = mel_spectrogram_transform(waveform) >>> inverse_melscale_transform = transforms.InverseMelScale(n_stft=1024 // 2 + 1) >>> spectrogram = inverse_melscale_transform(mel_spectrogram) """ __constants__ = [ "n_stft", "n_mels", "sample_rate", "f_min", "f_max", "max_iter", "tolerance_loss", "tolerance_change", "sgdargs", ] def __init__( self, n_stft: int, n_mels: int = 128, sample_rate: int = 16000, f_min: float = 0.0, f_max: Optional[float] = None, max_iter: int = 100000, tolerance_loss: float = 1e-5, tolerance_change: float = 1e-8, sgdargs: Optional[dict] = None, norm: Optional[str] = None, mel_scale: str = "htk", ) -> None: super(InverseMelScale, self).__init__() self.n_mels = n_mels self.sample_rate = sample_rate self.f_max = f_max or float(sample_rate // 2) self.f_min = f_min self.max_iter = max_iter self.tolerance_loss = tolerance_loss self.tolerance_change = tolerance_change self.sgdargs = sgdargs or {"lr": 0.1, "momentum": 0.9} if f_min > self.f_max: raise ValueError("Require f_min: {} <= f_max: {}".format(f_min, self.f_max)) fb = F.melscale_fbanks(n_stft, self.f_min, self.f_max, self.n_mels, self.sample_rate, norm, mel_scale) self.register_buffer("fb", fb) def forward(self, melspec: Tensor) -> Tensor: r""" Args: melspec (Tensor): A Mel frequency spectrogram of dimension (..., ``n_mels``, time) Returns: Tensor: Linear scale spectrogram of size (..., freq, time) """ # pack batch shape = melspec.size() melspec = melspec.view(-1, shape[-2], shape[-1]) n_mels, time = shape[-2], shape[-1] freq, _ = self.fb.size() # (freq, n_mels) melspec = melspec.transpose(-1, -2) if self.n_mels != n_mels: raise ValueError("Expected an input with {} mel bins. Found: {}".format(self.n_mels, n_mels)) specgram = torch.rand( melspec.size()[0], time, freq, requires_grad=True, dtype=melspec.dtype, device=melspec.device ) optim = torch.optim.SGD([specgram], **self.sgdargs) loss = float("inf") for _ in range(self.max_iter): optim.zero_grad() diff = melspec - specgram.matmul(self.fb) new_loss = diff.pow(2).sum(axis=-1).mean() # take sum over mel-frequency then average over other dimensions # so that loss threshold is applied par unit timeframe new_loss.backward() optim.step() specgram.data = specgram.data.clamp(min=0) new_loss = new_loss.item() if new_loss < self.tolerance_loss or abs(loss - new_loss) < self.tolerance_change: break loss = new_loss specgram.requires_grad_(False) specgram = specgram.clamp(min=0).transpose(-1, -2) # unpack batch specgram = specgram.view(shape[:-2] + (freq, time)) return specgram
[docs]class MelSpectrogram(torch.nn.Module): r"""Create MelSpectrogram for a raw audio signal. .. devices:: CPU CUDA .. properties:: Autograd TorchScript This is a composition of :py:func:`torchaudio.transforms.Spectrogram` and and :py:func:`torchaudio.transforms.MelScale`. Sources * https://gist.github.com/kastnerkyle/179d6e9a88202ab0a2fe * https://timsainb.github.io/spectrograms-mfccs-and-inversion-in-python.html * http://haythamfayek.com/2016/04/21/speech-processing-for-machine-learning.html Args: sample_rate (int, optional): Sample rate of audio signal. (Default: ``16000``) n_fft (int, optional): Size of FFT, creates ``n_fft // 2 + 1`` bins. (Default: ``400``) win_length (int or None, optional): Window size. (Default: ``n_fft``) hop_length (int or None, optional): Length of hop between STFT windows. (Default: ``win_length // 2``) f_min (float, optional): Minimum frequency. (Default: ``0.``) f_max (float or None, optional): Maximum frequency. (Default: ``None``) pad (int, optional): Two sided padding of signal. (Default: ``0``) n_mels (int, optional): Number of mel filterbanks. (Default: ``128``) window_fn (Callable[..., Tensor], optional): A function to create a window tensor that is applied/multiplied to each frame/window. (Default: ``torch.hann_window``) power (float, optional): Exponent for the magnitude spectrogram, (must be > 0) e.g., 1 for energy, 2 for power, etc. (Default: ``2``) normalized (bool, optional): Whether to normalize by magnitude after stft. (Default: ``False``) wkwargs (Dict[..., ...] or None, optional): Arguments for window function. (Default: ``None``) center (bool, optional): whether to pad :attr:`waveform` on both sides so that the :math:`t`-th frame is centered at time :math:`t \times \text{hop\_length}`. (Default: ``True``) pad_mode (string, optional): controls the padding method used when :attr:`center` is ``True``. (Default: ``"reflect"``) onesided: Deprecated and unused. norm (str or None, optional): If "slaney", divide the triangular mel weights by the width of the mel band (area normalization). (Default: ``None``) mel_scale (str, optional): Scale to use: ``htk`` or ``slaney``. (Default: ``htk``) Example >>> waveform, sample_rate = torchaudio.load("test.wav", normalize=True) >>> transform = transforms.MelSpectrogram(sample_rate) >>> mel_specgram = transform(waveform) # (channel, n_mels, time) See also: :py:func:`torchaudio.functional.melscale_fbanks` - The function used to generate the filter banks. """ __constants__ = ["sample_rate", "n_fft", "win_length", "hop_length", "pad", "n_mels", "f_min"] def __init__( self, sample_rate: int = 16000, n_fft: int = 400, win_length: Optional[int] = None, hop_length: Optional[int] = None, f_min: float = 0.0, f_max: Optional[float] = None, pad: int = 0, n_mels: int = 128, window_fn: Callable[..., Tensor] = torch.hann_window, power: float = 2.0, normalized: bool = False, wkwargs: Optional[dict] = None, center: bool = True, pad_mode: str = "reflect", onesided: Optional[bool] = None, norm: Optional[str] = None, mel_scale: str = "htk", ) -> None: super(MelSpectrogram, self).__init__() torch._C._log_api_usage_once("torchaudio.transforms.MelSpectrogram") if onesided is not None: warnings.warn( "Argument 'onesided' has been deprecated and has no influence on the behavior of this module." ) self.sample_rate = sample_rate self.n_fft = n_fft self.win_length = win_length if win_length is not None else n_fft self.hop_length = hop_length if hop_length is not None else self.win_length // 2 self.pad = pad self.power = power self.normalized = normalized self.n_mels = n_mels # number of mel frequency bins self.f_max = f_max self.f_min = f_min self.spectrogram = Spectrogram( n_fft=self.n_fft, win_length=self.win_length, hop_length=self.hop_length, pad=self.pad, window_fn=window_fn, power=self.power, normalized=self.normalized, wkwargs=wkwargs, center=center, pad_mode=pad_mode, onesided=True, ) self.mel_scale = MelScale( self.n_mels, self.sample_rate, self.f_min, self.f_max, self.n_fft // 2 + 1, norm, mel_scale )
[docs] def forward(self, waveform: Tensor) -> Tensor: r""" Args: waveform (Tensor): Tensor of audio of dimension (..., time). Returns: Tensor: Mel frequency spectrogram of size (..., ``n_mels``, time). """ specgram = self.spectrogram(waveform) mel_specgram = self.mel_scale(specgram) return mel_specgram
[docs]class MFCC(torch.nn.Module): r"""Create the Mel-frequency cepstrum coefficients from an audio signal. .. devices:: CPU CUDA .. properties:: Autograd TorchScript By default, this calculates the MFCC on the DB-scaled Mel spectrogram. This is not the textbook implementation, but is implemented here to give consistency with librosa. This output depends on the maximum value in the input spectrogram, and so may return different values for an audio clip split into snippets vs. a a full clip. Args: sample_rate (int, optional): Sample rate of audio signal. (Default: ``16000``) n_mfcc (int, optional): Number of mfc coefficients to retain. (Default: ``40``) dct_type (int, optional): type of DCT (discrete cosine transform) to use. (Default: ``2``) norm (str, optional): norm to use. (Default: ``"ortho"``) log_mels (bool, optional): whether to use log-mel spectrograms instead of db-scaled. (Default: ``False``) melkwargs (dict or None, optional): arguments for MelSpectrogram. (Default: ``None``) Example >>> waveform, sample_rate = torchaudio.load("test.wav", normalize=True) >>> transform = transforms.MFCC( >>> sample_rate=sample_rate, >>> n_mfcc=13, >>> melkwargs={"n_fft": 400, "hop_length": 160, "n_mels": 23, "center": False}, >>> ) >>> mfcc = transform(waveform) See also: :py:func:`torchaudio.functional.melscale_fbanks` - The function used to generate the filter banks. """ __constants__ = ["sample_rate", "n_mfcc", "dct_type", "top_db", "log_mels"] def __init__( self, sample_rate: int = 16000, n_mfcc: int = 40, dct_type: int = 2, norm: str = "ortho", log_mels: bool = False, melkwargs: Optional[dict] = None, ) -> None: super(MFCC, self).__init__() supported_dct_types = [2] if dct_type not in supported_dct_types: raise ValueError("DCT type not supported: {}".format(dct_type)) self.sample_rate = sample_rate self.n_mfcc = n_mfcc self.dct_type = dct_type self.norm = norm self.top_db = 80.0 self.amplitude_to_DB = AmplitudeToDB("power", self.top_db) melkwargs = melkwargs or {} self.MelSpectrogram = MelSpectrogram(sample_rate=self.sample_rate, **melkwargs) if self.n_mfcc > self.MelSpectrogram.n_mels: raise ValueError("Cannot select more MFCC coefficients than # mel bins") dct_mat = F.create_dct(self.n_mfcc, self.MelSpectrogram.n_mels, self.norm) self.register_buffer("dct_mat", dct_mat) self.log_mels = log_mels
[docs] def forward(self, waveform: Tensor) -> Tensor: r""" Args: waveform (Tensor): Tensor of audio of dimension (..., time). Returns: Tensor: specgram_mel_db of size (..., ``n_mfcc``, time). """ mel_specgram = self.MelSpectrogram(waveform) if self.log_mels: log_offset = 1e-6 mel_specgram = torch.log(mel_specgram + log_offset) else: mel_specgram = self.amplitude_to_DB(mel_specgram) # (..., time, n_mels) dot (n_mels, n_mfcc) -> (..., n_nfcc, time) mfcc = torch.matmul(mel_specgram.transpose(-1, -2), self.dct_mat).transpose(-1, -2) return mfcc
[docs]class LFCC(torch.nn.Module): r"""Create the linear-frequency cepstrum coefficients from an audio signal. .. devices:: CPU CUDA .. properties:: Autograd TorchScript By default, this calculates the LFCC on the DB-scaled linear filtered spectrogram. This is not the textbook implementation, but is implemented here to give consistency with librosa. This output depends on the maximum value in the input spectrogram, and so may return different values for an audio clip split into snippets vs. a a full clip. Args: sample_rate (int, optional): Sample rate of audio signal. (Default: ``16000``) n_filter (int, optional): Number of linear filters to apply. (Default: ``128``) n_lfcc (int, optional): Number of lfc coefficients to retain. (Default: ``40``) f_min (float, optional): Minimum frequency. (Default: ``0.``) f_max (float or None, optional): Maximum frequency. (Default: ``None``) dct_type (int, optional): type of DCT (discrete cosine transform) to use. (Default: ``2``) norm (str, optional): norm to use. (Default: ``"ortho"``) log_lf (bool, optional): whether to use log-lf spectrograms instead of db-scaled. (Default: ``False``) speckwargs (dict or None, optional): arguments for Spectrogram. (Default: ``None``) Example >>> waveform, sample_rate = torchaudio.load("test.wav", normalize=True) >>> transform = transforms.LFCC( >>> sample_rate=sample_rate, >>> n_lfcc=13, >>> speckwargs={"n_fft": 400, "hop_length": 160, "center": False}, >>> ) >>> lfcc = transform(waveform) See also: :py:func:`torchaudio.functional.linear_fbanks` - The function used to generate the filter banks. """ __constants__ = ["sample_rate", "n_filter", "n_lfcc", "dct_type", "top_db", "log_lf"] def __init__( self, sample_rate: int = 16000, n_filter: int = 128, f_min: float = 0.0, f_max: Optional[float] = None, n_lfcc: int = 40, dct_type: int = 2, norm: str = "ortho", log_lf: bool = False, speckwargs: Optional[dict] = None, ) -> None: super(LFCC, self).__init__() supported_dct_types = [2] if dct_type not in supported_dct_types: raise ValueError("DCT type not supported: {}".format(dct_type)) self.sample_rate = sample_rate self.f_min = f_min self.f_max = f_max if f_max is not None else float(sample_rate // 2) self.n_filter = n_filter self.n_lfcc = n_lfcc self.dct_type = dct_type self.norm = norm self.top_db = 80.0 self.amplitude_to_DB = AmplitudeToDB("power", self.top_db) speckwargs = speckwargs or {} self.Spectrogram = Spectrogram(**speckwargs) if self.n_lfcc > self.Spectrogram.n_fft: raise ValueError("Cannot select more LFCC coefficients than # fft bins") filter_mat = F.linear_fbanks( n_freqs=self.Spectrogram.n_fft // 2 + 1, f_min=self.f_min, f_max=self.f_max, n_filter=self.n_filter, sample_rate=self.sample_rate, ) self.register_buffer("filter_mat", filter_mat) dct_mat = F.create_dct(self.n_lfcc, self.n_filter, self.norm) self.register_buffer("dct_mat", dct_mat) self.log_lf = log_lf
[docs] def forward(self, waveform: Tensor) -> Tensor: r""" Args: waveform (Tensor): Tensor of audio of dimension (..., time). Returns: Tensor: Linear Frequency Cepstral Coefficients of size (..., ``n_lfcc``, time). """ specgram = self.Spectrogram(waveform) # (..., time, freq) dot (freq, n_filter) -> (..., n_filter, time) specgram = torch.matmul(specgram.transpose(-1, -2), self.filter_mat).transpose(-1, -2) if self.log_lf: log_offset = 1e-6 specgram = torch.log(specgram + log_offset) else: specgram = self.amplitude_to_DB(specgram) # (..., time, n_filter) dot (n_filter, n_lfcc) -> (..., n_lfcc, time) lfcc = torch.matmul(specgram.transpose(-1, -2), self.dct_mat).transpose(-1, -2) return lfcc
[docs]class MuLawEncoding(torch.nn.Module): r"""Encode signal based on mu-law companding. .. devices:: CPU CUDA .. properties:: TorchScript For more info see the `Wikipedia Entry <https://en.wikipedia.org/wiki/%CE%9C-law_algorithm>`_ This algorithm assumes the signal has been scaled to between -1 and 1 and returns a signal encoded with values from 0 to quantization_channels - 1 Args: quantization_channels (int, optional): Number of channels. (Default: ``256``) Example >>> waveform, sample_rate = torchaudio.load("test.wav", normalize=True) >>> transform = torchaudio.transforms.MuLawEncoding(quantization_channels=512) >>> mulawtrans = transform(waveform) """ __constants__ = ["quantization_channels"] def __init__(self, quantization_channels: int = 256) -> None: super(MuLawEncoding, self).__init__() self.quantization_channels = quantization_channels
[docs] def forward(self, x: Tensor) -> Tensor: r""" Args: x (Tensor): A signal to be encoded. Returns: Tensor: An encoded signal. """ return F.mu_law_encoding(x, self.quantization_channels)
[docs]class MuLawDecoding(torch.nn.Module): r"""Decode mu-law encoded signal. .. devices:: CPU CUDA .. properties:: TorchScript For more info see the `Wikipedia Entry <https://en.wikipedia.org/wiki/%CE%9C-law_algorithm>`_ This expects an input with values between 0 and ``quantization_channels - 1`` and returns a signal scaled between -1 and 1. Args: quantization_channels (int, optional): Number of channels. (Default: ``256``) Example >>> waveform, sample_rate = torchaudio.load("test.wav", normalize=True) >>> transform = torchaudio.transforms.MuLawDecoding(quantization_channels=512) >>> mulawtrans = transform(waveform) """ __constants__ = ["quantization_channels"] def __init__(self, quantization_channels: int = 256) -> None: super(MuLawDecoding, self).__init__() self.quantization_channels = quantization_channels
[docs] def forward(self, x_mu: Tensor) -> Tensor: r""" Args: x_mu (Tensor): A mu-law encoded signal which needs to be decoded. Returns: Tensor: The signal decoded. """ return F.mu_law_decoding(x_mu, self.quantization_channels)
[docs]class Resample(torch.nn.Module): r"""Resample a signal from one frequency to another. A resampling method can be given. .. devices:: CPU CUDA .. properties:: Autograd TorchScript Note: If resampling on waveforms of higher precision than float32, there may be a small loss of precision because the kernel is cached once as float32. If high precision resampling is important for your application, the functional form will retain higher precision, but run slower because it does not cache the kernel. Alternatively, you could rewrite a transform that caches a higher precision kernel. Args: orig_freq (int, optional): The original frequency of the signal. (Default: ``16000``) new_freq (int, optional): The desired frequency. (Default: ``16000``) resampling_method (str, optional): The resampling method to use. Options: [``sinc_interpolation``, ``kaiser_window``] (Default: ``"sinc_interpolation"``) lowpass_filter_width (int, optional): Controls the sharpness of the filter, more == sharper but less efficient. (Default: ``6``) rolloff (float, optional): The roll-off frequency of the filter, as a fraction of the Nyquist. Lower values reduce anti-aliasing, but also reduce some of the highest frequencies. (Default: ``0.99``) beta (float or None, optional): The shape parameter used for kaiser window. dtype (torch.device, optional): Determnines the precision that resampling kernel is pre-computed and cached. If not provided, kernel is computed with ``torch.float64`` then cached as ``torch.float32``. If you need higher precision, provide ``torch.float64``, and the pre-computed kernel is computed and cached as ``torch.float64``. If you use resample with lower precision, then instead of providing this providing this argument, please use ``Resample.to(dtype)``, so that the kernel generation is still carried out on ``torch.float64``. Example >>> waveform, sample_rate = torchaudio.load("test.wav", normalize=True) >>> transform = transforms.Resample(sample_rate, sample_rate/10) >>> waveform = transform(waveform) """ def __init__( self, orig_freq: int = 16000, new_freq: int = 16000, resampling_method: str = "sinc_interpolation", lowpass_filter_width: int = 6, rolloff: float = 0.99, beta: Optional[float] = None, *, dtype: Optional[torch.dtype] = None, ) -> None: super().__init__() self.orig_freq = orig_freq self.new_freq = new_freq self.gcd = math.gcd(int(self.orig_freq), int(self.new_freq)) self.resampling_method = resampling_method self.lowpass_filter_width = lowpass_filter_width self.rolloff = rolloff self.beta = beta if self.orig_freq != self.new_freq: kernel, self.width = _get_sinc_resample_kernel( self.orig_freq, self.new_freq, self.gcd, self.lowpass_filter_width, self.rolloff, self.resampling_method, beta, dtype=dtype, ) self.register_buffer("kernel", kernel)
[docs] def forward(self, waveform: Tensor) -> Tensor: r""" Args: waveform (Tensor): Tensor of audio of dimension (..., time). Returns: Tensor: Output signal of dimension (..., time). """ if self.orig_freq == self.new_freq: return waveform return _apply_sinc_resample_kernel(waveform, self.orig_freq, self.new_freq, self.gcd, self.kernel, self.width)
[docs]class ComputeDeltas(torch.nn.Module): r"""Compute delta coefficients of a tensor, usually a spectrogram. .. devices:: CPU CUDA .. properties:: Autograd TorchScript See `torchaudio.functional.compute_deltas` for more details. Args: win_length (int, optional): The window length used for computing delta. (Default: ``5``) mode (str, optional): Mode parameter passed to padding. (Default: ``"replicate"``) """ __constants__ = ["win_length"] def __init__(self, win_length: int = 5, mode: str = "replicate") -> None: super(ComputeDeltas, self).__init__() self.win_length = win_length self.mode = mode
[docs] def forward(self, specgram: Tensor) -> Tensor: r""" Args: specgram (Tensor): Tensor of audio of dimension (..., freq, time). Returns: Tensor: Tensor of deltas of dimension (..., freq, time). """ return F.compute_deltas(specgram, win_length=self.win_length, mode=self.mode)
[docs]class TimeStretch(torch.nn.Module): r"""Stretch stft in time without modifying pitch for a given rate. .. devices:: CPU CUDA .. properties:: Autograd TorchScript Proposed in *SpecAugment* :cite:`specaugment`. Args: hop_length (int or None, optional): Length of hop between STFT windows. (Default: ``win_length // 2``) n_freq (int, optional): number of filter banks from stft. (Default: ``201``) fixed_rate (float or None, optional): rate to speed up or slow down by. If None is provided, rate must be passed to the forward method. (Default: ``None``) Example >>> spectrogram = torchaudio.transforms.Spectrogram() >>> stretch = torchaudio.transforms.TimeStretch() >>> >>> original = spectrogram(waveform) >>> streched_1_2 = stretch(original, 1.2) >>> streched_0_9 = stretch(original, 0.9) .. image:: https://download.pytorch.org/torchaudio/doc-assets/specaugment_time_stretch_1.png :width: 600 :alt: Spectrogram streched by 1.2 .. image:: https://download.pytorch.org/torchaudio/doc-assets/specaugment_time_stretch_2.png :width: 600 :alt: The original spectrogram .. image:: https://download.pytorch.org/torchaudio/doc-assets/specaugment_time_stretch_3.png :width: 600 :alt: Spectrogram streched by 0.9 """ __constants__ = ["fixed_rate"] def __init__(self, hop_length: Optional[int] = None, n_freq: int = 201, fixed_rate: Optional[float] = None) -> None: super(TimeStretch, self).__init__() self.fixed_rate = fixed_rate n_fft = (n_freq - 1) * 2 hop_length = hop_length if hop_length is not None else n_fft // 2 self.register_buffer("phase_advance", torch.linspace(0, math.pi * hop_length, n_freq)[..., None])
[docs] def forward(self, complex_specgrams: Tensor, overriding_rate: Optional[float] = None) -> Tensor: r""" Args: complex_specgrams (Tensor): A tensor of dimension `(..., freq, num_frame)` with complex dtype. overriding_rate (float or None, optional): speed up to apply to this batch. If no rate is passed, use ``self.fixed_rate``. (Default: ``None``) Returns: Tensor: Stretched spectrogram. The resulting tensor is of the same dtype as the input spectrogram, but the number of frames is changed to ``ceil(num_frame / rate)``. """ if overriding_rate is None: if self.fixed_rate is None: raise ValueError("If no fixed_rate is specified, must pass a valid rate to the forward method.") rate = self.fixed_rate else: rate = overriding_rate return F.phase_vocoder(complex_specgrams, rate, self.phase_advance)
[docs]class Fade(torch.nn.Module): r"""Add a fade in and/or fade out to an waveform. .. devices:: CPU CUDA .. properties:: Autograd TorchScript Args: fade_in_len (int, optional): Length of fade-in (time frames). (Default: ``0``) fade_out_len (int, optional): Length of fade-out (time frames). (Default: ``0``) fade_shape (str, optional): Shape of fade. Must be one of: "quarter_sine", ``"half_sine"``, ``"linear"``, ``"logarithmic"``, ``"exponential"``. (Default: ``"linear"``) Example >>> waveform, sample_rate = torchaudio.load("test.wav", normalize=True) >>> transform = transforms.Fade(fade_in_len=sample_rate, fade_out_len=2 * sample_rate, fade_shape="linear") >>> faded_waveform = transform(waveform) """ def __init__(self, fade_in_len: int = 0, fade_out_len: int = 0, fade_shape: str = "linear") -> None: super(Fade, self).__init__() self.fade_in_len = fade_in_len self.fade_out_len = fade_out_len self.fade_shape = fade_shape
[docs] def forward(self, waveform: Tensor) -> Tensor: r""" Args: waveform (Tensor): Tensor of audio of dimension `(..., time)`. Returns: Tensor: Tensor of audio of dimension `(..., time)`. """ waveform_length = waveform.size()[-1] device = waveform.device return self._fade_in(waveform_length, device) * self._fade_out(waveform_length, device) * waveform
def _fade_in(self, waveform_length: int, device: torch.device) -> Tensor: fade = torch.linspace(0, 1, self.fade_in_len, device=device) ones = torch.ones(waveform_length - self.fade_in_len, device=device) if self.fade_shape == "linear": fade = fade if self.fade_shape == "exponential": fade = torch.pow(2, (fade - 1)) * fade if self.fade_shape == "logarithmic": fade = torch.log10(0.1 + fade) + 1 if self.fade_shape == "quarter_sine": fade = torch.sin(fade * math.pi / 2) if self.fade_shape == "half_sine": fade = torch.sin(fade * math.pi - math.pi / 2) / 2 + 0.5 return torch.cat((fade, ones)).clamp_(0, 1) def _fade_out(self, waveform_length: int, device: torch.device) -> Tensor: fade = torch.linspace(0, 1, self.fade_out_len, device=device) ones = torch.ones(waveform_length - self.fade_out_len, device=device) if self.fade_shape == "linear": fade = -fade + 1 if self.fade_shape == "exponential": fade = torch.pow(2, -fade) * (1 - fade) if self.fade_shape == "logarithmic": fade = torch.log10(1.1 - fade) + 1 if self.fade_shape == "quarter_sine": fade = torch.sin(fade * math.pi / 2 + math.pi / 2) if self.fade_shape == "half_sine": fade = torch.sin(fade * math.pi + math.pi / 2) / 2 + 0.5 return torch.cat((ones, fade)).clamp_(0, 1)
class _AxisMasking(torch.nn.Module): r"""Apply masking to a spectrogram. Args: mask_param (int): Maximum possible length of the mask. axis (int): What dimension the mask is applied on. iid_masks (bool): Applies iid masks to each of the examples in the batch dimension. This option is applicable only when the input tensor is 4D. p (float, optional): maximum proportion of columns that can be masked. (Default: 1.0) """ __constants__ = ["mask_param", "axis", "iid_masks", "p"] def __init__(self, mask_param: int, axis: int, iid_masks: bool, p: float = 1.0) -> None: super(_AxisMasking, self).__init__() self.mask_param = mask_param self.axis = axis self.iid_masks = iid_masks self.p = p def forward(self, specgram: Tensor, mask_value: float = 0.0) -> Tensor: r""" Args: specgram (Tensor): Tensor of dimension `(..., freq, time)`. mask_value (float): Value to assign to the masked columns. Returns: Tensor: Masked spectrogram of dimensions `(..., freq, time)`. """ # if iid_masks flag marked and specgram has a batch dimension if self.iid_masks and specgram.dim() == 4: return F.mask_along_axis_iid(specgram, self.mask_param, mask_value, self.axis + 1, p=self.p) else: return F.mask_along_axis(specgram, self.mask_param, mask_value, self.axis, p=self.p)
[docs]class FrequencyMasking(_AxisMasking): r"""Apply masking to a spectrogram in the frequency domain. .. devices:: CPU CUDA .. properties:: Autograd TorchScript Proposed in *SpecAugment* :cite:`specaugment`. Args: freq_mask_param (int): maximum possible length of the mask. Indices uniformly sampled from [0, freq_mask_param). iid_masks (bool, optional): whether to apply different masks to each example/channel in the batch. (Default: ``False``) This option is applicable only when the input tensor is 4D. Example >>> spectrogram = torchaudio.transforms.Spectrogram() >>> masking = torchaudio.transforms.FrequencyMasking(freq_mask_param=80) >>> >>> original = spectrogram(waveform) >>> masked = masking(original) .. image:: https://download.pytorch.org/torchaudio/doc-assets/specaugment_freq_masking1.png :alt: The original spectrogram .. image:: https://download.pytorch.org/torchaudio/doc-assets/specaugment_freq_masking2.png :alt: The spectrogram masked along frequency axis """ def __init__(self, freq_mask_param: int, iid_masks: bool = False) -> None: super(FrequencyMasking, self).__init__(freq_mask_param, 1, iid_masks)
[docs]class TimeMasking(_AxisMasking): r"""Apply masking to a spectrogram in the time domain. .. devices:: CPU CUDA .. properties:: Autograd TorchScript Proposed in *SpecAugment* :cite:`specaugment`. Args: time_mask_param (int): maximum possible length of the mask. Indices uniformly sampled from [0, time_mask_param). iid_masks (bool, optional): whether to apply different masks to each example/channel in the batch. (Default: ``False``) This option is applicable only when the input tensor is 4D. p (float, optional): maximum proportion of time steps that can be masked. Must be within range [0.0, 1.0]. (Default: 1.0) Example >>> spectrogram = torchaudio.transforms.Spectrogram() >>> masking = torchaudio.transforms.TimeMasking(time_mask_param=80) >>> >>> original = spectrogram(waveform) >>> masked = masking(original) .. image:: https://download.pytorch.org/torchaudio/doc-assets/specaugment_time_masking1.png :alt: The original spectrogram .. image:: https://download.pytorch.org/torchaudio/doc-assets/specaugment_time_masking2.png :alt: The spectrogram masked along time axis """ def __init__(self, time_mask_param: int, iid_masks: bool = False, p: float = 1.0) -> None: if not 0.0 <= p <= 1.0: raise ValueError(f"The value of p must be between 0.0 and 1.0 ({p} given).") super(TimeMasking, self).__init__(time_mask_param, 2, iid_masks, p=p)
[docs]class Loudness(torch.nn.Module): r"""Measure audio loudness according to the ITU-R BS.1770-4 recommendation. .. devices:: CPU CUDA .. properties:: TorchScript Args: sample_rate (int): Sample rate of audio signal. Example >>> waveform, sample_rate = torchaudio.load("test.wav", normalize=True) >>> transform = transforms.Loudness(sample_rate) >>> loudness = transform(waveform) Reference: - https://www.itu.int/rec/R-REC-BS.1770-4-201510-I/en """ __constants__ = ["sample_rate"] def __init__(self, sample_rate: int): super(Loudness, self).__init__() self.sample_rate = sample_rate
[docs] def forward(self, wavefrom: Tensor): r""" Args: waveform(torch.Tensor): audio waveform of dimension `(..., channels, time)` Returns: Tensor: loudness estimates (LKFS) """ return F.loudness(wavefrom, self.sample_rate)
[docs]class Vol(torch.nn.Module): r"""Adjust volume of waveform. .. devices:: CPU CUDA .. properties:: Autograd TorchScript Args: gain (float): Interpreted according to the given gain_type: If ``gain_type`` = ``amplitude``, ``gain`` is a positive amplitude ratio. If ``gain_type`` = ``power``, ``gain`` is a power (voltage squared). If ``gain_type`` = ``db``, ``gain`` is in decibels. gain_type (str, optional): Type of gain. One of: ``amplitude``, ``power``, ``db`` (Default: ``amplitude``) Example >>> waveform, sample_rate = torchaudio.load("test.wav", normalize=True) >>> transform = transforms.Vol(gain=0.5, gain_type="amplitude") >>> quieter_waveform = transform(waveform) """ def __init__(self, gain: float, gain_type: str = "amplitude"): super(Vol, self).__init__() self.gain = gain self.gain_type = gain_type if gain_type in ["amplitude", "power"] and gain < 0: raise ValueError("If gain_type = amplitude or power, gain must be positive.")
[docs] def forward(self, waveform: Tensor) -> Tensor: r""" Args: waveform (Tensor): Tensor of audio of dimension `(..., time)`. Returns: Tensor: Tensor of audio of dimension `(..., time)`. """ if self.gain_type == "amplitude": waveform = waveform * self.gain if self.gain_type == "db": waveform = F.gain(waveform, self.gain) if self.gain_type == "power": waveform = F.gain(waveform, 10 * math.log10(self.gain)) return torch.clamp(waveform, -1, 1)
[docs]class SlidingWindowCmn(torch.nn.Module): r""" Apply sliding-window cepstral mean (and optionally variance) normalization per utterance. .. devices:: CPU CUDA .. properties:: Autograd TorchScript Args: cmn_window (int, optional): Window in frames for running average CMN computation (int, default = 600) min_cmn_window (int, optional): Minimum CMN window used at start of decoding (adds latency only at start). Only applicable if center == false, ignored if center==true (int, default = 100) center (bool, optional): If true, use a window centered on the current frame (to the extent possible, modulo end effects). If false, window is to the left. (bool, default = false) norm_vars (bool, optional): If true, normalize variance to one. (bool, default = false) Example >>> waveform, sample_rate = torchaudio.load("test.wav", normalize=True) >>> transform = transforms.SlidingWindowCmn(cmn_window=1000) >>> cmn_waveform = transform(waveform) """ def __init__( self, cmn_window: int = 600, min_cmn_window: int = 100, center: bool = False, norm_vars: bool = False ) -> None: super().__init__() self.cmn_window = cmn_window self.min_cmn_window = min_cmn_window self.center = center self.norm_vars = norm_vars
[docs] def forward(self, specgram: Tensor) -> Tensor: r""" Args: specgram (Tensor): Tensor of spectrogram of dimension `(..., time, freq)`. Returns: Tensor: Tensor of spectrogram of dimension `(..., time, freq)`. """ cmn_specgram = F.sliding_window_cmn(specgram, self.cmn_window, self.min_cmn_window, self.center, self.norm_vars) return cmn_specgram
[docs]class Vad(torch.nn.Module): r"""Voice Activity Detector. Similar to SoX implementation. .. devices:: CPU CUDA .. properties:: TorchScript Attempts to trim silence and quiet background sounds from the ends of recordings of speech. The algorithm currently uses a simple cepstral power measurement to detect voice, so may be fooled by other things, especially music. The effect can trim only from the front of the audio, so in order to trim from the back, the reverse effect must also be used. Args: sample_rate (int): Sample rate of audio signal. trigger_level (float, optional): The measurement level used to trigger activity detection. This may need to be changed depending on the noise level, signal level, and other characteristics of the input audio. (Default: 7.0) trigger_time (float, optional): The time constant (in seconds) used to help ignore short bursts of sound. (Default: 0.25) search_time (float, optional): The amount of audio (in seconds) to search for quieter/shorter bursts of audio to include prior to the detected trigger point. (Default: 1.0) allowed_gap (float, optional): The allowed gap (in seconds) between quiteter/shorter bursts of audio to include prior to the detected trigger point. (Default: 0.25) pre_trigger_time (float, optional): The amount of audio (in seconds) to preserve before the trigger point and any found quieter/shorter bursts. (Default: 0.0) boot_time (float, optional) The algorithm (internally) uses adaptive noise estimation/reduction in order to detect the start of the wanted audio. This option sets the time for the initial noise estimate. (Default: 0.35) noise_up_time (float, optional) Time constant used by the adaptive noise estimator for when the noise level is increasing. (Default: 0.1) noise_down_time (float, optional) Time constant used by the adaptive noise estimator for when the noise level is decreasing. (Default: 0.01) noise_reduction_amount (float, optional) Amount of noise reduction to use in the detection algorithm (e.g. 0, 0.5, ...). (Default: 1.35) measure_freq (float, optional) Frequency of the algorithm’s processing/measurements. (Default: 20.0) measure_duration: (float or None, optional) Measurement duration. (Default: Twice the measurement period; i.e. with overlap.) measure_smooth_time (float, optional) Time constant used to smooth spectral measurements. (Default: 0.4) hp_filter_freq (float, optional) "Brick-wall" frequency of high-pass filter applied at the input to the detector algorithm. (Default: 50.0) lp_filter_freq (float, optional) "Brick-wall" frequency of low-pass filter applied at the input to the detector algorithm. (Default: 6000.0) hp_lifter_freq (float, optional) "Brick-wall" frequency of high-pass lifter used in the detector algorithm. (Default: 150.0) lp_lifter_freq (float, optional) "Brick-wall" frequency of low-pass lifter used in the detector algorithm. (Default: 2000.0) Example >>> waveform, sample_rate = torchaudio.load("test.wav", normalize=True) >>> waveform_reversed, sample_rate = apply_effects_tensor(waveform, sample_rate, [["reverse"]]) >>> transform = transforms.Vad(sample_rate=sample_rate, trigger_level=7.5) >>> waveform_reversed_front_trim = transform(waveform_reversed) >>> waveform_end_trim, sample_rate = apply_effects_tensor( >>> waveform_reversed_front_trim, sample_rate, [["reverse"]] >>> ) Reference: - http://sox.sourceforge.net/sox.html """ def __init__( self, sample_rate: int, trigger_level: float = 7.0, trigger_time: float = 0.25, search_time: float = 1.0, allowed_gap: float = 0.25, pre_trigger_time: float = 0.0, boot_time: float = 0.35, noise_up_time: float = 0.1, noise_down_time: float = 0.01, noise_reduction_amount: float = 1.35, measure_freq: float = 20.0, measure_duration: Optional[float] = None, measure_smooth_time: float = 0.4, hp_filter_freq: float = 50.0, lp_filter_freq: float = 6000.0, hp_lifter_freq: float = 150.0, lp_lifter_freq: float = 2000.0, ) -> None: super().__init__() self.sample_rate = sample_rate self.trigger_level = trigger_level self.trigger_time = trigger_time self.search_time = search_time self.allowed_gap = allowed_gap self.pre_trigger_time = pre_trigger_time self.boot_time = boot_time self.noise_up_time = noise_up_time self.noise_down_time = noise_down_time self.noise_reduction_amount = noise_reduction_amount self.measure_freq = measure_freq self.measure_duration = measure_duration self.measure_smooth_time = measure_smooth_time self.hp_filter_freq = hp_filter_freq self.lp_filter_freq = lp_filter_freq self.hp_lifter_freq = hp_lifter_freq self.lp_lifter_freq = lp_lifter_freq
[docs] def forward(self, waveform: Tensor) -> Tensor: r""" Args: waveform (Tensor): Tensor of audio of dimension `(channels, time)` or `(time)` Tensor of shape `(channels, time)` is treated as a multi-channel recording of the same event and the resulting output will be trimmed to the earliest voice activity in any channel. """ return F.vad( waveform=waveform, sample_rate=self.sample_rate, trigger_level=self.trigger_level, trigger_time=self.trigger_time, search_time=self.search_time, allowed_gap=self.allowed_gap, pre_trigger_time=self.pre_trigger_time, boot_time=self.boot_time, noise_up_time=self.noise_up_time, noise_down_time=self.noise_down_time, noise_reduction_amount=self.noise_reduction_amount, measure_freq=self.measure_freq, measure_duration=self.measure_duration, measure_smooth_time=self.measure_smooth_time, hp_filter_freq=self.hp_filter_freq, lp_filter_freq=self.lp_filter_freq, hp_lifter_freq=self.hp_lifter_freq, lp_lifter_freq=self.lp_lifter_freq, )
[docs]class SpectralCentroid(torch.nn.Module): r"""Compute the spectral centroid for each channel along the time axis. .. devices:: CPU CUDA .. properties:: Autograd TorchScript The spectral centroid is defined as the weighted average of the frequency values, weighted by their magnitude. Args: sample_rate (int): Sample rate of audio signal. n_fft (int, optional): Size of FFT, creates ``n_fft // 2 + 1`` bins. (Default: ``400``) win_length (int or None, optional): Window size. (Default: ``n_fft``) hop_length (int or None, optional): Length of hop between STFT windows. (Default: ``win_length // 2``) pad (int, optional): Two sided padding of signal. (Default: ``0``) window_fn (Callable[..., Tensor], optional): A function to create a window tensor that is applied/multiplied to each frame/window. (Default: ``torch.hann_window``) wkwargs (dict or None, optional): Arguments for window function. (Default: ``None``) Example >>> waveform, sample_rate = torchaudio.load("test.wav", normalize=True) >>> transform = transforms.SpectralCentroid(sample_rate) >>> spectral_centroid = transform(waveform) # (channel, time) """ __constants__ = ["sample_rate", "n_fft", "win_length", "hop_length", "pad"] def __init__( self, sample_rate: int, n_fft: int = 400, win_length: Optional[int] = None, hop_length: Optional[int] = None, pad: int = 0, window_fn: Callable[..., Tensor] = torch.hann_window, wkwargs: Optional[dict] = None, ) -> None: super(SpectralCentroid, self).__init__() self.sample_rate = sample_rate self.n_fft = n_fft self.win_length = win_length if win_length is not None else n_fft self.hop_length = hop_length if hop_length is not None else self.win_length // 2 window = window_fn(self.win_length) if wkwargs is None else window_fn(self.win_length, **wkwargs) self.register_buffer("window", window) self.pad = pad
[docs] def forward(self, waveform: Tensor) -> Tensor: r""" Args: waveform (Tensor): Tensor of audio of dimension `(..., time)`. Returns: Tensor: Spectral Centroid of size `(..., time)`. """ return F.spectral_centroid( waveform, self.sample_rate, self.pad, self.window, self.n_fft, self.hop_length, self.win_length )
[docs]class PitchShift(LazyModuleMixin, torch.nn.Module): r"""Shift the pitch of a waveform by ``n_steps`` steps. .. devices:: CPU CUDA .. properties:: TorchScript Args: waveform (Tensor): The input waveform of shape `(..., time)`. sample_rate (int): Sample rate of `waveform`. n_steps (int): The (fractional) steps to shift `waveform`. bins_per_octave (int, optional): The number of steps per octave (Default : ``12``). n_fft (int, optional): Size of FFT, creates ``n_fft // 2 + 1`` bins (Default: ``512``). win_length (int or None, optional): Window size. If None, then ``n_fft`` is used. (Default: ``None``). hop_length (int or None, optional): Length of hop between STFT windows. If None, then ``win_length // 4`` is used (Default: ``None``). window (Tensor or None, optional): Window tensor that is applied/multiplied to each frame/window. If None, then ``torch.hann_window(win_length)`` is used (Default: ``None``). Example >>> waveform, sample_rate = torchaudio.load("test.wav", normalize=True) >>> transform = transforms.PitchShift(sample_rate, 4) >>> waveform_shift = transform(waveform) # (channel, time) """ __constants__ = ["sample_rate", "n_steps", "bins_per_octave", "n_fft", "win_length", "hop_length"] kernel: UninitializedParameter width: int def __init__( self, sample_rate: int, n_steps: int, bins_per_octave: int = 12, n_fft: int = 512, win_length: Optional[int] = None, hop_length: Optional[int] = None, window_fn: Callable[..., Tensor] = torch.hann_window, wkwargs: Optional[dict] = None, ) -> None: super().__init__() self.n_steps = n_steps self.bins_per_octave = bins_per_octave self.sample_rate = sample_rate self.n_fft = n_fft self.win_length = win_length if win_length is not None else n_fft self.hop_length = hop_length if hop_length is not None else self.win_length // 4 window = window_fn(self.win_length) if wkwargs is None else window_fn(self.win_length, **wkwargs) self.register_buffer("window", window) rate = 2.0 ** (-float(n_steps) / bins_per_octave) self.orig_freq = int(sample_rate / rate) self.gcd = math.gcd(int(self.orig_freq), int(sample_rate)) if self.orig_freq != sample_rate: self.width = -1 self.kernel = UninitializedParameter(device=None, dtype=None)
[docs] def initialize_parameters(self, input): if self.has_uninitialized_params(): if self.orig_freq != self.sample_rate: with torch.no_grad(): kernel, self.width = _get_sinc_resample_kernel( self.orig_freq, self.sample_rate, self.gcd, dtype=input.dtype, device=input.device, ) self.kernel.materialize(kernel.shape) self.kernel.copy_(kernel)
[docs] def forward(self, waveform: Tensor) -> Tensor: r""" Args: waveform (Tensor): Tensor of audio of dimension `(..., time)`. Returns: Tensor: The pitch-shifted audio of shape `(..., time)`. """ shape = waveform.size() waveform_stretch = _stretch_waveform( waveform, self.n_steps, self.bins_per_octave, self.n_fft, self.win_length, self.hop_length, self.window, ) if self.orig_freq != self.sample_rate: waveform_shift = _apply_sinc_resample_kernel( waveform_stretch, self.orig_freq, self.sample_rate, self.gcd, self.kernel, self.width, ) else: waveform_shift = waveform_stretch return _fix_waveform_shape( waveform_shift, shape, )
[docs]class RNNTLoss(torch.nn.Module): """Compute the RNN Transducer loss from *Sequence Transduction with Recurrent Neural Networks* :cite:`graves2012sequence`. .. devices:: CPU CUDA .. properties:: Autograd TorchScript The RNN Transducer loss extends the CTC loss by defining a distribution over output sequences of all lengths, and by jointly modelling both input-output and output-output dependencies. Args: blank (int, optional): blank label (Default: ``-1``) clamp (float, optional): clamp for gradients (Default: ``-1``) reduction (string, optional): Specifies the reduction to apply to the output: ``"none"`` | ``"mean"`` | ``"sum"``. (Default: ``"mean"``) fused_log_softmax (bool): set to False if calling log_softmax outside of loss (Default: ``True``) Example >>> # Hypothetical values >>> logits = torch.tensor([[[[0.1, 0.6, 0.1, 0.1, 0.1], >>> [0.1, 0.1, 0.6, 0.1, 0.1], >>> [0.1, 0.1, 0.2, 0.8, 0.1]], >>> [[0.1, 0.6, 0.1, 0.1, 0.1], >>> [0.1, 0.1, 0.2, 0.1, 0.1], >>> [0.7, 0.1, 0.2, 0.1, 0.1]]]], >>> dtype=torch.float32, >>> requires_grad=True) >>> targets = torch.tensor([[1, 2]], dtype=torch.int) >>> logit_lengths = torch.tensor([2], dtype=torch.int) >>> target_lengths = torch.tensor([2], dtype=torch.int) >>> transform = transforms.RNNTLoss(blank=0) >>> loss = transform(logits, targets, logit_lengths, target_lengths) >>> loss.backward() """ def __init__( self, blank: int = -1, clamp: float = -1.0, reduction: str = "mean", fused_log_softmax: bool = True, ): super().__init__() self.blank = blank self.clamp = clamp self.reduction = reduction self.fused_log_softmax = fused_log_softmax
[docs] def forward( self, logits: Tensor, targets: Tensor, logit_lengths: Tensor, target_lengths: Tensor, ): """ Args: logits (Tensor): Tensor of dimension `(batch, max seq length, max target length + 1, class)` containing output from joiner targets (Tensor): Tensor of dimension `(batch, max target length)` containing targets with zero padded logit_lengths (Tensor): Tensor of dimension `(batch)` containing lengths of each sequence from encoder target_lengths (Tensor): Tensor of dimension `(batch)` containing lengths of targets for each sequence Returns: Tensor: Loss with the reduction option applied. If ``reduction`` is ``"none"``, then size (batch), otherwise scalar. """ return F.rnnt_loss( logits, targets, logit_lengths, target_lengths, self.blank, self.clamp, self.reduction, self.fused_log_softmax, )

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